# Why 24 bit audio and anything over 48k is not only worthless, but bad for music.



## keanex

http://people.xiph.org/~xiphmont/demo/neil-young.html
  
 Read em up boys.


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## ab initio

keanex said:


> http://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> Read em up boys.





While I do think this is a great article, i believe it has been posted here many times. You can get a sense for the age of the article just from the first sentence: 

'Articles* last month* revealed that musician Neil Young and Apple's *Steve Jobs discussed *offering digital music downloads of 'uncompromised studio quality''

The video he made afterwards is even better at explaining and demomnstrating some of the key concepts. It has also been linked to from this forum many times. It's great stuff!

Cheers

Cheers


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## michaelios

Yeah. Why buying lamborghini when speed limit is 40 mph? 

Отправлено с моего GT-I9100 через Tapatalk


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## kraken2109

Everyone on this board knows that, it's the rest of head-fi you have to convince. Which is difficult since head-fi themselves are sponsered by HDTracks...


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## ralphp@optonline

kraken2109 said:


> Everyone on this board knows that, it's the rest of head-fi you have to convince. Which is difficult since head-fi themselves are sponsered by HDTracks...


 

 Sometimes I wonder if there is any difference between sponsorship and outright bribery. In the case of high end audio publications similar lack of difference exists between advertising revenue and outright bribery.
  
 Truth, honesty and integrity are all just so 20th century. The 21st century is all about lies, dishonesty and corruption.


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## ProtegeManiac

michaelios said:


> Yeah. Why buying lamborghini when speed limit is 40 mph?


 
  
 Not the best analogy for discussing high-res audio. For most rich douchebags that statement is true, but for others, they have a racetrack within reasonable driving distance. While most a-holes stuck in LA traffic only take theirs to da clubbz, don't cuss at any Lambo on the freeway next to you because for all you know that might be one of the Andrettis going to Laguna Seca. It's not like they only take them out for the annual Bull Run up and down California.


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## bigshot

The limitation is in human ears. It doesn't matter how high a frequency you want your stereo to produce and how wide a dynamic range, it all comes down to whether human ears can hear it.

Audiophools love to spend lots of money pushing the decimal point further and further to the left and making the frequencies go higher and higher, but at a certain point, it all becomes moot because only bats can hear it.


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## cjl

One thing I disagree about (and it's stated correctly in the article, but the title here is misleading): 24 bit does not harm the music in any way. Having a 24 bit DAC can provide some advantages too - namely, it keeps the noise floor and signal to noise ratio acceptable even when using digital volume control. 24 bit file formats can also be useful when capturing recordings due to increased headroom without an audible noise floor, and 24+ bit formats are good for mastering/processing, again, to keep the noise from compounding during mastering. Of course, 24 bit music files are pointless for playback purposes - 16 bit is perfectly fine for that.


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## emailists

I am a bit aghast at some of the opinions displayed in this thread.  On a high resolution system, the benefits of 192K over 96K is clearly audible.  I use CHesky Jen Chapin recording of the same album I bought in 96 then 192 when it was released.  Just because someone can't hear the difference in their system doesn't mean it doesn't exsist.   I guess at one time just because people couldn't see the curvature of the earth led them to believe the earth was flat.


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## Steve Eddy

emailists said:


> I am a bit aghast at some of the opinions displayed in this thread.  On a high resolution system, the benefits of 192K over 96K is clearly audible.




Then it should be trivially easy for someone to demonstrate that under controlled conditions. No one has as far as I'm aware.

se


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## bigshot

emailists said:


> I am a bit aghast at some of the opinions displayed in this thread.  On a high resolution system, the benefits of 192K over 96K is clearly audible.


 
  
 Just as a bit of interesting trivia... 44.1K covers the full spectrum of frequencies that humans can hear- 20Hz to 20kHz, with a bit to spare. Higher sampling rates extend the frequency response higher, far beyond our ability to hear, but the core frequencies below 20kHz are rendered exactly the same at 44.1 as they are at 192. So whatever it is that you seem to think is clearly audible isn't audible with human ears. Perhaps a bat!
  
 However, it is possible that your equipment isn't designed to deal with super high frequencies and is adding distortion down in the audible range. So if you are positive you are hearing a difference, it is almost certainly noise, not music.


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## SilverEars

steve eddy said:


> Then it should be trivially easy for someone to demonstrate that under controlled conditions. No one has as far as I'm aware.
> 
> se


 
 The controlled condition, I really want is the most resolving audio system known to man.  Then, if it still cannot distinguished, and only then, I will side with the rebel alliance here.


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## ToddTheMetalGod

There's so many reasons to not use 24-bit over 16-bit, or 96/192 kHz over 44.1 kHz. No music, not even classical, has enough dynamic range to take advantage of more than 16-bit. Unless you can hear over 22 kHz than above 44.1 kHz is also useless (plus instruments don't have any important information in that register). Really, unless your DAC processes 24-bit data different resulting in better sound... then there is no reason to use it. But the reason it is processed differently on some DACs is because of the believe that Hi-Res is better. It's a viscous cycle.


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## bigshot

silverears said:


> The controlled condition, I really want is the most resolving audio system known to man.  Then, if it still cannot distinguished, and only then, I will side with the rebel alliance here.


 
  
 Why not just do a little googling and find out what additional benefits higher bitrates and sampling rates offer, then compare that to the thresholds of perception and you'll have your answer.
  
 higher bitrate = lower noise floor
 higher sampling rate = extended frequency response
  
 Since 16/44.1 already has a noise floor so low you would have to turn the volume up to hearing damage levels to hear it all, and since it has a frequency response that covers the full spectrum of human hearing, what can you possibly expect to hear?


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## Hutnicks

bigshot said:


> Why not just do a little googling and find out what additional benefits higher bitrates and sampling rates offer, then compare that to the thresholds of perception and you'll have your answer.
> 
> higher bitrate = lower noise floor
> higher sampling rate = extended frequency response
> ...


 

 The vibrations caused by the Freightliner driving by the studio during the recording session?


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## Steve Eddy

silverears said:


> The controlled condition, I really want is the most resolving audio system known to man.  Then, if it still cannot distinguished, and only then, I will side with the rebel alliance here.






se


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## emailists

I think people are missing the point that a sample rate only approximately twice the frequency rate (and getting close to 20K cycles is where that occurs) is low resolution.  Think of a fax machine sampling a line 1/10th an inch thick with a scanner whose resolution is 1/20th of an inch.  There would only be two samples (or dots) to represent the line.  Is the line on the fax there, yes, but it's highly pixelated.  And there have been experiments that show ultrasonic information in a recording does have an effect on fidelity.  If you're not hearing the benefits of higher resolution digital, then either the equipment is not resolving enough to allow it, the software (or downsampling) is not up to snuff, or the listener may not be cued into the benefit's of higher res files.   When I am color correcting video and I shift the hue slightly and toggle back and forth for a client, sometimes they can't see the color shift, while I can see it easily. Does that imply it's not there because some people can't see  (or hear) it?  No it doesn. But perhaps if I did that same color shift on my 15' DLP projection screen the person might be able to see it in that instance because the scale of the playback device makes things more easily discernible.  
  
 When I'm mixing audio I can usually easily hear a 1db increase, and 1 db is supposed to be the threshold of human hearing.  There are several times I've only raised a clip .5 db.   
  
 I also hate to bring this up but my theory is that many in the pro audio world have compromised their hearing by  exposure, so perhaps that is where some of the opinions about 44.1/16 audio quality come from, and possibly is the source of all the bad sounding recordings out there.   ALso ABX testing configs go through added passive or active components which could easily mask the fine details of higher res files.
  
 I should add that I am listening though monitors with edge of the art vapor deposited beryllium concentric drivers, (TAD) that act as point sources  and so perhaps that is why small sonic differences are so easily discernible.


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## bigshot

emailists said:


> I think people are missing the point that a sample rate only approximately twice the frequency rate (and getting close to 20K cycles is where that occurs) is low resolution.  Think of a fax machine sampling a line 1/10th an inch thick with a scanner whose resolution is 1/20th of an inch.  There would only be two samples (or dots) to represent the line.  Is the line on the fax there, yes, but it's highly pixelated.


 
  
 According to Nyquist, it only takes two points to perfectly reproduce a sound wave. The stair step and pixel analogy is not the right way to think about it. Digital audio is different than digital images. You are reconstructing the sound wave from the ground up. You aren't creating an image of it in pixels.
  
 The threshold of human perception for changes in volume in direct a/b comparison is between .5 and 1 dB, depending on the sound. For tones, the lower, for sparse and varied music, the higher.
  
 The resolution of 20-20 in a redbook audio file is EXACTLY the same as the resolution of the same frequencies in a 24/96 file. The only difference is the depth of the noise floor and the super audible frequencies (which have been proven to be not necessary for sound quality of music). Recorded music that has been mixed and mastered fits easily within 16/44.1 with room to spare all around.


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## ab initio

emailists said:


> I am a bit aghast at some of the opinions displayed in this thread.  On a high resolution system, the benefits of 192K over 96K is clearly audible.  I use CHesky Jen Chapin recording of the same album I bought in 96 then 192 when it was released.  Just because someone can't hear the difference in their system doesn't mean it doesn't exsist.   I guess at one time just because people couldn't see the curvature of the earth led them to believe the earth was flat.


 
  
 Prove it please. Citations needed.
  
 Why don't you take the Chesky Jen Chapin album you have, take the 192kHz and use foobar to downsample it to 44.1k. Then all you have to do is post your ABX log from foobar's ABX tool for 20 trials between the 44.1kHz and 192kHz versions of the same song. I think everybody here would be interested in seeing the results. The best part of all is that since you already have the album, you can do the test for free!
  
 cheers


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## ToddTheMetalGod

I'm going to go subjective for a second here, but I've tried listening to two recordings of the same master... one Hi-Res and one not (not blindly just regular listening) on many different systems with different DACs and could not tell the difference. 

Even if there is, it is likely out of human threshold of hearing.


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## SilverEars

Doesn't 24 bit mean there are 2^24 number of vertical levels or precision of the line-out signal?  And you use the analog volume knob which amplifies it.  This depends on how it was sampled in the first place.  If you sample the original analog 24-bit, wouldn't it make the output into 24-bit still precise?  If it was sampled at 16, then there is no point.  Also, Sampling frequency will make the signal reproduction much more accurate.  
  
 I guess the analog distortion would have more affect on the signal than such high signal conversion precision.


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## ab initio

silverears said:


> Doesn't 24 bit mean there are 2^24 number of vertical levels or precision of the line-out signal?  And you use the analog volume knob which amplifies it.  This depends on how it was sampled in the first place.  If you sample the original analog 24-bit, wouldn't it make the output into 24-bit still precise?  If it was sampled at 16, then there is no point.
> 
> Is this why we see 24-bit tracks floating around that says vinyl on it, since it was sampled 24-bit from vinyl?


 

 Not exactly. Dither allows signals to be recreated with much finer resolution than one might expect niavely from simply the number of discrete levels. For example, see the wiki page on dither: http://en.wikipedia.org/wiki/Dither
  
  
 As for the vinyl bit... yeah, folks like to record their vinyl records at absurd bitdepths and sampling rates because they like to support the harddisk industry.
  
 Cheers


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## SilverEars

ab initio said:


> Not exactly. Dither allows signals to be recreated with much finer resolution than one might expect niavely from simply the number of discrete levels. For example, see the wiki page on dither: http://en.wikipedia.org/wiki/Dither
> 
> 
> As for the vinyl bit... yeah, folks like to record their vinyl records at absurd bitdepths and sampling rates because they like to support the harddisk industry.
> ...


 
 Isn't that more of interpolation, or upconverting than recreation of the original signal?  If you sample the original signal at high sampling rate, it will be more precise than sampling at lower sampling rate and up converting.  up converting is padding it with samples that are guessed(filling in the gaps), not the original samples taken.


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## bigshot

silverears said:


> Doesn't 24 bit mean there are 2^24 number of vertical levels or precision of the line-out signal?  And you use the analog volume knob which amplifies it.


 
  
 How much dynamic range can you hear?


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## ab initio

silverears said:


> Isn't that more of interpolation, or upconverting than recreation of the original signal?  If you sample the original signal at high sampling rate, it will be more precise than sampling at lower sampling rate and up converting.  up converting is padding it with samples that are guessed(filling in the gaps), not the original samples taken.


 

 No, Dither is most certainly not interpolating. Dither exchanges quantization distortion (which shows up as energy concentrated at a few frequencies) for noise (which spreads the energy over many frequencies, resulting in noise with much lower peak amplitude).
  
 Cheers


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## SilverEars

Well anyway, on second thought, I have doubts that you need this much precision because the amp is gonna distort the signal, and that's has more affect on the signal.


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## ab initio

silverears said:


> Well anyway, on second thought, I have doubts that you need this much precision because the amp is gonna distort the signal, and that's has more affect on the signal.




In general, i think the physical transducer is going to distort the waveform more than either digitization or amplification. But yes, i think fussing over high def audio formats is one of hifi's lesser concerns.

Cheers


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## SilverEars

ab initio said:


> In general, i think the physical transducer is going to distort the waveform more than either digitization or amplification. But yes, i think fussing over high def audio formats is one of hifi's lesser concerns.
> 
> Cheers


 
 Well, when I A/B'd my DACs, I found one that my ears seems to favor, and sounded transparent.  I wonder what it could be, maybe the DAC's analog stage is done better than the others?


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## bigshot

Sounds like you got stuck with some defective DACs. I always check everything before it goes into my rig, and everything has always been transparent.


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## SilverEars

bigshot said:


> Sounds like you got stuck with some defective DACs. I always check everything before it goes into my rig, and everything has always been transparent.


 
 Which DAC do you use?


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## roamling

Here is my criteria for looking at high res audio:
  
 First of all I always try to get a 24bit album over a 16bit CD version or download, that is if the original recording was made in 24bit. 
 Second I try to get the recording at the sample rate it was recorded, if it was recored 48khz then I go for that, if its 192khz then its 192khz
 Third if the price for an album download is not right common sense applies....
  
 Other than that I am just enjoying high res music


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## bigshot

Just knock it all down to AAC 320 and save a load of disk space.


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## moriez

bigshot said:


> The limitation is in human ears. It doesn't matter how high a frequency you want your stereo to produce and how wide a dynamic range, *it all comes down to whether human ears can hear it.*
> 
> *Audiophools* love to spend lots of money pushing the decimal point further and further to the left and making the frequencies go higher and higher, but at a certain point, it all becomes moot *because only bats can hear it*.


 
  
 Lol, on point plus a good laugh


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## Currawong

ralphp@optonline said:


> kraken2109 said:
> 
> 
> > Everyone on this board knows that, it's the rest of head-fi you have to convince. Which is difficult since head-fi themselves are sponsered by HDTracks...
> ...


 
  
 I think you're confusing us with other sites. Sponsorship has zero bearing on the running of the site excepting that sponsors are except from most of the restrictions for Members of the Trade in the Terms of Service. Advertising is not handled by anyone who actively runs the site. In the case of most sponsors (such as the ones that don't post or don't have obvious banners), I'd have to actually ask Huddler if they are sponsoring or not. Actually, pre-Huddler, a company that had become successful through popularity on Head-Fi was _expected by the community _to pay for sponsorship to support it in turn.
  
 One of the nice things we have here is the choice to discuss what one likes, either in favour or against different approaches to audio gear. You guys should value that.


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## kraken2109

currawong said:


> I think you're confusing us with other sites. Sponsorship has zero bearing on the running of the site excepting that sponsors are except from most of the restrictions for Members of the Trade in the Terms of Service. Advertising is not handled by anyone who actively runs the site. In the case of most sponsors (such as the ones that don't post or don't have obvious banners), I'd have to actually ask Huddler if they are sponsoring or not. Actually, pre-Huddler, a company that had become successful through popularity on Head-Fi was _expected by the community _to pay for sponsorship to support it in turn.
> 
> One of the nice things we have here is the choice to discuss what one likes, either in favour or against different approaches to audio gear. You guys should value that.


 
 I understand what you're saying, but when head-fi have threads and even videos featuring Jude advertising an album from HDTracks where he himself tells you to buy the highest resolution version I can't help but feel it is influencing people in a negative way.
  
 Surely the idea of this forum is to educate people, not sell products from sponsors?


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## ralphp@optonline

kraken2109 said:


> I understand what you're saying, but when head-fi have threads and even videos featuring Jude advertising an album from HDTracks where he himself tells you to buy the highest resolution version I can't help but feel it is influencing people in a negative way.
> 
> Surely the idea of this forum is to educate people, not sell products from sponsors?


 

 +1
  
 Currawong: Please explain without tripping over your tongue.


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## Steve Eddy

bigshot said:


> How much dynamic range can you hear?




The instantaneous dynamic range of our ears is about 60dB.

se


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## Currawong

kraken2109 said:


> currawong said:
> 
> 
> > I think you're confusing us with other sites. Sponsorship has zero bearing on the running of the site excepting that sponsors are except from most of the restrictions for Members of the Trade in the Terms of Service. Advertising is not handled by anyone who actively runs the site. In the case of most sponsors (such as the ones that don't post or don't have obvious banners), I'd have to actually ask Huddler if they are sponsoring or not. Actually, pre-Huddler, a company that had become successful through popularity on Head-Fi was _expected by the community _to pay for sponsorship to support it in turn.
> ...


 
  
 So if Jude (or I) like something that happens to be made by someone who sponsors the site, he shouldn't talk about it?   That's basically what you're saying. 
  
 I like the music, personally, and appreciate the effort to make better recordings, since I have more expensive than average gear for a member here. The nice thing here is that we can discuss these topics freely and put forth out opinions (as long as we aren't being rude).


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## ralphp@optonline

currawong said:


> So if Jude (or I) like something that happens to be made by someone who sponsors the site, he shouldn't talk about it?   That's basically what you're saying.
> 
> I like the music, personally, and appreciate the effort to make better recordings, since I have more expensive than average gear for a member here. The nice thing here is that we can discuss these topics freely and put forth out opinions (as long as we aren't being rude).


 
 I believe that issue isn't whether or not you, Jude or anyone else on this site should be able to "talk" or "discuss" a given product on the site but rather whether you, Jude or anyone else on this site should be recommending fellow members purchase a product, particularly when, in the face of all the hard scientific evidence, there is no proof that the product is worthwhile other than on a purely subjective basis. And more to the point under discussion that a high resolution download sound superior to the CD without conducting proper comparative listening tests.
  
 Now of course the simple way around this is for you, Jude or anyone else on this site who makes such a recommendation to qualify that recommendation by clearly stating that the recommendation is based purely on subjective listening and that the recommendation runs counter to evidenced based objective reasoning. That I could live with 
	

	
	
		
		

		
		
	


	




 (remember that is the "Sound Science" section.)


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## fiascogarcia

I'm older, and I'm certain my hearing doesn't come close to hearing the upper and lower ends of the spectrum.  Yet I still feel that I notice a difference in the dynamics of many of the high def recordings.  If I have deluded myself into believing this, so be it.  I have posted many times the concept that "perception is reality", and that you cannot separate the psychology from the physics of how humans hear and perceive music.  You do not educate people if they only hear one side of an argument, you indoctrinate them.  I'm quite comfortable that most Head-fiers are intelligent enough to draw their own personal conclusions.


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## Steve Eddy

ralphp@optonline said:


> Now of course the simple way around this is for you, Jude or anyone else on this site who makes such a recommendation to qualify that recommendation by clearly stating that the recommendation is based purely on subjective listening and that the recommendation runs counter to evidenced based objective reasoning. That I could live with  (*remember that is the "Sound Science" section*.)




Except Jude isn't posting the reviews/videos/recommendations you're speaking of here in the Sound Science forum. Those are posted in the DBT Free Zone forums.

se


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## ralphp@optonline

steve eddy said:


> Except Jude isn't posting the reviews/videos/recommendations you're speaking of here in the Sound Science forum. Those are posted in the DBT Free Zone forums.
> 
> se


 

 Fair enough. Plus that explains my confusion since that is one section of the the forum that I avoid at all costs


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## Steve Eddy

ralphp@optonline said:


> Fair enough. Plus that explains my confusion since that is one section of the the forum that I avoid at all costs






se


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## cjl

silverears said:


> Well, when I A/B'd my DACs, I found one that my ears seems to favor, and sounded transparent.  I wonder what it could be, maybe the DAC's analog stage is done better than the others?


 

 It's entirely possible that both DACs are perfect to better than the limits of human perception, but one is outputting the signal at a slightly different level than the other (and very slight level differences are often perceived as quality differences). It's also possible (if you didn't do the test blind) that the differences are entirely psychological. That's why it's important when testing equipment by ear (if you want the most accurate results) to carefully level match with a digital multimeter (preferably to 0.1 dB or better) and to do the test blind. If you did both of these, and still heard a significant difference, then I agree with bigshot - one of the dacs was very poorly designed or defective.


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## keanex

bigshot said:


> Just knock it all down to AAC 320 and save a load of disk space.


 
 Mp3 LAME V0, save more space!


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## ab initio

ralphp@optonline said:


> I believe that issue isn't whether or not you, Jude or anyone else on this site should be able to "talk" or "discuss" a given product on the site but rather whether you, Jude or anyone else on this site should be recommending fellow members purchase a product, particularly when, in the face of all the hard scientific evidence there is no proof that the product is worthwhile other than on a purely subjective basis. And more to the point under discussion that a high resolution download sound superior to the CD without conducting proper comparative listening tests.
> 
> Now of course the simple way around this is for you, Jude or anyone else on this site who makes such a recommendation to qualify that recommendation by clearly stating that the recommendation is based purely on subjective listening and that the recommendation runs counter to evidenced based objective reasoning. That I could live with  (remember that is the "Sound Science" section.)




There are a couple points here:
-1- the most recent jude post that i think you are referencing is for the chesky binaural demo. This album is a collection of test tracks for testing your headphone audio equipment! Having the full final mix in all of its 24/192 overkill is exactly what you need to demonstrate whether or not hires formats are worthwhile downloads. a demo cd is the ideal candidate for one to purchase to test whether hires formats are audibly different.

-2- chesky is about the only studio i can think of that is releasing albums specifically catering to hifi enthusiasts using headphones. Whether or not you like the music styles or think that the 24/192 downloads are sonically superior is irrelevant. I am thankful that someone is producing headphone-centric music and I show support by purchasing the the demo album. I hope chesky continues releasing binaural recordings, i just hope that at some point he venutres outside the current box of genres offered and gets a little bit of vareity. Can't he get Rush to record a prog album for headphones? Anyways, the point here is that jude was encourging headfiers to support the production of quality mastered music specifically for headphones. While his suggestion to purchase the hires version isnt scientifically informed, it is economically informed--if headfiers dont buy binaural music, no one is going to make it.

-3- you neglect the possiblilty that jude is making the recommendation to buy hires in good faith. He simply might not have parsed through the scientific literature. Most people are not scientificly trained, and dont understand the difference between good science and good marketing. Furthermore, i bet most audiophiles want good sound, but dont want to spend the effort to understand the physics of good sound. I think a lot of audiophiles neglect science because it's too darn confusing to understand, especially with all the marketing noise out there designed to appear scientific to the naive. Before, you go on a crusade against every individual who doesnt understand the science aspect to audio systems, keep in mind that most folks simply cannot devote the time and mental capacity to study up on the subject. You have to patiently and persistently explain the science to the niave, because if you come off aggressive and condecending, you will be tuned out and everyone will be buying $10,000 power cables.

Cheers


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## ralphp@optonline

ab initio said:


> There are a couple points here:
> -1- the most recent jude post that i think you are referencing is for the chesky binaural demo. This album is a collection of test tracks for testing your headphone audio equipment! Having the full final mix in all of its 24/192 overkill is exactly what you need to demonstrate whether or not hires formats are worthwhile downloads. a demo cd is the ideal candidate for one to purchase to test whether hires formats are audibly different.
> 
> -2- chesky is about the only studio i can think of that is releasing albums specifically catering to hifi enthusiasts using headphones. Whether or not you like the music styles or think that the 24/192 downloads are sonically superior is irrelevant. I am thankful that someone is producing headphone-centric music and I show support by purchasing the the demo album. I hope chesky continues releasing binaural recordings, i just hope that at some point he venutres outside the current box of genres offered and gets a little bit of vareity. Can't he get Rush to record a prog album for headphones? Anyways, the point here is that jude was encourging headfiers to support the production of quality mastered music specifically for headphones. While his suggestion to purchase the hires version isnt scientifically informed, it is economically informed--if headfiers dont buy binaural music, no one is going to make it.
> ...


 

 A few points:
  
 1) Did you the read the other responses to my post you quoted and my follow-up responses?
  
 2) Bowers & Wilkins Society of Sound has issued several binaural recordings catering to headphone enthusiasts.
  
 3) Since the introduction of digital audio almost the entire high end audio world has slowly become anti-science because science can be used to prove that many of their cash cows, such as expensive digital audio cables or high resolution recordings, are simply a waste of money. Therefore we, the buying public, are being told again and again and again that the science doesn't matter or that the science is somehow incomplete or misleading. Also being anti-science puts one the position of being a good friend to the high end audio industry, e.g. review the history of the Computer Audiophile site for a clear example of this concept in action - less science means more buddies in the industry and more toys to play with.
  
 4) Just as there are several sections on the this forum that gleefully toss science aside (e.g. Cables, Power, Tweaks, Speakers, Accessories (DBT-Free Forum) and High-end Audio Forum), there are sections like this one (Sound Science) which embrace science and serve as a counter-balance to all the anti-science filling other parts the forum.
  
 5) If anything I did had the power to make anyone buy $10,000 power cables I would become a cable manufacturer in a heartbeat 
	

	
	
		
		

		
			





 since I would love to have my very own yacht.


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## ferday

ralphp@optonline said:


> 3) Since the introduction of digital audio almost the entire high end audio world has slowly become anti-science because science can be used to prove that many of their cash cows, such as expensive digital audio cables or high resolution recordings, are simply a waste of money. Therefore we, the buying public, are being told again and again and again that the science doesn't matter or that the science is somehow incomplete or misleading. Also being anti-science puts one the position of being a good friend to the high end audio industry, e.g. review the history of the Computer Audiophile site for a clear example of this concept in action - less science means more buddies in the industry and more toys to play with.


 
  
 it's nothing to do with digital, at all.  the high end audio world has always been anti-science.


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## bigshot

it depends on your definition of high end audio. Back when I started out in the 1970s, there was no such thing as audiophiles. We had hifi nuts back then. The approach was definitely scientific, because lots of hifi nuts built their amps from kits. The only luxury item status was the finish on the wood on the speaker cabinets.

I think the beginning of audiophoolery did start with digital audio. There were all kinds of crazy theories on why digital audio was lousy, none of which were based in science. I see lots of impressions from people that indicate that they are still thinking of digital audio the way they did about analog... generation loss, veils, etc. Back then, those things existed. Today they don't.


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## Steve Eddy

bigshot said:


> I think the beginning of audiophoolery did start with digital audio.




It started going off the rails a bit before that, namely with cables.

se


----------



## kraken2109

ab initio said:


> -3- you neglect the possiblilty that jude is making the recommendation to buy hires in good faith. He simply might not have parsed through the scientific literature. Most people are not scientificly trained, and dont understand the difference between good science and good marketing. Furthermore, i bet most audiophiles want good sound, but dont want to spend the effort to understand the physics of good sound. I think a lot of audiophiles neglect science because it's too darn confusing to understand, especially with all the marketing noise out there designed to appear scientific to the naive. Before, you go on a crusade against every individual who doesnt understand the science aspect to audio systems, keep in mind that most folks simply cannot devote the time and mental capacity to study up on the subject. You have to patiently and persistently explain the science to the niave, because if you come off aggressive and condecending, you will be tuned out and everyone will be buying $10,000 power cables.
> 
> Cheers


 
  
 My main point was that Jude is the 'frontman' of an audio forum and he shouldn't be spreading misinformation so a sponsor can make money. If it is true that he doesn't know anything about digital audio, it worries me that he is considered suitable for the job. I mean this in no disrespect to him.
 Point 3 doesn't really work seeing as this is an enthusiast forum. The focus should be on knowledge.
  


currawong said:


> So if Jude (or I) like something that happens to be made by someone who sponsors the site, he shouldn't talk about it?   That's basically what you're saying.
> 
> I like the music, personally, and appreciate the effort to make better recordings, since I have more expensive than average gear for a member here. The nice thing here is that we can discuss these topics freely and put forth out opinions (as long as we aren't being rude).


 
 You seem to have missed my point. I have no problem with a video saying 'hey look at this cool album guys'. My problem is when viewers are told to spend extra money on something pointless, which is the case in that video. This forum should be for education, not for the spread of misinformation and marketing snakeoil.
 Sure, I am biased against companies like HDTracks and you might not be. But I feel my dislike for them is justified since it seems to me like they're cheating people out of their money through lies.


----------



## ab initio

ralphp@optonline said:


> A few points:
> 
> 1) Did you the read the other responses to my post you quoted and my follow-up responses?
> *I fail to see the relevance here.*
> ...


 
 Cheers


----------



## bigshot

steve eddy said:


> It started going off the rails a bit before that, namely with cables.
> 
> se




Perhaps the introduction of hard sell commissioned salespeople at places like Pacific Stereo might be ground zero.

The anti-science bias in the other forums extends far beyond just double blind testing. When the split happened, that was the definition, but it's since evolved beyond that.


----------



## ferday

steve eddy said:


> It started going off the rails a bit before that, namely with cables.
> 
> se


 
  
 although i wasn't there (and hopefully none of you), it seems to me it "started" when the at-the-time inferior "discs" were trying to replace the audibly superior wax cylinders (1900's)...the format wars have never ended since then.  loudspeaker silliness has been around since the 40's at least.  it's all semantics anyways LOL but i enjoy the history stuff...there are a few fun to read (if not entirely accurate) history of hi-fi articles floating around the web
  
 i've been heavily into mountain biking for 25 years.  it's pretty easy to say 'back then we were just a bunch of bike nuts" but truth be told i was just a gear hound with less money to spend on gear, so it was substituted with passion. nostalgia tends to have a nice rosy hue doesn't it


----------



## ab initio

kraken2109 said:


> My main point was that Jude is the 'frontman' of an audio forum and he shouldn't be spreading misinformation so a sponsor can make money. If it is true that he doesn't know anything about digital audio, it worries me that he is considered suitable for the job. I mean this in no disrespect to him.
> Point 3 doesn't really work seeing as this is an enthusiast forum. The focus should be on knowledge.


 

 Well, like I said, he could be acting in good faith, naively spreading misinformation. But, as the the head of head-fi, I think it is his duty to back anyone who is providing a fairly unique service that caters directly to headphone users, which Chesky's binaural recordings do. Whether or not Jude is aware that scientific evidence suggests that 24/192 has no audible benefit over redbook CD for home listening, I do not know. But, since this is an internet forum for the general public and not a scientific journal, there is no reason to assume that anybody here has any grasp for the underlying physics of audio recording, reproduction, and hearing. If he was "Dr Jude" I might hold him to a higher standard, but for all I know, he's some dude who loves HiFi music and headphones. You shouldn't expect everybody discussing on these forums to have engineering degrees in electrical and mechanical engineering. People are here because they want to talk about music, their expensive HiFi equipment, and for MegaBuck system wankfests. I think the people who arrive at head-fi looking for answers are more likely lurking or dumped here after googling, rather than actively participating. Of course, there are exceptions.
  
 Cheers


----------



## ab initio

bigshot said:


> The anti-science bias in the other forums extends far beyond just double blind testing. When the split happened, that was the definition, but it's since evolved beyond that.


 
  
 True, but the rules don't prevent me from going into other forums and providing well-reasoned suggestions to those asking questions which start from false premises (e.g., here). You can bark math and numbers and provide proofs and textbook references all you want as long as you don't mutter ABX anywhere. The thick-skulled zealot might not like it, but it might help the casual googler that stumbles across the thread looking for answers. Who knows how many lurkers might benefit? I lurked for years before I signed up for a headfi account. There were some long and nasty discussions once-upon-a-time (namely, before I joined) regarding audibility of ps jitter, and I lurked and learned from the well reasoned arguments and filtered out the BS.  
  
 Cheers


----------



## bigshot

Well if I had to point to one difference between when I started out in hifi and today, I would have to say the difference involves the DIY attitude. In the 60s and 70s, if you wanted a really good sound system, you built a nice kit amp from Heathkit, you got plans for speaker cabinets and built them in your garage, and you read magazines to get circuit diagrams and to find parts suppliers. Now, high end audio is more like a religion. You go to a fancy store or website where a counsellor advises you to have faith in things you don't fully understand. All you have to do is peel back the long green and make a donation and hopefully the sound god smiles upon you. If not, pull out the wallet and make another donation.


----------



## Steve Eddy

kraken2109 said:


> My main point was that Jude is the 'frontman' of an audio forum and he shouldn't be spreading misinformation so a sponsor can make money.




What's the misinformation? That's all on the subjective side of the street. If someone says something sounds better to them, what's to argue? 

se


----------



## kraken2109

steve eddy said:


> What's the misinformation? That's all on the subjective side of the street. If someone says something sounds better to them, what's to argue?
> 
> se


 
 The misinformation is claiming something sounds better when it is physically impossible for said thing to sound better.
 If I told you I could hear sound at 30kHz, that would be misinformation, surely? Or would you just decide that's subjective?


----------



## Steve Eddy

kraken2109 said:


> The misinformation is claiming something sounds better when it is physically impossible for said thing to sound better.




"Sounds better" is purely subjective. Our subjective experiences are what they are regardless of what may be behind them and aren't necessarily an accurate reflection of the physical reality. In other words, we may subjectively perceive differences even when there are no actual audible differences. So how can you possibly say that it's physically impossible for something to "sound better" to someone? 

se


----------



## kraken2109

Rewording my post since it was misunderstood and deleted.
  
 I _personally_ believe that 'high definition audio' and all those who sell it claiming it is audibly different is, like many things in audio, simply a scam. I would loved to be proved wrong with some actual evidence, but at the moment all the evidence supports this view. I apologise if this upsets anyone, but this board is sound science and I know I'm not alone in this belief.


----------



## fiascogarcia

I'm wondering if the differences exist in my DAC.  Is it possible that a DAC  designed for 44.1 playback will sound just as good with a higher format sampling, but the reverse is not true?  Maybe a good processor designed for 44.1 will sound exactly like a converter built for higher formats?  Why not more discussion regarding the build and development of DAC processors?  Am I in the wrong forum?  I'm not an engineer, just wondering out loud.


----------



## kraken2109

fiascogarcia said:


> I'm wondering if the differences exist in my DAC.  Is it possible that a DAC  designed for 44.1 playback will sound just as good with a higher format sampling, but the reverse is not true?  Maybe a good processor designed for 44.1 will sound exactly like a converter built for higher formats?  Why not more discussion regarding the build and development of DAC processors?  Am I in the wrong forum?  I'm not an engineer, just wondering out loud.


 
 I'm not a DAC engineer either but I think this is a good route for the discussion to take.


----------



## Thad-E-Ginathom

kraken2109 said:


> Rewording my post since it was misunderstood and deleted.
> 
> I _personally_ believe that 'high definition audio' and all those who sell it claiming it is audibly different is, like many things in audio, simply a scam. I would loved to be proved wrong with some actual evidence, but at the moment all the evidence supports this view. I apologise if this upsets anyone, but this board is sound science and I know I'm not alone in this belief.


 
  
 I _personally _believe (If I might borrow your formula?) that there is _no such thing_ as "high definition," _or_ "High Resolution" audio.
  
 These are not technical terms. There is sample rate, and there is bit depth. The rest is invented by a collusion of marketing departments and tongues-out "audiophiles" that luuuurve those words.
  
 More: _High-bit-rate_ is a spin too. It is intended to make people think of their 44/48 as "low." It is not low. _Low_-bit-rate might have been there in the telephone industry, decades ago. High_er _might be accurate, even 44.2 is higher thatn 44.1, but, hey, it won't achieve the double marketing wammy of being saleable _and_ making people discontent with what they have.
  
 Does that mean that 16-bit, 44.1khz, is the bee's-knees-ultimate in music reproduction and should be set in stone as perfect for ever and cancel any other research? No, absolutely not. One day, PCM and DSD might seem as old-fashioned as a shellac 78rpm record does today. Who knows what will be discovered and developed in the future!
  
 Further more, when I listen to audio scientists (some of whom love music too) they tell that, no, 44.1/16 is _not_ ultimate. Sure it covers the entire audio frequency range, and just about all the entire dynamic range from _fff_ to _ppp,_ but there is more to it than that. There is the implementation of the filtering and reconstruction circuits. 48 is better, and some say 60 would have been ideal. 60, of course, is one standard that no-one gives us! 
  
 I'd like to be able to say, _my understanding is _and mean it... but I'm a maths dunce, formulae make my eyes spin, and this _is_ deeply mathematical stuff, so don't ask me to explain or justify this. The best I could do is point to the contributions of some people on other forums.
  
 Anyway, in the loose meaning of the phrase, _my understanding is... _that 48.1 is just fine but not perfect, 48 is better, 60 would be good (but we are not offered it anyway), but the reasoning is nothing to do with the usual misunderstanding of _sample_ rates, and just climbing the ladder of bigger numbers is senseless, except to the sales departments.


----------



## kraken2109

Quote:


thad-e-ginathom said:


> I _personally _believe (If I might borrow your formula?) that there is _no such thing_ as "high definition," _or_ "High Resolution" audio.
> 
> These are not technical terms. There is sample rate, and there is bit depth. The rest is invented by a collusion of marketing departments and tongues-out "audiophiles" that luuuurve those words.
> 
> More: _High-bit-rate_ is a spin too. It is intended to make people think of their 44/48 as "low." It is not low. _Low_-bit-rate might have been there in the telephone industry, decades ago. High_er _might be accurate, even 44.2 is higher that 44.1, but, hey, it won't achieve the double marketing wammy of being saleable _and_ making people discontent with what they have.


 
 Yes, I agree. I dislike marketing terms but using them was the easiest way of making my point.
  
    
 Quote:


thad-e-ginathom said:


> Does that mean that 16-bit, 44.1khz, is the bee's-knees-ultimate in music reproduction and should be set in stone as perfect for ever and cancel any other research? No, absolutely not. One day, PCM and DSD might seem as old-fashioned as a shellac 78rpm record does today. Who knows what will be discovered and developed in the future!
> 
> Further more, when I listen to audio scientists (some of whom love music too) they tell that, no, 44.1/16 is _not_ ultimate. Sure it covers the entire audio frequency range, and just about all the entire dynamic range from _fff_ to _ppp,_ but there is more to it than that. There is the implementation of the filtering and reconstruction circuits. 48 is better, and some say 60 would have been ideal. 60, of course, is one standard that no-one gives us!
> 
> ...


 

  
 I'd like some sources for this because I haven't heard any good arguments for 44.1/16 not being perfect.
 (This is not me calling you wrong, a liar, or insulting you in any way. *This is me attempting to learn more.* I felt the need to put this because tone is often hard to read online).


----------



## bigshot

Redbook was designed to be perfect. It should be perfect sound. If it isn't, something is wrong somewhere.

There are areas for improvement, but they are outside of the realm of 2 channel audio, and some of them don't involve fidelity, but rather euphonics.


----------



## cjl

There are good technical reasons why 16/48 or even 16/60 or so would be better in some ways, but they have nothing to do with the ability of the format to capture the audible spectrum. Rather, the slightly higher sampling rate allows for the use of an anti-aliasing filter that has a much shallower rolloff without cutting off anything in the audible frequency range, which is easier to design. With modern electronics, a filter that is audibly perfect in the passband up to 20kHz but is 100+dB down at the nyquist frequency (22.05kHz for 16/44.1) is doable, and that eliminates most of the reason behind wanting the slightly higher sample rate.


----------



## Thad-E-Ginathom

kraken2109 said:


> I'd like some sources for this because I haven't heard any good arguments for 44.1/16 not being perfect.
> (This is not me calling you wrong, a liar, or insulting you in any way. *This is me attempting to learn more.* I felt the need to put this because tone is often hard to read online).


 
  
 Probably, I have to plead guilty to hearsay on that one --- and perhaps even misinterpretation. Any correction will be a welcome update to my understanding. I suppose I am thinking that if 48, 60, mean easier and ?better? filtering, then, by assumption that that produces a ?better? result, and if there is such a thing as a better result, 44.1 can't have been perfect in the first place. 
  
 I have been trying to learn, with the linked article in the opening post being something of a wake up call, and the videos explaining stuff (like the stair-step misconception) that I had _never_ even begun to understand before.
  
 My loosely-called _understanding _is definitely a work in progress, despite a certain personal tendency to _talk_ as if it is set.
  
 The "marketing" stuff, by the way, was certainly not aimed at you. You just gave me the conversation jumping-in point, with the nicely-put "I believe..." These high-def/res labels are all over the place now: we'll all start talking like that soon. Until then, it's a regular rant of mine.


----------



## kraken2109

thad-e-ginathom said:


> Probably, I have to plead guilty to hearsay on that one --- and perhaps even misinterpretation. Any correction will be a welcome update to my understanding. I suppose I am thinking that if 48, 60, mean easier and ?better? filtering, then, by assumption that that produces a ?better? result, and if there is such a thing as a better result, 44.1 can't have been perfect in the first place.
> 
> I have been trying to learn, with the linked article in the opening post being something of a wake up call, and the videos explaining stuff (like the stair-step misconception) that I had _never_ even begun to understand before.
> 
> ...


 

 My understanding of sampling is that 44.1kHz is more than enough. In terms of filtering (e.g. anti-aliasing filtering) I've read that pretty much every modern DAC will oversample to get steeper filters, so there should be no need for a higher sampling rate.


----------



## bigshot

Yeah. It's not accurate to say that higher sampling rates are necessary to sound quality because they aren't.


----------



## Thad-E-Ginathom

There's a post by j_j on Gearslutz, about sample rates and the filtering slopes, in _their_ going-on-for-ever pono/sampling-rates thread. The trouble is, finding it amongst the thousands of others.
	

	
	
		
		

		
			





 ...


----------



## RazorJack

tl;dr version: DBT the benefits of 24 bit 192 kHz over CD quality, or stop fooling yourself.
  


ab initio said:


> As for the vinyl bit... yeah, folks like to record their vinyl records at absurd bitdepths and sampling rates because they like to support the harddisk industry.


 
  





 lol


----------



## Duncan

I think that some of the talk of hi-res being better is due to poor marketing / implementation of the original redbook standard...

The first CD players were based on the Philips TDA1540 DAC which was only 14 bit (still technically good enough, but - we're in it for the numbers! ), which, it was argued was why CD sounded so bright in the early days (ignoring brick wall filters, people didn't want to know or understand about them!)...

Anyhow, I'd rather have a well implemented 16 bit setup than a poorly implemented 24 bit one...


----------



## Steve Eddy

thad-e-ginathom said:


> There's a post by j_j on Gearslutz, about sample rates and the filtering slopes, in _their_ going-on-for-ever pono/sampling-rates thread. The trouble is, finding it amongst the thousands of others. ...




When you say j_j, do you mean jj as in James Johnston formerly of Bell Labs?

EDIT: Never mind, found him. It is ol' jj.

se


----------



## emailists

Andreas Koch who was involved in the creation of the standards for SACD states 
  
 that studies have shown humans can perceive frequencies up to 100k, however at a much lower amplitude
  
 (it may have been dynamic range as well)
  
 http://www.youtube.com/watch?feature=player_embedded&v=nj7d7Jnx0xc
  
 it's about 24 minutes in.   The entire panel  on the video consists of luminary electrical engineers,
  
 recording engineers, mastering people, etc.
  
 If anyone walked up to one of the people in this RMAF panel that designed or use these high res professional tools
  
  to record or transfer master tapes every day,  and stated  
  
 " 24 bit audio and anything over 48k is not only worthless, but bad for music."
  
 I can only imagine what their reaction would be.


----------



## Noah99

bigshot said:


> Just as a bit of interesting trivia... 44.1K covers the full spectrum of frequencies that humans can hear- 20Hz to 20kHz, with a bit to spare. Higher sampling rates extend the frequency response higher, far beyond our ability to hear, but the core frequencies below 20kHz are rendered exactly the same at 44.1 as they are at 192. So whatever it is that you seem to think is clearly audible isn't audible with human ears. Perhaps a bat!
> 
> However, it is possible that your equipment isn't designed to deal with super high frequencies and is adding distortion down in the audible range. So if you are positive you are hearing a difference, it is almost certainly noise, not music.


 
 I'm sorry to say but I just recently visited my audiologist and I was told by three doctors that the adult human ear can only hear up to about 8,000Hertz.
 Infants are able to hear from 20hz to 18khz but as we become older our frequencies range drops.
 Don't bother testing your ear from online downloaded frequencies because they will not turn out the same as a professional  Audiologist' equipment. in an acoustic room


----------



## Xenophon

Connect your flamethrowers, I don't care a whit but imo some of you people are overdoing it a bit.  You're entitled to your opinion and sure, most of it will be based upon science -so was an opinion about medicine dating back to the 15th century which I just read, btw.  The 'science' honchos of that time are now generally considered as idiots wearing pointed hats while they were just reasoning based on what was available in terms of 'knowledge' at that time- but the attitude and lack of tolerance of some toward 'non believers' would do the taliban proud.
  
 I'm sure this site can be criticised, yet although the owners are under no obligation whatsoever to provide you with any kind of platform, you have the liberty to express your views.  If nothing else, that deserves some respect imo.


----------



## cjl

noah99 said:


> I'm sorry to say but I just recently visited my audiologist and I was told by three doctors that the adult human ear can only hear up to about 8,000Hertz.
> Infants are able to hear from 20hz to 18khz but as we become older our frequencies range drops.
> Don't bother testing your ear from online downloaded frequencies because they will not turn out the same as a professional  Audiologist' equipment. in an acoustic room


 
 Most consumer grade gear can reproduce 10-15kHz just fine, and most good quality consumer gear can reproduce up to 20kHz just fine. Your audiologist was wrong, except for people who have damaged their hearing. I have tested myself up to about 19kHz just fine (I used to be able to hear up to 20kHz pretty clearly in high school, but now it's down to somewhere between 18.5 and 19kHz).


----------



## Currawong

emailists said:


> Andreas Koch who was involved in the creation of the standards for SACD states
> 
> that studies have shown humans can perceive frequencies up to 100k, however at a much lower amplitude
> 
> ...


 
  
 He also says that DSD is still good out to 100 kHz because it still has a noise level of -30dB of noise above 50 kHz (whereas PCM has a hard cut-off).  Try overlaying that much noise over regular music! It'd be unlistenable! 
  
 I think though that points out where "science" has its limitations: Not every scientist agrees completely on everything -- it isn't an arbiter of absolute truth by any stretch of the imagination.


----------



## esldude

emailists said:


> Andreas Koch who was involved in the creation of the standards for SACD states
> 
> that studies have shown humans can perceive frequencies up to 100k, however at a much lower amplitude
> 
> ...


 

 Well, Mr. Koch sounds like he is referring to Oohashi.  Which is of course not been replicated, and some others have shown his test subjects were hearing IM products not direct 100 khz sounds.   Mr. Koch also promotes another myth, that the human ear can hear above 20 khz for transient information.  Now that would be a very curious transducer indeed.  It would have a far higher bandwidth only for transients than it does for steady state.  I don't know of other transducers that work that way.  Speaker cones, microphone diaphragms etc. don't.  The ear starts with the movement of the ear drum.  The ear drum doesn't move at those frequencies.  Most transducers can have their transient response deduced from the max frequency steady state response as it is more or less the same.  Some have a bit lower transient response than max steady state frequency response.

 But in the audiophile world we have the magic transducer with 5 times the transient response of its steady state response.  And it does this with reduced dynamic range.  And the ear drum doesn't move at these rates except for the brief transient after which is apparently does not respond at all.  Quite the curious transducer from the standpoint of the physics involved.  Further while the upper steady state response of the ear drum has a very high threshold it apparently responds freely to these high frequency transients that are at very low absolute sound levels.  Even more curious.  Steady state, no response and a high threshold for what it will respond to on the upper end with super quick response to quite small level signals 5 times that high.  Wow!


----------



## BeyerMonster

kraken2109 said:


> I _personally_ believe that 'high definition audio' and all those who sell it claiming it is audibly different is, like many things in audio, simply a scam.


 
 Out of curiosity, do you have the same attitude towards vendors selling 256 kbps or even 128 kbps MP3s as CD-quality or "near CD-quality"?
  
 In the case of lossless high-res audio case, you can at least argue that their production and distribution costs are higher (storage/bandwidth).
 In the lossless cases, they're actually saving money via lower costs/bandwidth.
  
 Arguments for/against audible differences can be applied in either scenario.


----------



## BeyerMonster

cjl said:


> noah99 said:
> 
> 
> > I'm sorry to say but I just recently visited my audiologist and I was told by three doctors that the adult human ear can only hear up to about 8,000Hertz.
> ...


 
 It's entirely possible that his audiologist was right, but he misheard the doc.


----------



## Thad-E-Ginathom

steve eddy said:


> When you say j_j, do you mean jj as in James Johnston formerly of Bell Labs?
> 
> EDIT: Never mind, found him. It is ol' jj.
> 
> se


 
  
 Yes, of course being _a scientist_ doesn't make a person infallible, but  if I'm to take anybody's word, then the CV is certainly impressive. As is the capacity to patiently explain and put into layman, even non-mathematical, terms. Read some of his stuff on Hydrogen Audio too.
  
 Monty, of Xiph, still gets the credit for opening my eyes, though.
  
 Hey, I'm over sixty years old: why did nobody tell me all this stuff long, long ago! 
	

	
	
		
		

		
		
	


	




. Oh well, better late than never!


beyermonster said:


> Out of curiosity, do you have the same attitude towards vendors selling 256 kbps or even 128 kbps MP3s as CD-quality or "near CD-quality"?


 
   
 I hadn't really thought about that one. I don't really buy MP3, or any lossy audio, if I can help it,so the question hasn't arisen. Off-hand, no. It isn't such a fabrication as the high-res marketing.
  
 Quote:


> In the case of lossless high-res audio case, you can at least argue that their production and distribution costs are higher (storage/bandwidth).
> In the lossless cases, they're actually saving money via lower costs/bandwidth.


 
  
 When I am buying music, I would buy FLAC rather than MP3. I don't object to the higher cost for the reasons you state. 


> Arguments for/against audible differences can be applied in either scenario.


 
 I believe that there are many reasons why two sound samples _could_ sound different on the same system, before even getting into the statements of pure faith which inspire so many to become "high res" evangelists.
  
 In the first place, we have the people who heard some high-bit-rate music and declared it marvellous, without listening to a 44.1 copy of the same thing. In the second, we people who declare it marvellous because on one DAC in one system, they detected a difference. Many of these people will decry any sort of blind testing, formal or informal too, but even if the second group genuinely did hear something, it is no basis for a universal conclusion.
  
 Me? Old[ish] and with diminishing hearing that rolls off from 8K, so if someone points me at a tiny, tiny variation in the decay of some cymbal shimmer (assuming that it did exist), well, I probably didn't get most of that shimmer anyway.
  
  
 I can still talk, though!


----------



## Soundsgoodtome

Are there such equipment through the same receiving method that process bit depth differently? Example 16 vs 24 bit utilizing a different circuit or method of decoding resulting in better sound? 

Also what about bit depth setting differences? Playing 16-bit files while DAC is set to run on 24-bit via Windows setting or vice versa? Is there quality loss when DAC settings is set at 24bit and 16bit files are playing? 

This isn't counting WASAPI or ASIO taking control of device and changing the setting automatically to match. And not to be mistaken by sampling which does create quantization errors when going from one bit depth to another.


----------



## Duncan

soundsgoodtome said:


> Are there such equipment through the same receiving method that process bit depth differently? Example 16 vs 24 bit utilizing a different circuit or method of decoding resulting in better sound?
> 
> Also what about bit depth setting differences? Playing 16-bit files while DAC is set to run on 24-bit via Windows setting or vice versa? Is there quality loss when DAC settings is set at 24bit and 16bit files are playing?
> 
> This isn't counting WASAPI or ASIO taking control of device and changing the setting automatically to match. And not to be mistaken by sampling which does create quantization errors when going from one bit depth to another.


Yes,

The Micromega MyDac (to name one from _personal_ experience) uses different clocks depending on if receiving [a multiple of] either 44.1 or 48khz sample rates...


----------



## robertsong

Wow, this "Monty" character is an enormous crackpot. Whether he truly believes the stuff that wrote in that joke of an article, or whether he an agenda to pass on such propaganda is impossible to tell. He does not state any credentials what-so-ever, nor does he disclose his full name. Hmmm.
  
 My guess is this is somebody with nothing more than a computer programming background. No actual audio engineering experience. Am I right?
  
 PS: He his DEAD WRONG.


----------



## robertsong

keanex said:


> Mp3 LAME V0, save more space!


 
  
  
 That sounds good to you, eh? Cheers!


----------



## Steve Eddy

robertsong said:


> Wow, this "Monty" character is an enormous crackpot. Whether he truly believes the stuff that wrote in that joke of an article, or whether he an agenda to pass on such propaganda is impossible to tell. He does not state any credentials what-so-ever, nor does he disclose his full name. Hmmm.
> 
> My guess is this is somebody with nothing more than a computer programming background. No actual audio engineering experience. Am I right?
> 
> PS: He his DEAD WRONG.




Um, would you happen to have any sort of support for this empty-handed claim of yours? Or can I just come back and say "You are DEAD WRONG" and trump your "He is DEAD WRONG"?

se


----------



## Steve Eddy

esldude said:


> Well, Mr. Koch sounds like he is referring to Oohashi.  Which is of course not been replicated, and some others have shown his test subjects were hearing IM products not direct 100 khz sounds.




Beat me to it. Thanks.

se


----------



## elmoe

robertsong said:


> Wow, this "Monty" character is an enormous crackpot. Whether he truly believes the stuff that wrote in that joke of an article, or whether he an agenda to pass on such propaganda is impossible to tell. He does not state any credentials what-so-ever, nor does he disclose his full name. Hmmm.
> 
> My guess is this is somebody with nothing more than a computer programming background. No actual audio engineering experience. Am I right?
> 
> PS: He his DEAD WRONG.


 
  
 How about you offer some argument then? What are your credentials? Hmmm.


----------



## Steve Eddy

thad-e-ginathom said:


> Yes, of course being _a scientist_ doesn't make a person infallible, but  if I'm to take anybody's word, then the CV is certainly impressive. As is the capacity to patiently explain and put into layman, even non-mathematical, terms. Read some of his stuff on Hydrogen Audio too.




Don't get me wrong. I wasn't being critical of him. I've known him for years and just haven't seen much of him lately. 

se


----------



## Steve Eddy

elmoe said:


> How about you offer some argument then? What are your credentials? Hmmm.




Let's not forget that it's the argument that makes the argument, not the "credentials" of the person making the argument. Not even Einstein got away with that.

se


----------



## Thad-E-Ginathom

steve eddy said:


> Don't get me wrong. I wasn't being critical of him. I've known him for years and just haven't seen much of him lately.
> 
> se


 
  
 Not getting you wrong... speaking generally


----------



## Steve Eddy

thad-e-ginathom said:


> Not getting you wrong... speaking generally




Gotcha. 

se


----------



## Noah99

xenophon said:


> Connect your flamethrowers, I don't care a whit but imo some of you people are overdoing it a bit.  You're entitled to your opinion and sure, most of it will be based upon science -so was an opinion about medicine dating back to the 15th century which I just read, btw.  The 'science' honchos of that time are now generally considered as idiots wearing pointed hats while they were just reasoning based on what was available in terms of 'knowledge' at that time- but the attitude and lack of tolerance of some toward 'non believers' would do the taliban proud.
> 
> I'm sure this site can be criticised, yet although the owners are under no obligation whatsoever to provide you with any kind of platform, you have the liberty to express your views.  If nothing else, that deserves some respect


 
 Not to be rude, but I'm sure that no one else here has a Phd in Audiology. 
  
 I was also just stating a fact
  
 Plus after I was tested I was told I have a perfect ear and I just need to maintain it.
 I later asked them if I could be tested higher frequencies and it was not audible through my ear nor through hers.
 So next time, please visit a doctor who at least has a  masters, in stead of googling it or checking wikipedia.
  
 A first born can here frequencies 20hz to 18khz
 While adults can here only up to 8khz
 (I forget what the lower frequency was) 
  
 Also when you have a very wide bandwidth being played it will be audible to the ear because of the larger soundstage, and more of a 3D perspective.
  
 I hope this helps,
 Noah


----------



## robertsong

noah99 said:


> Not to be rude, but I'm sure that no one else here has a Phd in Audiology.
> 
> I was also just stating a fact
> 
> ...


 
  
  
 All one has to do is twiddle an eq to prove the 8khz limit wrong. But why would an audiologist's equipment produce different results from say online test? That seems a little odd. Did you ask him/her for the reason?


----------



## robertsong

I just tried the 12khz band on the graphic EQ in JRMC and swept it from 12db to -12db and the difference was not subtle.  I bet every single person in this thread could here that.  14khz on the other hand was much more subtle. 16khz? Forget about it.


----------



## robertsong

elmoe said:


> How about you offer some argument then?


 
  
 What I have to offer is nothing that has not been said in a 50 page thread by a poster who calls himself "gregorio", or out of one my books.
  
  
 I do my own listening, and believe that is how everybody should draw there own opinions conclusions. There is actually no "right" or "wrong" with the 192khz sample rate as the author implies. It's entirely a subjective thing and opinions vary.  What can't be denied is that there _is_ an audible difference between 24/192 and 16/44.1


----------



## elmoe

robertsong said:


> What I have to offer is nothing that has not been said in a 50 page thread by a poster who calls himself "gregorio", or out of one my books.
> 
> 
> I do my own listening, and believe that is how everybody should draw there own opinions conclusions. There is actually no "right" or "wrong" with the 192khz sample rate as the author implies. It's entirely a subjective thing and opinions vary.  What can't be denied is that there _is_ an audible difference between 24/192 and 16/44.1


 
  
 If the only way we'll get your opinion is through buying one of your books, you shouldn't post in this thread to start with.
  
 The difference can, and is denied by every DBT ABX test ever done by anyone. If your claim was anywhere near truthful then there would be countless DBT tests proving it, there isn't a single one. What can't be denied objectively is that there is in fact, absolutely NO DIFFERENCE between 24/192 and 16/44.1 files of the same recording/master to be heard by the human ear. If you'd like to deny this claim feel free to make your own DBT ABX test under the proper procedure and film it for us all to see. If it could be done it would've been a long time ago.


----------



## kraken2109

beyermonster said:


> Out of curiosity, do you have the same attitude towards vendors selling 256 kbps or even 128 kbps MP3s as CD-quality or "near CD-quality"?
> 
> In the case of lossless high-res audio case, you can at least argue that their production and distribution costs are higher (storage/bandwidth).
> In the lossless cases, they're actually saving money via lower costs/bandwidth.
> ...


 
 An mp3 is not 'near CD quality' so personally I would disagree with them - perhaps if they rephrased it to 'audibly close to CD quality' for a 320kbps mp3 I could understand, but just saying near CD quality is a bit ambiguous. Personally I struggle to tell the difference a lot of the time between a high bit rate mp3 (256+) and lossless, but in some cases I can and so can others so I have no problem with lossless audio at 44.1/16. Scientifically compare (e.g. spectrum analysis) an mp3 and lossless PCM and there is a large measurable difference within the audible range. This isn't the case with higher sampling frequencies.
  
 Sure, higher bandwidth costs more. My complaint is when it is claimed that it sounds better, or that there are problems with 44.1/16. As bigshot says, 44.1/16 was chosen for a reason.


----------



## Steve Eddy

robertsong said:


> What can't be denied is that there _is_ an audible difference between 24/192 and 16/44.1




Then demonstrate it with something more than hand-waving and empty claims.

se


----------



## Soundsgoodtome

It's as easy as doing an abx switching plug-in on Foobar, a capable DAC and decent phones.


----------



## Duncan

If we spin this around, and say that the ONLY reason that 24/xxx could be better than 16/xxx is because it offered more headroom / dynamic range, would that then make things easier for both sides of this to see where the other is coming from?

Personally, I'm happy to listen to 256kbps mp3, as much as I am a DSD file, can I genuinely tell the difference? in mastering, yes, quite possibly, but to these ears, only by virtue of better headroom on the higher res file...

Then again, for those producers that don't push everything deep into the red, redbook can still sound extraordinary.


----------



## esldude

robertsong said:


> Wow, this "Monty" character is an enormous crackpot. Whether he truly believes the stuff that wrote in that joke of an article, or whether he an agenda to pass on such propaganda is impossible to tell. He does not state any credentials what-so-ever, nor does he disclose his full name. Hmmm.
> 
> My guess is this is somebody with nothing more than a computer programming background. No actual audio engineering experience. Am I right?
> 
> PS: He his DEAD WRONG.


 

 Assuming we are talking about the same article he explains what the broad outlines of human hearing are.  None of that is controversial or unsupported.  His name is Christopher "Monty" Montgomery.  I believe he developed OGG encoding/decoding and works on H264 audio/video codecs.  I don't know if his background is mostly in programming or what.  Since his codecs work pretty well he knows something about what he is doing and talking about.
  
 As you appear to be very mistaken in saying he is dead wrong perhaps a more detailed critique by you is in order.


----------



## esldude

robertsong said:


> What I have to offer is nothing that has not been said in a 50 page thread by a poster who calls himself "gregorio", or out of one my books.
> 
> 
> I do my own listening, and believe that is how everybody should draw there own opinions conclusions. There is actually no "right" or "wrong" with the 192khz sample rate as the author implies. It's entirely a subjective thing and opinions vary.  What can't be denied is that there _is_ an audible difference between 24/192 and 16/44.1


 

 Then it appears Sir, you are in the wrong forum.  Everybody listening to form their own opinion with no right or wrong isn't how this particular sub-forum works.  If you have some credible evidence or explanation of how 192 vs 44 is audible now would be the time to enlighten us.


----------



## elmoe

duncan said:


> If we spin this around, and say that the ONLY reason that 24/xxx could be better than 16/xxx is because it offered more headroom / dynamic range, would that then make things easier for both sides of this to see where the other is coming from?
> 
> Personally, I'm happy to listen to 256kbps mp3, as much as I am a DSD file, can I genuinely tell the difference? in mastering, yes, quite possibly, but to these ears, only by virtue of better headroom on the higher res file...
> 
> Then again, for those producers that don't push everything deep into the red, redbook can still sound extraordinary.


 
  
 More headroom on hi-res files makes no sense though. Take a 192/24 file, downsample it to 44.1/16, the headroom will be exactly the same on both. Why would it even change? Is that coming from some assumption that because bit "depth" goes from 16 to 24, the "sound" also gains in "depth"??? lol...


----------



## Soundsgoodtome

Just to throw this in the mix, DSD files warrant 192 because of its sampling rate, correct?


----------



## kraken2109

duncan said:


> If we spin this around, and say that the ONLY reason that 24/xxx could be better than 16/xxx is because it offered more headroom / dynamic range, would that then make things easier for both sides of this to see where the other is coming from?
> 
> Personally, I'm happy to listen to 256kbps mp3, as much as I am a DSD file, can I genuinely tell the difference? in mastering, yes, quite possibly, but to these ears, only by virtue of better headroom on the higher res file...
> 
> Then again, for those producers that don't push everything deep into the red, redbook can still sound extraordinary.


 
 The only thing bit depth affects is signal to noise ratio, or the height of the noise floor. Higher than 16 can be useful in production when levels are low and need to raise them without bringing up noise to noticeable levels, but for playback there is no benefit in higher than 16. For a track to require more than 16 it would either have simply been mastered to an incredibly low level, or it would have to have more dynamic range than pretty much any speaker (or ears) can handle.


----------



## Duncan

elmoe said:


> More headroom on hi-res files makes no sense though. Take a 192/24 file, downsample it to 44.1/16, the headroom will be exactly the same on both. Why would it even change? Is that coming from some assumption that because bit "depth" goes from 16 to 24, the "sound" also gains in "depth"??? lol...


...did you not read the last line of your quote? For those producers that like to push the limits, there is more potential snr with 24 bit compared to 16, 144db vs 96db iirc...

_However_, that is immaterial if you don't push deep into the red...


----------



## Steve Eddy

And just to reiterate, the instantaneous dynamic range of our ears is about 60dB.

se


----------



## elmoe

duncan said:


> ...did you not read the last line of your quote? For those producers that like to push the limits, there is more potential snr with 24 bit compared to 16, 144db vs 96db iirc...
> 
> _However_, that is immaterial if you don't push deep into the red...


 
  


steve eddy said:


> And just to reiterate, the instantaneous dynamic range of our ears is about 60dB.
> 
> se


 
  
 ^ Enough said...


----------



## ralphp@optonline

duncan said:


> If we spin this around, and say that the ONLY reason that 24/xxx could be better than 16/xxx is because it offered more headroom / dynamic range, would that then make things easier for both sides of this to see where the other is coming from?
> 
> Personally, I'm happy to listen to 256kbps mp3, as much as I am a DSD file, can I genuinely tell the difference? in mastering, yes, quite possibly, but to these ears, only by virtue of better headroom on the higher res file...
> 
> Then again, for those producers that don't push everything deep into the red, redbook can still sound extraordinary.


 

 I think that you are misunderstanding the whole headroom/dynamic range thing. Think of the available headroom/dynamic range provided by 16 bit digital audio has a normal letter sized envelop (4" X 9-1/2") and think of the available headroom/dynamic range provided by 24 bit digital audio has a legal sized envelop (9" x 12"). Now think of the audio contained in these envelops as a normal sized business card. Someone mails you a normal sized business card in a letter sized envelop and you open the letter and pull out a normal sized business card. Now that same person mails you a normal sized business card in a legel sized envelop and you open the letter and pull out a normal sized business card. In either case the size of the business card is exactly the same, changing the size of the envelop does not change the size of the business card.
  
 Changing the size of the audio container, i.e. increasing the bit depth, does NOT increase the dynamic range of the recording, it just increases the amount the dynamic range of that the container can hold. So a recording with 60db of dynamic range will have EXACTLY the same dynamic range, i.e. 60db, in any and all audio containers with enough bit depth to hold 60db of dynamic range - increasing the bit depth of the container will NEVER change this fact. But then again, FACTS are the bane of high end audio.


----------



## kraken2109

If I remember correctly, it's also impossible (at least currently) to produce an analogue circuit with 144dB of signal to noise ratio.


----------



## Duncan

Ralph...

That is the difference between you and pretty much everyone else in this thread - you've explained it, not in an upfront, aggressive manner, but - in a way that actually makes objective sense...

I stand corrected on my thoughts re snr vs the peak recording level...

Thank You


----------



## ralphp@optonline

duncan said:


> Ralph...
> 
> That is the difference between you and pretty much everyone else in this thread - you've explained it, not in an upfront, aggressive manner, but - in a way that actually makes objective sense...
> 
> ...


 

 You are most welcome.
  
 Unfortunately there is just tons and tons of absolutely deliberate misinformation being spread around the internet and in print regarding high resolution audio since lots of people are looking to make lots of money by getting people totally confused. Get informed and you can then make informed buying decisions. In other words: Don't believe the hype.


----------



## elmoe

duncan said:


> Ralph...
> 
> That is the difference between you and pretty much everyone else in this thread - you've explained it, not in an upfront, aggressive manner, but - in a way that actually makes objective sense...
> 
> ...


 
  
 I apologize if you thought any of my posts where aggressive, they weren't meant as such. My general approach to the Sound Science forum is 'all in good fun'.


----------



## kraken2109

elmoe said:


> I apologize if you thought any of my posts where aggressive, they weren't meant as such. My general approach to the Sound Science forum is 'all in good fun'.


 
 It's often hard on the internet to judge tone.


----------



## Soundsgoodtome

You need hiRes tracks to hear it. 





kraken2109 said:


> It's often hard on the internet to judge tone.


----------



## kraken2109

soundsgoodtome said:


> You need hiRes tracks to hear it.


----------



## ab initio

ralphp@optonline said:


> Unfortunately there is just tons and tons of absolutely deliberate misinformation being spread around the internet and in print regarding high resolution audio since lots of people are looking to make lots of money by getting people totally confused. Get informed and you can then make informed buying decisions. In other words: Don't believe the hype.




It's so unfortunately true 

Cheers


----------



## Thad-E-Ginathom

Catching up wit a couple of pages of posts...
  
 I think there were some mistakes in numbers back there along the thread.
  
 My hearing is noticeable poor. Bad enough that if we converse, I may have to ask you to repeat yourself here and there, and, if there is a background noise like a crowded room, I will hardly hear at all. 
  
 According to last test, it rolls off from 8khz. Most healthy-eared adults can hear much more than that, with the theoretical maximum, probably only seen in youngsters, of 20khz.
  
 Pointy-hat medieval scientists would only be relevant to the subject if they had invented sampling theory, or if they had had any form of digital music.  Science is not trying to explain digital music: it invented it.
  
 Sampling theory is not the explanation of the music: the music exists because of the sampling theory.
  
 Somebody else said that, and I wish I could remember their exact words, because they put it better.


----------



## cjl

beyermonster said:


> It's entirely possible that his audiologist was right, but he misheard the doc.


 
 True enough


----------



## esldude

About the audiologist saying you don't hear above 8 khz.  Some 20 years ago, given a hearing test for job promotion, I asked about my hearing and was told it was pretty much perfect.  I asked about the highest frequency they tested at as having heard test tones over my hifi none of them sounded all that high in frequency during his test.  He told me they only tested to 8khz.  I inquired a bit more.  The professional audiologist idea as it was explained to me was in terms of listening to conversations, communicating, etc. there was no handicap whatsoever if hearing was only good to 8 khz and no more.  Further there was an abundance of data on hearing norms and by the time there is any significant damage that was non-trivial it would show up with reduced sensitivity at 8 khz.  Hence no need to test above 8 khz.  I guess the entire professional audiologist world and the researchers that back it up just never happened to learn about our transient discrimination ability to 100 khz.


----------



## Soundsgoodtome

Seems like their testing and meaning of perfect hearing is based on human speech.


----------



## cjl

esldude said:


> I guess the entire professional audiologist world and the researchers that back it up just never happened to learn about our transient discrimination ability to 100 khz.


 
 We don't have any ability to discriminate transients to 100kHz (unless you happen to be the world's only internet-commenting bat). Most people with good hearing can go up to 15-18kHz though, and younger people with good hearing can go up to 20.


----------



## bigshot

Nothing above 12-14kHz is particularly important to music, so most people have nothing to worry about.


----------



## ralphp@optonline

cjl said:


> We don't have any ability to discriminate transients to 100kHz (unless you happen to be the world's only internet-commenting bat). Most people with good hearing can go up to 15-18kHz though, and younger people with good hearing can go up to 20.


 

 While it is true that mere moral humans can only hear to up to 20kHz but this does not apply for all the golden eared audio writers. In fact their advertisers absolutely demand that anyone editing or writing in any publication in which they advertise have all of the following super human abilities:
  
 1) the ability to hear up to a minimum of 96kHz - necessary to review 192kHz high resolution files.
  
 2) Ears which can withstand 140db dynamic range - necessary to review 24 bit audio files
  
 3) the ability to hear pico second jitter - needed to tell the difference between asynchronous USB and non-asynchronous USB
  
 4) the ability to hear the difference between a $200 and $2,000 power cord - needed to sell cable advertising
  
 5) the ability to hear the difference between a $10 USB cable and a $500 USB cable - again, needed to sell cable advertising
  
 and most importantly:
  
 6) the ability to make the readers believe that any the five abilities listed above is real.


----------



## SilverEars

I just looked at my HE-6's specs
  
 Frequency Response: *8 to 65 KHz  *





  I got moor hertz.


----------



## Duncan

lol Ralph,

I remember a pair of Technics headphones that were launched 10-15 years ago, where the big promo emblazoned across the box was that their maximum frequency was 100khz...

Whether that was at -50db or not, immaterial, just goes to show your point that marketing does attempt to sell you the impossible...

Actually - here you go, info about them here (edit - the thread I posted was somewhat inane)


----------



## esldude

cjl said:


> We don't have any ability to discriminate transients to 100kHz (unless you happen to be the world's only internet-commenting bat). Most people with good hearing can go up to 15-18kHz though, and younger people with good hearing can go up to 20.


 

 Sorry, didn't have a facetious emoticon to deploy.
  
 Yes, as I commented earlier in the thread.  The idea our hearing stops with steady tones at 20 khz (or less) yet responds to transients of 100 khz is a rather loony idea.


----------



## esldude

ralphp@optonline said:


> While it is true that mere moral humans can only hear to up to 20kHz but this does not apply for all the golden eared audio writers. In fact their advertisers absolutely demand that anyone editing or writing in any publication in which they advertise have all of the following super human abilities:
> 
> 1) the ability to hear up to a minimum of 96kHz - necessary to review 192kHz high resolution files.
> 
> ...


 
 Actually all they need is #6 on your list.  And it is quite apparent they have that ability in spades.


----------



## ralphp@optonline

steve eddy said:


> What's the misinformation? That's all on the subjective side of the street. If someone says something sounds better to them, what's to argue?
> 
> se


 

 There is quite a bit to argue. To wit:
  
 Let's say that I come home after shopping at the local mall and tell my wife that I saw Abe Lincoln shopping in Macys - no someone who looked a lot like Lincoln but actually Lincoln himself in the flesh. Now hopefully my wife would just tell me that I'm either totally mistaken or completely crazy and things would end there. But now what if I was so insistent that I really saw Abe Lincoln that I decided to set up a booth and sell tickets at $100 each for your chance to see Lincoln alive and in the flesh. Hopefully people would realize that I was either completely crazy or some kind of scam artist. Soon ticket sales would fall off to zero. End of story.
  
 Now suppose I write for some super slick high end audio publication and I say that I can clearly hear the difference between a 24bit/96kHz digital audio file and a 16bit/44.1kHz digital audio (dithered and resampled from the 24bit/96kHz digital audio file) and what's more clearly stated that the high price of the 24bit/96kHz digital audio file was therefore totally justified. However I provide absolutely no proof other then my word of honor. And furthermore, that no one else is able to tell the difference between the two files in any ABX test.
  
 So please explain to me how the first scenario differs form the second scenario. Note: I used the example of high resolution digital audio files but I could have instead used the proclaimed differences between digital audio cables as an even better example.


----------



## Thad-E-Ginathom

Well, if you tell them that, if only they trusted their own ears, they could hear the same things, they would hear them too, and everybody is happy. And you get the money.
  
 All those of us that try to cling to any sort of truth do is to make people _unhappy_.
  
 How about... we all give up and join the _happy club?_


----------



## elmoe

Not at my wallet's expense.


----------



## ralphp@optonline

thad-e-ginathom said:


> Well, if you tell them that, if only they trusted their own ears, they could hear the same things, they would hear them too, and everybody is happy. And you get the money.
> 
> All those of us that try to cling to any sort of truth do is to make people _unhappy_.
> 
> How about... we all give up and join the _happy club?_


 
 Do you mean that we should all move to Colorado? I know that with a little herbal help I too can hear 100kHz, and not just transients.


----------



## cjl

ralphp@optonline said:


> Do you mean that we should all move to Colorado? I know that with a little herbal help I too can hear 100kHz, and not just transients.


 
 Nah - we already have enough people here in Colorado - the state is full. You all should move to Washington instead


----------



## SilverEars

thad-e-ginathom said:


> Well, if you tell them that, if only they trusted their own ears, they could hear the same things, they would hear them too, and everybody is happy. And you get the money.
> 
> All those of us that try to cling to any sort of truth do is to make people _unhappy_.
> 
> How about... we all give up and join the _happy club?_


 
 You know I actually like discussions on the forums outside the science.  We know it's subjective, but it's fun to try out different gears and talk about our subjective impressions.  Are they absolutely correct like my multimeter?  No, not too many things we people do are.  I think it's fun that there are things that we cannot be absolutely certain, and we can be subjective about it, and I think this is part of the hobby itself.  
  
 Even some  of us in here have gears at home, some expensive headphones.  Where did the idea to purchase that specific headphones come from?  Tyll's graphs?  
	

	
	
		
		

		
		
	


	




 I think we still need to listen to gear and see if we like the sound if want to keep them.


----------



## ralphp@optonline

silverears said:


> You know I actually like discussions on the forums outside the science.  We know it's subjective, but it's fun to try out different gears and talk about our subjective impressions.  Are they absolutely correct like my multimeter?  No, not too many things we people do are.  I think it's fun that there are things that we cannot be absolutely certain, and we can be subjective about it, and I think this is part of the hobby itself.
> 
> Even some  of us in here have gears at home, some expensive headphones.  Where did the idea to purchase that specific headphones come from?  Tyll's graphs?
> 
> ...


 

 Your post reminds me of something that should be remembered when having these kinds of discussions. Namely that NOT everything in high end audio is unscientific mumbo jumbo. There is quite a bit of high end audio principles and ideas that are firmly grounded in good science and engineering unfortunately there is even a greater amount that is just pure snake oil. The real trick is to learn enough to be able to tell the real from the snake oil.
  
 In the end this whole high resolution audio craze shows just how important good mastering is - in fact it is so important that good mastering pretty much trumps just about anything else. As in a high bit rate mp3 made from well mastered CD will sound much better than a high resolution flac file made from a poorly mastered recording.
  
 Pay attention to what actually matters and even a modest audio system will provide one with hours of listening pleasure.


----------



## Soundsgoodtome

It doesn't rain as much in Colorado. 





cjl said:


> Nah - we already have enough people here in Colorado - the state is full. You all should move to Washington instead


----------



## SilverEars

ralphp@optonline said:


> Your post reminds me of something that should be remembered when having these kinds of discussions. Namely that NOT everything in high end audio is unscientific mumbo jumbo. There is quite a bit of high end audio principles and ideas that are firmly grounded in good science and engineering unfortunately there is even a greater amount that is just pure snake oil. The real trick is to learn enough to be able to tell the real from the snake oil.
> 
> In the end this whole high resolution audio craze shows just how important good mastering is - in fact it is so important that good mastering pretty much trumps just about anything else. As in a high bit rate mp3 made from well mastered CD will sound much better than a high resolution flac file made from a poorly mastered recording.
> 
> Pay attention to what actually matters and even a modest audio system will provide one with hours of listening pleasure.


 
 I agree with what you say.  There are fundamental things that we as audio enthusiasts should be weary of.  Not many of us that love music will have a technical degree or have the common sense to recognize snake oil.  
	

	
	
		
		

		
		
	


	




  I think at most time it's more of common sense than anything else.  
  
 What I realized is that it being very scientific could detract you from trying out some great performing products.  If you are very skeptical, it's highly likely you will not believe most of anything, and will not try out things.  There are times merits to certain things the majority of these audiophile community commonly do, sometimes numbers of heads do come up with interesting outcomes.  Lots of these people here are honest people that will try to give you their hones opinions on things.


----------



## Steve Eddy

ralphp@optonline said:


> So please explain to me how the first scenario differs form the second scenario. Note: I used the example of high resolution digital audio files but I could have instead used the proclaimed differences between digital audio cables as an even better example.




Having hallucinations of Abraham Lincoln would be a sign of some seriously abnormal mental health issues. Subjectively perceiving sonic differences even when there are no actual audible differences is a normal part of the human condition. 

se


----------



## ralphp@optonline

steve eddy said:


> Having hallucinations of Abraham Lincoln would be a sign of some seriously abnormal mental health issues. Subjectively perceiving sonic differences even when there are no actual audible differences is a normal part of the human condition.
> 
> se


 

 You addressed the mental health aspects of my two scenarios but not the part about using false claims (whether knowingly or unknowingly) to make money.


----------



## cjl

soundsgoodtome said:


> It doesn't rain as much in Colorado.


 
 Colorado is better than Washington, without a doubt. I won't argue with that. I just said we already have too many people here


----------



## Steve Eddy

ralphp@optonline said:


> You addressed the mental health aspects of my two scenarios but not the part about using false claims (whether knowingly or unknowingly) to make money.




Unless someone makes an absolute objective claim regarding actual audibility, where's the false claim? If someone says 24/96 or whatever sounds better to them than 16/44, where's the false claim? Are you making the claim that these people are lying when they relate their own subjective experiences? If so, that's a pretty damn bold claim.

se


----------



## ToddTheMetalGod

Nearly all recording microphones roll off at 14 KHz. Whether you think you can hear a difference with over 44.1 KHz doesn't matter, the information doesn't exist for you to process. As for 24-bit, the human hear can hear approximately in volume steps of 1 dB... so unless you're listening at over 96 dB there is no benefit over 16-bit.

Edit: If you don't believe me about the microphones, show me a recording waveform with information above the 22 KHz possible with 44.1 KHz audio.


----------



## bigshot

> Even some  of us in here have gears at home, some expensive headphones.  Where did the idea to purchase that specific headphones come from?  Tyll's graphs?
> 
> 
> 
> ...


 
  
 I use measurements to choose what I buy. I also use them to correct for imbalances. Those graphs represent sound in a very precise and specific way. You just need to know how to read and apply them.
  
 Also, a skeptic is just as apt to try new things as anyone else. It's just that a skeptic wants proof of the result, not just a warm fuzzy feeling inside.


----------



## Xenophon

ralphp@optonline said:


> There is quite a bit to argue. To wit:
> 
> Let's say that I come home after shopping at the local mall and tell my wife that I saw Abe Lincoln shopping in Macys - no someone who looked a lot like Lincoln but actually Lincoln himself in the flesh. Now hopefully my wife would just tell me that I'm either totally mistaken or completely crazy and things would end there. But now what if I was so insistent that I really saw Abe Lincoln that I decided to set up a booth and sell tickets at $100 each for your chance to see Lincoln alive and in the flesh. Hopefully people would realize that I was either completely crazy or some kind of scam artist. Soon ticket sales would fall off to zero. End of story.
> 
> ...


 
 There are quite a bit of moral/legal differences between both scenarios.
  
 In scenario one I think (hope) the general opinion among the public will be that it is not, in fact, possible to see A. Lincoln shopping at Macy's.  Therefore if you truly believed this to be the case and act as described, you'd be judged mentally ill but there'd be no intention to deceive, no morally reprehensible component and -in most jurisdictions- when hauled in front of a judge for attempted fraud you'd be held not guilty for reasons of insanity.
  
 In scenario B the crucial difference -which is not taken into account as a possibility in your example, rendering it flawed- is on a moral level wether or not you, as a seller, actually believe that a difference exists.  Morally this has absolutely nothing to do with preponderance of scientific evidence.  If you genuinely believe there to be some kind of difference then morally there's no problem, just like in the example above.  If you believe there to be no difference whatsoever then you're committing a morally reprehensible act, irrespective of the objective truth.   Wether or not you're committing fraud depends essentially on the acceptability of your position to the judge or (in the US I guess) a jury.  And there you might have a tough time getting a conviction because apparently despite scientific evidence pointing toward there being no audible differences, not everyone's convinced, going by the sales numbers.  Not to mention the fact that the majority of people adhere to some type of religion which by definition implies the acceptance of some unprovable elements while judging a situation.


----------



## castleofargh

steve eddy said:


> ralphp@optonline said:
> 
> 
> > You addressed the mental health aspects of my two scenarios but not the part about using false claims (whether knowingly or unknowingly) to make money.
> ...


 

 are we talking about the power of high res audio, or about the very real effects that placebo can have on humans?
 both are very well documented now.


----------



## Steve Eddy

castleofargh said:


> are we talking about the power of high res audio, or about the very real effects that placebo can have on humans?
> both are very well documented now.




We're talking about whether or not someone who reports their subjective experience with something, whether it be high res audio or not, is making false claims. Some seem to be arguing that they are. 

se


----------



## bigshot

Well, they can claim anything they want, and they can believe it fervently with all their heart, but if it isn't true, it's false. Right?


----------



## ToddTheMetalGod

steve eddy said:


> We're talking about whether or not someone who reports their subjective experience with something, whether it be high res audio or not, is making false claims. Some seem to be arguing that they are.
> 
> se


Audio is the only hobby I've ever seen where people actively promote false information, even after being shown evidence that disproves them. When disproven in other hobbies other people accept the proof, not relentlessly fight it with subjective experience. I think I need to leave the sound science forum forever. Maybe this forum should be DBT free like the cables and tweaks section...


----------



## Steve Eddy

bigshot said:


> Well, they can claim anything they want, and they can believe it fervently with all their heart, but if it isn't true, it's false. Right?




You're saying their subjective experience isn't true? That they're subjectively experiencing something other than what they're reporting to be experiencing? I certainly hope that's not what you're saying, but I don't see you saying anything else here.

se


----------



## Steve Eddy

toddthemetalgod said:


> Audio is the only hobby I've ever seen where people actively promote false information, even after being shown evidence that disproves them. When disproven in other hobbies other people accept the proof, not relentlessly fight it with subjective experience. I think I need to leave the sound science forum forever. Maybe this forum should be DBT free like the cables and tweaks section...




I'm still trying to figure out where you guys are getting the notion that someone relating their subjective experience is a false claim.

se


----------



## ToddTheMetalGod

steve eddy said:


> You're saying their subjective experience isn't true? That they're subjectively experiencing something other than what they're reporting to be experiencing? I certainly hope that's not what you're saying, but I don't see you saying anything else here.
> 
> se


Subjective experience is affected by psychology. The brain has as much of an affect on what we hear as our ears do. Technically speaking, their belief that 24-bit audio is better is making it better.


----------



## Soundsgoodtome

cjl said:


> Colorado is better than Washington, without a doubt. I won't argue with that.


 

 Peyton Manning would disagree


----------



## Steve Eddy

toddthemetalgod said:


> Subjective experience is affected by psychology.




Absolutely.




> The brain has as much of an affect on what we hear as our ears do.




Well, I wouldn't say _as much_. Otherwise, we'd be all over the map in terms of what particular individuals hear with respect to others. One person hears a mockingbird! another hears a lion's roar when all the time it's a dog barking. But yes, I agree with what you're saying in essence.




> Technically speaking, their belief that 24-bit audio is better is making it better.




That can certainly be part of it. Though even without being told exactly what one may be comparing, just the expectation of some difference can have an influence.

Now, after all that, let me ask you this. What's really the only thing that matters at the end of the day when we're sitting listening to music for our own enjoyment than our subjective perceptions? What else on earth do we have to work with but that?

se


----------



## ProtegeManiac

toddthemetalgod said:


> *Audio is the only hobby I've ever seen where people actively promote false information, even after being shown evidence that disproves them.* *When disproven in other hobbies other people accept the proof, not relentlessly fight it with subjective experience. *I think I need to leave the sound science forum forever. Maybe this forum should be DBT free like the cables and tweaks section...


 
  
 Start drinking wine and hanging out with people who will suggest what wine to get, then come back here and tell us about all the BS 
	

	
	
		
		

		
		
	


	




 There was one blind test where "connoisseurs" picked Chilean over French, then Napa Valley and Australian in another. I know people who scoff at our drinking sweet raspberry+grape wine brought in in large vats with oak planks in them (or our coconut liqueur which without fruit flavors is technically our Moonshine), and only drink the mishandled French wine (tropical country, slow customs at port, unrefrigerated trucks) that are halfway to becoming red wine vinegar. It's atrocious that these people buy them for $200+ per bottle when you can just blow the same cash on cognac, which travels better.


----------



## Xenophon

protegemaniac said:


> Start drinking wine and hanging out with people who will suggest what wine to get, then come back here and tell us about all the BS
> 
> 
> 
> ...


 
 See bold:  this is the core of the entire issue.  Not the spending of the moneyed hoi polloi that is, but the 'we know better' attitude of people -wearing lab coats or not- who think they have a moral obligation to dictate (distinct from 'inform') what others should do.  If people want to blow their cash, me thinks it's their business and their prerogative to do so.  If they want some expert assistance they presumably have the disposable income to get it.  
  
 BTW:  back in the 18th-19th century some wines were sent on a sea voyage from France to Cochin, India and back to speed up their development and maturation in the tropical heat.  But obviously I agree that it's not a good idea to move wine too much and certainly not to expose it to huge temperature shocks.  Which is why my cellar remained in Europe and wasn't moved to India.


----------



## ab initio

xenophon said:


> See bold:  this is the core of the entire issue.  Not the spending of the moneyed hoi polloi that is, but the 'we know better' attitude of people -wearing lab coats or not- who think they have a moral obligation to dictate (distinct from 'inform') what others should do.  If people want to blow their cash, me thinks it's their business and their prerogative to do so.  If they want some expert assistance they presumably have the disposable income to get it.




But the problem isnt people blowing their cash. It's that those who blow their cash also run their mouths and justify blowing their cash with claims that what they blow their cash on is inherently superior because it cost them a lot of dough, telling budding audiophiles that they need to blow their cash on the same crap, and insulting anyone who knows better and isnt foolish enough (or rich enough) to blow their cash on audiofool's gold.

Cheers

 PS, im not sure that 24 bit is really as silly as the > 48kHz sample rates. Dynamic range and low noise floor can be beneficial when music has very soft bits and very loud bits.


----------



## Xenophon

ab initio said:


> But the problem isnt people blowing their cash. It's that *those who blow their cash also run their mouths and justify blowing their cash with claims that what they blow their cash on is inherently superior because it cost them a lot of dough, telling budding audiophiles that they need to blow their cash on the same crap,* and insulting anyone who knows better and isnt foolish enough (or rich enough) to blow their cash on audiofool's gold.
> 
> Cheers
> 
> PS, im not sure that 24 bit is really as silly as the > 48kHz sample rates. Dynamic range and low noise floor can be beneficial when music has very soft bits and very loud bits.


 
 Which is obviously equally pointless and patronising.  And demonstrates my intolerance of any proselytising activity I guess.  Everyone should do as they seem fit; I once had the opportunity to test 2 interconnects costing about 10k USD.  Sounded the same as my 30 GBP pair to me but then that was irrelevant, the friend who had purchased them was happy.
  
 I draw the line at purchasing modern date high-resolution 'studio master' versions of '30s mono recordings on their original 78 rpm format   Those artefacts will doubtless sound nice.
  
 Shocking revelation after plodding through this thread:  apparently my hearing, which according to my audiologist still extended to 17 kHz at age 39  measured 2 years ago has diminished to 14-15 kHz if jRiver is to be believed and provided no other major variables are at work.


----------



## Agharta

As I couldn't hear the difference, I played some cd sourced 16/44.1 files and the same songs in high resolution format to my pet cat.

He told me my music taste sucked.


----------



## Thad-E-Ginathom

ralphp@optonline said:


> Do you mean that we should all move to Colorado? I know that with a little herbal help I too can hear 100kHz, and not just transients.


 

 With the right active ingredient, you might be able to _see_ it, let alone hear it!
  
 (NB: post based on experience in long, long ago youth. No, I have not indulged in such "experiments" for more than four decades.)


----------



## ToddTheMetalGod

Sometimes audio should be subjective, such as when selecting headphones or amps to use or purchase. Other times, such as when selecting file formats, picking out DACs or turntables, and room treatment or damping for speakers you should be analyzing things objectively.


----------



## kraken2109

steve eddy said:


> I'm still trying to figure out where you guys are getting the notion that someone relating their subjective experience is a false claim.
> 
> se


 

  
_'The world's greatest sounding music downloads'_
 Doesn't sound like they're being subjective to me, sounds like they're making a factual claim.
  
*From their FAQ section:*
 Q: Will I really hear the difference between the various formats?
 A: You should hear a substantial difference when listening to the music on a home stereo. The music will sound cleaner, the bass will be tighter and you will notice a higher definition in all the instruments. If you are going to pay for digital music, you might as well own it in the highest-quality format available.
  
 This video also claims they sound better than CD, DVD and even blu-ray which is interesting since blu-ray can store 192/24...
 https://www.youtube.com/watch?v=7l8xKUjqKOM
  
  
 Please note I am using HDTracks as an example since they seem to be the most well known seller of high sample rate and high bit depth music.


----------



## ToddTheMetalGod

steve eddy said:


> I'm still trying to figure out where you guys are getting the notion that someone relating their subjective experience is a false claim.
> 
> se


The human mind is powerful and fills in missing information for the senses. If you believe something hard enough it can affect what you hear. Ever thought you saw scary things in the night and convinced you did? I don't believe in ghosts, but many people claim they're real. In my eyes, the difference between 24-bit and 16-bit audio is as real as poltergeists.


----------



## cer

steve eddy said:


> You're saying their subjective experience isn't true? That they're subjectively experiencing something other than what they're reporting to be experiencing? I certainly hope that's not what you're saying, but I don't see you saying anything else here.
> 
> se


 
 The problem is that the reported subjective experience is almost always directly linked to technical specifications like 24/196. And that IS very probably a false claim, as it is very probable that the experience has really nothing to do with those specifications. If someone claims that 24/196 sounds better than 16/44 to him/her, it's not the actual sound format that causes the subjectively different experience but the persons limited comprehension and prejudices about those formats. To put it differently - if a person claims that 24/196 sounds better, he or she is very probably just plain wrong. It's his own mind that causes the subjective difference, which, of course is very real. But the claim itself is fallacious. 
 "24/196 sounds best to me" - it's just not true - prejudices, the lack of understanding of basic digital audio principles and human hearing limitations, is what causes 24/196 to subjectively sound the best. I can easily live with a claim that "24/192 sounds best to me, but i don't really know why". That's fine, you can't obviously know everything about everything. Bigger numbers, peer pressure, obviously it's usually going to sound better if you don't know the underlying facts. But not knowing stuff doesn't make an incorrect claim true. And that, of course, is the very essential reason why we need objective confirmation when talking about technical specifications, because as technical specifications are not subjective, it is wrong to link subjective impressions to objective parameters without objective confirmation.


----------



## Thad-E-Ginathom

toddthemetalgod said:


> The human mind is powerful and fills in missing information for the senses. If you believe something hard enough it can affect what you hear. Ever thought you saw scary things in the night and convinced you did? I don't believe in ghosts, but many people claim they're real. In my eyes, the difference between 24-bit and 16-bit audio is as real as poltergeists.


 
  
 And yet, people experience poltergeists!
  
 People experience poltergeists, ghosts, psychic phenomena, and other stuff up to and including "god." I'd have to include myself in there somewhere, but I'm also an abject _disbeliever_ in other things that would appear on the same list.
  
 The experience is one thing: the explanation is another. Even an imaginary experience is an experience to the the person that has it. The mind _is_ strong.
  
 In audio, the _subjectivist_ camp is characterised by a complete lack of analysis, and even a denial of it. The last time I tried to explain to audio friends that, yes, I was hearing a difference at this gear-change iteration, but self-examination caused me to conclude that I was hearing a difference _because I had heard the same piece of music a number of times, and my brain was simply listening to different aspects of it_, I was unanimously advised to _trust my ears._ Sure, I trust my ears (subject to know hearing loss) but that does not meant that, without introspection at least, I trust my mind/brain.
  
 Why does the "subjectivist camp" (yes, I know, it's a generalisation, of course) _do_ that? That is a scientific question in its own right, and I'm sure that some of the marketing departments are on to it!


----------



## Agharta

Did you see that Bigfoot Diaries programme where they DNA rested hair that fervent 'believers' claimed was real sasquatch? It was heartbreaking. One of them was from a porcupine, for chrissake. A porcupine. And don't get me started on the chump who reckoned he'd shot an infant big foot and held it in his hands as it died. 

Some people just need to experience a reality that makes them different to the rest of us. It helps them through the night, clearly. 

If your mumbo jumbo Hi-res audio helps keep the infinite perspective vortex at bay, so be it. Just don't try and sell it to me. I'm too busy taking photos of the Loch Ness monster with me mate Elvis.


----------



## Hapster

Got annoyed when I read someone touting an android app that allowed 192khz playback and claiming it sounded much better.


----------



## ralphp@optonline

ab initio said:


> But the problem isnt people blowing their cash. It's that those who blow their cash also run their mouths and justify blowing their cash with claims that what they blow their cash on is inherently superior because it cost them a lot of dough, telling budding audiophiles that they need to blow their cash on the same crap, and insulting anyone who knows better and isnt foolish enough (or rich enough) to blow their cash on audiofool's gold.
> 
> Cheers
> 
> PS, im not sure that 24 bit is really as silly as the > 48kHz sample rates. Dynamic range and low noise floor can be beneficial when music has very soft bits and very loud bits.


 
 Your first paragraph is quite true and reflects one of my major objections to subjective audio claims, however the second paragraph is just not true - as I stated earlier if the music being recorded has less than 120db of dynamic range (the limit for 16 bit audio) than it makes no difference if the bit depth is increased from 16 bit to 24 bit - the dynamic range will remain unchanged. By the way NO music has a dynamic range greater than 120db, NONE.
  


agharta said:


> As I couldn't hear the difference, I played some cd sourced 16/44.1 files and the same songs in high resolution format to my pet cat.
> 
> He told me my music taste sucked.


 
 Play your cat some Charlie Parker - all hep cats love be-bop.
  


thad-e-ginathom said:


> With the right active ingredient, you might be able to _see_ it, let alone hear it!
> 
> (NB: post based on experience in long, long ago youth. No, I have not indulged in such "experiments" for more than four decades.)


 
 With some of the new hybrids available today one can not only hear it and see it but touch it and taste it as well.
  


kraken2109 said:


> _'The world's greatest sounding music downloads'_
> Doesn't sound like they're being subjective to me, sounds like they're making a factual claim.
> 
> *From their FAQ section:*
> ...


 
 Exactly my point - these claims about high resolution audio are not being made by just "someone" but by someone trying to get YOUR money based on these completely FALSE claims. Quite different from one's audiophile friend relating their experience.


----------



## ab initio

ralphp@optonline said:


> Your first paragraph is quite true and reflects one of my major objections to subjective audio claims, however the second paragraph is just not true - as I stated earlier if the music being recorded has less than 120db of dynamic range (the limit for 16 bit audio) than it makes no difference if the bit depth is increased from 16 bit to 24 bit - the dynamic range will remain unchanged. By the way NO music has a dynamic range greater than 120db, NONE.




16 bit has the dynamic range of 96dB. Using noise shaped dither, the dynamic range can be extended in some frequencies at the expense of bringing up the noise floor in others. 

I never said all 24 bits are necessary, rather, that extra dynamic range isnt a bad thing. 18 or 19 bits would probably be enough to cover all situations, but data has a bad habit of taking up chunks of 8bits. 

I have music that has sections mastered at -44 dB. They're from the 70s, so in this case the digital noise floor is below the tape hiss, but im not sure if that would be the case if it were recorded and mixed in a modern studio. The advice you give is good common sense, albeit not rigorously accurate. I just think it is poor form to preach blanket statements that arent 100 true.

Cheers

Cheers


----------



## kraken2109

ab initio said:


> 16 bit has the dynamic range of 96dB. Using noise shaped dither, the dynamic range can be extended in some frequencies at the expense of bringing up the noise floor in others.
> 
> I never said all 24 bits are necessary, rather, that extra dynamic range isnt a bad thing. 18 or 19 bits would probably be enough to cover all situations, but data has a bad habit of taking up chunks of 8bits.
> 
> ...


 

 Nobody in a modern recording environment would be recording something at -44dB, let alone the final master being anywhere near that level. Most professional recording is done at 24bit, and that makes sense since it allows levels to be changed without any (no matter how small) negative impacts on the final CD (16bit) version.
 The point we're making is that for *playback*, 16bit is more than enough.


----------



## Steve Eddy

kraken2109 said:


> _'The world's greatest sounding music downloads'_
> Doesn't sound like they're being subjective to me, sounds like they're making a factual claim.
> 
> *From their FAQ section:*
> ...




That's called marketing. 

The context of this particular discussion was an individual simply describing their subjective experience. Specifically to the accusation by others that such people are making false claims.

se


----------



## ralphp@optonline

kraken2109 said:


> Nobody in a modern recording environment would be recording something at -44dB, let alone the final master being anywhere near that level. Most professional recording is done at 24bit, and that makes sense since it allows levels to be changed without any (no matter how small) negative impacts on the final CD (16bit) version.
> The point we're making is that for *playback*, 16bit is more than enough.


 

 Thank you for helping me out. As you stated 24bit is very useful on the recording/editing side but unnecessary on the final playback side. By the way the fact that almost all recordings are edited at some point makes DSD pretty much another marketing hoax - as in to edit a DSD recording, the DSD must be converted to PCM, edited and then reconverted to DSD. (Of course for a straight conversion of analog to DSD, i.e. with no editing done in the digital domain, DSD is fine.) So remember once you go PCM you never come back!


----------



## Steve Eddy

toddthemetalgod said:


> The human mind is powerful and fills in missing information for the senses. If you believe something hard enough it can affect what you hear. Ever thought you saw scary things in the night and convinced you did? I don't believe in ghosts, but many people claim they're real. In my eyes, the difference between 24-bit and 16-bit audio is as real as poltergeists.




Fine. But again, this goes back to accusing people of making false claims when sharing their subjective experiences.

I guess some just don't seem to grasp what it is I'm trying to say so I'll just move on.

se


----------



## ab initio

kraken2109 said:


> Nobody in a modern recording environment would be recording something at -44dB, let alone the final master being anywhere near that level. Most professional recording is done at 24bit, and that makes sense since it allows levels to be changed without any (no matter how small) negative impacts on the final CD (16bit) version.
> The point we're making is that for *playback*, 16bit is more than enough.




You're making blanket statements. Artistic experession might lead an artist to put large swings in dynamic range in their music. Who are you to say otherwise?

I'm playing devils advocate here because i want you folks to make better, more rigours arguments. I think it's really important for folks to understand the difference between hires formats and redbook cd, and to understand how absolutely small any theoretical difference between the audibility of them actually is, and what bit depth and sampling rate affect.

Music doesn't have to have 120dB dynamic range to be limited by 16bits. Theoretically, anything with more than 96 dB will be affected. You guys would better server the public showing how those differences are vanishingly unimportant in 99.9% of all cases, and how they might only be detected in carefully designed examples.

At that point, it's pretty clear that anything more than cd quality is unnecessary. 

Cheers


----------



## ralphp@optonline

steve eddy said:


> That's called marketing.
> 
> The context of this particular discussion was an individual simply describing their subjective experience. Specifically to the accusation by others that such people are making false claims.
> 
> se


 

 No, I never disagreed with your assertion that if an individual claims to hear a difference, even when it is scientifically impossible for there to be any difference, then that individual believes that there is a difference. What I have been claiming all along is that when these purely subjective differences are used to make FALSE claims to sell a product that this is more than just "marketing" and should be called out for the garbage that it is. By the way my use of the word "false" in this case, and in the case of HDTracks marketing, is the use of purely subjective impressions being stated or implied as provable objective FACTS.
  
 In other words, if one wants to believe in magic fairy dust, fine but just don't try to sell me any especially by making claims about the wonderful magic properties. In that case the only magic property in the fairy dust is the ability to separate fools from their money.


----------



## ralphp@optonline

steve eddy said:


> Fine. But again, this goes back to accusing people of making false claims when sharing their subjective experiences.
> 
> I guess some just don't seem to grasp what it is I'm trying to say so I'll just move on.
> 
> se


 
 See my previous post. Private individuals as opposed to for profit corporate entities, even if the US Supreme Court has deemed them equal.
  


ab initio said:


> You're making blanket statements. Artistic experession might lead an artist to put large swings in dynamic range in their music. Who are you to say otherwise?
> 
> I'm playing devils advocate here because i want you folks to make better, more rigours arguments. I think it's really important for folks to understand the difference between hires formats and redbook cd, and to understand how absolutely small any theoretical difference between the audibility of them actually is, and what bit depth and sampling rate affect.
> 
> ...


 
 If the music actually had a dynamic range of greater than 96db then it would not matter whether the digital audio recording of it was 16 bit or 24 bit since no home audio system would be capable of playing it because of the massive amount of power required to reproduce it, which is just another valid reason why 24bit is not necessary for proper playback.


----------



## Steve Eddy

ralphp@optonline said:


> No, I never disagreed with your assertion that if an individual claims to hear a difference, even when it is scientifically impossible for there to be any difference, then that individual believes that there is a difference. What I have been claiming all along is that when these purely subjective differences are used to make FALSE claims to sell a product that this is more than just "marketing" and should be called out for the garbage that it is. By the way my use of the word "false" in this case, and in the case of HDTracks marketing, is the use of purely subjective impressions being stated or implied as provable objective FACTS.
> 
> In other words, if one wants to believe in magic fairy dust, fine but just don't try to sell me any especially by making claims about the wonderful magic properties. In that case the only magic property in the fairy dust is the ability to separate fools from their money.




But that's not what this particular discussion was about. It began with thus statement from kraken2109:

"My main point was that Jude is the 'frontman' of an audio forum and he shouldn't be spreading misinformation so a sponsor can make money."

To which I replied:

"What's the misinformation? That's all on the subjective side of the street. If someone says something sounds better to them, what's to argue?"

And kraken2109 again:

"The misinformation is claiming something sounds better when it is physically impossible for said thing to sound better."

The claim here is that someone simply expressing their subjective experience is misinformation. Which I disagree with. You came in quoting my "what's to argue" query when it comes to someone's subjective experience and said there's plenty to argue.

What I said had absolutely nothing to do with marketing or selling anything. It only had to do with someone's subjective experience being misinformation or a false claim. My only point being that there's nothing to argue when it comes to someone's subjective experience unless you're calling them a liar.

And I'm just going to leave it at that.

se


----------



## Agharta

Ego leads to all sorts of bad opinions.


----------



## ab initio

ralphp@optonline said:


> If the music actually had a dynamic range of greater than 96db then it would not matter whether the digital audio recording of it was 16 bit or 24 bit since no home audio system would be capable of playing it because of the massive amount of power required to reproduce it, which is just another valid reason why 24bit is not necessary for proper playback.


 

 I've heard the 16 bit noise floor using my sound isolating Senn hd280 headphones. Would I have wanted a 3 kHz 0dB single to come blaring through my headphones immediately after it? No. Would I have wanted a -30 dB single to come through my headphones immediately after it? Probably not.
  
 But it is incorrect to say that >96 dB is impossible with audio equipment. Most competently designed amplifiers can achieve 100dB signal-to-noise.
  
 You can't make absolutist statements that are demonstrably false. This is how audiophile zealots can make counter arguments that make it look like their position is more reasonable. You need to refute their points with rigorous arguments.
   
Cheers


----------



## Agharta

What is a noise floor?


----------



## ab initio

kraken2109 said:


> Nobody in a modern recording environment would be recording something at -44dB, let alone the final master being anywhere near that level. Most professional recording is done at 24bit, and that makes sense since it allows levels to be changed without any (no matter how small) negative impacts on the final CD (16bit) version.
> The point we're making is that for *playback*, 16bit is more than enough.


 
  
 Please take a look at this sample of one of my favorite King Crimson tracks, 'Lizard," which was on their aptly named album, "Lizard":
  
 http://www.head-fi.org/t/711374/does-frequency-response-or-csd-entirely-determine-sound-quality/15#post_10395918
  
 Here, I walk you through the math to calculate the different recorded amplitudes at different points in the music. there are parts at -44dB FS. One of the things I really like about this track is it's dynamic range. And pretty much  everything else about it 
	

	
	
		
		

		
			





.
  
 Not all music must be a live recording nor does it have to be dynamic rangeless uber-compressed in the studio.
  
 Just because rooms might be a bit noisy at ~30dB SPL, doesn't mean that the inside of a good pair of isolating headphones is. It's one of the things I like about my headphones vs my speakers.
  
 Cheers


----------



## ToddTheMetalGod

steve eddy said:


> Fine. But again, this goes back to accusing people of making false claims when sharing their subjective experiences.
> 
> I guess some just don't seem to grasp what it is I'm trying to say so I'll just move on.
> 
> se


 
 This is one of those arguments that have been going on since the early days of digital audio. We're trying to explain that the dynamic range of 24-bits is enough for the music and that the frequency response above 44.1 kHz is out human audibility... but nobody listens. They ask us where our scientific evidence is even though countless tests and measurements have gone on for probably 20 years. It's just getting old. Then people pull the argument that frequencies outside of the audible range "psychologically affect the listening through psycho-acoustics," which believe me does not happen, I can only hear up to 14 kHz strongly and 15 kHz barely and it drops straight off after that. If something is outside of hearing threshold, it's all the way out... there's no "sort of hearing it."
  
 I'm not saying someone's subjective experience is wrong, everyone can experience something differently, in fact there's huge differences between the hearing of average people (which probably results in our preferences in sound signature). But this is really something that is definitive and can't be argued. Unless the DAC is processing 24-bit audio differently than 16-bit or the recordings aren't identical, it is impossible to hear the difference between the two. It's been proven in many papers over the history of digital audio. 24-bit exists to lower the noise floor during recording. The audio engineers designing consumer digital audio chose 16-bit/44.1 kHz because it can reproduce audio perfectly during playback. They didn't simply settle for what existed at the time, they wanted to stomp tape out of existence in the consumer's eyes. Their salaries, well-being, and family depended on digital audio performing as expected... they weren't goofing off the entire time they were developing these standards.


----------



## fiascogarcia

That's it, I'm buying a Victrola!!


----------



## bigshot

steve eddy said:


> You're saying their subjective experience isn't true? That they're subjectively experiencing something other than what they're reporting to be experiencing?
> se




The reporting of their subjective impression may be accurate, but that doesn't make the claim true.


----------



## kraken2109

ab initio said:


> Please take a look at this sample of one of my favorite King Crimson tracks, 'Lizard," which was on their aptly named album, "Lizard":
> 
> http://www.head-fi.org/t/711374/does-frequency-response-or-csd-entirely-determine-sound-quality/15#post_10395918
> 
> ...


 

 This is certainly an interesting example. Unfortunately I don't own the track and I can't exactly analyse it through youtube. The question is, are you hearing the noise floor of 16bit digital audio or are you hearing the noise floor or the studio it was recorded in or perhaps the noise floor of the microphones or other recording equipment used? I'm not sure how we'd find out.
 With modern dither the noise floor of 16bit audio is even lower, so I wonder what it would sound like if that track was recorded today.
  
 Out of interest, I used Audacity to generate a 30 second 16bit WAV of silence. I'm not sure whether Audacity applied dither to this, and if so I don't know whether the dither in Audacity is a good one anyway, but I thought i'd share the file for those interested.
  
 https://www.dropbox.com/s/de46xlmsjx9arr6/30%20seconds%20of%20silence.wav
  
 Playing this file on my Denon receiver and AKG headphones, I have to turn the amp up to -10dB before I can just about hear noise. However, I can't tell you whether that noise is the noise floor of the track or the noise floor of something else in my system. I'd be interested to see how this track sounds to others.
  
 What I can say though is I tried listening to that king crimson track (sadly through youtube) at that volume level and it while the quiet opening was listenable (fairly loud), the louder sections were almost painful.
  
 EDIT: In an attempt to see if the noise was my system or 16bit, I generated another 30 seconds of silence and exported in 24bit. Even at 0dB on my receiver I can't hear noise with this one.
 https://www.dropbox.com/s/uen66lni39xvo1j/24bit%20silence.wav


----------



## bigshot

steve eddy said:


> That's called marketing. The context of this particular discussion was an individual simply describing their subjective experience. Specifically to the accusation by others that such people are making false claims. se




The false claim there is attributing the subjective experience to the difference in sound quality between redbook and high bit rate audio. It clearly wasn't due to that.


----------



## ralphp@optonline

ab initio said:


> Please take a look at this sample of one of my favorite King Crimson tracks, 'Lizard," which was on their aptly named album, "Lizard":
> 
> http://www.head-fi.org/t/711374/does-frequency-response-or-csd-entirely-determine-sound-quality/15#post_10395918
> 
> ...


 

 Either I'm confused, you're confused or we are both confused.
  
 Based on the link you provided I think that you are mistaking volume for dynamic range. Dynamic range is the difference between the lowest volume level and the highest volume level. So if you have a musical piece with the lowest volume being 1 db above total, dead silence (which is really not possible except out in space) and the loudest volume being 90 db above total, dead silence than the dynamic range of that musical piece would be 89db (90db - 1 db). Now if you raised the volume of the quietest passages to be audible in the real world, to say 30db, then the loudest parts would shifted to 120 db but the dynamic range would still be 89 db, even though the loud parts wold now be 30db above their previous peak of 90db. So I repeat, there is no music that humans can listen to without permanent hearing damage that needs more than 16 bits of dynamic range.


----------



## bigshot

ab initio said:


> You're making blanket statements. Artistic experession might lead an artist to put large swings in dynamic range in their music. Who are you to say otherwise?
> Cheers




96dB isn't just a wide dynamic range. It's a range that puts a sizeable chunk of the info across the line into inaudibility at any tolerable listening volume... not just comfortable, but tolerable.

Even if you were listening in a room with absolutely no noise, your ears and blood pulsing through them would create a noise floor... like the sound you hear when you're in a cave. Redbook audio in practice has to be boosted over its maximum range in order to hear everything... which pushes it up to the threshold of pain.

It doesn't really matter though, because even the most dynamic music doesn't exceed 40-50dB. Redbook has room to spare.


----------



## Steve Eddy

bigshot said:


> The reporting of their subjective impression may be accurate, but that doesn't make the claim true.




If the only claim is their subjective experience, how can it be anything other than true unless you've got a bunch of people running around lying about what their subjective experience is?

se


----------



## bigshot

kraken2109 said:


> Playing this file on my Denon receiver and AKG headphones, I have to turn the amp up to -10dB before I can just about hear noise. However, I can't tell you whether that noise is the noise floor of the track or the noise floor of something else in my system. I'd be interested to see how this track sounds to others.




On my speaker rig, the noise from cranking the amp way up is probably in the same area as the noise floor of redbook. 90dB is a hell of a lot.


----------



## bigshot

steve eddy said:


> If the only claim is their subjective experience, how can it be anything other than true unless you've got a bunch of people running around lying about what their subjective experience is?
> 
> se




It depends on how they state it. If they say, "High bitrate sounded better to me.", they're not drawing any conclusion; but "I see ghosts." and "High bitrate sounds better than CD." contain conclusions that are false. For some reason claims in audiophile press and advertising ALWAYS points straight at the conclusion.


----------



## ralphp@optonline

steve eddy said:


> If the only claim is their subjective experience, how can it be anything other than true unless you've got a bunch of people running around lying about what their subjective experience is?
> 
> se


 

 And where do we draw the line when purely subjective experience is being used as a marketing tool?
  
 If I believe that my audio system sounds 100 times better (remember it's purely subjective so I can assign a subjective numerical value to just how much better my audio system sounds) after I coat all the contact points in Wonder Coat (a proprietary mixture that I developed, but which is actually just plain tap water mixed with some liquid soap) that would be fine so long I am not trying to sell anyone else some Wonder Coat.
  
 Now what if I now am trying to sell Wonder Coat and the only proof can I offer is my purely subjective experience - again not a problem so long as everyone knows that Wonder Coat's magic is purely subjective.
  
 And finally, what if I am now trying to sell Wonder Coat and instead of clearly stating that Wonder Coat's magic is purely subjective I frame it so that it appears that Wonder Coat's magic is based on hard science (similar to the HDTracks marketing that was quoted earlier) - would this be okay? And I'm not referring to whether or not it would be morally or legally okay but to whether or not someone else has the right to call bulls*#t on my claims.
  
 In other words it is the turning of what is purely an individual's subjective experience/belief into a marketing tool (complete with quasi-scientific mumbo-jumbo - think any cable advertisement copy) that I believe we should be able to call BS on and rightfully demand hard, scientifically valid and repeatable proof.
  
 And guess what? There is no hard, scientifically valid and repeatable proof that 24 bit audio and sampling rates over 48k offers anything other than purely subjective improvements, which is why I am calling BS on HDTracks and their paid friends in the publishing world.


----------



## Hapster

ralphp@optonline said:


> And where do we draw the line when purely subjective experience is being used as a marketing tool?
> 
> If I believe that my audio system sounds 100 times better (remember it's purely subjective so I can assign a subjective numerical value to just how much better my audio system sounds) after I coat all the contact points in Wonder Coat (a proprietary mixture that I developed, but which is actually just plain tap water mixed with some liquid soap) that would be fine so long I am not trying to sell anyone else some Wonder Coat.
> 
> ...




PM sent, willing to buy some wondercoat, how much would a bottle cost?


----------



## Steve Eddy

ralphp@optonline said:


> And where do we draw the line when purely subjective experience is being used as a marketing tool?




I draw the line when they start trying to tell _other_ people what they'll hear.

se


----------



## Steve Eddy

hapster said:


> PM sent, willing to buy some wondercoat, how much would a bottle cost?




Save your money. Wonder Coat sucks. My Magic Coat is far superior.

se


----------



## ComradeDylie

ralphp@optonline said:


> if you have a musical piece with the lowest volume being 1 db above total, dead silence (which is really not possible except out in space)


 
  
 Oooo I think you might be on to something...we must start recording in space!!!!
  
 In all seriousness though, the sampling rate conundrum is the most interesting I think.  For two reasons:
  
 1.  There seem to be differing opinions about whether the higher sampling rates are destructive or not.  One camp says they are destructive for the reasons mentioned in the article, another says that, in fact, sampling at the higher rates will push aliasing, artifacts, etc outside of the audible range thus improving audio quality. Which is basically the totally opposite of the other camp which basically says it creates aliasing and artifacts inside the audible range.
  
 2.  When a trumpet or cymbal is heard live, the frequencies we cannot hear are in fact still present and hitting our bodies. Ergo, it may be more 'natural' sounding if these inaudible frequencies are included even if we cannot hear them.  Plus it could preserve high order harmonics that would otherwise be absent and although we cannot hear them, they could affect the waveform and provide a slightly different sound (perhaps in a similar way that the exact same note played through a tube amp and a solid state amp will have a different color to the sound)
  
 I didnt bother clicking the article link but I am assuming it is the same one I read a month or so ago...and there was at some point a comparison to a videophile who wants great Microwave, IR, UV, X-ray and gamma performance from his TV or something.  Even says to use an IR remote in pitch black to see if you can see it (you cant...obviously..)
  
 This is a broken argument.  Why?  Have you ever hung out around infrared light?  That you cant see?  Yeah, that light is friggin hot, and you feel its presence although you cannot see it.  So yeah, if during a desert scene in a movie your TV started putting off IR light to simulate the same heat you would feel in the desert; would this not be even more immersive and 'accurate'?  I vote yes. Similarly, we cannot see UV light, but we can see its effects when it hits that UV artwork.  If, during a black light scene in a movie, that UV light was to emit from the TV and hit your UV poster would this also not feel more accurate and realistic?  Again I vote yes.  This whole 'we cant see it so it doesnt matter' argument is a little lost on me because we use IR and UV all the time during our lives and NOTICE quite obviously that they are present.  So if we have the ability to preserve these effects then why not?  (disclaimer:  yeah you could get cancer or something, meh, thats not the point now is it!)
  
 Does this correlate to audio?  Probably.  They are both electromagnetic waves after all.  Although, in theory, it would also likely be much less obvious than with the light.
  
 About bit depth:
  
 Recording in space aside.... (come see me in 15-30 years, hopefully Ill have a studio up there by then lol)
  
 If mastering in 24 bit and distributing in 16 bit helps lower the noise created during mixing and editing then mastering in 32 bit and distributing in 24 bit would only seem to increase this effect and hopefully provide an even cleaner sounding track.  This I dont think is really a debate.  I think that would be a fact.
  
  Now the question is, would that same 32 bit master, sound the same in 16 bit release as it did in 24?  Im not sure, but I would think (especially with using clever techniques that may or may not exist yet) that the extra bit depth could be used specifically for the eliminate noise.  Something like having a large dynamic gap between the music and noise.  For example, use 18 bits for the music and 4 for noise and 2 for the gap:  loudest noise level at 00000F and have the quietest music be at 00003F and obviously the highest music level at FFFFFF.  I have no idea if thats even possible or if it would have any affect but hey its at least some food for thought.  
  
 Basically what I am saying is that it could have some positive effects that we only just now able getting to a point in hardware to be able to realize and to simply brush it off as being at inaudible frequencies or saying that a recording studio cant even get quiet enough to make use of the bit depth is a bit naive and kind of like saying "color tv?  why would anyone in the world want that?" or "HDTV?  why would someone want that?  you are just going to see a lot more pores and wrinkles on that older actress you thought was smoking hot.."
  
 I am not saying that Hi-Res is the real deal either, Im just playing devils advocate so we can have a proper debate discussion rather than all jumping on our fanboy bandwagons and saying junk like "oh yeah i totally hear a massive difference between 96 and 192" or "no way man it sounds the same to me",  that just isnt productive and gets nobody anywhere.  
  
 Notto mention most of our hearing is quite damaged in one way or another for one reason or another...
  
 Those darn kids have those text message tones that their teachers and parents cant hear...so would a 2 year old have a better chance at spotting differences?  Hell yeah one would!  But, they are two, not exactly the most articulate or experienced bunch are they?  So it would be pretty hard to test that too.  
  
 Basically the only real way to know is to record in space with best possible equipment, master in 64/768, publish in 32/384, let a 3-5 year old who has been raised with the sole goal of not damaging his or her ears listen to it through the worlds greatest source/dac/amp and phones also in space.  
  
 And who knows?  With all the genetic data we are generating and with the advances in bioengineering, maybe before too long we can splice some dog or bat into our DNA and actually be able to hear these frequencies.  So perhaps we will all be able to hear past 20 kHz in the future. Maybe then it will be the obvious choice?
  
 Ill shut up now


----------



## kraken2109

comradedylie said:


> About bit depth:
> 
> Recording in space aside.... (come see me in 15-30 years, hopefully Ill have a studio up there by then lol)
> 
> ...


 
 I know you're joking about the recording in space thing, but I hope you understand why that isn't possible. I'll give you a clue, it isn't something we can fix with technology.
  
 As mentioned before, the advantages in recording and mixing with a higher bit depth is that tracks can be made louder without raising the noise floor (digital). However there will always be the noise floor of the room, and in the most perfect room you'll have the noise floor of the microphone and recording equipment. Assuming a suitable mastered track, there should at least be one part that hits (or nearly hits) full scale. So to need more than 16 bits for playback there would have to be a part so quiet you'd struggle to hear it, unless you turned it up so loudly that the loud section was physically painful. Sure, you can use the argument of getting the noise of the digital noise floor as low as possible, but there's no point going further when, as said above, there are so many other noise floors to worry about.


----------



## ComradeDylie

kraken2109 said:


> I know you're joking about the recording in space thing, but I hope you understand why that isn't possible.


 
 Isnt possible yet.
  
 Humans couldnt do almost any of things that we can now, say 10,000 years ago.  
  
 Humans couldnt do most of the things we do on a regular basis, 3,000 years ago.
  
 Humans couldnt do many of the things we take for granted today, 300 years ago.
  
 Hell we couldnt even record digital music 100 years ago, now look at us go!
  
 Give it time, my friend, give it time.


----------



## ralphp@optonline

comradedylie said:


> If mastering in 24 bit and distributing in 16 bit helps lower the noise created during mixing and editing then mastering in 32 bit and distributing in 24 bit would only seem to increase this effect and hopefully provide an even cleaner sounding track.  This I dont think is really a debate.  I think that would be a fact.


 
 Disclaimer: I enjoyed most of your very amusing post and I am not trying to attack you. So with that said let me sharpen my sword 
	

	
	
		
		

		
			




  
 The bigger, larger, higher, etc. is better argument is one that audiophiles fall for each and every time and most of the time it is just not true. Increasing from 24 bit to 32 bit would not change anything expect for the advertising copy, as in "all our recordings are done in 32 bit, while almost everyone else is still using 24 bit". Just as a super tweeter which is capable of reproducing sound up to 40kHz does not sound any better than a tweeter which only goes to 20kHz.
  
 The bigger is better believe has lead to 2" thick power and speaker cables, 500 watt/channel amps etc. as manufacturers know that people will fall for "bigger is better" every time.
  
 A quick slightly off topic aside: speaking of high resolution video I find myself laughing at the recent push for 4k video. Laughing because more often than not, when I go to a friend or relative's house I find them watching a stretched standard definition channel on their giant HDTV - even when the same channel is available on a high definition channel. Point is most people don't know and don't care. The same can be said for our little discussion going on here - most people, as in 99% or so, will gladly listen to music on Pandora, streamed at 64k AAC+ - so who are we kidding with this high resolution nonsense?


----------



## Astropin

comradedylie said:


> Isnt possible yet.
> 
> Humans couldnt do almost any of things that we can now, say 10,000 years ago.
> 
> ...


 

 Except for the fact that sound waves require a medium, like air or water, to travel though.....now and forever.


----------



## bigshot

comradedylie said:


> Ergo, it may be more 'natural' sounding if these inaudible frequencies are included even if we cannot hear them.


 
  
 Good job! The "ergo" nearly completely covered up the circular logic!


----------



## ComradeDylie

astropin said:


> Except for the fact that sound waves require a medium, like air or water, to travel though.....now and forever.


 
 We also need air to breathe up there.  I imagine if we solve the air to breathe problem, the having no medium to hear sound in would not really be an issue.  
  
 Im thinking geosynchronous orbiting air balloon room type thing =)
  
  
 Quote:


ralphp@optonline said:


> The bigger, larger, higher, etc. is better argument is one that audiophiles fall for each and every time and most of the time it is just not true. Increasing from 24 bit to 32 bit would not change anything expect for the advertising copy, as in "all our recordings are done in 32 bit, while almost everyone else is still using 24 bit".



  
 Yeah Im totally with you there man, marketing is really evil stuff.  CD quality recordings using 32 bit -- who cares? But if you are producing 24 bit music it would make sense to mix in 32 bit.  Same as it made sense to mix in 24 bit for 16. Right?   Just seems that you would always want to mix at a higher bit rate than the music you are trying to produce.  On the other hand, we are already debating that 24 bits is more dynamic range than we need, so the necessity to mix in 32 bit could be smoke and mirrors since the junk would already be at low enough volume.  I am not a music producer, I do not claim to have any more than a rudimentary knowledge of mixing and mastering.  Just trying to extend the same logical strategies we use for CD res to hi-res
  


bigshot said:


> Good job! The "ergo" nearly completely covered up the circular logic!


 
 Well I hope you are referring to the fact that I said inaudible frequencies that we cannot hear.  Which isnt exactly circular logic, just poor use of adjectives.  Unless something else in there was circular (I sure hope not!).
  
  More or less I am referring to the audio version of the incandescent light vs fluorescent light vs sunlight phenomenon.  Basically humans feel more comfortable and our bodies respond much better to incandescent lighting because they generate heat and a more sun-like color whereas the fluorescent lights contain much more blue light and generate less heat. The miniature suns (Tommy Edison bulbs) cause a more natural reaction from our body than the fluorescents.  Thats why being in the office or hospital feels so uncomfortable. (amongst other reasons sure...)
  
 Not totally sure that it applies to audio, but theres no reason to say that it wouldnt.  If that extra stuff isnt there, it just isnt as accurate of a reproduction so perhaps it doesnt sound as true to life as it could if it were there.  I hope that isnt circular


----------



## Xenophon

ralphp@optonline said:


> And where do we draw the line when purely subjective experience is being used as a marketing tool?
> 
> If I believe that my audio system sounds 100 times better (remember it's purely subjective so I can assign a subjective numerical value to just how much better my audio system sounds) after I coat all the contact points in Wonder Coat (a proprietary mixture that I developed, but which is actually just plain tap water mixed with some liquid soap) that would be fine so long I am not trying to sell anyone else some Wonder Coat.
> 
> ...


 
 The absence of evidence for an occurrence  does not constitute evidence of absence.  E.g. the 'all swans are white' inductive reasoning example which held fine as a probabilistic 'proof' until some naturally black swans were discovered.  Which is why I call any categoric claim (you prudently kept 'subjective' in your statement) such as regarding the bitrates as premature and at best probabilistic using the knowledge now at our disposal.  Probabilistic models, even when extremely developed, are problematic because they typically imply the acceptance of a number of elements which may turn out to be based on quicksand if more data becomes available. 
  
 Of course, the above does not mean that someone claiming that there are benefits gets a free pass:  it's up to the claimant to provide evidence and in this case as well as with cables etc therein lies the difficulty for them of course.  I have no problems with them peddling their stuff as long as that's made clear.


----------



## bigshot

comradedylie said:


> Well I hope you are referring to the fact that I said inaudible frequencies that we cannot hear.


 
  
 No it was the whole idea that inaudible frequencies can make music sound better. How are they going to sound at all when they're inaudible?!
  
 By the way, there have been studies that have shown that super-audible frequencies have absolutely no impact on sound quality. In fact, music doesn't really contain any, because the only musical instruments capable of producing them are cymbals and triangles, and in those, auditory masking from lower harmonics totally obliterates the ones in the upper range of hearing. Inaudible frequencies would be doubly inaudible!
  
 Here is a "for instance" for you... This is an Apple Lossless file of a digital recording of Bach. The first half is the recording as it appears on the CD. The second half is just the frequencies above 12kHz.
  
 http://www.vintageip.com/test/freqresponsetest.m4a
  
 Silence... Yes, there isn't even anything above 12kHz in a lot of music, much less above 20kHz...


----------



## Hapster

bigshot said:


> No it was the whole idea that inaudible frequencies can make music sound better. How are they going to sound at all when they're inaudible?!
> 
> By the way, there have been studies that have shown that super-audible frequencies have absolutely no impact on sound quality. In fact, music doesn't really contain any, because the only musical instruments capable of producing them are cymbals and triangles, and in those, auditory masking from lower harmonics totally obliterates the ones in the upper range of hearing. Inaudible frequencies would be doubly inaudible!
> 
> ...


 

 "video will not play because file is corrupt"


----------



## bigshot

It's not a video. It's an Apple Lossless file. Play it in iTunes.


----------



## briskly

Bigshot, I think you need to use a steeper filter. When I look at it in SPAN, I can see you also trimmed off some response below 15khz.
  
 Not that I don't get your point and all about high frequencies. Though perhaps try a test just removing those higher frequencies instead?


----------



## bigshot

I used the 1/3 octave graphic EQ in Sound Forge and cut everything but the last three sliders 3 x -20dB. I could do the inverse, just cutting the top end, but there wouldn't be any audible difference.


----------



## BeyerMonster

> Here is a "for instance" for you... This is an Apple Lossless file of a digital recording of Bach. The first half is the recording as it appears on the CD. The second half is just the frequencies above 12kHz.
> ...
> Silence... Yes, there isn't even anything above 12kHz in a lot of music, much less above 20kHz...


 
DISCLAIMER: I installed Audacity about 5 minutes ago and have basically no prior experience doing this, so it's 50/50 that I did all of this wrong.
  
 I thought this claim was pretty interesting, so I wanted to try a little experiment for myself. I took a "modern" track I owned that I assumed would have some high frequency information, and did a teeny bit of analysis in Audacity for grins.
  
 This is the before:

  
 Original Waveform:

  
I applied this filter:

  
 This is the after:


----------



## ab initio

kraken2109 said:


> This is certainly an interesting example. Unfortunately I don't own the track and I can't exactly analyse it through youtube. The question is, are you hearing the noise floor of 16bit digital audio or are you hearing the noise floor or the studio it was recorded in or perhaps the noise floor of the microphones or other recording equipment used? I'm not sure how we'd find out.
> With modern dither the noise floor of 16bit audio is even lower, so I wonder what it would sound like if that track was recorded today.
> 
> Out of interest, I used Audacity to generate a 30 second 16bit WAV of silence. I'm not sure whether Audacity applied dither to this, and if so I don't know whether the dither in Audacity is a good one anyway, but I thought i'd share the file for those interested.
> ...


 
  
 HI kraken,
  
 Thank you for the noise test files. Normally, to test if i could hear the 16 bit noise floor, I set my DAC to 16 bit mode, and pause a song in winamp, that way I get the DAC's output through window's kmix which adds the dither. Now I have those files, it's much more convenient (and I don't have to futz with settings on my Linux machine!). Here, I will use your tracks and compare them to listening to music.
  
 Abstract:
 Here, I'm going to ABX the empty files kraken uploaded, seeing if I can differentiate between 16bit and 24bit empty files. Then I'm going to listen to music and see if my  eardrums are blown into my skull. Then I will repeat the ABX.
  
 Materials and methods:
 I listened from my CentOS linux server which has some extra fans running to keep my passively cooled raid card and all the hard disks in the machine cooled. I had a quieter listening environment in my old apartment when the server was in my bed room and I could listen in my office without my much quieter laptop+usb dac. Here, I'm using the machines motherboard line out into a Schiit Magni, and I'm using Paradox headphones with slight noise isolation. Software volumes are maxed. The schiit volume is set very loud, at about 2 o'clock. You can try and estimate the resulting SPLs using the data from this fantastic thread here. I found that the diagram of the volume pot at the bottom of the post reflects the knob on the Magni. However, I believe the sensitivity of the Paradox is less than the sensitivity listed in the thread for Fostex T50rp by direct comparison to the AKG 240s (My Paradox (modded T50rp) headphones are noticeably less sensitive than my pair of stock AKG240s, contrary to the listed data).    
  
 I played the files through foobar2000 using WINE on the centos system. Using the ABX tool, I tested whether I could differentiate the  between 16bit and 24bit empty files at a specific volume setting. 30ish trials were conducted per ABX test. Then, after the ABX test, I listened to King Crimson's Lizard without adjusting the volume.
  
 Next, I listened to the Telarc recording of Tchaikovsky's 1812 overture (and adjusting volume (lower) to accommodate the cannon blasts). Afterward, using these volume settings, I repeated the ABX of the 16bit and 24bit empty files at this lower  volume setting.
  
  
 Results:
 Here is my foobar ABX to show whether or not I can likely discriminate between the two files: 
  


Spoiler: Warning: Spoiler!



foo_abx 1.3.4 report
 foobar2000 v1.2.9
 2014/05/05 23:40:48
  
 File A: Z:\home\r*n\Downloads\30 seconds of silence (1).wav
 File B: Z:\home\r*n\Downloads\24bit silence.wav
  
 23:40:48 : Test started.
 23:40:58 : 01/01  50.0%
 23:41:10 : 02/02  25.0%
 23:41:16 : 03/03  12.5%
 23:41:19 : 04/04  6.3%
 23:41:23 : 05/05  3.1%
 23:41:27 : 06/06  1.6%
 23:41:36 : 07/07  0.8%
 23:41:43 : 08/08  0.4%
 23:41:47 : 09/09  0.2%
 23:41:51 : 10/10  0.1%
 23:41:59 : 11/11  0.0%
 23:42:04 : 12/12  0.0%
 23:42:17 : 13/13  0.0%
 23:42:22 : 14/14  0.0%
 23:42:25 : 15/15  0.0%
 23:42:32 : 16/16  0.0%
 23:42:35 : 17/17  0.0%
 23:42:40 : 18/18  0.0%
 23:42:45 : 19/19  0.0%
 23:42:47 : 20/20  0.0%
 23:42:51 : 21/21  0.0%
 23:42:54 : 22/22  0.0%
 23:42:58 : 23/23  0.0%
 23:43:02 : 24/24  0.0%
 23:43:05 : 25/25  0.0%
 23:43:08 : 26/26  0.0%
 23:43:12 : 27/27  0.0%
 23:43:15 : 28/28  0.0%
 23:43:18 : 29/29  0.0%
 23:43:28 : 30/30  0.0%
 23:43:39 : Test finished.
  
  ---------- 
 Total: 30/30 (0.0%)


  
  
 WIthout touching the volume settings, I played King Crimson's Lizard (16/44.1). According to replayGain, the track has a peak level of 0.828 (~ -2dB) from an album with a peak of 0.99 (~0 dB). The noise in the intro sounds louder than the dither noise in the empty 16bit file, and I'm of the opinion it is tape noise (The album was recorded in 1970). At this volume setting it's definitely loud and not something I do every day, but sometimes I want to play my music loud, feel the bass, and get lost for half and hour. This is not the first time I've listened to this song at this volume. 
  
 Another track I'm fond of is the Telarc recording of the 1812 overture (24/88.2). This is a more modern recording, and the background noise is quite low, and I believe it is due to the room noise picked up by the microphones. Here, the cannons are a bit louder, especially at the end, so when I relive the climax of caddyshack, i turned the volume down a bit. According to replay gain, the track and album peak level is 0.999941 ( ~0 dB ). Here Magni was set at 12:30. Afterward, I repeated the ABX at this lower volume setting:


Spoiler: Warning: Spoiler!



foo_abx 1.3.4 report
 foobar2000 v1.2.9
 2014/05/06 00:21:24
  
 File A: Z:\home\r*n\Downloads\30 seconds of silence (1).wav
 File B: Z:\home\r*n\Downloads\24bit silence.wav
  
 00:21:24 : Test started.
 00:21:38 : 01/01  50.0%
 00:21:45 : 02/02  25.0%
 00:21:52 : 03/03  12.5%
 00:22:02 : 04/04  6.3%
 00:22:12 : 05/05  3.1%
 00:22:19 : 06/06  1.6%
 00:22:28 : 07/07  0.8%
 00:22:38 : 08/08  0.4%
 00:22:42 : 09/09  0.2%
 00:22:45 : 10/10  0.1%
 00:22:57 : 11/11  0.0%
 00:23:08 : 12/12  0.0%
 00:23:25 : 13/13  0.0%
 00:23:29 : 14/14  0.0%
 00:23:40 : 15/15  0.0%
 00:23:49 : 16/16  0.0%
 00:25:06 : 17/17  0.0%
 00:25:23 : 18/18  0.0%
 00:25:29 : 19/19  0.0%
 00:25:44 : 20/20  0.0%
 00:25:56 : 21/21  0.0%
 00:26:08 : 22/22  0.0%
 00:26:12 : 23/23  0.0%
 00:26:23 : 23/24  0.0%
 00:26:27 : 24/25  0.0%
 00:26:37 : 25/26  0.0%
 00:26:49 : 26/27  0.0%
 00:27:01 : 27/28  0.0%
 00:29:50 : 27/29  0.0%
 00:30:08 : 28/30  0.0%
 00:30:37 : 29/31  0.0%
 00:30:47 : Test finished.
  
  ---------- 
 Total: 29/31 (0.0%)


  
 Conclusions:
 I just wanted to point out that there are some specific, worst-case, cherry-pickable examples, where one might find the noise floor detectable and be able to listen to a track at that volume setting. If King Crimson's LIzard were recorded in a modern studio, it is plausible that the 16bit/44.1k dither noise floor could be detectable during the quietest parts of the track. It would take a track from an album with big swings in dynamics, and would only be detectable in the quietest passages while listening at otherwise very loud levels.
  
 I'm not arguing for folks to buy hires audio. I'm asking you guys of the sound science forum to make more rigorous arguments, or to use the appropriate qualifiers on your statements, e.g., "for all practical purposes, 24 bit is unnecessary for listening" is a much better statement than "It's impossible for anyone to hear beyond 16 bit/44.1kHz". Even though redbook CD is dam near perfect for just about all audio playback situations, it only takes one single counter example, where it's slightly less than perfect to make the absolutist blanket statement false.
  
 Let me clarify how hard the noise floor of the 16 bit track is to hear: It's really hard to hear. I didn't notice the noise until I compared it directly against the 24 bit file using the instant ABX switcher. Also, it is pretty quite here in this small midwestern city, in the middle of the night, in my office which is half under ground, in my house which is set back up and off the street. While doing the ABX, I had to pause and wait for a train to pass, and a plane to pass. It's not worth worrying about. In both tracks that I listened to, the recording noise exceeds the noise floor of the file. 
  
 Cheers


----------



## bigshot

Analogue noise floors and room tones are going to be higher than digital noise floors. I don't know how you differentiate between them without creating a PCM file of silence to test with.


----------



## ab initio

bigshot said:


> Analogue noise floors and room tones are going to be higher than digital noise floors. I don't know how you differentiate between them without creating a PCM file of silence to test with.


 
 kraken created a silent digital PCM file with only the dither noise in both 16 and 24 bit depths. He linked them in this thread a few posts back. These are the tracks i ABX'd before listening to some music at the same volume settings.
  
 Of the two songs i listened to, the first was recorded in the fall of 1970. The noise was analog---I'm guessing tape. The second was a recorded performance, and im pretty certain the noise is room tones.
  
 Cheers.


----------



## bigshot

You turned up the silent track until you could hear the 16 bit noise floor, then played music at the same volume? Can you still hear? That must have been hella loud!


----------



## kraken2109

bigshot said:


> You turned up the silent track until you could hear the 16 bit noise floor, then played music at the same volume? Can you still hear? That must have been hella loud!


 

 Try it for yourself bigshot, I'd like some more opinions.


----------



## Currawong

Guys, how about we don't ruin the interesting discussion and experiments with nonsense?
  
 I've added Paul Graham's "How to Disagree" to the forum sticky as a reminder.


----------



## Agharta

Can someone tell me what a noise floor is? It sounds pretty cool, like an East German underground club or something.


----------



## kraken2109

agharta said:


> Can someone tell me what a noise floor is? It sounds pretty cool, like an East German underground club or something.


 
 Noise floor refers to the level where the noise resides in the signal, obviously lower is better. In digital audio the noise floor often mainly comes from dither because it sounds better than quantisation distortion.
  
 http://en.wikipedia.org/wiki/Noise_floor


----------



## cjl

bigshot said:


> It's not a video. It's an Apple Lossless file. Play it in iTunes.


 
 For people who run Windows and don't want to install any Apple software (like me), the latest version of VLC media player will play it just fine as well, though older versions (even fairly recent older versions) have trouble for some reason.
  
 As for the second half of that file? It isn't silent. It's filled with high frequency tones. I had to turn it down (from the level at which I listened comfortably to the first half) after 10 seconds or so because it was starting to give me a headache. That having been said, I doubt it would be noticeable if you removed that part from the original, since the frequency spectrum of that second half peaks at 17-19kHz, at a level ~40dB down from the main content. Also, it looks like your filter cut it off at more like 14-15kHz rather than 12 (if not even a bit higher - the spectrum of the second half rolls off below 17kHz, while the spectrum of the first half shows substantially more 15kHz content than 17, and more 13kHz content than either), which dramatically changes the audibility, since hearing sensitivity drops off pretty steeply in that upper octave of hearing range.


----------



## Agharta

Has anyone ever complained in real life about the noise floor on their music? That seems a bit sad.


----------



## cjl

agharta said:


> Has anyone ever complained in real life about the noise floor on their music? That seems a bit sad.


 
 I have, definitely.
  
 (Admittedly, I was listening to a cassette at the time...)


----------



## cjl

Here's Bigshot's file from the last page, except converted to flac, and with the second half removed and replaced with an extremely steep high pass filtered version of the first half (-120dB at 11,500Hz, -90dB at 11,900Hz, -0.5dB at 12000Hz).
  
 https://dl.dropboxusercontent.com/u/40020825/freqresponsetest.flac


----------



## bigshot

cjl said:


> As for the second half of that file? It isn't silent. It's filled with high frequency tones. I had to turn it down (from the level at which I listened comfortably to the first half) after 10 seconds or so because it was starting to give me a headache.




Let me give you a little wake up call regarding relative proportions...

Look at the waveform. Do you see the difference between the size of the first half and the second? That gives you an idea of how much sound exists up above 12kHz. It isn't empty. You can crank the heck out of the volume and blow your ears out on those frequencies, but at normal listening volume, you ain't going to hear that stuff, particularly buried under the rest of the spectrum.

My illustration was intended to point out how unimportant the frequencies above 12kHz are to listening to music at normal listening levels, not to deny that those frequencies exist altogether. Some people think that the octave above 10kHz contributes to "treble". Not even close. Having some kind of understanding about how different frequencies contribute to music makes it a lot easier to judge quality improvements. Some things mattter a lot. Some things don't mean jack diddly.

One other quick note. Everyone talks about 12kHz and 14kHz and 16kHz as if they are miles apart. That range is less than three notes difference. "Do Re Me". 10kHz to 20kHz is about one octave. It accounts for about 10% of the full range of human hearing, and the least important part of what we hear by far.


----------



## bigshot

agharta said:


> Has anyone ever complained in real life about the noise floor on their music? That seems a bit sad.




It was an issue in the analogue age. Keeping the noise floor down low required Dolby filtering for cassette tapes, and LPs had a noise floor much higher than anything we're discussing here.


----------



## ab initio

bigshot said:


> It was an issue in the analogue age. Keeping the noise floor down low required Dolby filtering for cassette tapes, and* LPs had a noise floor much higher than anything we're discussing here*.


 
 They still do! 
	

	
	
		
		

		
			




  
 Cheers


----------



## bigshot

The irony is that the audiophools who argue that 96dB of dynamic range in digital isn't enough are the same ones who happily sit down and listen to LPs with dynamic ranges that are a fraction of that.

Numbers in the abstract aren't helpful. They need to be put in context.


----------



## Thad-E-Ginathom

Yes, those who still sit down and listen to vinyl (rather than those who are new to it and blown away by the romance) are very used to the noise floor.
  
 In the digital world, I am think it a bit odd when I hear claims like, "I changed this blah or that blahblah and noticed in immediate lowering of the noise floor." If I could hear the noise floor at all (at least at non-injurious levels) I'd think my equipment was faulty.
  
 The numbers game is something I have never been very good at, and it was only a few days back that I actually realised how close (in "notes") those bigger Hz number are: every octave is doubling of frequency. 24khz to 48khz is just one octave; 48khz to 96khz is one more, and 96khz to 192khz just one more. Or would be, if we _could_ hear them.


----------



## cjl

bigshot said:


> Let me give you a little wake up call regarding relative proportions...
> 
> Look at the waveform. Do you see the difference between the size of the first half and the second? That gives you an idea of how much sound exists up above 12kHz. It isn't empty. You can crank the heck out of the volume and blow your ears out on those frequencies, but at normal listening volume, you ain't going to hear that stuff, particularly buried under the rest of the spectrum.
> 
> ...


 
  
 1) The waveform in that image is on a linear scale, but your hearing is on a log scale. You can't just point at it and say "look, the wave is small so it's clearly inaudible"
  
 2) Above 10kHz does contribute to treble. It's the extreme high end of the range, admittedly, but it is quite audible if there is something wrong in that frequency range
  
 3) 12k, 14k, and 16k are very different from a standpoint of audibility - even though they are only 3 notes (or less) apart, the steepness of the rolloff of human perception means that a 12kHz cut will be very audible in some music, while a 16kHz cut would be nearly inaudible in nearly all music.
  
 Here's an image of my file's waveform (with the correct 12k high pass filtering in the second half) on a log scale - you can clearly see the amount of high frequency content present in the second half. I'd also like to point out that you chose a sample (intentionally or not) that has relatively little high frequency content - I could relatively easily find several pieces of music that have substantially more content above 10kHz (and where it would be much more audible to cut all frequencies above 12k).


----------



## cjl

For example, here's another file processed the same way, with the second half a copy of the first with a sharp highpass at 12k (and this is even in a linear scale, not a log scale, so there's a LOT of high frequency content in this file...). I also took this same file and did the opposite (hard low pass filter at 12k) and listened to it, and it is VERY obvious with this example. I can do a full ABX and post logs if anyone wants, but it's a pretty obvious difference. I didn't try moving the cutoff higher to see where it becomes inaudible, but I suspect given my hearing, that level would be somewhere in the 16kHz range or so (maybe even a bit higher - this file has a lot of high frequency content in it).


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## bigshot

You keep applying absolutes to what I am saying. I'm not saying frequencies above 12kHz are not present in music or not audible at all. What I am saying is that they are unimportant to the enjoyment of music.

It's above the fundamentals and major harmonics of most musical instruments. It's far beyond the point where humans can't discern pitch any more. It's generally at a very low level compared to other frequencies, making it susceptible to masking. It's at the bleeding edge of human hearing where just about everyone's hearing starts to roll off. Without a direct A/B comparison, it's not likely anyone would even notice it missing.

Even the sub bass (20 to 40Hz) which is poorly served in many home stereos is more important than the octave above the treble range (10-20kHz).

People put WAY too much emphasis on numbers and not enough on sound. 12-14kHz sounds really nice and big and important on paper, but iin practice, it is barely significant at all. The only way to think this stuff is important is to keep looking at numbers on a page and not translate that into real world sound.

The most important frequencies to the reproduction of music are the ones covered by Fletcher Munson. Next in importance are the mids below that. Then the bass and treble at the ends. Then the sub bass. Dead last is the top octave. It accounts for 10% of the range and far less than that of music.


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## cjl

bigshot said:


> You keep applying absolutes to what I am saying. I'm not saying frequencies above 12kHz are not present in music or not audible at all. What I am saying is that they are unimportant to the enjoyment of music.
> 
> It's above the fundamentals and major harmonics of most musical instruments. It's far beyond the point where humans can't discern pitch any more. It's generally at a very low level compared to other frequencies, making it susceptible to masking. It's at the bleeding edge of human hearing where just about everyone's hearing starts to roll off. Without a direct A/B comparison, it's not likely anyone would even notice it missing.


 
 But that's precisely what I'm saying - in some music, it is GLARINGLY obvious if you apply a cut above 12kHz. It isn't subtle. In the song you posted, it would be pretty subtle, but that's more a function of the choice of music than anything else. I would not consider a system high-fidelity unless it could accurately reproduce up to at least 16kHz or so (preferably 20), because given certain musical samples, the difference is extremely obvious, and the >12kHz content is NOT at a low level even relative to other frequencies (and important to the enjoyment of the music, in my opinion). I'll post a sample file illustrating what I mean in just a minute...


----------



## ab initio

> I'll post a sample file illustrating what I mean in just a minute...


----------



## cjl

OK, here are two samples of the beginning minute or so of a song. One has been brickwalled at 12kHz, one is the original flac. I bet more than 90% of people could easily abx these two files (the 10% being those with significant high frequency hearing loss).
  
 https://dl.dropboxusercontent.com/u/40020825/Sample1_frtest.flac
 https://dl.dropboxusercontent.com/u/40020825/Sample2_frtest.flac
  
 EDIT: Here's two more samples (treated the same way - one is untouched, one lowpassed at 12kHz) from the same song, just to give a couple more examples...
  
 https://dl.dropboxusercontent.com/u/40020825/Sample3_frtest.flac
 https://dl.dropboxusercontent.com/u/40020825/Sample4_frtest.flac


----------



## cjl

Oh, and for what it's worth...
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.2
 2014/05/06 15:28:58
 File A: C:\Music\Sample1_frtest.flac
 File B: C:\Music\Sample2_frtest.flac
 15:28:58 : Test started.
 15:29:29 : 01/01  50.0%
 15:29:52 : 02/02  25.0%
 15:30:01 : 03/03  12.5%
 15:30:19 : 04/04  6.3%
 15:30:46 : 05/05  3.1%
 15:31:00 : 06/06  1.6%
 15:31:15 : 07/07  0.8%
 15:31:25 : 08/08  0.4%
 15:31:31 : 09/09  0.2%
 15:31:42 : 10/10  0.1%
 15:31:53 : 11/11  0.0%
 15:32:16 : 12/12  0.0%
 15:32:21 : 13/13  0.0%
 15:32:24 : 14/14  0.0%
 15:32:28 : 15/15  0.0%
 15:32:31 : 16/16  0.0%
 15:32:34 : 17/17  0.0%
 15:32:37 : 18/18  0.0%
 15:32:40 : 19/19  0.0%
 15:32:46 : 20/20  0.0%
 15:32:57 : 21/21  0.0%
 15:33:05 : 22/22  0.0%
 15:33:09 : 23/23  0.0%
 15:33:13 : 24/24  0.0%
 15:33:16 : 25/25  0.0%
 15:33:18 : 26/26  0.0%
 15:33:20 : 27/27  0.0%
 15:33:23 : 28/28  0.0%
 15:33:27 : 29/29  0.0%
 15:33:30 : 30/30  0.0%
 15:33:45 : Test finished.
  ----------
 Total: 30/30 (0.0%)
  
 (Note the timestamps - the difference isn't subtle, so I could determine which was which within a matter of a couple of seconds for the most part, usually without even playing the reference files)


----------



## bigshot

I'm afraid I can't play Flac files.
  
 But again, I'm talking about relative importance. I don't seem to be able to get that across to you. A double blind test completely misses the point of what I'm saying.


----------



## cjl

I know you're talking about relative importance, but you keep asserting quite enthusiastically that the upper octave doesn't contribute much to the music, and I strongly disagree. You keep acting as though your own aesthetic preference and thoughts on which factors matter is the only correct way to set up a system, and because you don't believe that upper octave is important, it clearly must not be important to anyone, with any genre or type of music. All I'm trying to show is that for certain music types and sound samples, this is clearly incorrect.
  
 As for the double blind test, it shows that the difference between the two files is clearly audible (especially when you account for how quickly I was able to discern the difference). Once audibility has been shown, it largely comes down to preference (including preference in music style). Honestly, I'd even dispute your assertion that the uppermost octave (10-20kHz) is less important than the lowermost octave (20-40Hz) - I'd rather have a system with rolled off bass below 40Hz, but good treble response up to 20kHz than a system with rolled off treble at 10k but perfect bass. Of course, ideally, you'd have a system with good response 20-20k.
  
 (Oh, and if you want a way to play flac files, VLC works quite nicely, and is compatible with Apple, Windows, and most versions of Linux... http://www.videolan.org/vlc/index.html)


----------



## ComradeDylie

bigshot said:


> No it was the whole idea that inaudible frequencies can make music sound better. How are they going to sound at all when they're inaudible?!
> 
> By the way, there have been studies that have shown that super-audible frequencies have absolutely no impact on sound quality. In fact, music doesn't really contain any, because the only musical instruments capable of producing them are cymbals and triangles, and in those, auditory masking from lower harmonics totally obliterates the ones in the upper range of hearing. Inaudible frequencies would be doubly inaudible!
> 
> ...


 
 Oh..thats not circular at all.  I didnt say better.  I said more natural.  Perhaps it would be better to say "feel" instead of sound.  Or that we would 'perceive' them to be more natural.  Whatever.  You clearly have old man hearing loss if you cant hear those high frequencies. Me and that other dude can.  Sorry bro, for your sound blindness =(.  lol kidding.  
  
 Can we get a piccolo, chimes, and cymbal only track up in here?  Try cutting out the high freqs out of that and tell me if it sounds right...oh wait...sound blind...
  
 And you know what else we dont need besides frequencies over 12khz?  Letters past W.  I mean who even uses them?  We only need the first 23.
  
 I like how we are talking about bit depth and sampling rates and if it makes a difference.  And then now we are on whether or not for sure, 100% _*AUDIBLE*_ frequencies matter or not...oh no...
  
 On the matter of relative importance:
  
 Relative importance of posting on head-fi compared to anything that actually matters:  nil.  But here we are.  So....exercise?  Grow a garden?  Develop new clean fuels? Learn something new?  Play with your kids?  Cook dinner instead of eating McDonalds...again!? Brush your teeth?  Vacuum?  Dust?  Help cure diseases?  Idk...its all relatively important stuff eh?  But here we are.


----------



## Xenophon

cjl said:


> OK, here are two samples of the beginning minute or so of a song. One has been brickwalled at 12kHz, one is the original flac. I bet more than 90% of people could easily abx these two files (the 10% being those with significant high frequency hearing loss).
> 
> https://dl.dropboxusercontent.com/u/40020825/Sample1_frtest.flac
> https://dl.dropboxusercontent.com/u/40020825/Sample2_frtest.flac
> ...


 
 Indeed easy, not a subtle difference.  At least in this sample it's clear that removing the >12kHz frequencies makes the music somehow soulless.


----------



## ab initio

Bigshot,
  
 How much do you think that one's personal preferences in music affect what aspects of sound reproduction they value the most? If I understand correctly, you are particularly fond of classical music, which, I think, tends to have much less content in the highest octave of human hearing, than, say, modern rock or metal, which feature much more percussion (notably the cymbal crashes).
  


bigshot said:


> The most important frequencies to the reproduction of music are the ones covered by Fletcher Munson. Next in importance are the mids below that. Then the bass and treble at the ends. Then the sub bass. Dead last is the top octave. It accounts for 10% of the range and far less than that of music.


 

 That said, I definitely agree with this ranking of relative importance. For reference, I listen to music which varies from late baroque through modern classical to 60s and 70s classic rock and progressive rock through modern progressive rock, progressive metal, and hard rock. I don't listen to really any pop, hip-hop, country, or electronic anything. I don't know how those genre's might affect my priorities for sound reproduction. Maybe folks who favor pop, hip hop, and/or electronic dance type genre's might place bass quantity (and quality? maybe not?) higher up in their priorities? I'm not sure if there are electronic genre's that would use oodles of super high frequency synthesized sounds? I would find that painful, but the dog might rock out to that.
  
 It's sort of funny (at least, I get a huge kick out of the fact) that the least important frequencies in music are the ones that make music files so stinking big. It's the highest octaves that take up all the disk space, just for the sake of satisfying Nyquist theorem for those frequencies. Every extra octave makes the file double in size. 
  
 Because this thread is lacking in sample calculations (at least to my satisfaction), let's do one:
  
 If we assume that we can hear from 20Hz to 20kHz, then we can hear a 10 octave range. The octaves are (in Hz):

```
Octave: Start: End: Nyquist:            1          20          40          80            2          40          80         160            3          80         160         320            4         160         320         640            5         320         640        1280            6         640        1280        2560            7        1280        2560        5120            8        2560        5120       10240            9        5120       10240       20480           10       10240       20480       40960
```
  
 Every time another octave needs to be captured in a recording, the resulting file size doubles. For the less technically inclined (do they every visit this forum?) we call a doubling in size with a unit increase "exponential growth", which is the the mathy way of saying something gets awfully expensive in a hurry. The theoretical minimum size of an uncompressed PCM file (per second) is the [Nyquist frequency] times [the bitdepth of the recording] times [the number of channels e.g., 2 for stereo]_._
  
 If we ignored the top octave, and only recorded music up to 11025 Hz (i.e., sampled at 22.05 kHz), we would cut down the file size by 50%. If the music is band limited below 11kHz, then the recording  with sampling rate of 22050 Hz would still have _exactly the same fidelity_ as a recording at 44100 Hz (or 88.2kHz, or 192 kHz, or 32 million THz, or whatever 30x super DSD is, etc.). This is throwing out 1 octave out of 10, i.e., the 10% of the range that bigshot mentioned above. Adding additional octaves beyond 20kHz extends the "musical range" by only a small fraction (_"musical range" is in quotes, because nobody can even hear those frequencies_) at the expense of doubling the file size for each extra inaudible octave.
  
 176.4 kHz audio increases the octave range of the recording by 20% over the 44.1kHz version, _all of it inaudible__ to humans_, at the expense of increasing the file size to 400%. Your pet bats will thank you, though!
  
 This is not a "glass is half full" vs "half empty" kind of thing. (well, one might say that 96kHz recordings are half full). For HiRes formats, this is a "glass is twice as big as they need to be (or four times, etc..)" kind of thing.
  
 I recommend that we change the term "High Resolution audio" to "Highly Inefficient audio". I bet it will sell even better! You could shorten it to "High-In" audio. I wonder how this will go over in the other forums.
  
 Cheers


----------



## cjl

ab initio said:


> It's sort of funny (at least, I get a huge kick out of the fact) that the least important frequencies in music are the ones that make music files so stinking big. It's the highest octaves that take up all the disk space, just for the sake of satisfying Nyquist theorem for those frequencies. Every extra octave makes the file double in size.
> 
> Because this thread is lacking in sample calculations (at least to my satisfaction), let's do one:
> 
> ...


 
 Interestingly, in my samples above, you can somewhat see this effect - they're all 16 bit, 44,100Hz files, but the ones lacking the >12kHz content compress into a FLAC much better because of the missing high frequency information. The low pass filtered files are around 85-90% the size of the full files, since the high frequency information is difficult to compress. They would be even smaller if I resampled them at 25kHz or so after applying the low pass filter (which would still perfectly preserve the 0-12kHz information).


----------



## bigshot

comradedylie said:


> And you know what else we dont need besides frequencies over 12khz?  Letters past W.  I mean who even uses them?  We only need the first 23.


 
  
 That is actually a perfect example. The top octave is to the rest of the frequency spectrum as the letters X, Q and Z are to the alphabet. Approximately 10% of the alphabet, but responsible for a tiny percentage of words.
  
 Most headphones and speakers don't perform well on the outer octaves of the spectrum of human hearing, neither in extension to the outer limits, nor in balance to the rest of the spectrum. Every day we listen to band limited response on TV, car stereos, portable stereos and radios and we don't complain. Is it nice to have balanced response all the way from 20Hz to 20kHz? Sure, why not? But it better not be a deal breaker for you or you won't find much to listen to until you get that USB port put in your skull! But the important thing is, if you can get balanced response within the range of 40Hz to 10kHz, you've gone the lion's share of the way to getting really good sound. I'd much rather have that than unbalanced with extension all the way out.
  
 Many audiophiles spend all their time worrying about the specks of dust and ignore the elephant sitting in the corner. Relativity is an important thing to have, but it's in short supply in audiophile forums, that's for sure! It might be a psychological thing.


----------



## cjl

bigshot said:


> That is actually a perfect example. The top octave is to the rest of the frequency spectrum as the letters X, Q and Z are to the alphabet. Approximately 10% of the alphabet, but responsible for a tiny percentage of words.


 
  
 And yet, despite not being present all the time, they are clearly and demonstrably necessary for an accurate reproduction of a text (or a piece of music).


----------



## Xenophon

ab initio said:


> Bigshot,
> 
> How much do you think that one's personal preferences in music affect what aspects of sound reproduction they value the most? If I understand correctly, you are particularly fond of classical music, which, I think, tends to have much less content in the highest octave of human hearing, than, say, modern rock or metal, which feature much more percussion (notably the cymbal crashes).


 
 Depends on the piece.  I also listen exclusively to classical and while it's true that if I put my DSP viewer in JRiver to display everything between 9-16 kHz, for many pieces it stays eerily...black.  But not for all by a long shot.  For instance I'm now listening to Albinoni's concerto strings Opus 9 , number 4 in A major (admittedly a somewhat disingenuous choice)  and there's still quite some activity between 8-14 kHz (visual estimate).  If I have the time later today I'll take a couple of pieces from it, apply filters and post them so you can hear for yourself.


----------



## bigshot

cjl said:


> And yet, despite not being present all the time, they are clearly and demonstrably necessary for an accurate reproduction of a text (or a piece of music).


 
  
 Absolute again. I'm speaking in practical terms, not armchair theory. We live in an imperfect world with imperfect transducers. You can get close as damn it in the core frequencies and it makes a big difference. And you can get close as damn it in the outside edges of human hearing and it doesn't make much difference at all. You can't have 100% of 100% all the time. You pick your battles. But it really helps if you have some perspective to choose which battles to fight and which aren't worth it. It's all relative.
  
 I'm talking about stereo systems to listen to music on here by the way.


----------



## bigshot

ab initio said:


> How much do you think that one's personal preferences in music affect what aspects of sound reproduction they value the most? If I understand correctly, you are particularly fond of classical music, which, I think, tends to have much less content in the highest octave of human hearing, than, say, modern rock or metal, which feature much more percussion (notably the cymbal crashes).


 
  
 I listen to a lot of different kinds of music, not just classical. Fidelity is fidelity, I suppose.
  
 But I'm not talking fidelity in an abstract sense... I'm talking about relative importance... and that is the concept that isn't getting across. I thought the alphabet analogy was perfect, but it whooshed right on by.
  
 One of the big problems I see in audiophila (and it's just as bad in the Sound Science forum as it is in the rest of Head Fi) is the tendency to look at numbers on a page and not relate them to sound. If I say "take a 1kHz tone at 45dB" how many people do you think would have an idea of what that would sound like? Would they know that it would drive them right out of the room, or would they nod their head in theoretical ignorance and say, "yes... yes..."? I see people going to the mat fighting about frequencies on a headphone chart that really don't mean much at all... and then totally overlook the remarkable performance of the same headphones through the entire range that matters. People buy headphones with V shaped response or "in your face" midrange and worry about the part of the chart all the way up at the very top. That makes no sense at all to me.
  
 Because somewhere in a book it says "humans hear from 20Hz to 20kHz", that doesn't mean that every single frequency in that range is just as important to a guy sitting in a chair listening to music. But the OCD cuts in and people start worrying about molehills while the mountains go unattended. The perfect example of this is the old saw, "Audiophiles worry about the last few percent." Yes, that is undeniably true... so true in fact that they usually have great performance where it doesn't matter and mediocre performance where it does!
  
 There are ways of prioritizing improvements in sound quality that don't involve factoring out to more and more decimal points. For instance, all things being equal, simply going from 2 channel stereo to 5:1 is a MASSIVE leap in sound quality. Add a good DSP to that and you are squaring that huge improvement. But people are more interested in speaking about noise floors or distortion at -90dB being theoretically audible. They worry about frequencies they won't even be able to hear in ten years! There are things about sound reproduction that most people don't even begin to address. Most of that involves room acoustics and sound fields.
  
 Crazy town. I just want music to sound realistic. That isn't hard, and it doesn't depend on frequencies I can barely hear. In fact, I can get spine tinglingly realistic sound out of a 1914 Caruso record and my upright Victrola. I'll give you a hint how that is... My Victrola has a horn designed to project sound in a specific direction using the shape of the room to put a real dimensional acoustic envelope around the recorded singer. That can have more of an impact on realism than the drastically reduced frequency response of an acoustic recording.
  
 I know these are things that aren't published as specs or discussed in hifi forums. But it's all relative to the degree of realism you achieve with your system.


----------



## ab initio

bigshot said:


> That is actually a perfect example. The top octave is to the rest of the frequency spectrum as the letters X, Q and Z are to the alphabet. Approximately 10% of the alphabet, but responsible for a tiny percentage of words.


 

 O Fortuna,
 velut Luna
 statu variabilis,
 semper crescis
 aut decrescis;
 vita detestabilis
 nunc obdurat
 et tunc curat
 ludo mentis aciem;
 egestatem,
 potestatem,
 dissolvit ut glaciem.

 Sors immanis
 et inanis,
 rota tu volubilis,
 status malus,
 vana salus
 semper dissolubilis;
 obumbrata
 et velata
 mihi quoque niteris;
 nunc per ludum
 dorsum nudum
 fero tui sceleris.

 Sors salutis
 et virtutis
 mihi nunc contraria;
 est affectus
 et defectus
 semper in angaria.
 hac in hora
 sine mora
 cordae pulsum tangite!
quod per sortem
 sternit fortem,
 mecum omnes plangite!
  





  
 Cheers


----------



## cjl

bigshot said:


> Absolute again. I'm speaking in practical terms, not armchair theory. We live in an imperfect world with imperfect transducers. You can get close as damn it in the core frequencies and it makes a big difference. And you can get close as damn it in the outside edges of human hearing and it doesn't make much difference at all. You can't have 100% of 100% all the time. You pick your battles. But it really helps if you have some perspective to choose which battles to fight and which aren't worth it. It's all relative.
> 
> I'm talking about stereo systems to listen to music on here by the way.


 
 I'm talking in practical terms too. I'll be the first to agree with you if you say that something like 256kbps lossy compression sounds very nearly the same as lossless, or that fancy cables are pointless, or that all well-designed solid state amps sound pretty much exactly the same (or, more accurately, do sound exactly the same within the limits of the loads and power levels they are designed to output). However, here you are claiming that in practical terms, the upper octave doesn't matter, and that is simply wrong. I've already provided examples of sound where the upper octave clearly matters, and cutting off the upper 8kHz of human hearing range strongly and noticeably affects the output. This isn't a subtle thing.
  
 (Any change that makes it sound like I've thrown a heavy blanket over my speakers, even when I'm listening casually with a ton of background noise (the A/C is running right now) is not something I'd consider "impractically small")


----------



## Xenophon

Never was big on the pomp of the Carmina Burana but it reflects the mood in the discussion here I guess.


----------



## GrindingThud

Indeed: http://m.youtube.com/watch?v=leUTSH_wKos


ab initio said:


> Cheers


----------



## ab initio

I'm guessing you guys didn't see what I did there....
  
 This only goes to prove that we _don't_ need Q, X, or Z.
  
 Cheers


----------



## cjl

I like this version better...
  
 https://www.youtube.com/watch?v=nIwrgAnx6Q8


----------



## Xenophon

Yes I did (but the relative prevalence of that letter in Latin is quite high, cherry picking).


----------



## GrindingThud

I saw and it made me laugh..... I still am. 
Edit.....never saw cjl's version....now I'm laughing harder. 


ab initio said:


> I'm guessing you guys didn't see what I did there....
> 
> This only goes to prove that we _don't_ need Q, X, or Z.
> 
> Cheers


----------



## bigshot

It seems to me, if you can do this without X, Q or Z then you don't need them at all!
  
 `Twas brillig, and the slithy toves
   Did gyre and gimble in the wabe:
 All mimsy were the borogoves,
   And the mome raths outgrabe.

 "Beware the Jabberwock, my son!
   The jaws that bite, the claws that catch!
 Beware the Jubjub bird, and shun
   The frumious Bandersnatch!"
  
 He took his vorpal sword in hand:
   Long time the manxome foe he sought --
 So rested he by the Tumtum tree,
   And stood awhile in thought.
  
 And, as in uffish thought he stood,
   The Jabberwock, with eyes of flame,
 Came whiffling through the tulgey wood,
   And burbled as it came!
  
 One, two! One, two! And through and through
   The vorpal blade went snicker-snack!
 He left it dead, and with its head
   He went galumphing back.
  
 "And, has thou slain the Jabberwock?
   Come to my arms, my beamish boy!
 O frabjous day! Callooh! Callay!'
   He chortled in his joy.


----------



## bigshot

cjl said:


> I'm talking in practical terms too. I'll be the first to agree with you if you say that something like 256kbps lossy compression sounds very nearly the same as lossless, or that fancy cables are pointless, or that all well-designed solid state amps sound pretty much exactly the same (or, more accurately, do sound exactly the same within the limits of the loads and power levels they are designed to output).


 
  
 OK. Well... you can have the practical side in those. I'm happy to go absolute on all of those things... AAC 256 sounds exactly the same as lossless, fancy cables are a total waste and all modern solid state amps sound the same.


----------



## limpidglitch

bigshot said:


> It seems to me, if you can do this without X, Q or Z then you don't need them at all!
> 
> `Twas brillig, and the slithy toves
> Did gyre and gimble in the wabe:
> ...


 
  
 My man!


----------



## ComradeDylie

Im glad my alphabet thing hit the spot, I was pretty proud of that one I thought it was spot on lol.
  
 Quote:


bigshot said:


> OK. Well... you can have the practical side in those. I'm happy to go absolute on all of those things... AAC 256 sounds exactly the same as lossless, fancy cables are a total waste and all modern solid state amps sound the same.


 
 Whoa whoa whoa.  Fancy cables are a waste?  Thats news to me.  Sounds like blasphemy.
  
 I agree to a point.  I rather have a nice flexible fancy cable that is durable and well sleeved and soft and comfortable and doesnt clink a whole lot than some POS cable that is going to fall apart after 3 months (Apple, Im looking at you).  Does it sound better?  Nah, I doubt it.  Are they electrically superior sure, doesnt mean you can hear it.  But a *total* waste? Surely not.  A good cable can be good for other non 'how it sounds' reasons that can matter (electrical superiority not being one).  
  
 I didnt think all amps sounded the same, just amps with matching/extremely similar transfer functions.  I havent really considered it though.  But as far as I am aware a tube amp with the same transfer function as a solid state amp would sound the same?  Are we saying all modern solid state amps have pretty much the same transfer function?  Are we saying SS amp A that puts out 1/10 the power of SS amp B (but is 25 times smaller...) will sound the same when listened to on an HE-6?  
  
 (Please read the amp part with full on deadly seriousness, no joking in there)
  
 Really guys 256?  Not even 320?  Well if thats true then...iTunes...holy crap..iTunes?!? did it right?  
  
 I think the file size point on the upper octave is the most valid made on its relative worthlessness.  And the biggest detractor for Hi-Res er I mean Hi-N music.  The biggest case for lossy files etc.  I mean if we drop on octave and use 256...damn I mean I could literally put 12 times more music on my iPod.  That rules, plain and simple.  
  
 5.1 vs Stereo.  Yeah its cool and all but stereo still sounds better in 2 channel i think.  Weeeeell, actually, the MCH Stereo DSP sounds really good but I dont care for DTS or Dolby DSP (for music).  The center channel is extremely overworked I feel.  Music that has been mastered for 5.1 sounds better but its hard to find things that will even play it lol.  And most of the songs I like arent in 5.1.  That is huge, I mean its hard for me to say that Storm Corrosion sounds so good, when I mean Im just not a huge fan of the music at the base.  So its been pretty hard for me to judge on that. 
  
 At the end of the day I am a hardware nerd.  I mean I want the fast processor, too much memory, too much storage and then I want to make it faster.  I have literally no need to, but its fun.  I do like the cool hardware aspect of the hi-res scene, and that is why I mess with it at all.  Lets let Pono Store come out and see if the music sounds better.  Im not saying use a Pono, but if I can get Reflektor off of there (they had an Arcade Fire signed Pono....) then I will.  Not because I think that the 24/96 part will make it better.  But as many people have stated SACDs often used completely different and superior masters, this is what I am hoping comes from Pono.  Really good, much less compressed, masters would be a god send.  If I have to have a 24/96 file to get that, then so be it.  I can always down sample it and run it through an AAC encoder lol.


----------



## ThePianoMan

I think 24/96 or even 192 is currently the only way to get better masters. I have heard well-mastered CD's sound amazing, even better than some DVD audio stuff. But on the flip side, I know i at least can tell a difference between many files played in plain CD and 96/192. Is it because of the higher sampling rate? It's much more LIKELY that the master used was just better quality. I know some people have reported HD tracks using lower quality masters to make the CD and mp3 sound worse. I think a lot of things would have to change in the recording industry for this to change. Because the masters, being the ultimate source of our audio files, are the area of the recording studios. It's an interesting pickle for sure...


----------



## BeyerMonster

thepianoman said:


> It's much more LIKELY that the master used was just better quality....I think a lot of things would have to change in the recording industry for this to change. Because the masters, being the ultimate source of our audio files, are the area of the recording studios.


 
 And THIS is why 99% of this thread is sadly meaningless. We're COMPLETELY at the mercy of the record companies/studios that distribute the content. While many recording engineers are likely perfectly capable and willing of churning out album after album of high quality content (musically and sonically), the people that write their checks are in the business of selling music, not producing the best quality music they can. If the "lie" of 24/192 is what it takes to get people to shell out more cash than the price of their favorite tracks on iTunes or Amazon MP3, it will happen, regardless of sonic impact... even if we end up downconverting it all to a lower resolution format such as 16/44.1 or 16/48 for our own usage/playback.
  
 My biggest fear is that 24/96 or 24/192 will end up being higher bitrate versions of the same squashed dynamics that we've become accustomed to.
 Dynamic range isn't everything, but have you guys tried looking for DR Database entries for HDTracks content? Some of it is not pretty and isn't even close to requiring the 24-bit part:
 http://dr.loudness-war.info/album/list?artist=&album=hdtracks


----------



## ThePianoMan

I am mildly curious as to where DSD fits into this, though I suspect it runs into the same basic problem of sampling size 44khz etc.
And as a trained musician myself, UCAN confirm the bit about hearing to anyone who might doubt it. Even overtones of the Tuba and Timpani, up to the highest piccolo note don't go beyond what CD quality can give you.


----------



## bigshot

comradedylie said:


> Sounds like blasphemy.


 
  
 Go check it all with controlled listening tests. You'll find out what I did. The stuff they talk about in hifi forums is usually the stuff that doesn't matter at all.
  
 P.S. Yes solid state amps. I am talking about amps that are designed *not* to have a sound. 256 AAC VBR is audibly transparent. I tested that too. And I can take my 5:1 system and dumb it down to a standard 2 channel system and it doesn't sound anywhere near as good as in 5:1. Same equipment. Same settings. Same everything. Come to LA and I will prove it to you.


----------



## esldude

thepianoman said:


> I am mildly curious as to where DSD fits into this, though I suspect it runs into the same basic problem of sampling size 44khz etc.
> And as a trained musician myself, UCAN confirm the bit about hearing to anyone who might doubt it. Even overtones of the Tuba and Timpani, up to the highest piccolo note don't go beyond what CD quality can give you.


 

 A solution in search of a problem that doesn't exist.


----------



## bigshot

Either that or marketing genius! How else you get everyone to buy Dark Side of the Moon YET AGAIN?


----------



## ThePianoMan

This thread is both uplifting and disheartening all at once, for so many reasons...


----------



## Agharta

I'm waiting till high res ear implants become available.


----------



## esldude

bigshot said:


> Either that or marketing genius! How else you get everyone to buy Dark Side of the Moon YET AGAIN?


 

 Well, you hit the nail on the head.  I owned the 8-track (yes I am getting old, listened to it mostly in a 1967 Camaro), then cassette (mostly listened in an early Honda Accord, yes I sold the Camaro and bought the Accord in its place), original issue LP, Harvest issue LP, and demo copy LP, then Mobile Fidelity LP, and finally MoFi gold Ultradisc CD.  I haven't sprung for the high rez version.  The CD from Mofi was the best followed by the Harvest LP.  MoFi Lp was good, but somehow not as nice as Harvest LP or gold CD.


----------



## Agharta

That probably explains why Neil Young decided to demo his Pono to a bunch of stoned rockers in his car and then strangely forgot to turn the engine on or drive anywhere.


----------



## cjl

comradedylie said:


> Whoa whoa whoa.  Fancy cables are a waste?  Thats news to me.  Sounds like blasphemy.
> 
> I agree to a point.  I rather have a nice flexible fancy cable that is durable and well sleeved and soft and comfortable and doesnt clink a whole lot than some POS cable that is going to fall apart after 3 months (Apple, Im looking at you).  Does it sound better?  Nah, I doubt it.  Are they electrically superior sure, doesnt mean you can hear it.  But a *total* waste? Surely not.  A good cable can be good for other non 'how it sounds' reasons that can matter (electrical superiority not being one).


 
  
 Sure, but if all you want is a reasonably well made cable that is electrically perfect (or at least perfect enough that it will not affect the signal in any perceptible way), you can get that for fairly cheap. I understand the people buying cables from places like Blue Jeans Cable for example, even though those still cost more than I would pay for cable, personally. At least they're genuinely just well-made cables that look a bit nicer than the ones from someplace like Monoprice (though they won't sound any different). The ones that I will never understand are the ones that are hundreds of dollars per foot, and claim that you will now be able to hear every heartbeat from the people in the booth of the recording studio where your recordings were made, because normal cables blur out the details (or other bogus nonsense like that).
  
  


comradedylie said:


> I didnt think all amps sounded the same, just amps with matching/extremely similar transfer functions.  I havent really considered it though.  But as far as I am aware a tube amp with the same transfer function as a solid state amp would sound the same?  Are we saying all modern solid state amps have pretty much the same transfer function?  Are we saying SS amp A that puts out 1/10 the power of SS amp B (but is 25 times smaller...) will sound the same when listened to on an HE-6?
> 
> (Please read the amp part with full on deadly seriousness, no joking in there)


 
 All modern solid state amps (both for headphones and speakers) will sound identical so long as they meet a certain set of criteria:
 1) The amp has an inaudibly low noise floor (this isn't really a problem for speaker amps, but it can be a problem for headphone amps, especially if you have sensitive, low-impedance headphones)
 2) The amp's output impedance is substantially lower (preferably by at least an order of magnitude) than the minimum impedance of the headphone/speakers being driven - again, this isn't a problem for most well-designed solid state speaker amps, but it is a problem with some headphone amps. Many tube amps also have a high output impedance, which can significantly affect the sound.
 3) The amp has both sufficient voltage swing and sufficient current sourcing capability to drive the speakers to a high level (preferably a couple dB above the maximum listening level) without significant distortion in the output signal. Once again, usually not a problem as long as the amp is designed to drive the load impedance that the speaker/headphones present, and the amp gets loud enough. This can be a problem with a few esoteric high end speakers, since the impedance drops to absurdly low levels in some speakers (which requires a lot of current to drive, and amps not designed for this enormous current requirement can have problems). If you can find an impedance curve for your speakers, you can see what the minimum impedance is, and compare it to your amp's rating. Chances are, if your speaker never drops below 4 ohms (which covers the majority of "8 ohm" speakers out there), pretty much any modern solid state amp with sufficient power can drive it just fine. If you do have an odd, high end, super low impedance set of speakers, a potential good option to drive them is actually a professional amp - many of them have significantly more current capability than amps designed for home use, without the audiophile price tag.
 4) Flat frequency response from 20-20k. Most inexpensive speaker amps do this just fine - interestingly, it's the higher end, more esoteric designs you have to watch out for (especially in headphone amps)
 5) Channel balance - usually not an issue in speaker amps, can be a problem with some headphone amps
  
 There are a couple more considerations as well,  but the interesting thing about it is, you usually don't run into any problems with low to mid range, mass-produced solid state amps driving relatively normal, well-designed speakers. Audible problems are much more likely to happen with a high end, "audiophile-oriented" design than with a normal mass produced amp. So, if you want a good accurate sound, get a mass produced, well built solid state amp. Preferably a recent one, since the extra features (especially things like room correction) have really improved in the past few years.
  


comradedylie said:


> Really guys 256?  Not even 320?  Well if thats true then...iTunes...holy crap..iTunes?!? did it right?
> 
> I think the file size point on the upper octave is the most valid made on its relative worthlessness.  And the biggest detractor for Hi-Res er I mean Hi-N music.  The biggest case for lossy files etc.  I mean if we drop on octave and use 256...damn I mean I could literally put 12 times more music on my iPod.  That rules, plain and simple.


 
 I'm not familiar with AAC, so I can't comment on it, but I can say that well-encoded 320 MP3 is completely transparent to my ear with the exception of a couple of very carefully picked samples (and even in those cases, I have to do a fast-switch comparison while listening extremely carefully to a tiny chunk of the file to hear any difference). As a result, I use 320kbps MP3 as my main listening format. I do have all of my files archived in either FLAC or WMA Lossless though (depending on when I ripped them - I originally ripped in WMA, but I have moved to FLAC for more recent files), just so I have a perfect archival copy of everything. As for the upper octave? Keep it. 256 or 320 lossy encoding will keep the majority of that high-frequency content, and will sound much, much better than a low-passed file where the top bit was discarded.
  
  


comradedylie said:


> 5.1 vs Stereo.  Yeah its cool and all but stereo still sounds better in 2 channel i think.  Weeeeell, actually, the MCH Stereo DSP sounds really good but I dont care for DTS or Dolby DSP (for music).  The center channel is extremely overworked I feel.  Music that has been mastered for 5.1 sounds better but its hard to find things that will even play it lol.  And most of the songs I like arent in 5.1.  That is huge, I mean its hard for me to say that Storm Corrosion sounds so good, when I mean Im just not a huge fan of the music at the base.  So its been pretty hard for me to judge on that.


 
  
 I actually agree with you on the 5.1 vs stereo, but I'll be the first to admit that my current listening room is pretty well set up for 2.1 (my sub is in an excellent place, as are the two mains), but my surround speakers are much less optimal (I don't have room currently to place them how I'd really like). As a result, surround content definitely isn't up to its full potential in my listening area for the time being. I'll likely redo my setup at some point when I have more space, but for the moment, it has to do. It works just fine for movies, but for music, I still prefer stereo on my setup.
  
  


comradedylie said:


> At the end of the day I am a hardware nerd.  I mean I want the fast processor, too much memory, too much storage and then I want to make it faster.  I have literally no need to, but its fun.  I do like the cool hardware aspect of the hi-res scene, and that is why I mess with it at all.  Lets let Pono Store come out and see if the music sounds better.  Im not saying use a Pono, but if I can get Reflektor off of there (they had an Arcade Fire signed Pono....) then I will.  Not because I think that the 24/96 part will make it better.  But as many people have stated SACDs often used completely different and superior masters, this is what I am hoping comes from Pono.  Really good, much less compressed, masters would be a god send.  If I have to have a 24/96 file to get that, then so be it.  I can always down sample it and run it through an AAC encoder lol.


 
 Heh - I definitely understand this attitude. I'm definitely a hardware nerd too, and I love looking at design articles, and seeing how things like the Benchmark DAC2 (for example) are designed to extract every bit of performance from modern technology, even though a lot of what is done on things like that is way, way past what is necessary for audible perfection. However, I am a little frustrated having to do this whole "I'm going to buy the high-res in the hope that they mastered it better" thing - the "low-res" formats are perfectly capable of sounding way, way better than they currently do on a lot of music, and I wish people would focus on getting the existing, perfectly adequate formats to sound as good as they can, rather than inventing new formats for no reason.


----------



## ComradeDylie

bigshot said:


> Go check it all with controlled listening tests. You'll find out what I did. The stuff they talk about in hifi forums is usually the stuff that doesn't matter at all.
> 
> P.S. Yes solid state amps. I am talking about amps that are designed *not* to have a sound. 256 AAC VBR is audibly transparent. I tested that too. And I can take my 5:1 system and dumb it down to a standard 2 channel system and it doesn't sound anywhere near as good as in 5:1. Same equipment. Same settings. Same everything. Come to LA and I will prove it to you.


 
 The blasphemy line and most everything out of my mouth has been tongue in cheek facetiousness where I actually mean the opposite lol.  Mostly I agree with you across the board (although not on the octave part, but Saint Hi-N saved you there and made it sound somewhat reasonable =). I just felt that some of your points needed further fleshing out so I try to go hyperbolic metaphorical to get someone I agree with to justify their point further in better terms.  
  
 On the cable thing, I mean I do own Silver Dragons but I got them used and it was mainly a bling thing.  Like idk a gold watch, I didnt get them thinking it would make a sound improvement.  Well, and the one I got new was only because it is pretty hard to find IEM cables terminated in 4XLR.  A lot of people think the SD is a great match for the JH13 so I did fall for that a little bit but it is a really nice cable and I like side by side coax cable design whereas most aftermarkets are the braided cable style.  
  
 On the 5.1 vs stereo thing, you are saying that you prefer to listen to a PCM track in DTS or Dolby DSP rather than the MCH Stereo DSP?  Or that you simply prefer having 5 speakers hit you instead of two?  I like the MCH Stereo DSP, and I mean since I have the 5 speakers (I got all 5 Floor Standing instead of the center and bookshelves for 3) it really pisses me off when only 2 are in use.  I think maybe the MCH Stereo DSP sounds so good for me is that all 5 speakers are the same and maybe the effect wouldnt be as good if I had opted for the standard set up.  Not sure there, but admittedly I never listen to normal stereo out of those guys just because of the waste factor.  However, I also dont ever hit the DTS or Dolby option for music ever either (anymore, now that Ive tried them out enough)


----------



## bigshot

comradedylie said:


> On the 5.1 vs stereo thing, you are saying that you prefer to listen to a PCM track in DTS or Dolby DSP rather than the MCH Stereo DSP?  Or that you simply prefer having 5 speakers hit you instead of two?  I like the MCH Stereo DSP, and I mean since I have the 5 speakers (I got all 5 Floor Standing instead of the center and bookshelves for 3) it really pisses me off when only 2 are in use.


 
  
 I use the 7:1 Stereo DSP on my Yamaha receiver. That is probably the same or similar to your MCH Stereo DSP. It improves the soundstage of 2 channel stereo and adds a pleasing liveness to the room. It also seems to take some of the curse off of being off axis to the mains.


----------



## ab initio

cjl said:


> 2) The amp's output impedance is substantially lower (preferably by at least an order of magnitude) than the minimum impedance of the headphone/speakers being driven - again, this isn't a problem for most well-designed solid state speaker amps, but it is a problem with some headphone amps. Many *tube amps also have a high output impedance*, which can significantly affect the sound.


 
 I have no idea why fancy tube amps, _especially the expensive ones_, do not have output transformers. It takes that high voltage, high impedance output, and turns it into a low impedance output suitable for driving enough current. it's not like the expensive amps are trying to save you money by omitting 10 lbs of iron core transfomer. I would have thought having the biggest, baddest output transformer would be an audiophile badge of honor, yet it seems audiophiles would much rather have their headphones operating electrically under-damped. 
	

	
	
		
		

		
			




  
  


cjl said:


> Sure, but if all you want is a reasonably well made cable that is electrically perfect (or at least perfect enough that it will not affect the signal in any perceptible way), you can get that for fairly cheap. I understand the people buying cables from places like Blue Jeans Cable for example, even though those still cost more than I would pay for cable, personally. At least they're genuinely just well-made cables that look a bit nicer than the ones from someplace like Monoprice (though they won't sound any different). The ones that I will never understand are the ones that are hundreds of dollars per foot, and claim that you will now be able to hear every heartbeat from the people in the booth of the recording studio where your recordings were made, because normal cables blur out the details (or other bogus nonsense like that).


 
  


> Originally Posted by *ComradeDylie* /img/forum/go_quote.gif
> 
> On the cable thing, I mean I do own Silver Dragons but I got them used and it was mainly a bling thing.  Like idk a gold watch, I didnt get them thinking it would make a sound improvement.


 
  
 Scientific studies have proven without a doubt that Japanese quartz has superior PRaT, but in an A/B sighted comparison, I think my Swiss mechanical's ETA 2836-2 sounds about 8 times smoother. The difference is so obvious, I don't even need to blind test to tell the difference! As a true audiophile, I obviously blame Jitter for the difference.
  




  
 Cheers


----------



## ComradeDylie

bigshot said:


> I use the 7:1 Stereo DSP on my Yamaha receiver. That is probably the same or similar to your MCH Stereo DSP. It improves the soundstage of 2 channel stereo and adds a pleasing liveness to the room. It also seems to take some of the curse off of being off axis to the mains.


 
  
 What are your opinions of the 7.1 dolby and dts music DSPs?


----------



## ferday

comradedylie said:


> What are your opinions of the 7.1 dolby and dts music DSPs?




I think they all have merits. Right now I'm enjoying audessy and THX, I like dts a lot but not a huge Dolby fan for music. I use height channels for my 7.1 instead of side surrounds, I like what it does to the stage


----------



## bigshot

comradedylie said:


> What are your opinions of the 7.1 dolby and dts music DSPs?


 
  
 I don't have those options with my Yamaha, except with native multichannel recordings. For those, I use the Neo music setting. I think that is DTS. I haven't figured that out entirely yet because most of the music I play is 2 channel. However, the opera blu-rays I have sound fantastic, so it must work with those settings.


----------



## ComradeDylie

bigshot said:


> I don't have those options with my Yamaha, except with native multichannel recordings. For those, I use the Neo music setting. I think that is DTS. I haven't figured that out entirely yet because most of the music I play is 2 channel. However, the opera blu-rays I have sound fantastic, so it must work with those settings


 
  
 The Neo setting is DTS, and it should work for 2 channel music.  The native multichannel should just auto play in whatever format it was produced in. Or at least thats how mine works lol


----------



## bigshot

Well then, I don't use it. I use Yamaha's 7:1 Stereo DSP to play 2 channel music in 5:1.


----------



## cjl

OK, one final set of test files, just because I was curious to try this with some other samples/genres including classical music (Bigshot's sample, specifically) and a copy I have of Rhapsody in Blue. Filenames state where I set the lowpass filter. It definitely isn't as obvious as it is with some other samples (such as the ones I posted earlier), but it definitely does make a difference. Personally, I can start hearing a difference once the cutoff is set to 14kHz (on both samples), and it becomes very obvious (and a clear degradation of sound quality) with the cutoff set to 12kHz. With the cutoff set to 16kHz or above, there was no audible difference to me.
  
  
 https://dl.dropboxusercontent.com/u/40020825/music%20test%20files/Classical%20test%20files.zip
  
 https://dl.dropboxusercontent.com/u/40020825/music%20test%20files/Rhapsody%20in%20Blue%20Test%20Files.zip


----------



## elmoe

Out of curiosity I tried the above. It was obvious up to 16kHz in my opinion. 16kHz was still very noticeable. At 18kHz I couldn't tell a difference though.


----------



## bigshot

Try cutting out the octave between 1kHz and 2kHz and see what it does to the music.


----------



## cjl

bigshot said:


> Try cutting out the octave between 1kHz and 2kHz and see what it does to the music.


 
  
 Sure, it'll be way more noticeable. That doesn't mean that 10-15kHz isn't important though.
  
 (Just for grins though, here it is, and the frequency spectrum is shown below. It sounds better than I thought it would, actually... https://dl.dropboxusercontent.com/u/40020825/music%20test%20files/1-2kHz%20cut.flac)


----------



## bigshot

Still can't play flac


----------



## ab initio

bigshot said:


> Still can't play flac




Is there no free player for mac that decodes flac?

Cheers


----------



## cjl

I pointed you to a free flac player a couple pages ago - it isn't hard to download. It would be a good thing to have, too, given the prevalence of the format. Because I'm feeling nice today though, here's an aiff, since I remember you like apple formats...
  
 https://dl.dropboxusercontent.com/u/40020825/music%20test%20files/1-2kHz%20cut.aiff


----------



## cjl

ab initio said:


> Is there no free player for mac that decodes flac?
> 
> Cheers


 
 VLC works on mac just fine.


----------



## Xenophon

+1 on the 14 kHz limit but then my hearing is going downhill fast once we're at 15 kHz...16 kHz I can't detect a difference anymore to be honest.  Cutting out the 1-2 kHz region is as disastrous as I thought it'd be given the places where most instruments are most of of the time.
  
 Bigshot:  do yourself a big favour and get the VLC-player.  I run it on my Mac, handles (almost) everything  under the sun in audio and video.


----------



## BeyerMonster

bigshot said:


> Try cutting out the octave between 1kHz and 2kHz and see what it does to the music.


 
 I think this thread has officially run its course.


----------



## Agharta

beyermonster said:


> I think this thread has officially run its course.




Where does the 'officially' come from?


----------



## Hapster

agharta said:


> Where does the 'officially' come from?


 
*Officially Origin: *
 1300–50;  Middle English  < Late Latin  officiālis  of duty, equivalent to Latin  offici ( um ) office + -ālis  -al


----------



## Agharta

That's the problem with high definition - it's officially for pedants.


----------



## bigshot

Today the weather changed and my sinuses kicked in. My ears plugged up and I'm not hearing much of anything. I have an interesting echo effect going from my Eustachian tube in one ear though. It sounds like surround sound.


----------



## ab initio

bigshot said:


> Today the weather changed and my sinuses kicked in. My ears plugged up and I'm not hearing much of anything. I have an interesting echo effect going from my Eustachian tube in one ear though. It sounds like surround sound.


 

 Is there a Sinus-Fi foobar plugin?
  
 Cheers


----------



## BeyerMonster

agharta said:


> Where does the 'officially' come from?


 
 Asserting that some frequencies near the upper ends of the hearing spectrum are unimportant to music and therefore unnecessary for proper reproduction of music is a reasonable hypothesis. You can explore that, create some experiments, and to some degree either confirm or refute. This is at least related to whether capturing/reproducing frequencies above 20 kHz are relevant to music and therefore whether higher sample rates are necessary or even useful.
  
 Suggesting that we try listening to music without a frequency that is near the middle of our hearing range and indisputably carries musical content is no longer remotely relevant to the OP, this thread, or the discussion about whether 24/192 or 24/96 offer meaningful differences as compared to 16/44.1.
  
 And yes, I misused "officially".


----------



## bigshot

You aren't reading what I am saying. I'm not saying that upper frequencies are unnecessary. I am saying that among the ten octave range of human hearing, the top octave is the *least* important to sound reproduction. To provide an example of the relative importance, I suggested comparing music with the top octave filtered out to music with an octave smack dab in the middle filtered out.
  
 Relative importance, not absolute statements of necessary/unnecessary.


----------



## cjl

bigshot said:


> You aren't reading what I am saying. I'm not saying that upper frequencies are unnecessary. I am saying that among the ten octave range of human hearing, the top octave is the *least* important to sound reproduction. To provide an example of the relative importance, I suggested comparing music with the top octave filtered out to music with an octave smack dab in the middle filtered out.
> 
> Relative importance, not absolute statements of necessary/unnecessary.


 
 So are you saying that any flaw in a sound reproduction system is only worth addressing if it is at least as significant a flaw as a complete band-stop filter from 1-2kHz?


----------



## bigshot

I'm saying that when you are looking at specs for a set of headphones, or working on getting your speakers to work well in the room, or choosing equipment, there are things that reap huge rewards if you pay attention to them, and things that really don't matter a lot. It isn't enough to get the right answer... you need the correct context too.
  
 Here is a for instance for you... I have seen quite a few people in the headphone forum up in arms when looking at response charts because of a high end roll off at the top end, but don't care at all about huge boosts in the upper midrange. Why? Because the chart they are looking at doesn't reflect Fletcher Munson, so the upper mid hump looks nice and flattish, while the precipitous drop at the end looks like the wrong end of a roller coaster. "Humans can hear to 20kHz!" they cry. "I want every single frequency!" Meanwhile the elephant in the corner goes completely ignored.
  
 It is really important to be able to quantify and prioritize things. Some things matter a lot, some hardly at all. It doesn't help when Sound Science people pipe up to pretty straightforward questions with long convoluted answers that include caveats about obscure exceptions to the rule that are highly unlikely to be the situation the questioner finds himself in. A normal person hears these sorts of confusing absolutist statements and only half understanding the context and relative importance, ends up chasing specs down to five, ten, twenty a hundred times beyond the threshold of hearing.
  
 It shouldn't start an argument to simply say that of the ten octaves that comprise the range of human hearing, the top octave is the least important to reproducing music. That seems like a pretty obvious thing if you are aware of what each octave of sound sounds like.


----------



## cjl

As I said before though, I even disagree with that final statement - I'll take a system which is flat from 40Hz-20kHz any day over one which is flat from 20Hz-10kHz (and the former is much easier to achieve too - making a tweeter flat out to 18-20kHz is much simpler than making a sub that can play flat to 20Hz, especially in a good size room and if you realize that THD isn't terribly important at those high frequencies). I agree that little dropoffs in the very extreme upper end of human hearing (>16kHz or so) don't matter, and I also agree that headphones which have both a huge midrange flaw and a rolled off treble will sound bad primarily because of the midrange, not the rolled off treble. However, the fact remains that the upper octave (especially the 10-16kHz region) is both audible and important to a wide range of music, and as a result, you cannot claim to have a true high-fidelity, near audibly perfect system unless it can recreate that part of the frequency range accurately. This is not some kind of fuss over nothing, this is not making a big deal out of some inaudible triviality, it is not chasing down specs that are five, ten, or twenty times beyond the threshold of human hearing. It is addressing a provably real, significantly audible portion of the music.
  
 As for your complaint about sound science? It's sound _*SCIENCE*_. It's about what is real, what is measurable, and what is empirically demonstrable. Unfortunately for your argument, that does actually include some caveats, some corner cases, and some situations that cannot be simplified down to a twelve-syllable anecdote. Looking at relatively high-end audio, many of these corner cases are not entirely uncommon either - it is entirely reasonable to expect that someone getting information in this section might own something that genuinely falls in one of those "exception" areas, and if they are not addressed, it could lead to incorrect information or a false impression. Besides, if I am interested in the legitimate technical and scientific details, I am free to discuss them. Nobody made you a moderator of this portion of the forum.


----------



## bigshot

Hoo boy!


----------



## cjl

(sorry if I sound a bit terse and irritable by the way - I've had a frustrating day at work...)


----------



## bigshot

My interest is in figuring out how to make home audio sound good... really good, not good to the nth degree crossing every t and dotting every i or "feel good" with self delusion. I realize that I'm a minority on the internet.


----------



## limpidglitch

Have your sinuses cleared up yet, bigshot?

 You should be able to play those FLAC files in QuickTime, if you're interested.


----------



## ComradeDylie

Borat's HiFi Country Western Sing-a-long:
  
  
  
 In my hobby there is problem
 And that problem is marketing
 It tells many many lies
 Because it knows it can
  
 Throw marketing down the well (c'mon!)
 So my hobby can be free (so my hobby can be free)
 We must make purchasing easy
 Then we have a big party
  
 In my hobby there is problem
 And that problem is the HiRes
 It take everybody money
 It never give it back
  
 Throw the HiRes down the well!
 So my hobby can be free (so my hobby can be free)
 You must grab it by its horns
 Then we have a big party
  
 If you see the HiRes coming
 You must be careful of its size
 You must grab it by its inaudible range
 And Ill tell you what to do
  
 Everybody!
  
 Throw the HiRes down the well! (throw the HiRes down the well)
 So my hobby can be free (so my hobby can be free)
 You must grab it by its horns (you must grab it by its horns)
 Then we have a big party (then we'll have a big party)
  
 Throw the HiRes down the well!  (throw the HiRes down the well!)
 So my hobby can be free! (so my hobby can be free!)
 You must grab it by its horns! (you must grab it by its horns!)
 Then we haaaaave a biiiiig paaaartyyyyyyy!


----------



## SunTanScanMan

^ rofl


----------



## Dark_wizzie

ralphp@optonline said:


> Sometimes I wonder if there is any difference between sponsorship and outright bribery. In the case of high end audio publications similar lack of difference exists between advertising revenue and outright bribery.
> 
> Truth, honesty and integrity are all just so 20th century. The 21st century is all about lies, dishonesty and corruption.


 
 Oh common. Don't say that people were somehow more moral and kinder in history's past. Any serious analysis proves otherwise. Have you read The Better Angels of Our Nature by Dr. Steven Pinker?


----------



## ralphp@optonline

dark_wizzie said:


> Oh common. Don't say that people were somehow more moral and kinder in history's past. Any serious analysis proves otherwise. Have you read The Better Angels of Our Nature by Dr. Steven Pinker?


 

 You're right, of course, but isn't mankind supposed to get more enlightened as time by and not less?


----------



## Dark_wizzie

ralphp@optonline said:


> You're right, of course, but isn't mankind supposed to get more enlightened as time by and not less?


 
 From our perspective, yes. But also depends on what subject matter. And sometimes change comes too slowly. 100 years is a long time compared to our lifespans but on the scale of the universe or even the evolution of humans, 100 years is barely anything.


----------



## Agharta

The past had at least one thing going for it - no Justin Bieber.


----------



## Thad-E-Ginathom

dark_wizzie said:


> Oh common. Don't say that people were somehow more moral and kinder in history's past. ... .... ...


 
  
 The good old days, when Snake Oil came in actual bottles?


----------



## Xenophon

ralphp@optonline said:


> You're right, of course, but isn't mankind supposed to get more enlightened as time by and not less?


 
 If one reads the classics then there's not much reason for optimism, frankly.  The very same concoctions that were peddled in roman times are still being sold now (hairloss 'cures', skin 'rejuvenating' creams.....).    Surprises me that none of these guys uses a slogan like 'Over 2000 years' of positive experience, proven  formula'.  If one thing amazes me then it is how nothing ever changes in that respect, they just latch on to new technology (the entire audiophile cables myth).


----------



## Agharta

Every time you blink, your brain has to remind you the world hasn't ended. No wonder we struggle to comprehend our place in history, the universe or audio science.


----------



## GrindingThud

I visited Bath England once and visited the Roman bath house. I was amazed that in 70AD people threw coins in fountains and wells...they excavated a bunch from the well there, along with notes written on lead scraps that basically say.....to my girlfriend who cheated on me, die bitch. I laughed when I saw it and realized the human condition is changed little from 2000 years ago.
http://en.wikipedia.org/wiki/Curse_tablet


----------



## Agharta

It's just called Bath, no need for the England bit. Have to groan everyone I watch an American film and they refer to London, England or Paris, France.


----------



## limpidglitch

agharta said:


> It's just called Bath, no need for the England bit. Have to groan everyone I watch an American film and they refer to London, England or Paris, France.


 
  
 When you saw the film _Paris, Texas_, did you groan then?
 I believe there is a London in Ontario.
 Even Bergen can be found in Belgium, the Netherlands and Germany, probably in the states as well.


----------



## Agharta

Except nobody but an American would wonder whether you meant London in England or London in Ontario.


----------



## limpidglitch

agharta said:


> Except nobody but an American would wonder whether you meant London in England or London in Ontario.


 

 So what's the problem then?
 You said they were american films.


----------



## GrindingThud

So how would you differentiate it from Bath Maine? ...other than we're talking ancient Roman baths....
Paris Arkansas 
London Ohio




agharta said:


> Except nobody but an American would wonder whether you meant London in England or London in Ontario.


----------



## ComradeDylie

In fact, Arkansas has them both lol and for that matter most every other state does too.  Paris, Arkansas or London, Arkansas, or England, Arkansas.
  
 Cant forget about Nederland, Texas.  For those of you wondering Nederland is how you really call The Netherlands.
  
  
 Ill admit though,  I havent seen any place called *København in the states.*


----------



## GrindingThud

Lol, but only in the US do we see this nonsense. At least when it started there was New London and New York.



comradedylie said:


> In fact, Arkansas has them both lol and for that matter most every other state does too.  Paris, Arkansas or London, Arkansas, or England, Arkansas.
> 
> Cant forget about Nederland, Texas.  For those of you wondering Nederland is how you really call The Netherlands.
> 
> ...


----------



## GrindingThud

So what were we talking about.....24/48 audio?


----------



## limpidglitch

grindingthud said:


> So what were we talking about.....24/48 audio?


 

 Boooring.

 I checked Wikipedia and noticed that a Bergen, Kentucky has been changed into Burgin. I assume it makes perfect sense if you're a Kentuckian.


----------



## ComradeDylie

grindingthud said:


> Lol, but only in the US do we see this nonsense. At least when it started there was New London and New York.


 
  
 Whoa! Whoa! Cool your jets me mon, it was Nieuw Nederland and Nieuw Amsterdam way before it was New York.  And if you notice all of the stuff around present day New York is named after places in Nederland.  For example, Brooklyn (Breukleun or something), Harlem (Haarlem)  The Bronx (Jonas Bronck's Plantation), Yonkers (Jonkers),  Queens (even though England has Queens too but they dont have Queensday like they do in Nederland), I mean the whole place is all Dutch.  Then you got New Sweden down there in Delaware, New France up there in Quebec and you had New England too up there in Mass.  New Spain was somewhere, maybe Florida.  Our whole country was named this way!  Well heck, most of North and South America probably.  And random other places they discovered.  You've got Nova Scotia which is New Scotland, down there by New Zealand (Zeeland is a province in Nederland) they have New Britain and New Ireland down there by Papua New Guinea and there is a New Hanover there too for the Germans.  A handful of nations "discovered" most of the world and just named everything New (Insert place in their country)  it really isnt all that creative but what are ya gonna do.  I feel as though there should be a New Portugal somewhere,  probably would be near Brazil?


----------



## castleofargh

so you're telling me to go accross the US instead of wasting money travelling the world?
  
 "it's a small world after all, it's a small world after all, it's a small small world" ^_^


----------



## Agharta

The problem with saying "London, England" vs "London, Ontario" or "Paris, France" vs "Paris, Texas" is that it gives both cities an equal status, when clearly and unarguably the USA versions are secondary and inferior. It's an example of US cultural ignorance, as if the rest of the globe is somehow irrelevant.


----------



## Dark_wizzie

agharta said:


> The problem with saying "London, England" vs "London, Ontario" or "Paris, France" vs "Paris, Texas" is that it gives both cities an equal status, when clearly and unarguably the USA versions are secondary and inferior. It's an example of US cultural ignorance, as if the rest of the globe is somehow irrelevant.


 
 Ontario isn't a state in USA.
 Stop bashing USA and let's get on topic.


----------



## cjl

agharta said:


> The problem with saying "London, England" vs "London, Ontario" or "Paris, France" vs "Paris, Texas" is that it gives both cities an equal status, when clearly and unarguably the USA versions are secondary and inferior. It's an example of US cultural ignorance, as if the rest of the globe is somehow irrelevant.


 
 So, when I say that I have visited Anchorage, AK, or Denver, CO, am I being culturally ignorant against my own country?


----------



## OddE

limpidglitch said:


> When you saw the film _Paris, Texas_, did you groan then?
> 
> I believe there is a London in Ontario.
> 
> Even Bergen can be found in Belgium, the Netherlands and Germany, probably in the states as well.




Paris, Texas is one of the most beautifully strange movies I've ever seen. 

I do think, however, that the oldest Bergen of them all is in Norway - founded in 1070AD.

If not the oldest, at least it is the most beautiful.


----------



## Agharta

cjl said:


> So, when I say that I have visited Anchorage, AK, or Denver, CO, am I being culturally ignorant against my own country?




Yes. Why do you need to identify the state?


----------



## castleofargh

I guess a delivery guy might care more than we do about those "small" details ^_^.


----------



## kraken2109

Well this went downhill fast


----------



## Agharta

Is that downhill fast, Oh? 





castleofargh said:


> I guess a delivery guy might care more than we do about those "small" details ^_^.




That is an address, and has nothing to do with how you reference a town in normal speech.


----------



## kraken2109

How about we get back to talking about audio?


----------



## Thad-E-Ginathom

agharta said:


> Is that downhill fast, Oh?
> That is an address, and has nothing to do with how you reference a town in normal speech.


 
  
  
 Did you mean to say "... how you *refer to* a town in normal speech?"
  





  
 (But hey, I'm not sure if I put that question mark in the right place!)


----------



## Dark_wizzie

Just ignore the troll and let's move on. Although TBH, the topic of this thread is long finished, we already know the answer.


----------



## Thad-E-Ginathom

Yeah, true... and sorry.
  
 Somehow, though, repetition on this subject doesn't seem to get anywhere. As you say, we know the answer ---and we also know what the marketing people will do, regardless of the answer.


----------



## Golden Ears

So many interesting and thoughtful comments.  I am slowly going through them.
  
 I'm one of the weirdos that enjoys hi-end car (there are very very very few cars that qualify) , hi-end home audio (many good systems qualify) , hi-end headphones(very few systems qualify) , and gasp..even hi-end Pro Audio (it does exist BTW- and there are very few systems)
  
 Many believe Redbook is it and anything else is a waste. (I agree there are great sounding CDs out there)
  
 IIRC
  
 The parameters of the CD were defined in such a way that SQ was not the primary consideration.
  
 The max Playtime of a CD was selected as the length of Beethoven's 9th symphony.
 The Size of a CD was selected as what could fit into a Single DIN car stereo unit. *
  
 As such given the pit spacing and lasers available at a consumer price at the time... that decided the Redbook standard as 16/44.1
  
 Had Blu-ray technology been available at the time I would see no reason why they would not have gone with 24/192 (even if 24/96 or 24/88.2 sounds better) because it would have made for great marketing.
  
 CD was marketing. They marketed the advantages of an optical format that (At the time) was nearly scratch and skip proof as they original discs were laminated between *two* sheets of polycarbonate (I used to sell CD's when they first came out in the early 80's by taking a CD out of the player- vigorously scrubbing it on a stone pebble wall and then inserting it back into the player and it would play back without skipping (you could not do this now with the way CDs are made) this demo sold CD players like hot cakes. And it had nothing to do with sound quality. In fact people did not even care... they just wanted freedom from damaged LPs.
  
 So if you gave Marketing people the chance for saying that a digital file could go out to an inaudible 96kHz with a noise floor 20 times that of human hearing... and it could fit in the dash of your car so you no longer had to buy tapes for your car... they would have gone with that..
  
 If that had been the standard.... which IMHO it would have if Blu-ray had been around at the time.....because it offers strong advantages for marketing. And if a rival company *SIMULTANEOUSLY* came out with REDBOOK-  and both released their standards of 16/44.1 and 24/192  I think the higher sample rate would have won. Historically you can look towards DVD-A vs SACD. (yes I know it isn't apples to apples) but in this case---higher numbers were assumed to be better.
  
  
 So would there have been an army of people arguing that to down sample from say 24/88.2 (Lets make it an integer down sample) to 16/44.1 sounded better??? I highly doubt it. I don't think they could have gotten any traction at all.
  
 So let's go to the history.....
  
 Downloaders.... Napster offered MP3 heavily compressed. And many said "it is CD quality...you can not hear the difference" and initially on crap ear buds a lot of people could not or felt it was a small price to pay for carrying more songs. Well it's a more than a decade later... and many people do now recognize a difference.
  
 If blu-ray was the initial format at 24/192.... what would people have settled on as the best format of compression? would it have been 24/48, 24/96, 16/48, (assuming integer compression) it would not have been 16/44.1 (IMHO there is nothing Magical about the 44.1 number).
  
 Well we would *STILL* get a lot of people arguing that 320kbs is great or Ogg Vobis is the best...and some arguing for 24/48 and some for 24/96. and some for 24/192.
  
*The point is...CD 16/44.1 would certainly NOT be the number we would be looking at. *
  
 A lot of time and effort has gone into to make the 16/44.1 format sound good. (if anyone cares... I happen to like the Wadia Digitmaster Algorithm which as it interfaces with the Wadia 860 series of machines and some prefer the Wadia 861. The benefits of digital volume control over analog at a lower price point make sense. at the high end, I like analog volume controls.) Which is why I think given the human effort in 44.1 in listening trials to get it to sound right ... 88.2 should have some more benefits than 96.
  
 So let's take a look... CD might have been 14 bit too.
  
 http://www.research.philips.com/technologies/projects/cd/
  
 The above is worthwhile reading.
  
 So I don't doubt for a second if Blu--ray were introduced initially there would be a bunch of people arguing over which codec sounds best...most would just opt for the highest number...
  
*What do I like*? I'm still undecided, but I think that* 24/88.2 has something to offer in that it is an integer of 44.1 so all the years spent trying to get 44.1 to sound right* ..some of that is preserved. and 88.2 goes up to 44.1kHz sonically which should be enough. Here is a file many might find and be familiar with.. "Riders on the Storm " Doors LA Woman in 88.2  give that a listen..  compare that to the 44.1.
  
 Not a lot of high quality audio components will freak out when fed a 44.1kHz sine wave....for instance Spectral probably wouldn't care, neither would Audio Research amps.
  
 In my personal experience , for example, loudspeakers tweeters that only go to 20kHz or even 25kHz don't sound as accurate as tweeters that go beyond 35kHz in part because the tweeters tend prominently beam more and some can look pretty ragged in frequency response near their extremes. So Tweeters capable of going well beyond 20kHz probably aren't a ragged in frequency response as tweeters that only go to 20kHz* AT 20kHz or even at 14Khz.*
  
 I may have been nicknamed Golden Ears as a kid in audio shops, and I doubt my hearing is as good now, but I never claimed to have ultrasonic hearing like a bat. But sonically.... for whatever reason....I think signals are more believable if they are not limited to below 20kHz in the playback format chain.
  
*So there is a sonic reason for overkill.* When it comes to speakers tweeters.
  
 I also think that some people mistake noise as signal. for instance when Solid State amplifiers came out... if you look at old reviews...of Solid state gear. Some people thought the highs from Solid state were "More Natural sounding than Tubes" but these very same ears from the same reviewers might never say that today. Not because their ears are older and their hearing is lesser, but because now they can better discern noise from signal. At the time some of the glare of Solid State gear seemed to add to the electric guitar "crispness"  of rock. I felt that way at the time. And now listening to the same thing years later- I miss some of the.. (Shall I say err.. gritty musicality) of the added solid state glare to the higher guitar solos.
  
 Lastly....
  
 Some people have mentioned... that you need an ultra high resolution system to hear the difference. My personal thoughts are that  super high resolution systems seem to benefit less from high resolution because they have the best DACs for making 44.1 sound good. Whereas... often lesser Mid-fi gear that has lesser quality  DACs suffer more from Rebook than hi-rez systems. Of course to hear what is really going on.. a high rez system is better. For me one of the luxuries of a true high resolution system is that CD Rebook sounds better on it than on a lesser system. 
  
 As a system tuner at hi-fi shows. I often demo 95% of the time with Redbook. Because that is what most people will play. I don't whip out  master tape reel to reel of a recording... it is not representative of the format available to the consumer. Does Reel to Reel sound better... ABSOLUTELY! But I often go through demos playing about 10-15 cuts of Redbook and then just so people can hear- I'll play a 24/96 file. Then they can hear what hi-rez will do.
  
 Now as someone who did High end DJ in the 1970's and 80's (A weird animal to say the least- but in a sense it is what I do NOW at RMAF- CES- T.H.E. SHOW now in small scale)  I used to hear other DJ's spin vinyl and 95% of the time it sounded awful on their systems (except for  some of Richard Long and Associates later systems) . Just dreadful. Their cartridges were never set up right- they used crappy styli, they had  crap systems...
  
 But IIRC they were the first group of people to say digital was awful. And they noticed it on lower resolution PA systems. I also noticed it. I slowly stopped buying digital music. Digital playing back real instruments lacked that "boogie factor" that makes people dance to real instruments. Digital does not suffer much from digital synthesized dance music...but it tended to butcher a lot of non digital dance music at the time. Digital does the least damage to digitally synthesized  instruments....and *that is what becomes popular... the Music that sounds best on the system used for playback*. For instance Rap and Hip hop became popular on tiny home HT speakers and subs... or pretty much  the same speaker driver compliment in cars- because that was the musical genre that was least damaged by these music systems.
  
 There will be many who select say 1990's music mixed in 16/48 (to do their tests)  and they will cry out that Redbook is just fine.. but that is in part because the original format was limited to 16/48. 
  
 What do I think would have been best as our format? I think 32/90kHz because 32 bit would have given engineers lots of latitude (even though the best we can output out of our DACs is 20-22 bit because of thermal noise) and 32 bit could allow for nearly all attenuation to be done digitally bringing better sound to lower cost gear. I picked 90kHz because tweeters that can do 35-45kHz sounded pretty good to me ...even admitting that some of that could be do to ringing of tweeters and distortion..but if in the end it sounded closer to live unamplified music...I'll take the euphonic distortion. realistically if we had more perfect drivers 32/60kHz or 32/80kHz would probably be best.
  
 These are practical reasons for good sound in my opinion given what we have and are limited to in speakers and amplifiers and the abilities of the average person doing the initial mix down in the studio. Yes there are great recordings that were made with early digital... but there are as a percentage extremely few of them...in so much it was either luck or EXTREME SKILL that made them. Giving engineers more latitude and headroom  is a good thing, and not having to crush it afterwards probably will help us to hear the original mix.. or in my opinion even better if the original mix played through your system sounds like the live venue of unamplified instruments. In Deference to the passing of Harry Pearson.. HP was right on this goal of the recording sound lichee live event. I can't stand it that magazines want the playback to sound like the recording the mixer heard. I want to hear the musicians and venue and audience  materialize in my room- *I don't want to feel like I am a recording engineer sitting at a mixing console.* YMMV .
  
 As for 24/192 in my personal experience  I have not heard enough material on 24/192 in direct comparison to 24/96 to hear the difference (I have not integer down sampled 24/192 to 24/96 to hear it- though on two of my systems limited to 24/96 I may have to in the future...one of my systems can handle DSD 256) . What I do hear nowadays is a lot of the remasters in 24/192 sound less natural on high resolution systems than earlier 24/96  remasters. Kinda saddened about the direction we are headed...sonically as remasters are IMHO "over spiced" to sound a bit hyped on neutrally sourced components.
  
  
  
  
  
  
  
  
  
  
  
  
 *They had digital Laser Discs which were deemed too large for a consumer format (the size of an LP) and sadly... they had a fantastic analog track that the laser could read. Ever want to a comparison get John Carpenters "The Thing" and listen to both the digital track and the analog track...the analog track is astoundingly good.


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## bigshot

I'm afraid that was WAY too much for one post. I didn't get through it. I got as far as "sound quality wasn't a primary concern" and I stopped. The whole purpose of PCM was to create a *perfect* sound reproduction format... one that met or exceeded every threshold of human perception. The length of time a disk held was only a consideration in the design of the disk itself. The audio format was already established.


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## ralphp@optonline

bigshot said:


> I'm afraid that was WAY too much for one post. I didn't get through it. I got as far as "sound quality wasn't a primary concern" and I stopped. The whole purpose of PCM was to create a *perfect* sound reproduction format... one that met or exceeded every threshold of human perception. The length of time a disk held was only a consideration in the design of the disk itself. The audio format was already established.


 

 The entire post is filled with all the standard audiophile counter arguments to the hard science of digital audio. Apparently the previous 300+ posts, many of which very ably debunk each point that Golden Ears brings up (yet again). My advice to Mr. Ears (excuse me for assuming that Golden Ears is male) is to spend lots and lots and lots of money on all the latest audiophile approved pieces of digital audio playback equipment and to find DSD versions of all his favorite recordings so that he can rest assured of obtaining the highest possible sound quality. Remember that DSD is now the current audiophile flavor of the month and as such it is pretty much a given that PCM, even 32/384, is dead. Long live DSD!!


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## Golden Ears

bigshot said:


> I'm afraid that was WAY too much for one post. I didn't get through it. I got as far as "sound quality wasn't a primary concern" and I stopped. The whole purpose of PCM was to create a *perfect* sound reproduction format... one that met or exceeded every threshold of human perception. The length of time a disk held was only a consideration in the design of the disk itself. The audio format was already established.


 

 Then I did not recall correctly. Sorry about that.
  
 Sorry about the long post.
  
 If you read the Phillips link.. you'll see that people were tinkering with 14 bit prior to 16 bit. no necessarily for the CD format though. Clearly... few people able to discern well between audio gear  given the choice between analog Master tape an CD would select CD- many would select LP over CD. In that respect CD and PCM has failed and is not a perfect sound reproduction format. Also few would select even 24/192  or any PCM rate over the Analog Master tape.
  
 Given that... I think it is fair to say..the codec or sample rate etc..etc. was/is not optimized. Which is why for the life of me I can not understand why anyone would think the CD is perfect or that REDBOOK is good enough or that something isn't wrong with digital in it's current state. Not that digital playback  won't improve..to the contrary... the DACs have improved, but I am still not convinced that there isn't a better way to package digital to help the playback devices manage it better. For instance I think every block of info  should have a sequencing number.. so it could be reclocked perfectly- as as the bits get closer together in time..this might become more important.. there should also be extensive error sum checking like in computers so info isn't as easily lost.
  
 What I hear personally is that 24/88.2 sounds pretty amazing. I haven't got a ton of 24/48 but what I have doesn't sound as good as 24/88.2
  
  
  
 Thanks for your corrections... always appreciated..


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## Golden Ears

ralphp@optonline said:


> The entire post is filled with all the standard audiophile counter arguments to the hard science of digital audio. Apparently the previous 300+ posts, many of which very ably debunk each point that Golden Ears brings up (yet again). My advice to Mr. Ears (excuse me for assuming that Golden Ears is male) is to spend lots and lots and lots of money on all the latest audiophile approved pieces of digital audio playback equipment and to find DSD versions of all his favorite recordings so that he can rest assured of obtaining the highest possible sound quality. Remember that DSD is now the current audiophile flavor of the month and as such it is pretty much a given that PCM, even 32/384, is dead. Long live DSD!!


 

 I haven't really preferred DSD. I like PCM better. I am male.
  
 There are many reasons not to believe audiophiles. not  all audiophiles are chasing after the same aspects of sound reproduction. There are any "audiophile systems " that to my ears do not sound real.
  
 When "Stereo review" magazine was out.. it just measured gear. and posted distortion figures. If that is "science" well all it really proved as that we placed the wrong weighting on the measurements.
  
 At the same time... I think more damage is likely done  to the signal in the DAC than the A to D converters.
  
 It's not that I don't enjoy digital. At the moment I don't listen to Lp because I sold my better TT. I am amazed at how much resolution the better DACs wring out of REDBOOK, but after 15 years... I don't think we will get to the sound of the Analog master tape or the live experience. Records took a very log time to sound good and LP payback took longer to be able to get good sound from those grooves.  Even if digital moved forward at twice the pace of Vinyl we won't get great sound anywhere near a master tape  until 2040.
  
 IMHO So long as 24/48 doesn't sound absolutely live... it isn't the correct format.


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## castleofargh

golden ears said:


> ...


 
 there is the pro side and the consumer side, pros need what they need and most likely already have it.
 consumers need easy to use stuff that someone from marketing succeeds to make them think was great and new while not asking any sacrifice from them(because we don't like to give up anything we already have).
 most people buying albums don't know much about sound, don't really understand what bitdepth and sample rate numbers mean, or actually have the wrong understanding that it is like video. so all in all I think you're talking about knowledgeable people who were never concerned by storage space, and that's a very small minority even on headfi.
 when mp3 came out it was pretty bad(the first versions of the codec didn't leave me with a dream like experience, now I use mp3 on all portable gears). but is was small and most people couldn't tell or didn't care, you could as you said get it on napster with your 28 or 56k in 30mn a song or something ^_^. and that was enough to make it the dominant format for years. not audio quality.
 so again I'm not convinced that people would'nt have gone for lower resolution pcm if only for the huge gain of storage space. having 48 or 50khz instead of 44.1 wouldn't have changed a thing, maybe a few albums would have had one less song on CDs but that's it. it was a choice made for us and could have been anything.
 now going to 24/96 is already a huge step to take, and I'm not sure even now that a lot of people are ready to take it. and I'm saying that when thank you audio gods, we can use flac. in wave it would just be horrible.
 and in the end it's still more of a psychological problem than a real need for quality, we're still with systems that can barely keep 16/44 fidelity up to our ears. when all my sources will do 130db snr at the amp output and redbook is the weak link, I'll probably go for 24bit, because hey why not? and when that day comes, we'll probably all have 10tera drives in in phone or at least super high internet for streaming, so all will be solved.  
 we're not there yet.


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## ralphp@optonline

golden ears said:


> I haven't really preferred DSD. I like PCM better. I am male.
> 
> There are many reasons not to believe audiophiles. not  all audiophiles are chasing after the same aspects of sound reproduction. There are any "audiophile systems " that to my ears do not sound real.
> 
> ...


 

 I notice that you say a few things which quite true and then some other things that run counter to those statements. For example you correctly state " I don't think we will get to the sound of ... the live experience." which is absolutely true. But the missing piece of the statement "the Analog master tape or" kind of implies that analog master tape is something to strive for, which in the case of older analog recordings is, again, very true but it also incorrectly implies that analog master tape is basically equivalent to the live experience, which is not true.
  
 The real sticking point of the whole high resolution business isn't whether or not high resolution sounds like "the live experience" but rather whether or not high resolution sounds better than standard resolution, i.e. CD or 16bit/44.1kHz.  I say this because recordings will never sound as good as live, just a fabulous picture of a beautiful sunset will never be as good as seeing that sunset live. The good part is that in the case of high resolution digital audio versus standard resolution digital audio the two are quite easy to compare side by side, whether it be in sighted listening tests or double blind listening tests.


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## bigshot

golden ears said:


> Clearly... few people able to discern well between audio gear  given the choice between analog Master tape an CD would select CD- many would select LP over CD.


 
  
 Well, most people haven't had the opportunity to directly compare the playback of a 24 track master mix down and a CD. Fortunately, I have. I have acted as a producer at recording sessions and sound mixes and have worked as a post production supervisor and a sound editor as well. I worked in the analogue era and recorded a record that was recorded on 24 track tape and distributed on both LP and CD. It was the transition period when digital audio was gradually replacing analogue.
  
 I was as skeptical as anyone about digital audio back then, and I was worried that we were delivering a four track ADAT to the record label instead of the traditional four track analogue tape. I asked my engineer to rack up the original 24 track and the ADAT digital bounce down side by side so I could compare them. I closed my eyes and he switched back and forth on the board for me. Absolutely no difference.
  
 The record was released on CD in the US and on vinyl in Australia (I think) because they hadn't gone all digital there yet. I got check disks on both formats. The difference between them was like night and day. Again, being the skeptical sort, I took my CD check disk and the ADAT master back to the studio we recorded and mixed at and had them rack the two up side by side so I could compare. No difference. I asked the engineer what sort of CD player he was using to play it back. He told me it was a midrange consumer Yamaha.
  
 There ya go!


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## Golden Ears

I only used analog master tape as it is generally accepted as being higher in the hierarchy of SQ...with of course "Live" at the top of the hierarchy . It is also easier to replay a master tape than to gather musicians. 
  
 So if you shoot  for an accomplish one goal... Say being able to make CD sound as good as or better than  a ferrous tape tape cassette played over a old Hitachi tape player.. then you could try to beat say a Tandberg 3014a playing a hi bias cassette, and then onward up the ladder...to reel to reel, 3rd gen masters, 2nd ten masters, master tape and then live.
  
 I think some reproduced music can get very close to live. The issue is that there are limitations . Certainly it would be easier to reproduce...say the strumming of a persons fingernails on a sofa cushion than say a human voice or an entire orchestra. The more familiar one is with something within the better  more discerning ranges of our  hearing the harder it is to fill the listener into thinking it is live and not a reproduction.
  
 I have heard some systems  that sound like a real person is singing.. but oddly the voice is wrong....and when you meet the singer they sound very different though both sound very live. I have also heard a system reproduce a singers voice very closely enough to be fooled hat it is the one and the same person. 
  
 There is a point where no matter how high the sample rate or bit rate...we can not hear any improvement (even if we try to fool ourselves) the issue is... does it all sound live at the point of no more improvement? Or do we hit a glass ceiling?
  
 If I am to understand the whole point of this thread to mean.. the glass ceiling of digital improvement is 24/48. I could see that as being pertinent in terms of wasted resources, storage, processing.  And certainly if extending the bit and sample rate does nothing more to improve the sound...then it is a waste. 
  
 I also think that given the sound of Victrolas of yesteryear ... recording engineers (at the time) might have thought that there was no point in trying to cut better laqueurs because there was no playback gear at the time that could take advantage of increased precision. We might have a vinyl standard for reproduction that would be much worse as a result today.
  
 So I understand the reason for 24/48 being sufficient just as people might have though it was a waste to cut records with better precision or lower wow and flutter etc. at the time.


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## bigshot

golden ears said:


> At the same time... I think more damage is likely done  to the signal in the DAC than the A to D converters.


 
  
 The damage to the sound that transducers cause is many orders of magnitude above any damage from DACs. DACs today are audibly transparent. In the early years of digital audio there were problems, but oversampling and caching, which are standard in just about all DACs today, have totally solved that.


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## RRod

32-bits will be just a convenience for computer interfacing for, well, ever, b/c yeah I'm not willing to put my ears into any kind of system that could actually deliver 32 bits above the noise floor  Already at 16-bits my most dynamic music is just at the edge of me needing to adjust the pot mid-song, which is the point where music listening gets annoying instead of enjoyable. Extra bits are certainly useful if the end-user wants to do his own processing, but that's pretty specialized a justification for throwing out extra 1s and 0s to the masses. Let the engineers do their thing and give the people what they can actually use, which keeps everything efficient. There HAS to be a stopping point for our PCM standard, because 64bit/6.144MHz that is legitimately delivered will just kill people and many bats before them.

 It's funny that it seems that people who actually *test*, in a controlled manner, formats against each other get flak for not "enjoying the music," and yet it is exactly the people calling for higher and higher specs that "sound better to me" that are shifting focus from real problems, like mastering, that affect musical enjoyment, to non-problems like sampling rate.


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## bigshot

ralphp@optonline said:


> recordings will never sound as good as live, just a fabulous picture of a beautiful sunset will never be as good as seeing that sunset live.


 
  
 The goal of recording isn't to recreate the live experience. That isn't possible, because there are so many aspects of a live experience that can't be captured by microphones or cameras.
  
 The goal of recording is to capture music and present it in a manner that is organized to provide the optimal recorded experience. That means that the engineer takes the limitations of recorded sound and works to play to the strengths of the process, creating a virtual experience that may actually be better than live in some respects.
  
 Some people think that great sound is made by pointing a mike at a performer and just recording it direct with no manipulation. But that is like trying to shoot a photograph without adjusting shutter speed or aperture or framing the picture in the viewfinder. You just end up with a lousy reproduction. The reason that mixing boards have all those equalizers and channels and volume pots and processors is so the engineer can make the sound *better*. Those tools aren't the problem. Whether or not the tool is effective at improving the sound is entirely up to the judgement of the engineer, not the equipment used.


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## ralphp@optonline

bigshot said:


> The goal of recording isn't to recreate the live experience. That isn't possible, because there are so many aspects of a live experience that can't be captured by microphones or cameras.
> 
> The goal of recording is to capture music and present it in a manner that is organized to provide the optimal recorded experience. That means that the engineer takes the limitations of recorded sound and works to play to the strengths of the process, creating a virtual experience that may actually be better than live in some respects.
> 
> Some people think that great sound is made by pointing a mike at a performer and just recording it direct with no manipulation. But that is like trying to shoot a photograph without adjusting shutter speed or aperture or framing the picture in the viewfinder. You just end up with a lousy reproduction. The reason that mixing boards have all those equalizers and channels and volume pots and processors is so the engineer can make the sound *better*. Those tools aren't the problem. Whether or not the tool is effective at improving the sound is entirely up to the judgement of the engineer, not the equipment used.


 

 Incredibly well stated. Thanks!


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## Golden Ears

bigshot said:


> Well, most people haven't had the opportunity to directly compare the playback of a 24 track master mix down and a CD. Fortunately, I have. I have acted as a producer at recording sessions and sound mixes and have worked as a post production supervisor and a sound editor as well. I worked in the analogue era and recorded a record that was recorded on 24 track tape and distributed on both LP and CD. It was the transition period when digital audio was gradually replacing analogue.
> 
> I was as skeptical as anyone about digital audio back then, and I was worried that we were delivering a four track ADAT to the record label instead of the traditional four track analogue tape. I asked my engineer to rack up the original 24 track and the ADAT digital bounce down side by side so I could compare them. I closed my eyes and he switched back and forth on the board for me. Absolutely no difference.
> 
> ...


 

 According to that line of reasoning , there is no reason to buy a CD payer better than  midrange consumer Yamaha. Or perhaps that the Analog playback could be bested by a midrange Yamaha CD player. I do see your point from your perspective and accept it.
  
 From my perspective..
 A fair test...yes... given the gear used.
 But definitive.... I might not look at it that way.
  
 Also I don't know what music was on the album? Was it well mic'd?  Was it considered an excellent master SQ wise? Or was it average? Was it live instruments mic'd in a good acoustic or a recent rap album with tons of compression ?
  
 For the vast majority of compressed pop stuff we hear... we could get away with a lot worse codecs than 24/48 and not loose much of anything sonically - but also we feel less from the music. I think the music that is destroyed less by low fidelity becomes more popular because it is not as sonically fatiguing as music that has been harmed in the reproduction process.... take for instance..Deadmau5  you could likely compress it a lot and not greatly destroy the musical impact of that pop. But take a well recorded  orchestra and compress it the same amount...it suffers.
  
 Jut because some people do not notice something in a  recording does not mean it is not there.  For example sometimes on better systems you will hear instruments or singers or percussion that you never realized was in the original recording. I am in NO WAY saying you have a less than optimal system...I am just saying that certain things become unearthed through better playback systems.
  
 I have played back things for people and people have sworn up and down that I was playing some "New Version" with added tracks, because they heard things that they previously never knew were there. 
  
 Let's say ... in the not too far future. some guy figures out a way to take 32 bit 192kHz digital ...process it and then get really great 24/48 out of it that sounds closer to live than either the 24/48 or 32/192 derived from the original feed. But he needs the headroom and information in a 32/192 signal to do it. Wouldn't it be a shame for him to only have a few 32/192 files to convert to this live sounding, processed,  24/48?
  
 I understand the argument for 24/48 NOW/Today , but I just wonder if we might miss something in the future if we needed up standardizing on a  format with lower resolution.


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## RRod

That guy already has 32/192 at his fingertips, and we're all fine with him using it for the headroom and archiving it for future remastering. But why do the millions of people listening to the music also have to deal with that same file, when a properly downsampled and dithered 16/48 version will be inaudibly different to them?


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## bigshot

golden ears said:


> According to that line of reasoning , there is no reason to buy a CD payer better than  midrange consumer Yamaha. Or perhaps that the Analog playback could be bested by a midrange Yamaha CD player.


 
  
 That is certainly correct... for pop music, for classical music, for jazz... for any kind of music. The reason for that is that the midrange Yamaha is designed to produce sound that is more accurate than any human ears can hear.
  
 The reason studios record 24/96 is so that they have latitude beyond the range of human hearing to make broad corrections to the sound. But once the mix is locked down there is no reason to lay down a file with sound quality beyond 16/44.1. 16/44.1 is audibly transparent to human ears.
  
 When it comes to getting the most out of a home playback system, we are extremely fortunate nowadays. The back end of the chain has been perfected and mass produced at a very reasonable price. Midrange CD players and amps today have specs that rival or exceed the equipment I would work with in professional sound studios back in the waning days of analogue. That should be a liberating feeling for audiophiles! Imagine having a $120 one pound blu-ray player in your home that sounds as good as the fabulously expensive 24 track two inch decks they used to record the music, that were as big as a washing machine.
  
 So if the electronic chain is perfect and inexpensive, why aren't all people's systems perfect sounding?
  
 Because of the wild card that still remains... Transducers and physical sound. Transducers (speakers and headphones) have not gotten significantly better like electronics have. And even if they were perfect, the acoustics of the average living room would make them sound awful anyway. The areas that audiophiles should be focusing on are speakers, room acoustics and audio processing to correct for both speakers and acoustics.
  
 I have a funny story about a friend of mine who owned an audiophile equipment storefront business. I went to visit him and he invited me to audition his system. He had a beautiful midcentury modern living room, tastefully decorated like something out of Architectural Digest. But the speakers were pushed out of the way against the wall, and the couch wasn't even situated in the proper listening position. The sound was mediocre and muddled, so I got up and walked to different points in the room. It was different wherever I stood. He asked me what I thought, and I told him honestly. He dejectedly agreed with me. Six months later he called me up all excited... "STEVE! STEVE! You have to come over and hear my system now! I've moved all the furniture out of the living room and put up acoustic panels!" I asked him what happened to the furniture. "My wife left me and took all of it with her. The stereo sounds great now!"
  
 If someone is serious about great sound, they need REALLY GOOD speakers (that ain't cheap), they need a good sized room that is treated acoustically and set up to have as good acoustics as practically possible, they need multichannel sound to create a lifelike dimensional sound field, and they need signal processing- specifically equalization and DSPs- to correct for the remaining acoustic deficiencies in the room. But instead, audiophiles worry about wires and bitrates and sampling rates and high end DACs that don't make a lick of difference. All of these things may have mattered back in the analogue era, but today? No.
  
 Understanding how home audio works is the best way to optimize if for peak performance. You know what matters and what doesn't.


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## mikeaj

Well said, bigshot. Some people know of the above but don't believe it, but a lot of people don't even realize what's going on. Some speaker / headphone measurements and at different positions really should be a wake-up call.
  
  


rrod said:


> Extra bits are certainly useful if the end-user wants to do his own processing, but that's pretty specialized a justification for throwing out extra 1s and 0s to the masses. [snip]


 
  
 Now you're making me wonder how many times in the history of ever have people clamored for higher bit depths (and/or sampling rates) for more headroom to apply their EQs, crossfeed, or whatever else. I'm pretty sure the format camp tends to heavily be purists who don't believe in some thing, or at least there's some very statistically significant correlation.


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## bigshot

The thing is, acoustics and room treatment aren't absolutes like specs are. You have to balance acoustics with livability. I see photos of home theater setups that look completely sterile and ugly, and I wonder how anyone would want something that looks like a mall shoebox theater in their home. Then I read the description and find out the theater has been banished to a damp basement or drafty garage. Concrete adds a whole other level of problems to getting decent sound.
  
 The photos of listening rooms and home theaters that I find the most interesting are ones that look like normal living rooms. I get lots of great ideas for furniture placement and prioritized room treatment from those, as opposed to brute force modified basements. I am really happy with my room because it looks like a really fun living room, but at the push of a button the screen comes down the lights dim and the sound kicks in and it is a first class screening room. It took a lot of balancing and compromising and experimenting to get there.
  
 The room is the vast unexplored territory in high end audio. People spend all their time focused on black boxes in a rack and totally ignore the aspect that makes or breaks the sound of the entire system.


----------



## RRod

mikeaj said:


> Well said, bigshot. Some people know of the above but don't believe it, but a lot of people don't even realize what's going on. Some speaker / headphone measurements and at different positions really should be a wake-up call.
> 
> 
> 
> Now you're making me wonder how many times in the history of ever have people clamored for higher bit depths (and/or sampling rates) for more headroom to apply their EQs, crossfeed, or whatever else. I'm pretty sure the format camp tends to heavily be purists who don't believe in some thing, or at least there's some very statistically significant correlation.


 
  
 I saw it mentioned as a rationale on here once before. I would never "clamor" for it, as 16bits already fills my pot and I don't think throwing my music/movies through an EQ or HRTF or two will degrade much. Still, it's a better rationale than "I want my square waves purtier."


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## bigshot

Most end user EQ doesn't even get close to the broad corrections a mixer makes. As long as you EQ subtractively and don't get into clipping, there is no reason why you would need higher rates.


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## Golden Ears

I also want to make sure that the room with the speakers is "livable" or you make an audio graveyard that seldom gets used.
  
 I do hear very large differences with DACs. I have  tested the Wadia 171, aRcam RDAC, Lynx Studio Hilo DAC, Apogee Mini DAC, Apogee  Duet, Apogee Duet 2, Meridian Explorer, Wadia 860, Chord Qute, E.A.R. , ..and well thought I would wish them to all be the same and "perfect" they all sound very different to me.The Hilo for example has incredible High frequency sound staging, but doesn't do vocals as well as some others. The Explorer has good tonality but does't do depth very well,  and so on.
  
 For those who can enjoy music to the fullest and can use lamp wire, a $120 Blu-ray, and Sony speakers form Best buy... in some ways they are far luckier than I am.
  
 I found that as my system got better I was able to enjoy a wider range of music and more songs on each album became enjoyable. even songs which seemed to be useless before had meaning- mostly because the fidelity allowed me to understand what was going on. Instead of being nicknamed "Golden Ears" maybe they should have called me "deaf guy" because I do need a certain amount of fidelity to really enjoy music. Women with their superior hearings can party to a clock radio (Seen it happen at Wellesley college way too many times) you would never see guys playing air guitar to a clock radio.
  
 It's not audio snobbery- in a way it is a financial hardship. And just like a handicapped person might derive great joy from having a well fitted high performance prosthetic arm on so do my ears and brain enjoy music more with a great audio system.
  
 I have spent way to much time with DACs to think they sound the same. In the 1990's I had pretty much given up on trying to listen to digital without being distracted so I started a digital system for my car. And hunting for a CD player I ended up buying a Wadia 860...which I could not bear to chop up to fit into my car....and it sounded good enough for the home.
  
 not every $$$$ Cd player sounds great. I heard some DCS systems that I did to like at all. This is personal preference since some people seem to love them. I liked a $2500 DAC I used at a show a few weeks ago that I would prefer to listen to a highly stylized  a $20,000 DAC we had at a show a few years ago. So to me they really don't sound the same.
  
 This site , I would assume, is filled with those looking for better sound. not just a better deal on headphones group buys. And I can say with certainty you could play a song on my Wadia and ten hot swap in other CD players and I could tell you blind every time which one was my Wadia 860x.  I can't say I could identify every player dow to the model number  100% of the time (it's not a skill I care to try to learn) but I could tell you which was was better or worse in terms of being closer to live sound  fairly quickly with source material I know well.
  
 Its true that speakers an the room add much more distortion... but distortion is something we can here through other things. For instance the ear typically hears the louder of two sounds,  but distortion is so small...how could it possibly hear small amounts of distortion, particularly when the room distortion is so much higher? Well we can. We just can. We don't need a perfect listening room to hear it.
  
 But then if you put  it in a  subway station at rush hour.....we can' tell.. there are limits.
  
 I for instance, have pretty limited appreciation for classical art. I couldn't identify the valuable excellent paintings from average ones.  It doesn't make me less of a person, it just means that those things don't concern me enough to feel or learn the difference. For some people seeing art in a  book is just as good as seeing it in person. for me... there is no difference. I'd say "all those pictures look pretty much the same , I don't see the reason for using a more expensive canvass or paints or brushes."
  
 and there are likely a lot of people like me. But I would to say that canvasses or brushes or paints did not make a difference in artwork. or that all paints have evolved and canvases and even brushes that there could be no further improvement and that the lighting in a  museum makes more of a difference in appreciating the artwork than the artist.. Even though I can't tell the difference in art  , I will not doubt those who can and do for a living. There are many people who sell art successfully without an appreciation for it. 
  
 So I say , if it is worth it to record at very high rez and then make available the best sounding format..that is ha should be done. but unlike analog where you can get an incrementally better playback system- digital is locked.... so you should be careful not to lock in the resolution for playback at too low a rate....unless you want to buy the entire catalog again like is being done now with HD tracks.


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## SpaceQuest

bigshot said:


> The goal of recording isn't to recreate the live experience. That isn't possible, because there are so many aspects of a live experience that can't be captured by microphones or cameras.
> 
> The goal of recording is to capture music and present it in a manner that is organized to provide the optimal recorded experience. That means that the engineer takes the limitations of recorded sound and works to play to the strengths of the process, creating a virtual experience that may actually be better than live in some respects.
> 
> Some people think that great sound is made by pointing a mike at a performer and just recording it direct with no manipulation. But that is like trying to shoot a photograph without adjusting shutter speed or aperture or framing the picture in the viewfinder. You just end up with a lousy reproduction. The reason that mixing boards have all those equalizers and channels and volume pots and processors is so the engineer can make the sound *better*. Those tools aren't the problem. Whether or not the tool is effective at improving the sound is entirely up to the judgement of the engineer, not the equipment used.



 



I think the goal of recording Live concerts IS to capture the live athmo, TOO !
Could be done by a Multichannel Recording, ( Recording Engeneers in the past
didn't have that and have to use stereo setups ) Read you have a home theatre,
why did all big companys put DSP power in their amps to simulate those live rooms ?
Could it be, that they want to give you a experience back that bad recordings missed ?
For good recordings, i prefer to switch those DSPs off. 

Some great Recordings are recorded by simply pointing a mic at a performer,
DIRECT with NO manipulation. I really prefer this method, but it is not always the best.

Yes, i can drive to a recording session and plug my Browner Mics plus a Vovox Cable
into my MetricHalo and NO more Mastering is needed, if is is a DeathMetal session in a old
fabric Hall. In a Studio, im going to select the mics more carefully to get rid of room modes and use the Mic specs for doing EQ. So, if i do the job with a well selection of mics i have NOT to EQ something, too. There are sadly only a few recording engeneers, and these are well trained who use only a pure setup. Mastering Studios love their work. Using a EQ at a recording is always a compromise; well setuped levels, well choosen equipment is best.

The better the equipment you are using, the better and often faster you can work with
less stress to the sound.
Think of a magnifier glass to see details sharp, with muddy equipment you only
see mud. This is why the recording and playback chain is so important. The bad thing
about a very good chain is the price...


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## castleofargh

rrod said:


> mikeaj said:
> 
> 
> > Well said, bigshot. Some people know of the above but don't believe it, but a lot of people don't even realize what's going on. Some speaker / headphone measurements and at different positions really should be a wake-up call.
> ...


 

 square waves are mighty important, else you can't shape the song to look like you're playing super mario on an oscilloscope.


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## bigshot

golden ears said:


> I do hear very large differences with DACs. I have  tested the Wadia 171, aRcam RDAC, Lynx Studio Hilo DAC, Apogee Mini DAC, Apogee  Duet, Apogee Duet 2, Meridian Explorer, Wadia 860, Chord Qute, E.A.R. , ..and well thought I would wish them to all be the same and "perfect" they all sound very different to me.The Hilo for example has incredible High frequency sound staging, but doesn't do vocals as well as some others. The Explorer has good tonality but does't do depth very well,  and so on.
> 
> For those who can enjoy music to the fullest and can use lamp wire, a $120 Blu-ray, and Sony speakers form Best buy... in some ways they are far luckier than I am.


 
  
 The thing about high end equipment is that it is MUCH more likely to be deliberately hobbled to create a "house sound" than midrange equipment is. The market for midrange equipment is very competitive, and the manufacturers are all vying to make the most accurate and perfect design possible. They manufacture in mass quantities, and they don't want to be stuck with a bunch of unsalable inventory if tastes fluctuate..
  
 High end audio is an entirely different market. They vie to make *unique* sound. They actually WANT to sound different. They manufacture in small batches at very high markups, so they can cater to specific requests for imbalances. A lot of audiophiles refuse to use equalizers and try to create combinations of colored equipment that cancel out the imbalances in their listening room. It is a very costly, imprecise and ineffective way of addressing the problem, but that is what they choose to do. So boutique manufacturers create different response curves, so people can mix and match to find the combo that works for them.
  
 My approach is different. I want to start from an accurate baseline and do my corrections as the last step in the chain using EQ. That way, any transparent player that I plug in sounds exactly the same, because they are all totally neutral and they are all going through the same EQ correction. I can plug in a $120 Sony blu-ray player, a $40 Coby CD player or an iPod and they all sound *perfect*. If your player is colored and your amp is colored too, then the only possible combination that works is the one you've settled on. God forbid your player should give up the ghost on you! You're back to swapping a half dozen different brands and models in to find the one that works with your peculiar sound signature. For me, it's easy. I had a $120 Sony blu-ray player and I bought an Oppo BDP103D to replace it. All of my DSPs and EQ curves worked with the Oppo right out of the box.
  
 Transparency and accuracy is a very good thing! It can save you a lot of money and effort.
  
 Recording studios use studio monitors that are carefully calibrated to have a flat response. The engineer mixes and balances to that standard. Every good studio uses the same sort of set up, so you can start a mix in New York and finish in LA without hearing completely different things on each coast. The closer you can get your home system to matching the calibration and presentation used in recording studios, the more music will sound *right* to you. My primary goal in my system was to try to duplicate the sound I hear in the studio as closely as possible. I use many of the same kinds of speakers, and they are calibrated to the same sort of response. As a result, I can put on SACDs or BD-A disks of classical multichannel mixes recoded in DSD and they will sound the best they possibly can. And I can put on CDs of Caruso records recorded acoustically before WWI that have been mastered by Mark Obert Thorne and they will sound the best they possibly can.
  
 I always hear people saying that a particular component or system is better at playing audiophile recordings than rock... or better at classical than jazz. That just tells me that there is an imbalance in the system that different kinds of recordings or music reveals. If they were truly balanced, *everything* would sound great.


----------



## bigshot

spacequest said:


> I think the goal of recording Live concerts IS to capture the live athmo, TOO !
> Could be done by a Multichannel Recording, ( Recording Engeneers in the past
> didn't have that and have to use stereo setups ) Read you have a home theatre,
> why did all big companys put DSP power in their amps to simulate those live rooms ?
> ...


 
  
 Generally, recorded live music that seems live has been recorded with numerous mikes and mixed together to sound live. Classical music is an exception to that sometimes, because classical music is balanced by the conductor, so the engineer doesn't have to do as much. But even then, the sound field of a multichannel mix isn't recorded with 5 mikes in the same positions as your speakers. That sound field is manufactured in the mix.
  
 I always work with Neumann and Shure mikes (plus whatever the mixer is used to working with) and I've found as good as a U-87 is, the placement of the mike and the processing (EQ, compression, etc) are MUCH more important than the quality of the mike itself. Also, if you ask the engineer about his mike pres, he will tell you all about the sophisticated and clean noise gates and processing they do on the fly. A lot of things are happening under the surface. It isn't just a flat recording with a straight line from mike to record.
  
 DSPs are the future of audio in the home. The difference that the proper DSP makes in my system is like night and day. I have a Yamaha AV receiver and Yamaha is on the forefront of DSP design. Someday, you will get equipment with the right DSP and you will try it, and never go back to bypass again.


----------



## SpaceQuest

bigshot said:


> Generally, recorded live music that seems live has been recorded with numerous mikes and mixed together to sound live. Classical music is an exception to that sometimes, because classical music is balanced by the conductor, so the engineer doesn't have to do as much. But even then, the sound field of a multichannel mix isn't recorded with 5 mikes in the same positions as your speakers. That sound field is manufactured in the mix.
> 
> DSPs are the future of audio in the home. The difference that the proper DSP makes in my system is like night and day. I have a Yamaha AV receiver and Yamaha is on the forefront of DSP design. Someday, you will get equipment with the right DSP and you will try it, and never go back to bypass again.



 


Oh, i own a "Big Block Yamaha" (Z11), but the DSP processing is only real fun for some big Blockbusters. Using these
Soundfields for Music i tent to let them off or use only a quad field for reproduction. I like pure sound !

At my Studio i use 2 Lipinski L-707 with 2 Pass X-250 and no DSP. For Multichannel Production i use a
TC Electronic System 6000 + the icon remote as DSP processor and for Surround Processing a z-K6 K-Surround
Processor. So these DSPs are more enjoyable for me than the Yamaha stuff.


----------



## Dark_wizzie

bigshot said:


> The thing about high end equipment is that it is MUCH more likely to be deliberately hobbled to create a "house sound" than midrange equipment is. The market for midrange equipment is very competitive, and the manufacturers are all vying to make the most accurate and perfect design possible. They manufacture in mass quantities, and they don't want to be stuck with a bunch of unsalable inventory if tastes fluctuate..
> 
> High end audio is an entirely different market. They vie to make *unique* sound. They actually WANT to sound different. They manufacture in small batches at very high markups, so they can cater to specific requests for imbalances. A lot of audiophiles refuse to use equalizers and try to create combinations of colored equipment that cancel out the imbalances in their listening room. It is a very costly, imprecise and ineffective way of addressing the problem, but that is what they choose to do. So boutique manufacturers create different response curves, so people can mix and match to find the combo that works for them.


 
 How much would a great neutral speaker cost then?


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## bigshot

There are no neutral speakers. Even if there were, the second you put one in your living room, it wouldn't be neutral any more. The neutrality is calibrated in EQ. A good speaker is one that can produce a wide range of frequencies loud without distortion. The size of the room also affects that, because a large room requires much more volume than a small one. If they can do loud clean, then you can EQ them into doing whatever you want.
  
 Generally, you can spend a lot on a set of low distortion ribbon or planar magnetic speakers, like the Magnepans and then supplement them with a high end subwoofer; or at the other end of the spectrum, you can haunt thrift stores until a nice old 70s 12 inch cabinet monitor turns up and do it on the cheap.
  
 I kind of split the difference. My mains are super sweet hand me down 12 inch 6 way studio monitors from the 70s that my brother had built custom for his McIntosh rig, and my subwoofer is a very pricey Sunfire by Bob Carver.


----------



## bigshot

spacequest said:


> Oh, i own a "Big Block Yamaha" (Z11), but the DSP processing is only real fun for some big Blockbusters.


 
  
 The most useful Yamaha DSPs are the "Stereo to 7.1", which perfectly re-channels stereo to take advantage of a true center and sub, and the Vienna Hall ambience, which can make notoriously dry Toscanini recordings and make them sound perfectly listenable.


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## Golden Ears

I happen to go against the audio grain of no EQ. I have had amazing results with parametric EQ with the digital programmable settings of analog filters used in McIntosh car audio. 
  
 I do wonder why there is not more parametric eq for  the homes. It is extremely rare to wander into anyones listening room at a house and see it optimized at all for sound. 
  
 The rooms transfer function layers sonically atop the original recorded venue... and can cause weird effects. sometimes more mid and near field listening can stop some of this but the RT60 of your room gets tossed out.
  
 There are times when I wish you could have a separate speaker for each instrument. And each speaker somewhat optimized for each instrument .. this is entirely unrealistic.   So the best you can do is try to set up to minimize the need for EQ and EQ as needed as little as possible like a sort of micro surgery.  
  
 One issue with multiple sound sources like stereo is that there are double the number of reflections which are slightly out of time as there is a gap between the stereo speakers that introduces some time smear. It isn't like the sound was emanating from a  tightly clustered single point a large  room like musicians in a  band. Stereo introduces some smear..mono does not ---particularly for a soloist. When you start doing multi channel all the primary arrival times are correct but he reflected arrival times are all askew. the more channels you have the worse it is ...unless each channel had one dedicated musician.
  
 Higher sample rates can hold full resolution 24/48  for multiple channels... 
  
 In regards to neutral... the correct speaker in the appropriate space, can sound fairly neutral. But put a huge speaker in a tiny room or vice versa.. OR a bright speaker in a  bright room... and the output is not neutral. It is a good thing that not all speakers are neutral or every one with a non neutral room would never have a chance  for balanced sound. You can not take  a huge speaker in a a tiny room and EQ it into a good fit. nor can you do the reverse.
  
 all of this talk is nice in theory... 
  
 In practical applications where we have imperfect transducers and associated  imperfect electronics and imperfect listening rooms the only place truly neutral components find a completely useful home is in the homes of audio reviewers where they can put a non neutral component into their seemingly neutral system of neutral components and tell you how it deviates from neutral. A lot of the reviewers systems are not as musical as one might expect them to be. For them their system has to serve two purposes..as a reviewing tool and hopefully for them to enjoy the music. Certainly having say apogee grands in your system might sound extremely nice, but you would not be able to review many amplifiers. so a lot of great gear that shines best when paired with only a few components can't work well in a reviewers system. So many reviewers are delighted when they get a new neutral component that sounds even slightly better than another older neutral component  because that is one of the few times they can upgrade that system.
  
 The place to find good systems.... in "reviewers homes", are not in gear reviewers homes... they are  in the homes of the "record reviewers"... who are not bound by only having neutral components. Their systems stay optimized and are not constantly disrupted. 
  
 Te worst thing for music isn't sample rates that are too high..it is sample rates that are too low. MP3 decimated the music industry.... it's impact was worse than MTV and Hometheater put together.


----------



## bigshot

Above a certain bitrate, AAC doesn't truncate the high frequencies. At 256, it gets up to 19kHz or so from what I've been told. At 320 it is all the way up to the edge of hearing. It saves space by eliminating masked high frequencies (i.e.: ones you can't hear).
  
 My theory is that it is a LOT easier to make a big speaker sound good in a small room than it is to make a small speaker sound good in a big room. You can always rein it in, but if the oomph ain't there, no amount of EQing will put it back. (naturally, I'm not talking about PA systems in a closet here).


----------



## Dark_wizzie

So... Do speakers have relatively higher distortion relative to headphones? Also... what if somebody had the budget to pick how large they want a music room to be? Would the sound quality be better in a large room (with the right speaker setup of course)?
 And finally, I know EQ can do great things but I find it hard to do. Like, if I google "How to use EQ" I get basically nothing for the home user. Do I tune by ear, then how will I know the sound is truly flat? And do I use a parametric EQ instead of a normal EQ with sliders? (Parametric is like a line you tweak right, so it's like having a million sliders?) I'd say most people don't use EQ, all people know to do is to buy more expensive gear to get better sound. That's kind of tragic but I can see why it happens. I sit down with the Foobar EQ and I'm thinking... "Great. Now what?" I learned not to drag the sliders above 0dB to prevent clipping though. 
	

	
	
		
		

		
			





 Somebody suggested that 24bit makes sense for people who do a lot of EQ and other postprocessing. Just how much post-processing, EQ, etc must be done to a 16bit file to allow the noise floor to raise high enough for me to hear it? It sounds kindda loony to me.
  
 I don't think MP3 is what killed sound quality. MP3 is lossy compression but you can always opt to get a lossless version to bypass any questions of whether the MP3 version is any good. It's the loudness wars and DR getting squashed. I don't see how that's directly related to people beginning to use iPods loaded with MP3 files.


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## Golden Ears

I don't think a lot of the goo high end gear is far from neutral at all. Over the past few years both high end tube pre-amps and solid state pre-amps are moving towards a more neutral middle ground.
  
 I will say this... and I am sure it won't go down well. When I take an entire system of neutral components where each component has no signature sound . and put them all together  I tend to get "nothing".  a rather lifeless boring studio sound that does not sound live at all.
  
 In a studio things seem to be  are chosen for :
  
 1. Reliability and serviceability
 2. As a tool to process sound
 3. As a tool to inspect sound
  
 and they really aren't choosen for listening enjoyment.
  
  
 Studio gear will try to make mixing mistakes sound a bit more obvious so it is easier to catch them before the final mix down. There are very few studio speakers that make their way into peoples homes as reference high end gear. It certainly isn't because studio gear isn't expensive enough....as something that looks like the same speaker as a consumer model might cost 2-3 times as much and hopefully has better crossover components inside and better magnets, spiders etc. ONE WOULD HOPE....err..
  
 And..
 If studio speakers were so good at making music enjoyable and life like- you might expect to see more reviewed in consumer audio journals.
  
 My first speaker was a studio monitor, I changed to a high end audio speaker The Pyramid Met -3 by Dick Sequerra, and a few pairs of speakers later I got some 3/5 BBC monitors..and found them involving and not terribly accurate to making music sound more alive.
  
 The last pair of Studio monitors I still own are Genelec s30D which have their own 24/96 DAC built in as well as a couple of other neat features like automatic digital switching, individual dip switches for every driver gain, trip-amplified with class a/b amps, and a whole bunch of bass shelving controls so you can really fit the monitor to the room. They output 122 db...and have a Decca derived ribbon tweeter. Without stock cables they are lack luster ho hum.. but can really sing with better power cables (better harmonics, cleaner more powerful defined bass, clearer mids, more information, less listener fatigue etc) ... probably the most dramatic conversion I have heard with good power cables.  This studio monitor is one of the only studio monitors that The Absolute Sound reviewed and liked.  I don't care much for the onboard DAC and just feed them a nice signal from a  tube pre-amp to the analog inputs.. but it is nice to know there is a DAC there in case of an emergency..or if you just want to simply play them off a laptop with a optical to AES-EBU converter.
  
 I at one time wished they had 24/192 DACs but recently I have come to like 24/96 more. I do think that some (not all) electronics can have trouble with higher sample rate artifacts.
  
 I also wonder if 24/48 might sound better on them because at the time of manufacture a lot of studios were using 48kHz and that might be the most optimized codec in the speaker. I'll have to try that.
  
 So that is my take on studio gear, I find of the systems that sound the most alive...they were not all sourced completely  from neutral components.. which are like cooking with salt and pepper only... instead of using a few interesting stronger  spices to get appropriate sonic accents which lend to the perception of live music.
  
 I don't find that the better high end gear is very  far from neutral I do think it resolves significantly without distorting the sound stage.... and might have some zing somewhere which if used appropriately in the right circumstance can help a system. 
  
 You can find high end speakers in mastering rooms. For instance I have seen Dunlavys in several mastering rooms.
  
 It would be interesting to hear what people who have made excellent audiophile grade recordings think about higher sample rates. Cookie Marenco of Blue Coast records uses higher sample rates.. and people think she gets amazing results in her living room recording.


----------



## bigshot

dark_wizzie said:


> So... Do speakers have relatively higher distortion relative to headphones? Also... what if somebody had the budget to pick how large they want a music room to be? Would the sound quality be better in a large room (with the right speaker setup of course)?
> And finally, I know EQ can do great things but I find it hard to do. Like, if I google "How to use EQ" I get basically nothing for the home user. Do I tune by ear, then how will I know the sound is truly flat? And do I use a parametric EQ instead of a normal EQ with sliders? (Parametric is like a line you tweak right, so it's like having a million sliders?)


 
  
 Speakers generally have higher distortion than headphones, but they sound much better.
  
 My room is about 22 wide by 30 long I would guess. It's a pretty good size. A little bit larger wouldn't be bad. Maybe 28 by 35. It would get me a little further from the walls all around. But a smaller room doesn't require as big speakers to fill it, so it's a trade off. I personally like enough room for the music to open up in and a soundstage scale that relates to the size of actual performers. My soundstage is about 20 feet wide and 8 feet tall. Here is a photo of my room....
  

  
 I know what you are talking about with EQ. There is a long learning period where you figure out what the numbers mean in terms of sound and train yourself to think one frequency band at a time. The most accurate way to calibrate is with tone sweeps, but you can get really close by ear with music if you train yourself and analyze and take your time. It took me a few months of parallel parking to get mine right. I'm not sure which is best... parametric or graphic. It depends on the problems you are correcting. A parametric is great for overall curves, and a 31 band graphic equalizer is better if you have a lot of narrow spikes and dips. I wish I had both, but I just have a parametric right now.
  
 Balancing 5.1 systems adds a whole new level of complexity, because the volume of each channel affects the EQ. And the EQ affects the relative volume. You have to keep bouncing back and forth between the gain and the tone until they are both balanced at the same time.
  
 If you set up a good parametric or graphic equalizer, I would be happy to guru you through it here. I have a few inexpensive CDs that I use to EQ by ear. If you had those and an equalizer, I could let you know how to start. Then you could let me know the problems you run into and I could guide you to the right direction to solve them. It's kind of fun actually and it makes a HUGE improvement in your sound. Equalization and DSPs are the two secrets of my system. Properly applied, you can squeeze audiophile performance out of relatively humble equipment.


----------



## bigshot

golden ears said:


> I will say this... and I am sure it won't go down well. When I take an entire system of neutral components where each component has no signature sound . and put them all together  I tend to get "nothing".  a rather lifeless boring studio sound that does not sound live at all.


 
  
 Oh no! You haven't heard a calibrated flat response then!
  
 First of all, in order to bring the sub bass up to neutral, it takes a LOT of power. Most single cabinet speakers plunked down in a living room don't get down anywhere near where my Carver sub goes. It can be made to produce flat all the way down to 16Hz. If I put on an organ CD, the walls shake. I actually have to dip the resonant frequency of my room using EQ to prevent that, It isn't totally critical to have a flat response very far below 30Hz, because you are getting below the range where we discern pitch, but it does help to have a balanced relative volume down there so it fills in properly. A lot of times you will hear systems where the pluck of an acoustic bass doesn't connect to the low end of the note... as if there is a gap between the pluck and the thump. You'll also hear descending bass lines (in Beatles songs particularly) where each bass note is supposed to be the same volume, but as it goes down you can hear the volume going up and down. These are clear indications that your bass isn't flat.
  
 With the upper mids, balanced response is absolutely critical. A sound engineer friend of mine demonstrated auditory masking to me on his reference system once. He asked me to close my eyes and listen for a difference. I did and all of a sudden the treble disappeared completely. Then it came back again. He asked me to estimate what frequency band he had changed. I guessed somewhere around 6-7kHz. He said, "Watch what I'm doing this time." and he reached for 2kHz and dialed it up just a few dB. I couldn't detect any difference in the mids but the treble completely disappeared. He explained that a frequency response imbalance can affect the octave divisions above it by masking out the harmonic frequencies. When he dialed up 2kHz, it killed 4kHz and 8kHz, essentially neutering the treble.
  
 Narrow spikes sprinkled through the core mids can wreak havoc with high frequencies and upper mids. That's why a balanced response through the core frequencies is so important. Especially in the octaves around 4kHz where the ear is most sensitive. (More info: http://en.wikipedia.org/wiki/Auditory_masking) When you have a balanced response, your bass is full and defined, the treble is clear and sparkling, and your midrange doesn't jump out at you. Best of all, the sound is clearer, because the subtle details of the mix are revealed, instead of being masked. We normally thing of clarity as being related to distortion. It is, but it is just as related to response.
  
 High sampling rates are totally unnecessary because 44.1 is capable of perfectly recreating up to and just beyond the range of human hearing- 22kHz. Frequencies beyond that can't be heard, much less be heard as an element of music. Carefully conducted double blind listening tests have proven that super audible frequencies add nothing to the quality of recorded music. However it can be possible to discern the difference between high sampling rate material and redbook. A lot of audio equipment isn't designed to reproduce super audible frequencies and end up distorting down an octave or two in the audible range. In these cases, high sampling rates are worse, not better. That is the only difference a human can discern. Recording studios occasionally need super audible frequencies for harmonic effects. But most studios rarely go above 96. Some I have worked with use 48, which is a standard for DVD release and they have no problem producing first class sound.
  
 For a home hifi listener, high bit rates and high sampling rates are just packing peanuts to pad out file sizes. They do absolutely nothing for human ears. That is a proven fact. And I've verified it myself with careful testing of my own.
  
 By the way, 48 doesn't sound better than 44.1. It's just that redbook standard is 44.1 and DVD standard is 48. You use whatever your end product is going to be released on. I believe 48 was chosen to do brick wall filtering without oversampling. It has nothing to do with the sound of the extra couple of notes worth of super audible frequencies.


----------



## Dark_wizzie

bigshot said:


> Speakers generally have higher distortion than headphones, but they sound much better.
> 
> My room is about 22 wide by 30 long I would guess. It's a pretty good size. A little bit larger wouldn't be bad. Maybe 28 by 35. It would get me a little further from the walls all around. But a smaller room doesn't require as big speakers to fill it, so it's a trade off. I personally like enough room for the music to open up in and a soundstage scale that relates to the size of actual performers. My soundstage is about 20 feet wide and 8 feet tall. Here is a photo of my room....
> 
> ...


 
 Nice room. Unfortunately I cannot currently afford a good listening room with acoustic treatment and good speakers, so I went with a headphone instead. But in the future when I'm older and I have a REAL job I hope to have such a room. I want it to be treated and calibrated very well for computer use... I listen to music at my keyboard and mouse and I can use the speakers for none-music activities. I read that you live down in southern California... Too far for me to drive to. Bay Area here... Maybe on day I'll pop in to that room if your offer is still open by then. 
	

	
	
		
		

		
		
	


	




 (Although I don't see myself being in Socal anytime soon.)

 Today I only run two Rokits (my sub was from Logitecsh z2300 but the control pod might have died) in my kitchen... entire computer desk with speakers are in the corner of the room. Hard to fix these issues, so I put on my headphones instead.
  
 Do the most distortion free speakers offer distortion the human ear cannot hear? And are headphones the same?
  
  
 Anyways, about the whole studio monitors topic... Isn't the point of monitors to have accurate reproduction of sound so the artist knows what he's doing to his track? If the artist's track sounds boring and lifeless on that setup, then we need to blame the artist instead....


----------



## bigshot

The general threshold of audibility for distortion is around 1%. Speakers can get up to 4%, but the kind of distortion isn't objectionable. Some distortion only affects narrow bands in the response. These are less serous than broadband distortion. So I guess it depends whether higher levels of distortion in speakers are audible or not.


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## stv014

The distortion of dynamic speakers and headphones depends significantly on the frequency and level of the signal. Generally, the distortion is higher at lower frequency and/or higher SPL. Therefore, the woofer may easily have a few % of distortion while playing loud bass, but at the same time it is only tenths of a percent for the midrange speaker, and maybe even less for the tweeter. It also depends on the SPL, as already mentioned. Good headphones have lower distortion than speakers at the same SPL, although loudspeakers with multiple drivers have the advantage of reduced IMD (i.e. a distorting woofer will not "rattle" the upper midrange and treble, because the higher frequencies are played by separate drivers). On the other hand, bad passive crossovers add some distortion themselves.


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## SunTanScanMan

bigshot said:


>


 
 That's a great looking listening room - I do like that it's not super modern looking. It's homely and cozy, so I can imagine it's perfect for relaxing and listening to music.
  
 Would the same principles apply for desktop speakers? That is, would it be worth using an equaliser to the effect you mentioned? I generally listen to headphones and speakers at low volumes. Once I got used to it, I found all the details there, plus it was relaxing to enjoy the music. With speakers I also have to be considerate to my neighbours.
  
 ---
  
 I remember when I first heard samples of 24-bit music on music streaming services and being slightly disappointed. Perhaps I had unrealistic expectations, but that feeling has been repeated since. So I've never really explored them further. Not to mention the premium price that they demand. Ultimately, perusing these forums, I've come to belief that 'good enough' is enough for me. I've not the patience or the time to chase the last ounce of SQ.
  
 I used to stream FLAC quality, which I enjoyed, but even then I could find very little, if any differentiation from 320kbps. So switched to streaming the latter. I'm content as I can save money to buy more CDs especially when they're so cheap. When I get the CDs, I rip to ALAC as I found it the most convenient way. Tried ripping Box sets using EAC and I mucked up the labelling, not to mention it took an absolute age to rip. _So lesson learned - waste less time ripping, chasing SQ, find good music, enjoy more time listening_. *So that's good enough for me*.
  
 I do think it's quite easy to get caught up with 'chasing the dragon' - a mixture of gadgets, and wanting to exploit them 'to their potential' can get addictive fast. I know I went crazy for a period saving up and buying the headphones I have now. It's fun switching around, but I still feel it's excessive. I was interested in them, and had to satisfy my curiosity. Luckily I was satisfied after 4.


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## RRod

I just wanna know if the marlin has any effect on frequency response.


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## MacacoDoSom

no!


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## castleofargh

rrod said:


> I just wanna know if the marlin has any effect on frequency response.


 

 no, it is balanced by the giant rabbit with horns on the left(yeah I'm a pro when it comes to animals).


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## bigshot

JACKALOPE!
  
 In the back, I have a mugwump and a mars attacks alien that absorb stray cosmic waves!


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## DiscoProJoe

Great article. That was the one I found while googling a few months ago. The section I've copied and pasted below says it all. MP3s at 192 kbps rule!  
	

	
	
		
		

		
			




  
 ------------------------------------------------------------
 When does 24 bit matter?
  
 Professionals use 24 bit samples in recording and production for headroom, noise floor, and convenience reasons.
  
 16 bits is enough to span the real hearing range with room to spare. It does not span the entire possible signal range of audio equipment. The primary reason to use 24 bits when recording is to prevent mistakes; rather than being careful to center 16 bit recording-- risking clipping if you guess too high and adding noise if you guess too low-- 24 bits allows an operator to set an approximate level and not worry too much about it. Missing the optimal gain setting by a few bits has no consequences, and effects that dynamically compress the recorded range have a deep floor to work with.
  
 An engineer also requires more than 16 bits during mixing and mastering. Modern work flows may involve literally thousands of effects and operations. The quantization noise and noise floor of a 16 bit sample may be undetectable during playback, but multiplying that noise by a few thousand times eventually becomes noticeable. 24 bits keeps the accumulated noise at a very low level. Once the music is ready to distribute, there's no reason to keep more than 16 bits.


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## limpidglitch

bigshot said:


> JACKALOPE!
> 
> In the back, I have a mugwump and a mars attacks alien that absorb stray cosmic waves!


 
  
 Jackalopes are lame.
 Wolpertingers are the real deal.


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## castleofargh

wow you guys! I'm so jealous. in france we're stuck with the dahu http://en.wikipedia.org/wiki/Dahu


 our creature is sooo lame compared to yours.


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## bigshot

Here is my Mugwump!


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## mmeysarosh

For the vast majority of consumers, its generally correct for a variety of reasons. In comparing 16 bit to 24 bit recording depth, people can hear dynamics beyond a the 16 bit level, only a limited number of playback devices are capable of resolving that data in analog performance. The iPhone 6 was Apple's first device that had enough performance from its headphone output to resolve 17-18 bits worth of data. All prior models really couldn't do more than 16 bit, if not slightly less.
  
 The absolute top DAC devices on the market achieve at best 21 bits when measures 20Hz-20Khz. Even then, very few amps and speakers combos are capable of handling more than 18 bits of dynamic data while keeping distortion in check. Some of the most powerful home speakers ever built have no more than 118db, just below 20 bits of dynamic range. As for sampling rates above 48Hkz, its primary benefit is moving the roll off filters outside the audio band. Poorly implemented filters will affect phase response and have some audible effects in parts of the audible range. Modern filters have improved significantly in performance and up-sampling is no longer as necessary for good audio quality.
  
 Engineers will often use regular home gear to listen to a finished product. You have to ensure that the majority of the target audience can playback the recording with no issue, and this means mastering the output to fit within the performance range of their playback device. Something that 16/44 does for the overwhelming majority of people. As for the audiophile group who actually own equipment that perform beyond the norm? Its far too small of a group of people to take the cost of mastering above 16 bit to be worthwhile. Recording at 24 bit does exactly as one had mentioned, provide more dynamic range to deal with issues later in the mastering.
  
 Many consider vinyl recordings to sound better than their CD counterparts. Quite often, they are correct. The digital copy is mastered for the car or headphones, while the vinyl edition is mastered for a turntable paired to a complete stereo. Engineers understand and adapt their mastering process accordingly.


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## bigshot

People can't hear beyond a 16 bit level at the same time. Their ears shift sensitivity as volume rises and lowers. In practice, 40dB of dynamics is about the most someone could hear at one time.


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## mmeysarosh

Human impulse dynamic hearing can reach 120db. Granted you wouldn't want to do it all day, unless you really would like to eventually shut out the rest of the world.


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## Steve Eddy

mmeysarosh said:


> Human impulse dynamic hearing can reach 120db. Granted you wouldn't want to do it all day, unless you really would like to eventually shut out the rest of the world.




No, the instantaneous dynamic range of human hearing is what Bigshot says. Basically, you're not going to hear much below about 40dB of what average levels are at the time.

se


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## mmeysarosh

While you will only be able to discern sound up to 40db below the RMS power of a track, you can withstand significantly more above that in a short impulse. Again, no one would use it as it has limited practical purpose. Another thing to note is that our hearing has significantly different dynamic response capability throughout the audio band as well. The limits of low level audibility is lower for a signal at 1Khz than it would be at 10Khz or 100hz.


----------



## bigshot

mmeysarosh said:


> While you will only be able to discern sound up to 40db below the RMS power of a track, you can withstand significantly more above that in a short impulse.


 
  
 If you've ever been to a shooting range, you know what happens to your ears with loud short impulses. With the very first shot, they start to close down and you can't hear quiet stuff any more.
  
 But it isn't about earsplitting transient peaks that make you flinch. It's about listening to music. Peak level for comfortable listening to music at a relatively loud volume isn't going to get anywhere near 120dB. Especially if you are working with an ambient room tone that is creating a noise floor of its own. 60dB is probably already into the range of overkill for music.


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## castleofargh

and it's a false problem anyway as recorded albums don't actually use all that much dynamic.


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## RRod

castleofargh said:


> and it's a false problem anyway as recorded albums don't actually use all that much dynamic.


 
  
 Lowest RMS I've found in my music is -66dB (3s value, max peak normalized to 0dB).


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## castleofargh

yup I'm also in the 60/65db at best in my library.


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## Music818

Per my understanding, the sampling rate is talking about the numbers of sample acquired per second. The more the samples acquired, the higher the accuracy when converting back to analog signal as compare with the original signal. Is it correct?
 By therory(not talking about audible or not), higher sampling rate should get a better resolution than lower one. Right?


----------



## castleofargh

music818 said:


> Per my understanding, the sampling rate is talking about the numbers of sample acquired per second. The more the samples acquired, the higher the accuracy when converting back to analog signal as compare with the original signal. Is it correct?
> By therory(not talking about audible or not), higher sampling rate should get a better resolution than lower one. Right?


 

 if the analog conversion was done stupidly like on a good old barrel organ with paper sheets and holes in it, then you would be perfectly right. in reality the digital to analog process is a little more than adding dots and pray that it will make an ok line. I suggest you to go read about that conversion if you're interested.
 so in effect, the added precision isn't automatic, and the value of it is far below what pro highres people imagine(aka audible).
 the most obvious effect of having higher sampling rate is to be able to record higher frequencies. but hearing ultrasounds is a very doubtful theory and so are the perks of recording them.
 so benefits= yes . useful ones= mehhh
 (of course I'm saying this for the recorded music on our album and us music listeners, else there are great uses for high sampling in audio).


----------



## limpidglitch

music818 said:


> Per my understanding, the sampling rate is talking about the numbers of sample acquired per second. The more the samples acquired, the higher the accuracy when converting back to analog signal as compare with the original signal. Is it correct?
> By therory(not talking about audible or not), higher sampling rate should get a better resolution than lower one. Right?


 
  
 A similar question was asked (and answered) here.


----------



## Music818

Quote:


castleofargh said:


> if the analog conversion was done stupidly like on a good old barrel organ with paper sheets and holes in it, then you would be perfectly right. in reality the digital to analog process is a little more than adding dots and pray that it will make an ok line. I suggest you to go read about that conversion if you're interested.
> so in effect, the added precision isn't automatic, and the value of it is far below what pro highres people imagine(aka audible).
> the most obvious effect of having higher sampling rate is to be able to record higher frequencies. but hearing ultrasounds is a very doubtful theory and so are the perks of recording them.
> so benefits= yes . useful ones= mehhh
> (of course I'm saying this for the recorded music on our album and us music listeners, else there are great uses for high sampling in audio).


 
 Thanks for the reply. Do you mean the "sampling rate" has some special method using in audio's ADC process? That's higher sampling rate doesn't has more data acquired per second? Appreciate if there's any information can be shared.


----------



## castleofargh

music818 said:


> Quote:
> 
> 
> castleofargh said:
> ...


 
 http://xiph.org/video/ should be a good start.


----------



## RRod

music818 said:


> Quote:
> Thanks for the reply. Do you mean the "sampling rate" has some special method using in audio's ADC process? That's higher sampling rate doesn't has more data acquired per second? Appreciate if there's any information can be shared.


 
  
 It does mean more data is acquired per second. What that extra data allows the ADC to do is to capture higher frequencies before running into the problem known as "aliasing." The usefulness of this depends entirely on how useful those higher frequencies are, and if we're talking about human hearing, they aren't very useful.


----------



## Music818

rrod said:


> *It does mean more data is acquired per second.* What that extra data allows the ADC to do is to capture higher frequencies before running into the problem known as "aliasing." The usefulness of this depends entirely on how useful those higher frequencies are, and if we're talking about human hearing, they aren't very useful.


 
 Oh! That's what I want to clarify. Thankyou!
 Higher sampling rate -> more data acquired per sec. -> higher accuracy (terms may not correct, may use resolution or something else).


----------



## RRod

music818 said:


> Oh! That's what I want to clarify. Thankyou!
> Higher sampling rate -> more data acquired per sec. -> higher accuracy (terms may not correct, may use resolution or something else).


 
  
 Yes, you can get a more exactly reproduced analog waveform if that waveform has high frequency (> 22kHz) components. The problem is that we can't hear those higher frequencies, so while the waveform will be measurably more accurate, it's not *audibly* better to our ears.


----------



## Music818

rrod said:


> Yes, you can get a more exactly reproduced analog waveform if that waveform has high frequency (> 22kHz) components. The problem is that we can't hear those higher frequencies, so while the waveform will be measurably more accurate, it's not *audibly* better to our ears.


 
 Got it and as I mentioned, I'm not talking audible signal, just want to clarify the sampling rate issue. Thanks!


----------



## Music818

castleofargh said:


> http://xiph.org/video/ should be a good start.


 

 Interesting...... Thank you!


----------



## mmeysarosh

Certainly, and even a better example is loud concert. The conversation you have afterwards is oddly loud for good reason. I'm also quite sure that if you measured the sustained dynamic range of hearing across various frequencies, it would also show some significant curves with wide variations among a large number of people. From a point of silence to a loud shot, an average person would have able to hear up to 120db in range. You will lose your hearing if subjected to that kind of abuse in any regularity. Again, not very usable or practical
  
 You really wouldn't want even attempt a signal that would have that much of a dynamic impulse over the RMS power of the track from a reproduction standpoint. If a track is recorded at 24 bit with an RMS power of around -64db and was being played at a RMS volume at 80db. If you would send out a sample at -24db, the signal signal that would drive nearly any system into protection, overload, or physical damage. Its true if 16 bit recordings just lowered the RMS power to -16db RMS to -24 RMS, they would have plenty of room for most recordings out there. The loudness or recordings silliness has unfortunately given us otherwise.


----------



## Skyyyeman

Check out the lucid article on sampling rate by John Siau, Chief Engineer at Benchmark Media Systems, Inc., maker of audiophile and pro audio digital equipment (including the Benchmark DAC2):
  
 http://benchmarkmedia.com/blogs/news/14949325-high-resolution-audio-sample-rate?utm_source=Application+Notes&utm_campaign=72152862aa-


----------



## RRod

skyyyeman said:


> Check out the lucid article on sampling rate by John Siau, Chief Engineer at Benchmark Media Systems, Inc., maker of audiophile and pro audio digital equipment (including the Benchmark DAC2):
> 
> http://benchmarkmedia.com/blogs/news/14949325-high-resolution-audio-sample-rate?utm_source=Application+Notes&utm_campaign=72152862aa-


 
  
 He puts an awful lot of stock in the audibility of frequencies above 20kHz, ignoring the fact that most people have to amp things WAY up to hear anything up there. If there are arguments to be made for these high frequencies, they probably need to look beyond the ear.


----------



## mmeysarosh

Setting the sampling rate this high was a solution to move the effect of the filters out of the audio band, and sometimes quite audibly so. You would still have a filter, but its effect in audible sounds was next to nothing since it would begin its work at the 40Hkz range. One of the many times this became evident was through the development of DSD. In order to deal with the noise rise and frequency gained, the engineers developed a filter that began to take effect in the upper 20's of FR response. This created a difference that in measured response in the hi frequency domain and had PCM engineers as their filters would start in the upper and of the audible band. Realizing the effects of their filters had, development went further to enhance them.
  
 Gordon Rankin and Charles Hanson both have spoken about this at length. Both are lead engineers for their respective audio companies.


----------



## coinmaster

bigshot said:


> The limitation is in human ears. It doesn't matter how high a frequency you want your stereo to produce and how wide a dynamic range, it all comes down to whether human ears can hear it.
> 
> Audiophools love to spend lots of money pushing the decimal point further and further to the left and making the frequencies go higher and higher, but at a certain point, it all becomes moot because only bats can hear it.


 
 Bats need good music too, ya'know.


----------



## dazzerfong

skyyyeman said:


> Check out the lucid article on sampling rate by John Siau, Chief Engineer at Benchmark Media Systems, Inc., maker of audiophile and pro audio digital equipment (including the Benchmark DAC2):
> 
> http://benchmarkmedia.com/blogs/news/14949325-high-resolution-audio-sample-rate?utm_source=Application+Notes&utm_campaign=72152862aa-


 

 That article is forgetting one thing when it comes to the 'alias': use a low-pass filter! Bam, problem solved: anything higher than what you want your drop-off frequency to be will be inconsequentially small. Also, there's a reason all his postulations come down to 'what-if': if it's true, he wouldn't need to go 'what-if' and just say it as fact.
 Oh, and this line:
  
_What if the conductor had moved his baton a little faster in 1951? Would we have a slightly higher sample rate on the CD? It is entirely possible!_
  
 Huh? Tempo of a song has NOTHING to do with a sampling rate, at least not with PCM. With DSD, I might see why, considering the principle of DSD is on delta-sigma modulation (ie. differential height between frequencies, as opposed to pulse-code modulation), and hence a faster song with more rapid changes will work the DSD. A faster song will just have faster decaying waves, but the frequency of each wave will more or less remain the same (unless we're talking about digital manipulation, in which case, yes, if speed up, you can very possibly affect the frequency if you're not careful. Hence, the high-pitch associated with sped-up sequences).
  
 His explanation on bit rates, however, is a bit better, however, it fails to acknowledge that human ear's dynamic range of 130 dB is pointless when your ears start popping when you reach even 10% of that. What's worse is that, yes, the SNR of 16-bit will never reach 24-bit, but _who cares_. Yes, you get a better 'spec', but it's a useless spec in the long run. Unless you need something with a dynamic range of > 96 dB, then come back to me and you have your argument.
  
 I'll be honest, I have to give him some credit at least. While his grasp on the principals of signals and DSP is top-notch, his obsession with specs is taking over his acknowledgement of human limitations in reality. So, in other words, a fellow engineer, and I respect that: taking the approach that _might_ prove to be better later even though, as of now, doesn't really do anything.


----------



## jcx

we've gone over this a lot  - there is reason to be suspicious of 44.1 as possibly too low - hints, but no generally accepted controlled listening test "proof"
  
 but its hard to see that higher sample rate is "bad" - if nothing else it relaxes bit depth requirements 
  
 increased sample rate at the same bit depth does lower audio frequency noise by spreading some of the quantization noise over inaudible ultrasonic frequencies - noise shaped dither can take advantage of the extra bandwidth to push even more audio frequency noise beyond our hearing range
  
 16/96 would be the much better direction in my opinion for hi rez instead of the stupid 24/44.1 Beatles offering
  
http://www.meridian-audio.com/w_paper/Coding2.PDF


----------



## dazzerfong

jcx said:


> we've gone over this a lot  - there is reason to be suspicious of 44.1 as possibly too low - hints, but no generally accepted controlled listening test "proof"
> 
> but its hard to see that higher sample rate is "bad" - if nothing else it relaxes bit depth requirements
> 
> ...


 

 I'll settle for the guys mixing the stereo up properly: please, my ears bleed when I hear stereo, and with mono, it sounds stale.
  
 Also, having skimmed that article a while ago, I'm a bit suspicious how they arrived to this conclusion:
  
_One is forced to conclude that there is some real and much anecdotal evidence to suggest that the 20kHz bandwidth provided by a PCM channel using a sampling rate of 44.1kHz is inadequate_
  
 Ummmmm, what? The _only_ claim he makes for that, is that _the author has experienced listening tests which showed that the sound is degraded by the presence of normal (undithered) digital anti-alias and anti-image filters_. Yeah no: it's an opinion. Pointless in your argument.
  
 What's worse, _a super-HF-capable chain has yet to be developed to the same level of performance as the current reference_. In other words, even _if_ there was a difference, our equipment's limited anyway. Fix that first _then_ worry about the sampling rate. Or not. It's pretty hard to disprove something that can't be disproven because you lack the equipment to test it.
  
 However, the paper _does_ have a point near the end, where _'increase the sample rate by a margin sufficient to move the phase, ripple and transition regions further away from the human audibility cut-off_' will eliminate problems with imperfect filters. Well, do that when you record, but lop off all the excess frequencies once it's already converted to digital. It's not like you could hear it anyway.
  
 All in all, everyone's relying on 'it's better to be safe than sorry' as opposed to actual, you know, proof that it actually makes a difference.


----------



## bigshot

Whenever you draw a line, there's always going to be some joker who points ten feet beyond it and says it belongs there.
  
 Redbook was designed to be audibly transparent. It has always proven to be audibly transparent in controlled listening tests. Unless you plan to listen to music with something other than ears, it's perfect.


----------



## jcx

I was considering the thread title and the technically poor responses to the recent:
 Quote:


music818 said:


> Per my understanding, the sampling rate is talking about the numbers of sample acquired per second. The more the samples acquired, the higher the accuracy when converting back to analog signal as compare with the original signal. Is it correct?
> By therory(not talking about audible or not), higher sampling rate should get a better resolution than lower one. Right?


 
  
 higher sample rate does give more resolution in the conventional audio bandwidth - no supersonic hearing required  - actual audio band S/N does increase with increased sample rate at the same bit depth
  
 and dither taking advantage of the increased bandwidth above audio can make the S/N improvement huge enough that only 11-12 bits would be needed at 96 K -  SACD DSD give 120 dB audio S/N with just 1 bit and 2.8 MHz sample rate
  
  
 and on the "audible" frequency front Stereophile's 50+ year old male from industrial societies demographic almost certainly doesn't need more than 44.1k sample rate  - but I don't think it is possible or prudent to declare no human can tell the difference with current evidence


----------



## bigshot

jcx said:


> and on the "audible" frequency front Stereophile's 50+ year old male from industrial societies demographic almost certainly doesn't need more than 44.1k sample rate  - but I don't think it is possible or prudent to declare no human can tell the difference with current evidence


 
  
 Ten more feet to the right, eh?


----------



## RRod

jcx said:


> I was considering the thread title and the technically poor responses to the recent:…
> 
> higher sample rate does give more resolution in the conventional audio bandwidth - no supersonic hearing required  - actual audio band S/N does increase with increased sample rate at the same bit depth
> 
> ...


 
  
 People asking if "more samples give better resolution" usually aren't asking about s/n or noise shaping.
  
 It's also not incredibly hard to play 20k tones and see if people can hear them. What evidence exactly do you need that audiology and tons of ABX testing hasn't provided?


----------



## jcx

I think you have drawn a hard line that is not justified by the science - I do own Fastl, have read a fair amount on psychoacoustics - the data just isn't that complete
  
 I quite agree that home listening to commercial recorded music hasn't been shown to require higher than 44.1k - it is likely fine for 99+% of adults that can afford to purchase audio systems where any question could arise
  
 being in the Stereophile reader demographic myself I don't claim to hear higher than ~14k today
  
 but the graphs in the textbooks are mostly from few dozens of college students - show me 10k+ extended beyond the standard frequency range audiograms from both sexes, toddlers to adults from environments with low noise exposure then we would have more to estimate what human hearing limits might be
  
 if nothing else 20kHz is just too "round" a number to be trusted - I expect up to ~24kHz hearing that has been occasionally claimed would likely be confirmed
  
 if only for a 3rd sigma tail of prepubescent girls living in preindustrial quiet environments it still would invalidate the doctrinaire "no human can hear..." hard line drawn here


----------



## RRod

jcx said:


> I think you have drawn a hard line that is not justified by the science - I do own Fastl, have read a fair amount on psychoacoustics - the data just isn't that complete
> 
> I quite agree that home listening to commercial recorded music hasn't been shown to require higher than 44.1k - it is likely fine for 99+% of adults that can afford to purchase audio systems where any question could arise
> 
> ...


 
  
 Even if some minute % of people can hear up to 24000, we're splitting hairs. Are those 3 semitones of musical content at frequency levels that require high volume to hear even as single tones going to suddenly open music up for these people? I doubt it. And I think the point where you are talking about testing toddlers for high-end hearing ability, we're jumping the shark a bit.


----------



## C.C.S.

I have one small question about equalization that I would like to direct to anybody who is very experienced with it. I'm using a parametric EQ app to correct small imbalances in the response of my current pair of headphones, including a dip in the mids, centered on 2 kHz. Because this was the only significant dip that needed to be corrected, I corrected it by +4 dB, but I've also been told it's best to use subtractive EQ only, so I set global gain to -4 dB, in order to drop the entire curve by 4 dB.
  
 Is this the same, in practice, as subtractive EQ, or should I reset global to 0 dB and build the same curve using subtraction?


----------



## bigshot

jcx said:


> I quite agree that home listening to commercial recorded music hasn't been shown to require higher than 44.1k


 
  
 uh... isn't that what we're talking about here?! Do we have people here listening to super audible tones on their home stereos?!


----------



## castleofargh

jcx said:


> I was considering the thread title and the technically poor responses to the recent:
> Quote:
> 
> 
> ...


 
  

 as pure "can human do this?" kind of question, I understand your concern.
 but for audio, who give a pooh? if we were to even consider going past 44khz only for people with extended hearing, then we should also make sure all microphones extend correctly above 20khz, then hire only sound engineers that can themselves hear ultrasounds(lol), so that they could make a good mix. redo all the headphone measurements because many of the actual mics in the dummy heads aren't reliable in the trebles already.
 all this for a few young kids who can hear ultrasounds? and wouldn't music with rich loud ultrasound content damage their ears faster than the usual music?
  
 if we push the default sample rate, I wish we would do it for a real reason, not for kids who can hear ultrasounds and the 3 guys who were raised in a bunker and nobody could talk.
  
  
  
 the SNR "problem" on 16/44, I must say I don't understand it. well I do, we can do better, so why not do better? but in practice, people need to have a pretty great sound system to even pretend to be concerned by quantization noise. and then, it's not like we have a lot of conclusive ABX for 16/44 vs 24/96. so to me that answers the SNR problem pretty well on its own.
  
 from my own experience being mostly a portable audio guy with IEMs(and it's been like that since in ear came to be), this noise floor thing is not a joke, but more like a lame shame. take a DAP, read the specs: woot SNR -97db that's great! (well actually no it's not, it's barely enough for 16/44).
 oh but wait, how was that measured on a DAP? full output? I think I've seen some doing it for a simulated 80 or 90db into 32ohm load, are some going for 1V? IDK really how it's done on DAPs. I've seen specs with SNR going up with the gain switch changed to high, so I guess they really just push as much as they can to get a loudest signal voltage value as a way to increase the SNR ratio.
 anyway that's all great, but what about my IEM that goes at 90db with 0.02v(that's actually not the most sensitive stuff I own) and comes closer to 8ohm than to 32?
 if the measured noise doesn't also go down with the volume knob(and some component noises in amps do just that), I might end up with a noise that could be close to 35db louder(if the measure was actually done for a volume set at about 1V). add to that, me not listening to music loudly and certainly not at 90db, and I end up picking up a noise that is very much audible compared to the music. and that's how it is most of the times for me with DAPs.
 so I really can't imagine how something near -96db is my problem in audio? I feel like we have the perfect cars and fight about getting even more horse power to go faster, while not even looking at our good old square tires made out of wood. at least that's what it looks like in my audio world where the top IEMs/CIEMs are multidrivers with super high sensi, and most sources still don't have low impedance or low noise, and actually tend to have bad specs into low impedance loads. but all I read about on the net is: file format, audiophile µSD card, "-96db isn't enough we need highres", "I want to upgrade my DAC", silver cable, and "OMG 0.5ms jitter!".
 I don't get it.
  
 and on speakers, pretty much half the people I visited had a very clear hum on their system(usually the more expensive and powerful, the louder the hum...).
  
 so they all just dismiss that? dismiss the 1%distortion of many headphones, and focus on -96db quantization noise and 22khz+ signal?
 from my perspective yes 24bit audio and anything over 48k is worthless. untill the rest is fixed it is.


----------



## bigshot

castleofargh said:


> and on speakers, pretty much half the people I visited had a very clear hum on their system(usually the more expensive and powerful, the louder the hum...).


 
  
 I don't have any hum. What are you referring to?


----------



## castleofargh

bigshot said:


> castleofargh said:
> 
> 
> > and on speakers, pretty much half the people I visited had a very clear hum on their system(usually the more expensive and powerful, the louder the hum...).
> ...


 

 anything really, from buzzing from RFI into the amp, to ground loop, to the noise of other electric appliances turnig ON or OFF, to the classic power line hum at 50hz (I guess you guys get some at 60hz instead ^_^). think of all the reasons that made somebody come up with a ludicrous snake oil product.
 I'm not saying you can't avoid them, you sure can. same with my IEMs, I have a portable amp that gives me a great clean background out of almost anything. so if you really go for good specs(real ones, not headfi ones) you get them. what I'm saying is that we will end up debating high res or mp3 or whatever over dinner because somebody will say that I like audio gadgets, and then I will be offered a demonstration with 2 different masters(lol) on a sound system that has some matter of audible hum. I find that very strange as a thought process to be so concerned about the not so audible when there is something audible under their nose and they don't seem to notice it.


----------



## bigshot

I've found that ground loop problems and RF interference are things you either have or you don't. If you have it, you just track down the bugaboo and get rid of it. I haven't had a problem with that in decades though. I think the standards for electricians have gone up. Stuff like that only shows up in old houses.


----------



## Don Hills

dazzerfong said:


> _What if the conductor had moved his baton a little faster in 1951? Would we have a slightly higher sample rate on the CD? It is entirely possible! ... _


 
   
No, the sample rate was fixed at 44.1 (originally 44.056) KHz for a different technical reason. They would have had to develop new tape machines, and it was one step too far. Instead, they used modified video recorders. The sample rate was determined by how many samples they could fit on a video scan line.


----------



## dazzerfong

don hills said:


> No, the sample rate was fixed at 44.1 (originally 44.056) KHz for a different technical reason. They would have had to develop new tape machines, and it was one step too far. Instead, they used modified video recorders. The sample rate was determined by how many samples they could fit on a video scan line.


 
 Ah yes, I stand corrected on that. Thanks for pointing that out.


----------



## bigshot

I worked for the sound mixer who recorded the first television program recorded digitally... Barry Manilow's Copacabana special. My boss retired his Nagra and recorded on those old two part Sony Beta PCM machines. He had his shipped from Japan. He was one of the first professionals working in digital.


----------



## Thad-E-Ginathom

skyyyeman said:


> Check out the lucid article on sampling rate by John Siau, Chief Engineer at Benchmark Media Systems, Inc., maker of audiophile and pro audio digital equipment (including the Benchmark DAC2):
> 
> http://benchmarkmedia.com/blogs/news/14949325-high-resolution-audio-sample-rate?utm_source=Application+Notes&utm_campaign=72152862aa-


 
  
 I skim-read, but he seems to be happy to settle for 88KHz.
  
 If I remember rightly, digital pioneer JJ suggests that around 60Khz would be best. Again, IIRC, Lavry also talks of an optimum, and so does Monty. It is not that _higher sample rates is bad._ Or good. But that there is an optimum, beyond which, for various reasons, more is no better, and may be worse.
  
 Neither the music industry nor the hifi audio industry (obviously both have to world together, not only in the name of _music for fun and profit,_ but also so we can actually play the music) has taken any notice of _optimum. _A nice, trendy-looks-digital sequence of 48, 96, 192 and so on, looks much better to the marketing guys. 96 looks better than 48; 49 doesn't, whether it is or not.
  
 Because of the numbers game, music/audio faces the worst period in its history: huge investment in manufacturing and huge costs to customers. It suits the marketing guys. The real engineers, rather than those who are on the leashes of their marketing men, must be truly sick.
  
  


rrod said:


> He puts an awful lot of stock in the audibility of frequencies above 20kHz, ignoring the fact that most people have to amp things WAY up to hear anything up there. If there are arguments to be made for these high frequencies, they probably need to look beyond the ear.


 
  
 I don't see why 44.1 should be set in stone. I hear that I don't hear over 20kHz (in fact, being a bit old and a bit more deaf, I personally hardly make it into double figures) but I also hear that there are, or may be,  engineering benefits that arise from higher (but not ever-increasing) sample rates. If it is easier for the engineers to bring us 20Hz-20kHz at sample rates of 48, 60, or even 96, then let it be so, but lets stick to the technical and engineering realities, not the night-and-day differences that are caused by spending money ...or that might actually be there, but be caused because the DAC doesn't treat the sample rates equally.
  


dazzerfong said:


> I'll settle for the guys mixing the stereo up properly: please, my ears bleed when I hear stereo, and with mono, it sounds stale.
> 
> ... ... ..


 
  
 And I'll settle for properly-mastered music, an end to "loudness-wars" compression ...and proper research into whatever the next generation of better music recording/reproduction might be.


----------



## stv014

> Originally Posted by *Thad-E-Ginathom* /img/forum/go_quote.gif
> 
> I don't see why 44.1 should be set in stone. I hear that I don't hear over 20kHz (in fact, being a bit old and a bit more deaf, I personally hardly make it into double figures) but I also hear that there are, or may be,  engineering benefits that arise from higher (but not ever-increasing) sample rates. If it is easier for the engineers to bring us 20Hz-20kHz at sample rates of 48, 60, or even 96, then let it be so, but lets stick to the technical and engineering realities, not the night-and-day differences that are caused by spending money ...or that might actually be there, but be caused because the DAC doesn't treat the sample rates equally.


 
  
 The digital filters in modern DACs can cleanly reconstruct 0 to 20 kHz at 44.1 kHz sample rate without much difficulty. Roll-off and imaging can be limited to the 20-24 kHz range with a ~2 ms impulse response length.


----------



## RRod

thad-e-ginathom said:


> I don't see why 44.1 should be set in stone. I hear that I don't hear over 20kHz (in fact, being a bit old and a bit more deaf, I personally hardly make it into double figures) but I also hear that there are, or may be,  engineering benefits that arise from higher (but not ever-increasing) sample rates. If it is easier for the engineers to bring us 20Hz-20kHz at sample rates of 48, 60, or even 96, then let it be so, but lets stick to the technical and engineering realities, not the night-and-day differences that are caused by spending money ...or that might actually be there, but be caused because the DAC doesn't treat the sample rates equally.


 
  
 Engineering benefits are all good and fine, and few of us have any issue with recording at 24/192 or whatever. The issue is telling people they need, on the user end, more than 16/44.1 to get a "realistic" musical experience. What people are missing for that experience are good speaker setups, not lower noise floors or higher frequencies. Even then, their attempts can be thwarted by bad mastering, which can happen at all sample specs.


----------



## Thad-E-Ginathom

rrod said:


> Engineering benefits are all good and fine, and few of us have any issue with recording at 24/192 or whatever. The issue is telling people they need, on the user end, more than 16/44.1 to get a "realistic" musical experience. What people are missing for that experience are good speaker setups, not lower noise floors or higher frequencies. Even then, their attempts can be thwarted by bad mastering, which can happen at all sample specs.


 
  
 I don't object to 16/44.1, either!
  
 I do worry about those "lower noise floors" that people "hear." I can turn everything up to full, and, whether on speakers or on headphones, I hear silence. If I didn't, I would be looking for the fault. I did not have to pay $-thousands for that, either: my source is PC/ODAC.
  
 (oh... yes: I have tried _playing a file_ of silence. I'd hate to think that my low noise floor was just the result of an inactive device 
	

	
	
		
		

		
			





 )
  
 So, whether it's cables or bit depths, I don't really believe in these perceived lowered noise floors unless there was something wrong before. Or, at least, as I retain a certain unwillingness to tell people that they are lying about their experience 
	

	
	
		
		

		
		
	


	




, I treat their association of cause and effect with due suspicion, and I consider that the "cables inside their heads" might be coming into play. Needless to say, I apply the same suspicions to my own perceptions


----------



## bigshot

Recordings have noise floors too. Put a mike in a recording booth and turn on the air conditioning on a hot day, and you have a noise floor. It's very low, but it is there above the noise floor of the digital file. People who report hearing the noise floor of 16 bit are actually hearing the room tone in the recording studio, or the noise floor of the master tape (if it is an analogue recording).


----------



## ShadowSkulkerer

So...  Please don't mistake this as arguing.  I'm 21, quite ignorant on the matter and I'm really not up on science and physics and things.  I just recently got into mid-fi type stuff.  You can see my profile for my equipment.  I'm very happy with it btw.  But can someone explain to me why the 24bit/192khz Phil Collins album Face Value I bought on Pono Music sounds better than the songs I ripped from Phil Collins Greatest Hits CD.  I listened through my Fiio X3 and Headphones to the CD In the Air Tonight and the Pono version and I thought the Pono one was clearer and more detailed.  And I'm not saying various parts had more punch... 
  
   Allow me to explain. I've been listening to different things and reading these forums for a while.  Mostly over on MadLustEnvy's thread for gaming.  I've been collecting Genesis's albums from Duke onward, and from reading about sets and releases I discoverd most of the CDs were 1994 digital remasters and many thought them inferior.  They had volume raised on certain parts with an overall effect of brighter sound and loudness being mistaken for improved audio quality.  I sought out the original JVC or WEA corp CD releases and listened to them and was impressed.  They were more... natural.  And of course the sound that the band created not being altered was great.  I try to not be taken in by marketing... Anyway I thought the 24 bit Phil Collins songs sounded fantastic.  Could this be the source or something?


----------



## spook76

shadowskulkerer said:


> So...  Please don't mistake this as arguing.  I'm 21, quite ignorant on the matter and I'm really not up on science and physics and things.  I just recently got into mid-fi type stuff.  You can see my profile for my equipment.  I'm very happy with it btw.  But can someone explain to me why the 24bit/192khz Phil Collins album Face Value I bought on Pono Music sounds better than the songs I ripped from Phil Collins Greatest Hits CD.  I listened through my Fiio X3 and Headphones to the CD In the Air Tonight and the Pono version and I thought the Pono one was clearer and more detailed.  And I'm not saying various parts had more punch...
> 
> Allow me to explain. I've been listening to different things and reading these forums for a while.  Mostly over on MadLustEnvy's thread for gaming.  I've been collecting Genesis's albums from Duke onward, and from reading about sets and releases I discoverd most of the CDs were 1994 digital remasters and many thought them inferior.  They had volume raised on certain parts with an overall effect of brighter sound and loudness being mistaken for improved audio quality.  I sought out the original JVC or WEA corp CD releases and listened to them and was impressed.  They were more... natural.  And of course the sound that the band created not being altered was great.  I try to not be taken in by marketing... Anyway I thought the 24 bit Phil Collins songs sounded fantastic.  Could this be the source or something?




As a huge Genesis fan from ''Trespass' in 1970 to 'Wind and Wuthering'. That is quite easy to answer, IT IS THE MASTER/REMASTER. Take a listen to the Steven Wilson remixes of Jethro Tull, Yes, King Crimson and XTC and you will begin to understand with older recordings 16/44 lossless is just the beginning of the story. I have listened to at least 4 different masters of King Crimson or early Rush to find the least digitally compressed and best mix for my tastes. 

People think they buy a 24/192 that they have the best recording and I laugh. A good example is from those charlatans at HDTracks. They have a 24/196 of 'The Yes Album' by Yes but it is not the glorious Steven Wilson remix based upon the track listing but an earlier remaster. 

Bit debth is not the issue but the master or remaster is. Research the fan forums carefully and trust your ears to find the best remaster/remix. 

As a general rule any remaster from the advent of the iPod until about 2012 is overly digitally compressed. At least with progressive rock I have noticed remasters/remixes after 2012 have stopped that trend but you have to cautious.

p.s. If you ever want to discover the true Genesis sound (with Gabriel and Hackett still as members) the best CDs of 'Trespass' to 'Selling England by the Pound' are the 2013 Japanese remasters.


----------



## castleofargh

shadowskulkerer said:


> So...  Please don't mistake this as arguing.  I'm 21, quite ignorant on the matter and I'm really not up on science and physics and things.  I just recently got into mid-fi type stuff.  You can see my profile for my equipment.  I'm very happy with it btw.  But can someone explain to me why the 24bit/192khz Phil Collins album Face Value I bought on Pono Music sounds better than the songs I ripped from Phil Collins Greatest Hits CD.  I listened through my Fiio X3 and Headphones to the CD In the Air Tonight and the Pono version and I thought the Pono one was clearer and more detailed.  And I'm not saying various parts had more punch...
> 
> Allow me to explain. I've been listening to different things and reading these forums for a while.  Mostly over on MadLustEnvy's thread for gaming.  I've been collecting Genesis's albums from Duke onward, and from reading about sets and releases I discoverd most of the CDs were 1994 digital remasters and many thought them inferior.  They had volume raised on certain parts with an overall effect of brighter sound and loudness being mistaken for improved audio quality.  I sought out the original JVC or WEA corp CD releases and listened to them and was impressed.  They were more... natural.  And of course the sound that the band created not being altered was great.  I try to not be taken in by marketing... Anyway I thought the 24 bit Phil Collins songs sounded fantastic.  Could this be the source or something?


 

 in those situation take the practical test, convert the highres to what the other one is and listen if they also sound different. it's a much better way to find out than to trust any of us ^_^.


----------



## RRod

shadowskulkerer said:


> So...  Please don't mistake this as arguing.  I'm 21, quite ignorant on the matter and I'm really not up on science and physics and things.  I just recently got into mid-fi type stuff.  You can see my profile for my equipment.  I'm very happy with it btw.  But can someone explain to me why the 24bit/192khz Phil Collins album Face Value I bought on Pono Music sounds better than the songs I ripped from Phil Collins Greatest Hits CD.  I listened through my Fiio X3 and Headphones to the CD In the Air Tonight and the Pono version and I thought the Pono one was clearer and more detailed.  And I'm not saying various parts had more punch...
> 
> Allow me to explain. I've been listening to different things and reading these forums for a while.  Mostly over on MadLustEnvy's thread for gaming.  I've been collecting Genesis's albums from Duke onward, and from reading about sets and releases I discoverd most of the CDs were 1994 digital remasters and many thought them inferior.  They had volume raised on certain parts with an overall effect of brighter sound and loudness being mistaken for improved audio quality.  I sought out the original JVC or WEA corp CD releases and listened to them and was impressed.  They were more... natural.  And of course the sound that the band created not being altered was great.  I try to not be taken in by marketing... Anyway I thought the 24 bit Phil Collins songs sounded fantastic.  Could this be the source or something?


 
  
 "Greatest Hits" compilations are notorious for falling prey to loudness-war-type mastering**. As argh suggested, you can make your own down-conversion of the Pono tracks and compare them at 16/44.1 to the greatest hit's track. Any difference are then due almost entirely to the mastering of the CD, not the format itself.
  
 **One of my hopes for the Ponostore (which doesn't seem to be coming true) was the availability of such compilations in well-mastered versions, for those of us who aren't collectors of a particular group.


----------



## ShadowSkulkerer

So the version on the CD was probably poor. Because I think the other albums I have on CD sound fantastic... I suppose I could convert the 24bit one but that may not be as good a test as finding the 1981 issue CD of Face Value, because it would be doing encoding a second time... Just a theory.


----------



## C.C.S.

If you're encoding lossless to lossless, then it shouldn't be any problem. You can convert it down to 16 bits without anything funky happening.


----------



## interpolate

edit


----------



## bigshot

interpolate said:


> When working with 24-bit it offers a wider range of possible values and reduced truncation of sine waves and as the noise floor is lower, it can be pushed way below -100dBFS meaning what our ears can hear is enhanced.


 
  
 How can it enhance what our ears hear if it is below the threshold of human perception? More isn't better if your ears can't hear it.


----------



## interpolate

edit


----------



## cjl

interpolate said:


> When working with 24-bit it offers a wider range of possible values and reduced truncation of sine waves and as the noise floor is lower, it can be pushed way below -100dBFS meaning what our ears can hear is enhanced.


 
 Here's the fun part: when properly dithered, 16 bit already offers a noise floor that is at -100dBFS or lower.


----------



## cjl

interpolate said:


> ... subsonic and supersonic audio...


 
 Hang on, now I have to worry about whether my speakers are moving through the air at below or above the speed of sound? This audio business gets more complicated all the time...


----------



## kraken2109

interpolate said:


> When working with 24-bit it offers a wider range of possible values and reduced truncation of sine waves and as the noise floor is lower, it can be pushed way below -100dBFS meaning what our ears can hear is enhanced. So whether all of this is a placebo effect for the masses and pseudo-science for the needlessly pedantic. At the moment my equipment isn't brilliant however touch MDF everything is very listenable and swapping it to a cheaper alternative or fashionably over-priced candy-coloured headphones, the difference will be probably noticeable to me.
> 
> I prefer the FLAC format whilst 16-bit doesn't quite wow me, the 24-bit FLAC format is much less sibilant and reverb/ambience has much more sonic room at least to my ears.


 
 That's not how it works.


----------



## interpolate

edit


----------



## interpolate

Gone back to playing with lego and straws.


----------



## bigshot

Did you mean "sonic room" or "sonic boom"? Because hyper-sonic sound would definitely result in a sonic boom.


----------



## interpolate

No I think I meant Go away.
  
 See that's really mature isn't it.
  
 The only reason I edited my posts, so they wouldn't be constant quotable derisive comments for all the family. 
  
 Just end it, the jokes over now.


----------



## bigshot

I was being helpful and offering a good punch line to the joke!
  
 Well, in any case... Inaudible is inaudible, and focusing on theoretical improvements that human ears can't hear isn't going to get anyone an inch closer to getting better sound quality out of their music.


----------



## castleofargh

seems like I missed something fun.


----------



## interpolate

Sorry I got a bit (1-bit DSD) agitated. Bad day....For what it's worth, I bookmarked that website.


----------



## Harry Manback

In response to: 


> How can it enhance what our ears hear if it is below the threshold of human perception? More isn't better if your ears can't hear it.


 
  
 I agree with you on the surface.  I would like to see how the following would play out however:
  
 Produce an executable file that will select and playback 1 of 2 files at random.  One file would be "high res" and the other would be redbook.  Playback would be on the same equipment which is generally agreed to be capable of playing the "high res" file without downsampling.  The files would be at least an entire album in length, and preferably longer.  Over time, track how long each file was listened to.  This needs to be done over a span of at least a few weeks.  I would like to see if one or the other produces more listening fatigue than the other. 
  
 I'd like to see the same done for sampled music and analog.
  
 I am a software developer, and I would agree to write something to do this, if enough people would use it.  I don't want to waste the time for only 6 or 8 people to use it.
  
 I agree that science backs up the Nyquist theory.  It is a fact and not a theory however, that our understanding continues to grow as time passes.  There may be facets of audiology that we stumble upon that add another measurable dimension.  I've been listening to vinyl lately, and there is something that is different that I can't put into words really.  I concede that it may be placebo, but I do not concede that analog audio may have some advantages over sampled music (along with all it's deficiencies).
  
 I hope with all the attention that "high res" music is getting lately, leads to some university studies to clarify whether or not it has any useful benefits for listeners.  If it is different, that difference is measurable.  We may just need a different tool to measure and quantify any new types of differences that may be found.


----------



## bigshot

For the purposes of playing back recorded music in my home, I find that it's a LOT more effective to focus on the range of sound that human ears can actually hear. It might be interesting from a theoretical standpoint to study super audible frequencies, but it isn't going to make Dark Side of the Moon sound any better. There are always plenty of things within the audible range to work on, so it isn't like I'm lacking for stuff to optimize.
  
 By the way, a long time ago, the AES did a study on super audible frequencies and sound quality. The results were that frequencies above 20kHz added nothing to the perception of sound quality. Later studies indicated that they could measure brain wave activity reacting to super audible frequencies, but there was no conscious benefit from having them or perceived degradation from not having them. To me, that means they don't matter.


----------



## Thad-E-Ginathom

harry manback said:


> ... ... ...
> 
> I agree that science backs up the Nyquist theory.
> 
> ... ... ...


 
  
  
 It is not a *theory*, it is a *theorem*.
  
 Nothing "backs it up," it is the foundation on which the whole edifice is built. Science does not back it up: it _is_ the science.


----------



## Roly1650

harry manback said:


> In response to:
> 
> I agree with you on the surface.  I would like to see how the following would play out however:
> 
> ...



You're making this way more complicated than it needs to be. Your last paragraph uses the word difference or its derivative 4 times, so I offer you a difference or null test, which requires no university studies, no measurements, no blind listening tests.

Take your favorite 24 bit track and bounce it down to 16 bit, leaving the original sample rate. Now bounce the 16 bit file back up to 24 bit. A moments thought will tell you that by doing that you've lost any of the 24 bit "magic" in the file. Now load the original 24 bit file into an audio editor like Audacity. Alongside it load the bounced down/up second file. Invert this file using the editing tools and subtract the result from the original file. What's left is the difference between a 24 bit file and a 16 bit file of the same recording. This file will be playable, what do you hear? Anything worthwhile that you'd hear if the rest of the track was present? If the difference is any more than about 65dB the answer is no. Do that half a dozen times with your favorite music and you'll come to a conclusion and you'll then know what's worth worrying about and what isn't.

Difference or null tests can be used for all sorts of audio items, cables, amps, etc., beats the hell out of tedious listening tests, measurements etc. and puts a lie to the old chestnut that you can't test anything with music, music is too complicated.


----------



## ralphp@optonline

roly1650 said:


> You're making this way more complicated than it needs to be. Your last paragraph uses the word difference or its derivative 4 times, so I offer you a difference or null test, which requires no university studies, no measurements, no blind listening tests.
> 
> Take your favorite 24 bit track and bounce it down to 16 bit, leaving the original sample rate. Now bounce the 16 bit file back up to 24 bit. A moments thought will tell you that by doing that you've lost any of the 24 bit "magic" in the file. Now load the original 24 bit file into an audio editor like Audacity. Alongside it load the bounced down/up second file. Invert this file using the editing tools and subtract the result from the original file. What's left is the difference between a 24 bit file and a 16 bit file of the same recording. This file will be playable, what do you hear? Anything worthwhile that you'd hear if the rest of the track was present? If the difference is any more than about 65dB the answer is no. Do that half a dozen times with your favorite music and you'll come to a conclusion and you'll then know what's worth worrying about and what isn't.
> 
> Difference or null tests can be used for all sorts of audio items, cables, amps, etc., beats the hell out of tedious listening tests, measurements etc. and puts a lie to the old chestnut that you can't test anything with music, music is too complicated.


 

 What you will hear in the above is the sound of MONEY


----------



## OddE

ralphp@optonline said:


> What you will hear in the above is the sound of MONEY




-Which, incidentally, is one of a select few songs (if not the only one) in 7/4 time which has charted...


----------



## ralphp@optonline

odde said:


> -Which, incidentally, is one of a select few songs (if not the only one) in 7/4 time which has charted...


 

 True but I wrote "MONEY" not "MONEY©"


----------



## castleofargh

time is money. we all know that one.
 so if bits are also money, does that mean that 100% of an audio signal is money?


----------



## ralphp@optonline

castleofargh said:


> time is money. we all know that one.
> so if bits are also money, does that mean that 100% of an audio signal is money?


 

 That all depends on how much jitter there is - the more jitter, the less money.


----------



## Roly1650

ralphp@optonline said:


> What you will hear in the above is the sound of MONEY




I swear when I do a 16/24 difference test, I faintly hear, "We gotta move these refrigerators, we gotta move these color tv's".

I know the words, the title escapes me, but it may be appropriate.......


----------



## ralphp@optonline

roly1650 said:


> I swear when I do a 16/24 difference test, I faintly hear, "We gotta move these refrigerators, we gotta move these color tv's".
> 
> I know the words, the title escapes me, but it may be appropriate.......


 

 All I hear are Neil Young songs


----------



## Thad-E-Ginathom

All I hear are redemption songs.
  
 (google tells me I misquoted that, but hey, it seemed to fit 
  
  


> the more jitter the less money


 
  
 The more people talk about jitter, the _more_ money! Gets spent!
  
 There are some samples on Hydrogenaud.io of music with various levels of jitter applied. Those who are convinced that their latest multi-$k DAC or cable has reduced the jitter they hear will not want to go and listen to them, and will certainly not want to blind-test them against the no-jitter sample.  Others may find them very useful for finding out what jitter does actually sound like, rather than what they think it sounds like, or imagine that that they hear.
  
 I never knew, and yes, at _extreme_ levels, it does sound a bit like the word suggests.


----------



## interpolate

____{Harumph}_________


----------



## Harry Manback

I guess that this wasn't the right thread to post to. I think that the sample rate is much more important.


----------



## castleofargh

harry manback said:


> I guess that this wasn't the right thread to post to. I think that the sample rate is much more important.


 

 at least we do know that the prices have nothing to do with the audibility. but hey it costs me about 10euros to get the 0.5mo epub version of a book I can buy for 3€ at the nearby store. so maybe I should be happy that audio only tries to screw me a little bit.
  
 samples and bits both could have some pros and cons and ultimately, except for a few parameters along the chain, one can be transferred into the other(like DSD). the main argument for higher samples is the low pass filter, but it's still not that clear if we should care. and nowadays most DACs oversample anyway before applying the filters so it doesn't really change a thing.


----------



## bigshot

harry manback said:


> I guess that this wasn't the right thread to post to. I think that the sample rate is much more important.


 

 Your headphones make the difference more than anything.


----------



## Duncan

bigshot said:


> Your headphones make the difference more than anything.


Billion percent yes, cannot make a silk purse out of a sows ear...


----------



## dprimary

While higher sample rate has some uses in production. There has not been a single documented case of it being audible. Bob Katz a well known mastering engineer  got in this same discussion on the old mastering email list about 17 years ago as 96k convertors appeared Bob claimed it was the Holy Grail and Jim Johnston from ATT Labs told him there was no way 96k made any difference. I will have to dig up the old emails but the plan to test this developed over about a one year period. The discussion was fascinating to say the least and very informative. I have heard that Bob Katz has   written about it in his book. 
  
 While they did not agree they are both very professional and respectful of the other's viewpoint, working together spending a considerable amount of time and effort to put together a single variable test. Ultimately as Bob put together the music samples he announced to the list Jim is right I have tested and listen to all these samples and I can't tell the difference. He did invite anyone on the list to still come to his studio and test them themselves if they wished.
  
 I have nothing but respect for both of them.


----------



## dazzerfong

bigshot said:


> Your headphones make the difference more than anything.


 
 I'd having argued that the mastering of the song is the top.


----------



## Duncan

dazzerfong said:


> I'd having argued that the mastering of the song is the top.


Whilst I would agree that if mastered terribly it'll sound bad on whatever you listen with, that is different to the bitrate discussion, I assume where bigshot was going with the point is that it would be better to spend (throwing random numbers out here) $1000 on a pair of headphones to listen to 320kbps files, rather than $200, listening to FLAC (deliberately ignoring the hi-res point here)...


----------



## dazzerfong

duncan said:


> Whilst I would agree that if mastered terribly it'll sound bad on whatever you listen with, that is different to the bitrate discussion, I assume where bigshot was going with the point is that it would be better to spend (throwing random numbers out here) $1000 on a pair of headphones to listen to 320kbps files, rather than $200, listening to FLAC (deliberately ignoring the hi-res point here)...


 
 Oh, of course. Transducers + song makes by far the biggest differences: everything else is just absolute nitpicking.


----------



## bigshot

dazzerfong said:


> I'd having argued that the mastering of the song is the top.


 

 I'm assuming they listen to well mastered music, but yes, you are correct in that.


----------



## interpolate

320K MP3 is pretty good although I started ripping my CD's as FLAC at 16-bit with no compression. In reality my brain and ears won't be able to differentiate between the CD or mid-quality rendered MP3 however you don't know unless you try.


----------



## kraken2109

interpolate said:


> 320K MP3 is pretty good although I started ripping my CD's as *FLAC at 16-bit with no compression*. In reality my brain and ears won't be able to differentiate between the CD or mid-quality rendered MP3 however you don't know unless you try.


 
 FLAC is compressed. Just lossless compression, like a zip or rar file.


----------



## interpolate

Yes I am aware. Or was that just for clarification?


----------



## Joe Bloggs

I think it's bad in the sense that anything over 48k gets progressively harder to process (other than just decode and spit out), especially on portable hardware. And DSD is impossible to process without interpolating / decimating it first, which leads many audiophiles to cry foul (but then many audiophiles cry foul over any processing on general principle :rolleyes: )

If it isn't obvious to most here already, processing at the end of the digital chain before amplification and transduction is essential for high fidelity--it's physically impossible for loudspeakers to attain optimal phase and frequency response, in a real room, without equalization and more advanced digital room correction processing. I'll leave the equivalent elaboration on headphone audio to somebody else... :rolleyes:


----------



## kraken2109

interpolate said:


> Yes I am aware. Or was that just for clarification?


 
 Your post said _'I started ripping my CD's as FLAC at 16-bit with no compression'_


----------



## bigshot

He meant lossy compression


----------



## interpolate

Yes sorry, I should have been more exact between dictionary and lossy frequency rejection. To be honest, I think the naming convention for these do get lost in translation.
  
 I was referring to Compression Size 0 which essentially means the codec does not try to reduce the file in size by rejecting only minimum of repeat detail AFAIK (around 50%). To do this I used Mediamonkey which I licensed. 
  
 You have to be sharp and on your toe with this forum. 
	

	
	
		
		

		
			




  
 How MP3 encoders work is a little bit harder to explain....
  
 http://www.soundonsound.com/sos/may00/articles/mp3.htm
  
 So I'll let someone else do it on my behalf.


----------



## caml

interpolate said:


> I was referring to Compression Size 0 which essentially means the codec does not try to reduce the file in size by rejecting only minimum of repeat detail AFAIK (around 50%).


 
  
 Why don't you use maximum compression while encoding to FLAC ? It is a lossless format, you won't lose anything in terms of sound quality. Flac compression 0 or compression 8 both produce the exact same result once decoded and only differ in encoding time, cpu usage and of course resulting file size. The only reason not to use max compression ratio with flac or other lossless encoders is when you want to minimize cpu usage while playing, which is no concern on a full size computer but might be relevant on a portable player.


----------



## limpidglitch

caml said:


> Why don't you use maximum compression while encoding to FLAC ? It is a lossless format, you won't lose anything in terms of sound quality. Flac compression 0 or compression 8 both produce the exact same result once decoded and only differ in encoding time, cpu usage and of course resulting file size. The only reason not to use max compression ratio with flac or other lossless encoders is when you want to minimize cpu usage while playing, which is no concern on a full size computer but might be relevant on a portable player.


 
  
 And the reduction in cpu usage on playback is pretty insignificant.
  
Decoding speed vs. compression ratio:

  
 Source: *lvqcl *at HA


----------



## interpolate

It is going to be on a media player and the less computation used the better.


----------



## bigshot

Not necessarily. If you have enough oomph to just be able to do the job, that's all that matters. Extra oomph ain't gonna make it any better.


----------



## interpolate

I'm not understanding that graph really is the X axis KB/s and Z axis compression used?
  
 I was thinking I might use a little compression setting to get more space like Setting 2. This will mean I can get more music on the memory card although it's still going to take a while to fill a 64GB SD card.
  
 Also I have lots of 320K MP3 from Google Play Music. It's my CD collection I started to encode to FLAC fornat, so maybe I'll step back the size. 
  
 I'm tied between two players at the moment - iBasso DX50 or Fiio X3 mk. 2. I can buy the iBasso as they seem to be in stock whereas the FiiO isn't till the 28th of the month. Both are just about as capable as each other. Different DAC processors, software and hardware however I'm not sure which is better in the long run.


----------



## limpidglitch

The Y-axis shows decoding speed times real time, the X-axis shows the compression factor as compressed/uncompressed*100.
 So top right is low cpu usage and low compression, bottom left is high cpu usage and high compression. FLAC -4 or 5 looks like a good compromise

 A more extensive study done on a larger corpus can be found at xiph.org (which also explains why FLAC -3 could be a poor choice)


----------



## nick_charles

dprimary said:


> While higher sample rate has some uses in production. There has not been a single documented case of it being audible.
> 
> 
> *Not quite true, there was a paper from two researchers at McGill  (Sampling Rate Discrimination: 44.1 kHz vs. 88.2 kHz, Pras & Guastavino, 2010). The stats and methods are at least open to debate but you'll have to go to HydrogenAudio to see an extensive debate on this and the AES will charge you $20 for a copy. Attempts to get the raw data from the authors have met with stony silence...*


----------



## bigshot

Stony silence isn't peer review


----------



## nick_charles

bigshot said:


> Stony silence isn't peer review


 
  
 The paper itself *was* peer reviewed and was presented at an AES meeting. Conference papers are sometimes given a (little) bit of slack as long as they are likely to be of interest but not being privy to the review process it is pointless to speculate on how critical a reading it got, I had some concerns about some of the stats and some HA members had concerns about the methods but it was at least interesting...and on a personal note it is a bit cheeky to ask for the raw data, I certainly would not give up mine readily without a quid pro quo...


----------



## bigshot

I would think that if one person's study came up with results that no one else have ever found before, that person would want to be completely transparent with how they reached those results. It's been five years and no one is rushing out to try to duplicate their test to verify it, so I am assuming other researchers are figuring something in Denmark may be past its expiration date.


----------



## safulop

This whole thread is like a poster for the problems that occur when people who aren't experts in a subject try to figure stuff out for themselves.  The amount of misunderstanding of digital audio on here is just staggering.
  
 But I digress.  Since I am one of those who understands why CDs sound precisely as good as they need to, in theory, it's funny that I also love vinyl records.  Why?  Well for one thing I like the euphonic distortions; the little bits of high-frequency noise add a brightness and verve to the sound.  But mostly, the reason is that vinyl records generally are cut from nicer-sounding masters than their CD counterparts.  So if I were a record company, I would treat HDTracks the same way and provide a better master.  That would give all the audiophiles a reason to pay the big bucks for a million kHz/million bit sound file, though not for the reasons they generally think.
  
 I think the fact that vinyl has the best masters distracted everybody back in the day and really fueled the debates over CD sound, prolonging the discussion for much longer.  Now maybe we are seeing the same thing on HDTracks.
  
 I think it would be pointless to compare a downsampled version of a track from HDTracks to the original.  But it may not be pointless to compare the official CD version or iTunes version or Spotify download or whatever to the same song on HDTracks.  Who knows if they are actually different?


----------



## fiascogarcia

safulop said:


> This whole thread is like a poster for the problems that occur when people who aren't experts in a subject try to figure stuff out for themselves.  The amount of misunderstanding of digital audio on here is just staggering.
> 
> But I digress.  Since I am one of those who understands why CDs sound precisely as good as they need to, in theory, it's funny that I also love vinyl records.  Why?  Well for one thing I like the euphonic distortions; the little bits of high-frequency noise add a brightness and verve to the sound.  But mostly, the reason is that vinyl records generally are cut from nicer-sounding masters than their CD counterparts.  So if I were a record company, I would treat HDTracks the same way and provide a better master.  That would give all the audiophiles a reason to pay the big bucks for a million kHz/million bit sound file, though not for the reasons they generally think.
> 
> ...


 

 As one of those who is definitely not an expert, I was curious how it came about that studios used better masters for cutting albums vs more recent masters used for cd's?  It's almost counter intuitive to technological progression.  Thanks!


----------



## arnyk

nick_charles said:


> The paper itself *was* peer reviewed and was presented at an AES meeting. Conference papers are sometimes given a (little) bit of slack as long as they are likely to be of interest but not being privy to the review process it is pointless to speculate on how critical a reading it got, I had some concerns about some of the stats and some HA members had concerns about the methods but it was at least interesting...and on a personal note it is a bit cheeky to ask for the raw data, I certainly would not give up mine readily without a quid pro quo...


 
  
 IME it is not unusual for there to be a more detailed discussion of the analysis of the data including presentation or mention of the raw data.
  
 At the time of the McGill paper, there was AFAIK no peer review of anything but the abstract of conference papers.
  
 At a later conference, a related paper was given peer review and an award, and resulted in the following errr, lively discussion:
  
 https://secure.aes.org/forum/pubs/conventions/?ID=416


----------



## arnyk

fiascogarcia said:


> As one of those who is definitely not an expert, I was curious how it came about that studios used better masters for cutting albums vs more recent masters used for cd's?  It's almost counter intuitive to technological progression.  Thanks!


 
  
 One word: mistakes.
  
 Some master tapes were destroyed or overwritten by accident or even intentionally.
  
 However, there is no general trend or rule that  LPs are made from better masters than LPs. In fact most large volume LPs are made from cutting masters that are at least 1 (or more) generations further removed and contain a lot of processing to make them suitable for cutting onto vinyl. This work may be done in either the digital or analog domain. At some point in time the vast majority of all LPs were cut from digital masters for obvious reasons.
  
 Once a master tape is transcribed into the digital domain, any additional copying and processing  should be and usually is done in the digital domain, which has the advantage of ensuring that any audible changes are in some sense intentional.


----------



## safulop

From my limited knowledge of recording engineering, I can say that pop/rock CDs from the beginning were afflicted by "the loudness wars", which results in highly compressed music with restricted dynamic range operating at the highest decibel allowed on the medium.  I have a CD player with a level meter, and when I play rock or pop music (which is a lot of the time) you can usually see the meter is pegged to the max for most of the song.  When you look at the signal waveform in analysis software (try Audacity), you can see that it is maxed out almost looking clipped the whole time.  One engineer said these songs are basically "pink noise carrying a song."  Their auditory fidelity is dreadful.
  
 Now if I want to cut a record of that same song, I cannot use that master.  The cutting lathe will burn up, and even if a lathe could be beaten into submission to make the cut, the resulting record would play so poorly from all the tracking distortions that every customer would return it.  So you have to do a new master that has the compression dialed out and the decibels backed off.  In other words, now it sounds like something an audiophile might actually like.  The process of creating special masters for vinyl was in recent times carried out for rock bands like The Mars Volta, for example.  More than one of their records even credits someone for "mastering for vinyl."  
  
 When it comes to the music of older bands (e.g. Led Zeppelin), the opposite process was in play.  What I mean is, the original vinyl records sound awesome, they have wonderful masters.  When they put them onto CD years later they were "digitally remastered," which is really just a euphemism for "digitally ruined."  I've done comparisons of the spectral content in Led Zeppelin songs from vinyl vs. remastered CD from their famed "box set", and there are huge differences in the long-term average spectrum due to the EQ being set differently.  A major difference I saw was the CD version has a much higher boost at 2 kHz, which is precisely the range where your hearing is most sensitive and this will tend to make the song sound more present in a noisy environment like your car.  So, basically, an effort was made to make old Zeppelin records sound more like new records -- louder and more intrusive.  The result if you play it on your hifi system is not pleasant.  Suddenly Robert Plant sounds like he is shouting through a police radio.  Nasty CDs!
  
 Another thing they often did when remastering for CD was to actually remix the songs.  This was done for the Zeppelin I believe, and it has definitely been done for old Yes records among others.  In this process, the old-fashioned use of stereo was often eschewed in favor of a more modern balance, which has the effect of "mono-izing" a lot of older music where it was deemed unwanted to have, say, the drums on the left and guitars on the right.  I agree it is not a modern type of mix, but it is also not authentic to the original sound if you change it for a "remaster."
  
 So to sum up what I said earlier, if I were a record company and I'm sending the new Celine Dion album or Rush album to HDTracks (yes they are available there), I would instruct a mastering engineer to dial back the compression, so that audiophiles who bother with HDTracks could get a real treat instead of just snake oil for their money.


----------



## safulop

I'd like to add that while I agree with Arny that there is no "general rule" that LP masters should be better, in my experience with pop/rock music I have never heard a record whose master sounded worse than that of the corresponding CD.  Occasionally they use the same master basically (yes I've checked using long-term spectral analysis), but I've never heard a CD that had a *better* master than the record, in pop/rock music.
  
 Now in classical and jazz, perhaps all bets are off, because the rules for how the mastering is done are completely different.  I tried a back-to-back listening test with Lang Lang live in Vienna on my LP versus streaming from Spotify, and I actually could not hear any difference!  Except for the ticks and pops, of course.


----------



## safulop

Oh I also forgot to mention another reason so many CDs of older records sound like dog-patch.  They often tried in vain to get rid of the tape hiss, which was never a problem on vinyl because it's usually quieter than the surface noise anyway.  The result of this lovely "Dolby-izing" was that many CDs of old records (from Jon & Vangelis to John Denver) sound so muffled you can't believe that a record company was serious when they sold it to you.


----------



## safulop

I thought of a couple more points of interest - 
  
 on the subject of digital masters, the problem of over-compression became so rampant that some artists have begun supervising remasters of stuff that was mastered digitally in the first place. We normally think of "digital remastering" as necessary to salvage old Hank Williams records, but it is being used more and more to correct poorly done digital masters.  Mike Oldfield did this for his 1991 album "Heaven's Open," and Rush specifically admitted that their 2002 album "Vapor Trails" was so badly a victim of the loudness wars that it suffers from digital distortion.  It was remastered for online only in 2013.
  
 on the subject of LP masters being superior, Spotify agrees with me!  Numerous albums are posted there with the specific labeling "LP master" because they may be aware of which CDs have a bad reputation.  Of course, being Spotify, they also realize that you don't actually have to play the LP master on your turntable to hear its many benefits.


----------



## RRod

All you say is sensible, but it doesn't indicate a need for a new end-user format, simply a different attitude from musicians, producers, engineers, and, especially, the listening public.


----------



## safulop

rrod said:


> All you say is sensible, but it doesn't indicate a need for a new end-user format, simply a different attitude from musicians, producers, engineers, and, especially, the listening public.


 
 Yes this would solve everything, but you're talking about changing the whole world.  I'm saying, we could reserve stuff like HDTracks to be like the niche marketplace for audiophile-grade masters, as vinyl now is, and let all else be equal.  Objectivists and subjectivists could make "strange bedfellows" wanting the same sound files but for different reasons, yielding the same increased enjoyment of the music.


----------



## RRod

safulop said:


> Yes this would solve everything, but you're talking about changing the whole world.  I'm saying, we could reserve stuff like HDTracks to be like the niche marketplace for audiophile-grade masters, as vinyl now is, and let all else be equal.  Objectivists and subjectivists could make "strange bedfellows" wanting the same sound files but for different reasons, yielding the same increased enjoyment of the music.


 
  
 A few issues I have:
 1) I don't think I should have to pay any premium price to get music not mastered like s@#$
 2) I've been unimpressed with the information available on masters from sites like HDTracks and the Ponostore™®☭
 3) Audiophile-grade masters should be the norm, not a niche. Not that I have any idea how the audiophile world can change that


----------



## safulop

rrod said:


> A few issues I have:
> 1) I don't think I should have to pay any premium price to get music not mastered like s@#$
> 2) I've been unimpressed with the information available on masters from sites like HDTracks and the Ponostore™®☭
> 3) Audiophile-grade masters should be the norm, not a niche. Not that I have any idea how the audiophile world can change that


 
 Oh I agree completely, especially with point 2.  HDTracks (I have not used it) seems to be hit-or-miss just like Spotify, you have no idea what you are really getting.  I suppose my suggestion is just a coping strategy for dealing with your point 3.  I am perfectly happy playing my records, but it is crazy to me that in 2015 with all the world's supply of record pressing machines dating from the 1980s and prior, *vinyl* is how audiophiles can get the best-sounding masters?? A technology that was "perfected" in the 1950s? I mean, DVD-Audio and SACD were on the right track but didn't get anywhere, maybe HDTracks can provide for us.  The music business in general certainly won't.


----------



## arnyk

rrod said:


> All you say is sensible, but it doesn't indicate a need for a new end-user format, simply a different attitude from musicians, producers, engineers, and, especially, the listening public.


 
  
  
 Agreed that the listening public should stop buying bad sounding products. They never have because too many of them are irrationally overcome by "The new sound".


----------



## Thad-E-Ginathom

rrod said:


> A few issues I have:
> 1) I don't think I should have to pay any premium price to get music not mastered like s@#$
> 2) I've been unimpressed with the information available on masters from sites like HDTracks and the Ponostore™®☭
> 3) Audiophile-grade masters should be the norm, not a niche. Not that I have any idea how the audiophile world can change that


 
 Market forces.
  
 What we want is the properly-mastered 16/44.1 version, please, and then everybody will be happy. Oh, except the record company, because they can't charge me more.
  
 But market forces fail when the market has drunk the kool aid and want nothing but... kool aid.


arnyk said:


> Agreed that the listening public should stop buying bad sounding products. They never have because too many of them are irrationally overcome by "The new sound".


 
  
 And the new numbers. Even the relatively sane are to be found casually talking about "better" quality when what they actually mean is higher bit-rate. It has almost got to the point where this stuff has _entered the language.  _No. It  _has _entered the language: if _High-res audio_ is not in the dictionary yet, it soon will be, and that in itself will be an argument for it being real. 
  
 There is no such word as _updation. _Oh yes there is: battle lost, nonsense wins the day.


----------



## RRod

thad-e-ginathom said:


> Market forces.
> 
> What we want is the properly-mastered 16/44.1 version, please, and then everybody will be happy. Oh, except the record company, because they can't charge me more.
> 
> ...


 
  
 Certainly new releases of old material can sell. I mean how many version of DSotM are out there? But yeah, the incentive just isn't there to make an audiophile version that doesn't also have the hi-res baggage attached. The whole "hi-res" versus "better" thing really gets to me too. I have some mind-blowingly great sounding orchestral stuff on CD, so know all too well that the decision to make something sound bad is often a deliberate one, and that this can happen even at 24/192.


----------



## fiascogarcia

thad-e-ginathom said:


> Market forces.
> 
> What we want is the properly-mastered 16/44.1 version, please, and then everybody will be happy. Oh, except the record company, because they can't charge me more.
> 
> ...


 

 Apple will begin streaming in 256 on their new site at the end of this month, loading their streaming app on every iphone, so it isn't going to get any better soon.


----------



## RRod

fiascogarcia said:


> Apple will begin streaming in 256 on their new site at the end of this month, loading their streaming app on every iphone, so it isn't going to get any better soon.


 
  
 A well-mastered 16/44.1 album converted to 256AAC or even MP3 will sound better than stuff that's badly mastered at 24/192.


----------



## cjl

fiascogarcia said:


> Apple will begin streaming in 256 on their new site at the end of this month, loading their streaming app on every iphone, so it isn't going to get any better soon.


 

 256 AAC is more than adequate for nearly every piece of (two channel) music ever recorded, and it is almost never audibly distinguishable from 16/44 PCM (or higher, for that matter). Good mastering and less dynamic compression would do far more for music quality than any upgrade in bitrate from 256AAC (unless you use the additional bits for more channels).


----------



## Thad-E-Ginathom

cjl said:


> ... ... ... (unless you use the additional bits for more channels).


 
  
 I do accept the possibility of something entirely new in audio technology, let alone basic improvements in the technology that we have now. _16/44.1PCM 2-channel is sufficient _does _not_ mean it is the be-all and end-all of recording technology for ever more.
  
 The tragedy of the lunatic climb through the PCM sample rates and the DSD multipliers is that it is such a waste of resource that could be applied to audio research.
  
 I suppose, the fact is that it doesn't actually take much resource to produce these so-called-high-res copies. It's a new medium for the record companies to resell the same stuff without even having to buy machines to manufacturer it.


----------



## safulop

arnyk said:


> Agreed that the listening public should stop buying bad sounding products. They never have because too many of them are irrationally overcome by "The new sound".


 
 I think that audiophiles hear things differently from the average person, and we are constantly complaining that they are so wrong in their hearing.  I know a lot of friends who love that Led Zeppelin remastered box set.  I use the CD cases for coasters.  But I refuse to judge other people's preferences.  There's just no sense in judging a preference.
  
 From tape saturation (1950s) to Phil Spector's "wall of sound" (1960s) to the loudness wars, the music industry has defined itself around generating products that sound "good" on low-fi equipment.  My daughter plays music straight out of the screen speaker on an iPhone!  We're going backwards in some ways.  But it is not irrational, it's just a different priority system than we share in this community.  I think we can do better if we stop complaining about the world as it is and work toward a system where we can get access to the best-sounding products we want to hear.  Yes even if they cost more.


----------



## arnyk

safulop said:


> I think that audiophiles hear things differently from the average person


 
  
 That's because about 40 years ago there came to be enough audio gear that was sonically transparent, that a new form of audiophilia came to be which we now call Placebophilia.
  
 Placebophilia is the love of placeboes, which characterizes a lot of people who behave an a similar fashion as audiophiles, but are not the same. Another confusing term is Music Lover. Three different groups of people, sometimes overlapping.


----------



## blades

ralphp@optonline said:


> Sometimes I wonder if there is any difference between sponsorship and outright bribery. In the case of high end audio publications similar lack of difference exists between advertising revenue and outright bribery.
> 
> Truth, honesty and integrity are all just so 20th century. The 21st century is all about lies, dishonesty and corruption.


 
  
 Actually it is all about money.


----------



## safulop

arnyk said:


> That's because about 40 years ago there came to be enough audio gear that was sonically transparent, that a new form of audiophilia came to be which we now call Placebophilia.
> 
> Placebophilia is the love of placeboes, which characterizes a lot of people who behave an a similar fashion as audiophiles, but are not the same. Another confusing term is Music Lover. Three different groups of people, sometimes overlapping.


 
 Well, I'm not talking about that, I'm talking about how average people "can't hear" deep into the music, so they don't hear differentiated instrumentation or the subtle transients and spectral changes that define all the different sounds in the mix.  My wife once asked me why a stereo has two speakers!  She actually cannot hear that there are different sounds coming from the two channels.


----------



## Toom

Wood and trees comes to mind.


----------



## arnyk

safulop said:


> Well, I'm not talking about that, I'm talking about how average people "can't hear" deep into the music, so they don't hear differentiated instrumentation or the subtle transients and spectral changes that define all the different sounds in the mix.  My wife once asked me why a stereo has two speakers!  She actually cannot hear that there are different sounds coming from the two channels.


 
  
 How do you distinguish between perceptions of that kind that are real and those that are illusions, since they can be either?


----------



## Thad-E-Ginathom

safulop said:


> Well, I'm not talking about that, I'm talking about how average people "can't hear" deep into the music, so they don't hear differentiated instrumentation or the subtle transients and spectral changes that define all the different sounds in the mix.  My wife once asked me why a stereo has two speakers!  She actually cannot hear that there are different sounds coming from the two channels.


 
  
 She probably hears as well as you do. She is probably just as aware of the stereo image as you are.  It is not obligatory to take a technical interest in music playback to enjoy it.   If she connects, emotionally, with the music, she is doing all that the composers and performers asked her to do. She is not obliged to humour the hifi manufacturers, or their customers.
  
 Not meaning to be rude (people always say that when they are about to be!) I'm calling you at as a bit of an audio snob on this. Who cares how much technicality somebody know about or appreciates: I don't.
  
 Now I have to admit to being a snob about music itself: what you call the majority of people probably prefer to listen to music that does not require any deep listening. Most pop music is shallow rubbish (which is not to say that some of it is not pleasant occasionally of course) and  there is nothing below the surface to appreciate.
  
 Now I will reveal the true extent of _my_ snobbery: I'm horrified (and bored to the teeth with it) that _Hotel California  _is a common hifi test track! Hotel Bloody California! Sheesh... why buy hifi, at huge cost, arguing about the 0.01% difference, if _that_ is what people are going to listen to on it! Buy a boom box and be happy with a full wallet.
  
 And no, I'm not a classical-jazz only freak, and I feel the same about sculptured-sound jazz tracks that seem to have recorded _just_ to show off female vocals and double bass sounds _as reproduced on "hifi" equipment_. Yes, I do love classical music, and If I am, personally, going to check out hifi, there is no bigger challenge in terms of sound stage and dynamic range than a symphony --- but you are just as likely to find me listening to the _Grateful Dead_ at home.
  
 (This post is part of my _How Audiophiles Made Me Hate The Eagles _series 
	

	
	
		
		

		
			





 )


----------



## Toom

Very true. Nothing funnier than reading a long extensive review of another high end IEM or headphone, only to discover the author listens exclusively to some ***** metal, pop or musical soundtracks.


----------



## Roly1650

thad-e-ginathom said:


> She probably hears as well as you do. She is probably just as aware of the stereo image as you are.  It is not obligatory to take a technical interest in music playback to enjoy it.   If she connects, emotionally, with the music, she is doing all that the composers and performers asked her to do. She is not obliged to humour the hifi manufacturers, or their customers.
> 
> Not meaning to be rude (people always say that when they are about to be!) I'm calling you at as a bit of an audio snob on this. Who cares how much technicality somebody know about or appreciates: I don't.
> 
> ...



Rather Hotel California, (or the Dead), than Norah fecking Jones or Diana Krall, two more demo favorites from hell. Show me a system that can't do a passable job of reproducing either of those and I'll tell you it ain't worth buying. And why would you want a system that reproduces either of those well anyway? Music to sleep by and much, much worse than any sins The Eagles committed, but I do accept that HC is past its sell by date as demo material.

There's plenty of popular music that requires more than a modicum of concentration, it's not all Gangnam Style pap. A well recorded popular album is way more rewarding than a mediocre classical recording. The recording/mixing/mastering is the key, regardless of genre and there's as much skill, (and lack of) in classical as popular music. And using reverse snobbery, us popular music fans are fed up with having to subsidize the minority classical fans, who love to exhibit such pointless, muddle headed elitism. 

(This post is part of my "Why Are Classical Music Snobs Such Nutcrackers" suite, (it's a trilogy). The subtitle reads, "Let Them Have Norah Jones and Diana Fecking Krall To". Second in the series was "Why Does The World Need Yet Another Recorded Version Of Dead Composers Music?" Subtitled, "When Will Practice Make Perfect, So They Can Give Up?" The final part is titled "Why Did The Dead Composers Society Write Such Depressing Music?" Subtitled, "Is Faultless Reproduction Worth It For Music To Commit Suicide To, Won't a Boombox Do?".)


----------



## castleofargh

I test all my gears with:
 for voices
  
  and with :
 to test if a device is "fast".
  
 to test for dynamic I leave it to metallica.


----------



## Toom

castleofargh said:


> I test all my gears with:
> 
> for voices
> 
> ...




I understand Pono is going to make these available in 192/352 shortly.


----------



## interpolate

Your music taste scares me...
	

	
	
		
		

		
		
	


	



  
 I find the track In Your Heart by Moby is a great test track.


----------



## Thad-E-Ginathom

And all this... at higher sample rates, deeper bit depths, umpteen-times DSD!
  
 I despair!


----------



## Toom

thad-e-ginathom said:


> And all this... at higher sample rates, deeper bit depths, umpteen-times DSD!
> 
> I despair!


 
  
 It never ends. Capitalism eats itself.


----------



## interpolate

However the major question is, does Neil Young sound any better in PCM 384/32 sound?


----------



## RRod

interpolate said:


> However the major question is, does Neil Young sound any better in PCM 384/32 sound?


 
  
 http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php


----------



## nick_charles

interpolate said:


> However the major question is, does Neil Young sound any better in PCM 384/32 sound?


 
  
  


rrod said:


> http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php


 
  
  
 or indeed could Neil Young tell the difference ?


----------



## interpolate

When I first joined this topic I was clueless.
  
 Now I know why I am clueless, so some progress.


----------



## fiascogarcia

interpolate said:


> When I first joined this topic I was clueless.
> 
> Now I know why I am clueless, so some progress.


 

 That is a terrific comment!! 
	

	
	
		
		

		
		
	


	




  So I found out that although I think I know what I like, I may be wrong to like what I like, or may not even know what it is I like.


----------



## safulop

thad-e-ginathom said:


> She probably hears as well as you do. She is probably just as aware of the stereo image as you are.  It is not obligatory to take a technical interest in music playback to enjoy it.   If she connects, emotionally, with the music, she is doing all that the composers and performers asked her to do. She is not obliged to humour the hifi manufacturers, or their customers.
> 
> Not meaning to be rude (people always say that when they are about to be!) I'm calling you at as a bit of an audio snob on this. Who cares how much technicality somebody know about or appreciates: I don't.
> 
> ...


 
 Well I'm not sure I can respond to all that but let me just say I know people come in many different stripes of music appreciation.  I know people who never listen to music for its own sake, they only use it as a backdrop for driving or housecleaning or a party.  Such people do not use headphones.  (My wife self-admittedly belongs in this category, btw).  I also know music-lovers who listen to music as a constant backdrop to their day and who carefully curate their playlists etc, but who have no concern about audio quality.  They are just as happy listening to a song through the iPhone screen speaker as to a nice stereo.  Then there are music-lovers who can appreciate high-quality audio but do not require it to enjoy music playback. In my experience a lot of musicians fall into this group.  Finally there are music-lovers who would just as soon not bother to have music playing unless they are listening to high-quality playback, because they can't stand it otherwise.  I put myself in this latter category.
  
 I think that there is also a group of "audiophiles" who actually don't love music, they just buy audiophile-grade recordings to use to listen to their equipment.  This is a particularly interesting attitude but who am I to judge?  I am not a snob in the sense that I don't give a darn which of these stripes anyone falls into.  I don't preach or convert or proselytize or complain about other people's choices, as many audiophiles do.   But I think it would be disingenuous to try to insist that everyone is really the same in these respects.  If failing to do so is "snobbery" then so be it.
  
 On the subject of Hotel California, I still love the song and its original vinyl master, in spite of it being played so many times.  I think that compared to modern radio pop it has a very delicate sound and a good dynamic range.  I do think that it is musically interesting, here's a snippet about why: https://en.wikipedia.org/wiki/Hotel_California#Harmonic_structure


----------



## upstateguy

(didn't read through all 35 pages so if this has already been posted please disregard)
  
 I've never heard a difference but there is a difference:


----------



## safulop

The only difference I see is that the one on the left has double the frequency range.  All that stuff above 20 kHz is useless and doesn't add anything to the sound quality.  That's why you can't hear the difference, it is all inaudible, what little of it there is.  The original recording doubtless did not use equipment that could record above 20 kHz anyway, so what all that stuff even is I have no idea.  It surely was not part of the recording process.


----------



## upstateguy

safulop said:


> The only difference I see is that the one on the left has double the frequency range.  All that stuff above 20 kHz is useless and doesn't add anything to the sound quality.  That's why you can't hear the difference, it is all inaudible, what little of it there is.  The original recording doubtless did not use equipment that could record above 20 kHz anyway, so what all that stuff even is I have no idea.  It surely was not part of the recording process.


 
  
 take a look at it at 320.  I've never heard a difference but maybe you can feel it like you can with sub bass?


----------



## dazzerfong

upstateguy said:


> take a look at it at 320.  I've never heard a difference but maybe you can feel it like you can with sub bass?


 
 Temporal masking prevents that. Under certain circumstances, even with extra stuff in the audible range, you won't be able to register it in your brain due to other maskers.


----------



## castleofargh

with the graphs on the same scale it would be easier to tell something.
 the signal above 20khz is all below -90db on that graph, so even if we pretended like it matters, it's silly quiet and outside of the audible range. so obviously,


----------



## Thad-E-Ginathom

safulop said:


> Well I'm not sure I can respond to all that but let me just say I know people come in many different stripes of music appreciation.  I know people who never listen to music for its own sake, they only use it as a backdrop for driving or housecleaning or a party.  Such people do not use headphones.  (My wife self-admittedly belongs in this category, btw).  I also know music-lovers who listen to music as a constant backdrop to their day and who carefully curate their playlists etc, but who have no concern about audio quality.  They are just as happy listening to a song through the iPhone screen speaker as to a nice stereo.  Then there are music-lovers who can appreciate high-quality audio but do not require it to enjoy music playback. In my experience a lot of musicians fall into this group.  Finally there are music-lovers who would just as soon not bother to have music playing unless they are listening to high-quality playback, because they can't stand it otherwise.  I put myself in this latter category.
> 
> I think that there is also a group of "audiophiles" who actually don't love music, they just buy audiophile-grade recordings to use to listen to their equipment.  This is a particularly interesting attitude but who am I to judge?  I am not a snob in the sense that I don't give a darn which of these stripes anyone falls into.  I don't preach or convert or proselytize or complain about other people's choices, as many audiophiles do.   But I think it would be disingenuous to try to insist that everyone is really the same in these respects.  If failing to do so is "snobbery" then so be it.
> 
> On the subject of Hotel California, I still love the song and its original vinyl master, in spite of it being played so many times.  I think that compared to modern radio pop it has a very delicate sound and a good dynamic range.  I do think that it is musically interesting, here's a snippet about why: https://en.wikipedia.org/wiki/Hotel_California#Harmonic_structure


 
  
 Well, you certainly did respond, and very fully! I guess I went off on a tangent after the initial assertion that technical knowhow wasn't necessary to enjoy music. Yes, indeed there are a whole spectrum of people enjoying, or even loving music in different ways. I'm somewhere near your camp: bad sound annoys me, it is a source of irritation.
  
 It's a shame about Hotel California: I used to like it, but I didn't choose how often I've listened to it, and it just didn't turn out to be one of those songs that I, personally, _could_ listen to that often and still enjoy. I had a rant: but if anybody asks me what they should  use to test equipment they are going to listen to, I would say they should use whatever they like to listen to!


----------



## kraken2109

Does it annoy anyone else that spek only has a linear scale for frequency rather than log like every other tool in audio?


----------



## interpolate

A linear rather than a logarithmic scale?


----------



## arnyk

interpolate said:


> A linear rather than a logarithmic scale?



 



Yes, the frequency scale is linear. Common with spectrogram software, because log frequency axis are often harder to interpret.


----------



## inthere

upstateguy said:


> (didn't read through all 35 pages so if this has already been posted please disregard)
> 
> I've never heard a difference but there is a difference:


 
  
 Propaganda.These are not the same master recording, the one on the right has been compressed and/or limited.


----------



## interpolate

Whatever happened to gut feeling and enjoying music? There is better applications for spectrogram images alas to discuss and disprove audio myths, there is always room for it in the mix.  I reckon if you get the equipment right too it will also make an overall difference.


----------



## Toom

interpolate said:


> Whatever happened to gut feeling and enjoying music? There is better applications for spectrogram images alas to discuss and disprove audio myths, there is always room for it in the mix.  I reckon if you get the equipment right too it will also make an overall difference.


 
  
 This is head-fi. You can never have the equipment right.


----------



## interpolate

We need a Game of Thrones Meme for that.


----------



## Toom

Haha excellent - that's my new avatar sorted!


----------



## upstateguy

inthere said:


> upstateguy said:
> 
> 
> > (didn't read through all 35 pages so if this has already been posted please disregard)
> ...


 
  
 Are you sure about that?


----------



## hdtv00

Um yea if he isn't sure I am. How in this day and age can you not spot compression. Oh is it because it's become so common place. I start with how it sounds. I try and get high res versions and if they don't sound better or good I move on. I don't care what it's labeled it has to sound good.


----------



## safulop

hdtv00 said:


> Um yea if he isn't sure I am. How in this day and age can you not spot compression. Oh is it because it's become so common place. I start with how it sounds. I try and get high res versions and if they don't sound better or good I move on. I don't care what it's labeled it has to sound good.


 
 Well you still didn't say how you could tell by staring at the comparison figure.


----------



## hdtv00

Um because it's compressed....dude it's the internet just do a search audio compression and view pic results. Actually search Loudness Wars Compression and then you'll see, read an article on it too. It's destroyed modern music almost completely if you ask me.
  
 Anyway another thing I'll add is as much as I am into high res , really I got into high res because of the surround formats. What it can do to a properly mixed nice recording is amazing. Whole other level greatness.
  




  
 Anyway having my new setup with Aune T1 feeding a bottlehead crack the sound is beyond amazing even when listening to a plain cd. I wouldn't trade my setup and headphones for anything I can't see how the sound could be better honestly. Well ok more sub bass but anyway. Cheers.


----------



## dazzerfong

hdtv00 said:


> Um because it's compressed....dude it's the internet just do a search audio compression and view pic results. Actually search Loudness Wars Compression and then you'll see, read an article on it too. It's destroyed modern music almost completely if you ask me.
> 
> Anyway another thing I'll add is as much as I am into high res , really I got into high res because of the surround formats. What it can do to a properly mixed nice recording is amazing. Whole other level greatness.
> 
> ...


 
 There's only one problem with your analysis: what you're seeing is a magnitude vs. time response. What that graph was showing was time vs. frequency response. It's incredibly easy to notice compression in time domain, but not frequency domain, as your examples point to.
  
 In fact, if you make the Y-axis the same on that referenced graph and lop off everything after 20k, you'll see that there actually isn't much difference. Take a look at the following graph: the FLAC one is blurry because I stretched it such that 20k aligns with the 20k of the MP3, but even then:
  

  
 This isn't compression going on here, or even if it is, it's so damn hard to see in frequency domain, you really should use time domain (as all your examples do). I mean, of course, all the stuff above 22k is dead for MP3, but who cares?
  
 That being said, yes, audio compression is a big problem, but probably because of the music I listen to, I don't face it as often as you guys might (assuming you guys being rock fans or whatnot).


----------



## interpolate

The purpose of compression is to compressing the top frequencies for a fixed time although this leaves the quieter frequencies alone as determined by the threshold set on the compressor. What really kills dynamics is a brickwall limiter where the whole audio is fed into a limiter which basically increases the perceived volume for a set time and attenuating any peak transients by the said amount past the digital ceiling (threshold). This is where limiters that work on the K-System excel at giving you the best of the both worlds. TC Electronics use a LU (Loudness Units) Volume system which is similar however uses different techniques.
  
 https://www.soundonsound.com/sos/sep11/articles/loudness.htm
  
 You can't really tell from a waveform display what frequencies are being boosted or subtracted, you need a spectrogram or use some other form of spectrum analysis. Essentially all this pernickety behaviour belongs to people who work closely with audio recording. The trade off is people get better sounding recordings unless you listen to loud dance music where loud is proud.
  
 Compression essentially, reduces the loudness peak volumes at a set level by a set time. This cause a reduction in dynamic range whenever it's used. There are different forms of macrodynamics however compression is better added to music during the recording process rather than later. AFAIK


----------



## dazzerfong

interpolate said:


> The purpose of compression is to compressing the top frequencies for a fixed time although this leaves the quieter frequencies alone as determined by the threshold set on the compressor. What really kills dynamics is a brickwall limiter where the whole audio is fed into a limiter which basically increases the perceived volume for a set time and attenuating any peak transients by the said amount past the digital ceiling (threshold). This is where limiters that work on the K-System excel at giving you the best of the both worlds. TC Electronics use a LU (Loudness Units) Volume system which is similar however uses different techniques.
> 
> https://www.soundonsound.com/sos/sep11/articles/loudness.htm
> 
> ...


 
 But here's the thing: it's a hell of a lot easier to spot brickwalling when it's just magnitude as opposed to frequency, especially when a spectrogram has 3 things going on at once which just confuses an inexperienced person more.


----------



## arnyk

dazzerfong said:


> But here's the thing: it's a hell of a lot easier to spot brickwalling when it's just magnitude as opposed to frequency, especially when a spectrogram has 3 things going on at once which just confuses an inexperienced person more.


 
  
  
 There's a real syntactic glitch above. Magnitude and frequency are both orthogonal and co-existing properties. You always have both, so there is no either/or relationship between them.
  
 Spectrograms are plots of frequency versus magnitude versus time,. If you have something else, then it is not a spectrogram.


----------



## interpolate

It's like a new whole language this topic.


----------



## FFBookman

I wonder how many people in "sound science" actually create and/or mix music in a professional or semi-pro environment?
  
 Most of what I read here is discussed after the fact, trying to reverse engineer and ultimately disprove what is so easily proved in a production environment.  Higher bitrates, sensible compression, better ADCs and DAC's, the differences between digital simulations and analog realities -- all easily understood and adjusted for in a production environment. But take it to the listening side and people get crazy.
  
 Every time I watch a behind the scenes type of documentary with legendary record producers explaining their secrets, I picture people on this board heckling them and posting long rants about how little these people know about sound "science".
  
 The real _sound science_ IS music. Making the best sounding, beloved melodies and arrangements that move people emotionally is what's really important.


----------



## StanD

@FFBookman Electrical Engineers design the hardware that recording engineers use. Let's not forget who came first in line. No EE's no music,etc. An EE has a much better understanding of what's going on than the layperson who can easily make up stuff as they go along.


----------



## jcx

music production is much more a craft - little calculation, lots of judgments based in experience, lore - and the results are not tested against absolute standards - just what "sounds good" or "will sell"
  
 artistic choices are made choosing mics, their placement, in mixing/mastering - with the expectation that the EE designed equipment just works - or can be chosen for "flavor" where a analog tape or a tube preamp may not be transparent
  
 some distortions, dynamic compression or colorations are now expected in certain instruments, musical genres "sound"
  
 "sounds good", "will sell" are based in part in psychoacoustics, and rely on audio illusions and culturally evolved hyper realism with close micing, reverb, compression, highlighting, rearranging a synthetic audio soundscape that no audience member could ever have experienced
  
 between the mics and speakers/headphones a huge and "scientific" part is Electrical/Electronic Engineering, transducer signal conditioning, signal processing and deliberate sound effects boxes and plugins
  
 there EE knowledge coupled with some Psychoacoustics Science - tested, accepted principles does give firmer answers to some questions - unless you have those domains solid they then some discussions may seem dismissive of "the human element"
  
  
 but yes this thread title is unfortunate - 24/96 really is what's used as a minimum today, has practical advantages In The Studio
  
 16/44 so far hasn't been proven inadequate by the standards that textbook writers accept - there could be room still for doubt - but it really has to at esoteric levels given the decades of digital audio practice and the seeming large desire to "prove" 16/44 inadequate for consumer music distribution


----------



## StanD

Wasn't it recording engineers and producers that started the "Loudness Wars?" Yep, real geniuses.


----------



## Steve Eddy

stand said:


> Wasn't it recording engineers and producers that started the "Loudness Wars?" Yep, real geniuses.




When you want your song to sound louder than a competitor's song, it _was_ genius. 

se


----------



## StanD

steve eddy said:


> When you want your song to sound louder than a competitor's song, it _was_ genius.
> 
> 
> 
> ...


 
 Yet some audiophiles worship these geniuses. Is it one of those love/hate things?


----------



## FFBookman

I wouldn't say 16/44 is "inadequate". It's just a compromise based on several things - 1978 technology, price to market, the storage size of an optical disc.
  
 I own many 16/44 recordings that sound great. But if they were professionally recorded to tape or recorded at higher than 16/44, then I know there can be a better quality made available someday.
  
 Do the EE's on this board hold onto any other digital standard from 35 years ago as being "all we ever need" ?   We had 16-bit PC's running at 3 mhz 35 years ago, we had VHS, and we had the compact disc.  Two of those have been upgraded hundreds if not thousands of times, one has not.
  
 16/44 was a good first digital music standard. We just should have moved beyond it 20 years ago.  SACD + the Internet screwed up those plans.


----------



## StanD

ffbookman said:


> I wouldn't say 16/44 is "inadequate". It's just a compromise based on several things - 1978 technology, price to market, the storage size of an optical disc.
> 
> I own many 16/44 recordings that sound great. But if they were professionally recorded to tape or recorded at higher than 16/44, then I know there can be a better quality made available someday.
> 
> ...


 
 Have you evolved in the last 35 years? No? That means you can hear the same or slightly worse due to age. If you can't hear any better than humanly capable, then Red Book is more than good enough.


----------



## Steve Eddy

stand said:


> Yet some audiophiles worship these geniuses. Is it one of those love/hate things?




Shouldn't paint with such a broad brush. And most of the pressure to compress comes from the artists themselves and from higher up. 

se


----------



## maverickronin

ffbookman said:


> Do the EE's on this board hold onto any other digital standard from 35 years ago as being "all we ever need" ?


 
  
  
 Quote:


stand said:


> Have you evolved in the last 35 years? No? That means you can hear the same or slightly worse due to age. If you can't hear any better than humanly capable, then Red Book is more than good enough.


 
  
 Damn.  I turn the page and you've already beaten me to it..
  
 Maybe in a few decades when the Machines enslave us and entrap us within the Matrix @FFBookman will be able to persuade them to add an AES/EBU jack in the back of his head as well.


----------



## StanD

maverickronin said:


> Damn.  I turn the page and you've already beaten me to it..
> 
> Maybe in a few decades when the Machines enslave us and entrap us within the Matrix @FFBookman will be able to persuade them to add an AES/EBU jack in the back of his head as well.


 
 He who hesitates is lost, 
	

	
	
		
		

		
			




 Agent Smith might say that it's inevitable, only I have an expiration date and can't wait that long for my upgrade.


----------



## castleofargh

ffbookman said:


> I wonder how many people in "sound science" actually create and/or mix music in a professional or semi-pro environment?
> 
> Most of what I read here is discussed after the fact, trying to reverse engineer and ultimately disprove what is so easily proved in a production environment.  Higher bitrates, sensible compression, better ADCs and DAC's, the differences between digital simulations and analog realities -- all easily understood and adjusted for in a production environment. But take it to the listening side and people get crazy.
> 
> ...


 
 audio pros *manipulate the sound*, so of course the requirements can be bigger than just what we need for the final product. you explain it so badly that I wonder what you yourself are doing in a studio? just by playing around at home with some audio softwares I have been faced with why 16/44 is a bitch and make many things harder/limited. yet you only ever explain it with your "better is better is better" and whatever fallacy you come up with at the time. 
 how cool must that be, to be so right that you can even reject blind testing methods as being defective, and at the same time still not be ashamed to post your bogus propaganda. I wish I had such confidence.
  
 when I do some post processing on my pictures and will start messing around with the colors, I turn my 8bit pictures into 16bit per color channel so for RVB that's (2^16)³ colors(if I'm not wrong), who the hell needs so many? nobody looking at a picture for sure.
 I do that because if I work on 8bit and do a lot of successive changes and use a lot of layers I already worked on, let's say on the levels, I may end up with some "banding" in the final result(several values end up approximated to the same nearest available value blah blah blah not pretty). so using a higher bit depth is a way to lose less while doing some manipulations and the result looks clean. I will still output my picture in 8bit per channel and get the visual benefits of my work in 16bit. but to understand that you need to understand how things work and so far you have shown little signs of it being your case.
  
 the same way audio in 24bit instead of 16 lets you master several channels that may not all be perfectly adjusted for the same loudness yet, so importing them in a 24bit project without much care gives more headroom to do things without much fear of losing data.
 and many DSPs work better or simply need high sampling to work, then high sampling rate makes perfect sense. and overall, it is logical to work onto a bigger better stuff as we know that our manipulations are going to be destructive. so obviously it is interesting and kind of important for the workflow to use audio in high res. it has nothing to do with some audible superiority of high res. or whatever nonsense you wrote about emotion and moving people. the subjective choices of the guy doing the mastering don't justify using 24bit, the technology used in the mastering does. apple and oranges. 
 just like me doing my stuff in photoshop with 16bit per channel has nothing to do with the resolution output. and so your argument is a parody of reasoning where you try to justify what the pros are doing with the wrong motives for doing it just to serve your opinion. (who said "as always"?).
  
 you want to show that you can hear the difference clearly between 16/44 and 24/96? organize a well controlled blind test with a few guys from the objective cult in your town to testify, that would go a long way into us listening to what you badly try to say. if they also can identify the differences when they failed on their own gears, it would suggest that something is wrong with your system playing one of the formats. if you're alone to succeed, it would make you mighty famous as the golden ear of golden ears. and of course if you failed like everyone else, it would show that you're pushing something even your body doesn't believe.
 you know you won't get much success when going to cape canaveral and asking to be let into the spaceship saying "I know I can fly that stuff I don't need to prove it to you!". well it's the same here, just keeping on claiming you can do something and never ever trying to demonstrate it, you just look silly(and annoying).
  
 and in case you haven't noticed, nobody in here is trying to make pros use 16/44 for their job. because we understand that there is a difference in needs between production and listening.


----------



## old tech

Several posters in this thread have asked for proof that there is no sonic playback benefit from 24/96 or 24/192 compared with CD red book 16/44. As far as I am aware, the Moran and Meyer study has yet to be refuted, despite being a peer reviewed paper for the past eight years. It places the burden of proof back on to those who claim otherwise in controlled tests using the same master material. A link to an overview of the study is provided below.

http://drewdaniels.com/audible.pdf


----------



## StanD

I don't think FFBookman understands the difference between the nuances of the production process and the final delivered product. That's why he went on about 16/44.


----------



## arnyk

ffbookman said:


> I wouldn't say 16/44 is "inadequate". It's just a compromise based on several things - 1978 technology, price to market, the storage size of an optical disc.
> 
> I own many 16/44 recordings that sound great.
> 
> ...


 
  
 16/44 is not a compromise, it is an example of shot-in-the-dark technical overkill. When people finally understood enough about human hearing to make intelligent compromises, they invented MP3 and its descendants.
  
 The idea of analog tape sounds better ignores the actual facts of the matter. Analog tape has about 14 bits of dynamic range on the best day of its life. As soon as 16/44 and similar digital recording became feasible tools (which was about 10 years before the introduction of the CD) people abandoned analog tape in droves in favor of better sounding digital.
  
 On the day that the CD was introduced, digital with similar sample rates and with similar linear encoding was already the rule for first rate original recording and production of master tapes.
  
 What the above ignores is that the so-called 16 bit PCs of 35 years ago were on that day emulating 32 bit PCs. The only thing that was limited to 16 bits were some address registers in the CPU chip. The basic architecture supported 32 bit arithmetic from day 1 of the PC.  
  
 The above ignores the fact that good engineers don't upgrade to obtain better numbers, they upgrade to obtain real world performance benefits that serve some practical purpose.
  
 The fact that PCs are actually deep into performance overkill for most things that most people do is indicated by the fact that smart phones are very worthwhile tools, but have only a tiny fraction of the CPU power of even low end desktop PCs.


----------



## jonstatt

There are several debates merged into one here.
  
 Firstly, compression which is normally in the form of limiting. The loudness wars are a disaster for those that truly care about the quality of the music they listen to. Some examples of CD vs vinyl differences (in favour of vinyl) are Muse's new album Drones, and the particularly well known Daft Punk album, Random Access Memories.
  
 Here is the Muse info as per the DR database. It is not a "massive" difference but I have A/Bd the CD and vinyl and it is not hard to hear it.
  
http://dr.loudness-war.info/album/list?artist=muse&album=drones
  
 Also notice the hires version is no better and it is still afflicted by this compression. In fact the vinyl version is the ONLY version to have the improved DR. Considering that technology wise vinyl is the most restricted in true DR capability, this is rather ironic.
  
 Here is the Daft Punk one
  
http://dr.loudness-war.info/album/list?artist=daft+punk&album=random+access+memories
  
 Same again, ONLY the vinyl version has the better dynamic range. Once again it is very easy to hear the difference.
  
  
 So we are using technologies that have vast DR capability, and then throwing it all away! But this isn't really what should be the focus of this thread, because we are not discussing the music carrying medium and it's capabilities, but the disaster of the modern music industry and what it actually puts onto those mediums.
  
  
  
  
  
  
 Secondly, the much more inflammatory debate about whether you can "hear" the difference between "hi-res" and CD. There are some that insist they can, and then science articles insisting that we cannot. Is it like my Dad who says he cannot see the difference on his 32 inch TV between HD and SD, yet I can see it clearly? Or is it truly that our ears are not resolving enough. Do we vary enough from human to human, just like we do with our eyes (huge variances between human to human), that we simply cannot know unless we do a controlled experiment specifically with a room full of people from these forums that insist they can hear the difference. By the way, I am one of those that believes they can hear the difference.
  
 Here is an interesting forum link from a highly respected mastering engineer Steve Hoffman. And the second link is a pointer to where he himself has made production versions on SACD and CD of the same album and claims the "decay" of the sound is truncated in the CD version.....a point I know will meet with vehement objections on here.
  
http://forums.stevehoffman.tv/threads/what-sounds-just-like-the-master-tape-cd-vinyl-sacd-or-an-open-reel-tape-copy.133328/
  
http://forums.stevehoffman.tv/threads/what-sounds-just-like-the-master-tape-cd-vinyl-sacd-or-an-open-reel-tape-copy.133328/#post-3062841


----------



## RRod

> Originally Posted by *jonstatt* /img/forum/go_quote.gif
> 
> Here is an interesting forum link from a highly respected mastering engineer Steve Hoffman. And the second link is a pointer to where he himself has made production versions on SACD and CD of the same album and claims the "decay" of the sound is truncated in the CD version.....a point I know will meet with vehement objections on here.
> 
> ...


 
  
 Might as well start the objections 
	

	
	
		
		

		
		
	


	




 There's always the possibility that the CD layer on an SACD is deliberately mastered differently than the DSD layer. And then there's the whole sighted bias thing.


----------



## jonstatt

rrod said:


> Might as well start the objections
> 
> 
> 
> ...


 
  
 Sure, but remember in this case the post was written by the mastering engineer himself insisting that he didn't do anything differently between the layers. He created and mastered both SACD and CD himself.


----------



## RRod

jonstatt said:


> Sure, but remember in this case the post was written by the mastering engineer himself insisting that he didn't do anything differently between the layers. He created and mastered both SACD and CD himself.


 
  
 Questions like volume matching still come up. The sighted evaluation is the elephant in the room anyway. I saw a talk online recently by the head of a hi-res recording business. Everything was refreshingly scientific and reasonable, right until he finally discussed his preference for hi-res, which came down to "I just feel less fatigue while mastering with it." Now that certainly matters to him and is a fine reason to keep doing what he does, but it's not particularly strong evidence of format superiority. It would be nice for a truly rock-solid, large sample, reviewed-all-to-hell study to come out and put a nail in this stuff, I will say (Seems like M&M wasn't enough to convince ppl). Fine if it comes out in hi-res favor, just please fix it engineer pplz!


----------



## old tech

What Steve Hoffman say's is all and well, but can he really tell the difference in a controlled DBX? I guess not. Remember that Meyer and Moran is actually a peer review paper and after being out there for eight years it still hasn't been refuted. Why not suggest to Steve that he contact the Boston Acoustic Society and take up the challenge?

The other comment I have make about Steve, I don't know him personally and I'm sure he is a genuine guy, but he does have a bias towards analogue production and reproduction (not many studio engineers do these days) and is known for mixing music for "warmness". I hardly think he is an authority on accuracy of "sound decay".


----------



## arnyk

rrod said:


> Questions like volume matching still come up. The sighted evaluation is the elephant in the room anyway. I saw a talk online recently by the head of a hi-res recording business. Everything was refreshingly scientific and reasonable, right until he finally discussed his preference for hi-res, which came down to "I just feel less fatigue while mastering with it." Now that certainly matters to him and is a fine reason to keep doing what he does, but it's not particularly strong evidence of format superiority. It would be nice for a truly rock-solid, large sample, reviewed-all-to-hell study to come out and put a nail in this stuff, I will say (Seems like M&M wasn't enough to convince ppl). Fine if it comes out in hi-res favor, just please fix it engineer pplz!


 
  
 The assertion that Hi Rez causes less listener fatigue could be tested in an ABX test:
  
http://www.nousaine.com/pdfs/Flying%20Blind.pdf
  
 Not expecting any of its advocates to do so, just saying...


----------



## jonstatt

rrod said:


> Questions like volume matching still come up. The sighted evaluation is the elephant in the room anyway. I saw a talk online recently by the head of a hi-res recording business. Everything was refreshingly scientific and reasonable, right until he finally discussed his preference for hi-res, which came down to "I just feel less fatigue while mastering with it." Now that certainly matters to him and is a fine reason to keep doing what he does, but it's not particularly strong evidence of format superiority. It would be nice for a truly rock-solid, large sample, reviewed-all-to-hell study to come out and put a nail in this stuff, I will say (Seems like M&M wasn't enough to convince ppl). Fine if it comes out in hi-res favor, just please fix it engineer pplz!


 
  
 The main argument that seems to be the primary objection with the M&M study, is that they didn't use any recordings that started in the digital domain at the higher bit depths/rates to start with. However this is an argument mainly stated by those who have "something to lose" if the study was true. For example, see this forum post from AIX
  
http://www.avsforum.com/forum/112-surround-music-formats/981557-thoughts-meyer-moran-sacd-dvd-vs-cd-audio-study.html#post12917422
  
 Now of course, if the original recording was digital and 16 bit / 44.1kHz or even 48kHz then it would be very silly to expect to hear a difference whether it is put on CD or SACD etc. Some of the recordings used originated from analogue tapes. However it is well known that quite a few modern artists choose to record these days on 2 inch analogue tape, which is both expensive and most certainly has the resolution capability that exceeds CD (not withstanding whether this difference is audible or not).
  
 At the end of the day, my primary reason for buying vinyl is because of the loudness wars, and my primary reason for buying SACD is because I am often assured of superior mastering. I tend to research online on the forums for opinions/comparisons and the dynamic range database and then choose which format I want to buy in.
  
 If hi-res audio really is snake oil (for our ears), then couldn't any of us who have purchased hi-res equipment take legal action against the industry for false advertising? For example , Sony UK say this on their website
  
 "High-Resolution Audio works by converting analogue sound to digital at a higher and more precise rate than CDs. High-Resolution Audio can have a rate of 24bit/96Hz, capturing more of the details and subtleties in your music. Hear your tracks exactly as they were recorded with no compromise on sound."
  
 So if I cannot actually hear any more detail or subtleties in my music, I want my money back not just for the equipment but all the music purchases that I have made because of what the industry promised that are in fact "lies".


----------



## icebear

jonstatt said:


> ....
> If hi-res audio really is snake oil (for our ears), then couldn't any of us who have purchased *hi-res equipment take legal action against the industry for false advertising?* For example , Sony UK say this on their website
> 
> "High-Resolution Audio works by converting analogue sound to digital at a higher and more precise rate than CDs. High-Resolution Audio can have a rate of 24bit/96Hz, capturing more of the details and subtleties in your music. Hear your tracks exactly as they were recorded with no compromise on sound."
> ...


 
  
 Their lawyers will tell that your equipement is just insufficient to reveal all the goodness in the high rez files and that your ears are not high rez approved either.
 I'm afraid the odds are against you, very weak position in this case
	

	
	
		
		

		
			





.


----------



## RRod

> Originally Posted by *jonstatt* /img/forum/go_quote.gif
> 
> If hi-res audio really is snake oil (for our ears), then couldn't any of us who have purchased hi-res equipment take legal action against the industry for false advertising? For example , Sony UK say this on their website
> 
> ...


 
  
 Doubtful. For instance, nothing in your first quote there is untrue, at least if the files are actually hi-res and not upsampled Redbook. It's actually a craftily well-worded passage: the higher spec does in fact capture more of the details of the original wave form, and you will hear the tracks as they were recorded with no compromise in sound, because 16/44.1 is already enough to give you that.


----------



## arnyk

rrod said:


> Doubtful. For instance, nothing in your first quote there is untrue, at least if the files are actually hi-res and not upsampled Redbook. It's actually a craftily well-worded passage: the higher spec does in fact capture more of the details of the original wave form, and you will hear the tracks as they were recorded with no compromise in sound, because 16/44.1 is already enough to give you that.


 
  
 You can't capture details that aren't there to begin with.  
  
 Upsampling is often overtly fraudulent at the functional level. Understanding the process that is usually used, may make the fraud become even more apparent. Basically, the original real samples are interleaved with samples that are zeroes, and the resulting stream of samples at the higher rate is low pass filtered at the Nyquist frequency of the original data. This has the effect of filling in the time slots that were zeroes with estimated values.


----------



## old tech

jonstatt said:


> The main argument that seems to be the primary objection with the M&M study, is that they didn't use any recordings that started in the digital domain at the higher bit depths/rates to start with. However this is an argument mainly stated by those who have "something to lose" if the study was true. For example, see this forum post from AIX
> 
> http://www.avsforum.com/forum/112-surround-music-formats/981557-thoughts-meyer-moran-sacd-dvd-vs-cd-audio-study.html#post12917422
> 
> ...


 
 Apart from technical claims by marketing departments, SACDs are not entirely a cynical exercise.  As you say, with SACDs you can be (mostly) assured of superior mastering.  This allows labels to indentify and sell into a niche market which would otherwise be swamped by consumers who don't really care.  Same with records these days, putting aside the hipsters, labels know that people who stick with this format are chasing better produced sound that are not victims of the loudness wars.
  
 In regard to 2" analogue tape recording exceeding CDs, that may be the case with the best of them but the reason they are not used anymore is because 24bit digital recording is superior, and an advance in audio production.  There was a time when analogue tape was better than digital (just as there was a time when film photography was still better than digital), but the cross over happened before the millenium.
  
 As for the marketing material by Sony and others, read it carefully.  While it certainly carries an inference that hi res is superior on playback material, it does not actually state it as a fact in regard to what we hear.  The statement is actually correct to say 24bit and higher sampling rates captures more of the sublteties in the music (though probably more relevant to the bit rate), it fails to say that 16bit is already at the limit of what our ears and brains can resolve and the sampling rate of CDs is already outside the range of human hearing, so a higher one would not make a difference, even if assuming there was music content at these higher frequencies and they were captured by studio mikes and equipment.


----------



## FFBookman

stand said:


> I don't think FFBookman understands the difference between the nuances of the production process and the final delivered product. That's why he went on about 16/44.


 

 You guys are cute.  I assume none of you have actually mixed real music before.  You all pat yourself on the back feeling smarter than others yet you hide behind bad science and out of context testing to explain away what musicians, producers, engineers, and even juts plain old listeners have heard for decades now.
  
 Prove it prove it prove it you shout.   Listen and you should have all the proof you need.  
  
 Your listening tests are highly flawed. They should all be thrown out because their results are rubbish.  AB tests your memory and your ability to sort out what answer you think someone wants, not your actual hearing.  Where are the tests measuring the emotional and physical response of the person as the music plays?  Where are the tests that have any other data point than one made as an arbitrary choice by a person doing something they are never asked to do in the natural world?   You have none.
  
 A Null test would come closer to convincing me I was crazy if it didn't go through the same ADC/DAC channels and show on the same workstation that is lying to you in the first place. Same thing with oscilloscopes. No one mixes music on scopes. No one good mixes music staring at a screen drawing of a waveform. Sound is not visual, and you don't have the math for much of it still.  That's why real reverbs still sound far superior to digital ones.
  
 Mix a song, go ahead, go for it.  I know it's below you geniuses but just try it.
  
 Start with a live instrument if you want, get a decent interface, a nice room, and record it at 24/96. Or use samples and drum loops at 24/96. Make sure to apply liberal use of pan, delay, reverb, EQ, and all other tricks in the arsenal of music production.
  
 First choice you have to make is resolution. Then do your whole mix, make sure you layer plenty of parts and work very carefully on the interaction of the instrument and voices. 
  
 Then it's time to get it mastered and ship it to the "consumer".  Are you going to downsample, dither, and anti-alias your music ?  Will you hear a difference?  
  
 Can you even talk to a mastering engineer without calling him crazy for believing is such voodoo magic as 24bit audio?
  
 You guys are cute.  Have fun at my expense, I'm not the one calling the world flat.
  
 Here's my proof ----  1400k > 320k.  5800k > 1400k.   More data = more sound = more accuracy.   Prove that wrong.
  
  
 I have a 24bit brain, sorry you don't.  [Ok probably more like 20 bit, but def more than 16)


----------



## maverickronin

old tech said:


> Apart from technical claims by marketing departments, SACDs are not entirely a cynical exercise.  As you say, with SACDs you can be (mostly) assured of superior marketing.  This allows labels to indentify and sell into a niche market which would otherwise be swamped by consumers who don't really care.  Same with records these days, putting aside the hipsters, labels know that people who stick with this format are chasing better produced sound that are not victims of the loudness wars.


 
  
 When I see SACDs that are just upsampled from Redbook I have hard time not assuming it's a cynical marketing exercise to reel in suckers with more money than sense.
  
 You could build a brand image of superior sound by emphasizing recording and mastering techniques instead of  superfluous formats.  When you find that companies are sometimes simply transcoding content to the new format it's hard to take their commitment to quality seriously.


----------



## maverickronin

ffbookman said:


> Here's my proof ----  1400k > 320k.  5800k > 1400k.   More data = more sound = more accuracy.   Prove that wrong.


 
  
 How pray tell, do you determine the limit of the human ear to discern that accuracy?


----------



## old tech

maverickronin said:


> How pray tell, do you determine the limit of the human ear to discern that accuracy?


 
 Spot on.  As I keep banging on, there is an unrefuted peer review paper which does not support we can hear 24bit over 16 or 96khz sampling rate over 44.1. Rather than some here trying to convince us otherwise, why don't they contact the AES with a challenge to refute their paper?


----------



## old tech

maverickronin said:


> When I see SACDs that are just upsampled from Redbook I have hard time not assuming it's a cynical marketing exercise to reel in suckers with more money than sense.
> 
> You could build a brand image of superior sound by emphasizing recording and mastering techniques instead of  superfluous formats.  When you find that companies are sometimes simply transcoding content to the new format it's hard to take their commitment to quality seriously.


 
 I dunno.  I have come across many SACDs that sound better than their CD counterparts which supports the better mastering contention but I also have come across CDs which sound better to my ears than their SACD counterparts - eg Dire Straits Brothers in Arms which sounds compressed on the SACD.  The thing I find disappointing is that all the specialist CD labels such as MFSL, DCC etc have disappeared.  It seems that it is difficult convincing the average audiophile that investing in a more expensive format and playback device is unecessary to obtain the benefits of a well mastered final production.


----------



## parfaitelumiere

It's so funny to read the first link.
 When I knew about the hi-res files, and read about hot it worked, I found it quite as a rubbish, because it appeared immediately same a a tv showing visible colors, UV and Infrareds, even we don't see them.
 I can easily ear differences between MP3 files and 16 bit files (tried a game, and made 6/6, with PM-3 and Imac soundboard), but made no differences between 16 bit and 24 bit files, and I'm not planning to buy 150K$ stuff to try to listen any difference, so I decided to use only 16 bit 48 Khz limit files, not too heavy.


----------



## StanD

ffbookman said:


> You guys are cute.  Have fun at my expense, I'm not the one calling the world flat.
> 
> Here's my proof ----  1400k > 320k.  5800k > 1400k.   More data = more sound = more accuracy.   Prove that wrong.
> 
> ...


 
 Prove you're right, you can't. Anecdotes don't qualify.
 24 bit brain, that's discontinuous. Perhaps you shouldn't dither more than 1 bit.


----------



## upstateguy

stand said:


> ffbookman said:
> 
> 
> > You guys are cute.  Have fun at my expense, I'm not the one calling the world flat.
> ...


 
  
 Stan, it would be a slippery slope to claim that more data does not equal more accuracy?


----------



## castleofargh

upstateguy said:


> stand said:
> 
> 
> > ffbookman said:
> ...


 
  in context it's about audibility threshold and anything below it will sound the same. else it isn't an audibility threshold ^_^. so we should be looking into those thresholds for different parameters, but how could bookman do that when he rejects all proper testing methods?
 instead he goes for one of his favorite fallacy "2 is bigger than 1 so I win", that tries to put us in the group of people saying that higher resolution can't have more data. of course no such group ever existed and it was never the reason of the argument. we disagree on audibility alone and the audible benefits of those extra data.


----------



## upstateguy

castleofargh said:


> upstateguy said:
> 
> 
> > stand said:
> ...


 
  
 Can't argue with that.  ABX works fine for me.  DBT, effective, but tougher to implement. 
  
 OTOH, Bookie did mention nulling in a previous post.


----------



## RRod

For sample/bit spec comparisons, nulls are pretty enlightening and easy to do, though one has to be careful turning up the volume on high-res material that one cannot hear but that will happily break the headphones/speakers


----------



## maverickronin

old tech said:


> I dunno.  I have come across many SACDs that sound better than their CD counterparts which supports the better mastering contention but I also have come across CDs which sound better to my ears than their SACD counterparts - eg Dire Straits Brothers in Arms which sounds compressed on the SACD.


 
  
 That pretty much what I was getting at.  The format itself doesn't serve as an indicator of quality in other areas.  Some are good, some are bad, and you have to hear them to find out.  It's not any different than picking out releases in normal formats.


----------



## cjl

ffbookman said:


> Your listening tests are highly flawed. They should all be thrown out because their results are rubbish.  AB tests your memory and your ability to sort out what answer you think someone wants, not your actual hearing.  Where are the tests measuring the emotional and physical response of the person as the music plays?


 
 If one sample gives someone a different emotional response than a different one, that's ABXable. ABX doesn't tell you what method you should use to listen for a difference, nor does it place limits on the environment used, the samples used, or the time taken. It simply removes the non-audible cues for which sample is which. If you can't hear the difference in a double blind test, you can't hear the difference. Period.


----------



## StanD

upstateguy said:


> Stan, it would be a slippery slope to claim that more data does not equal more accuracy?


 
 If it can't be heard by humans then it's wasted storage space and computational resources. Perhaps if we were WhaleBats it might be more meaningful.


----------



## upstateguy

stand said:


> upstateguy said:
> 
> 
> > Stan, it would be a slippery slope to claim that more data does not equal more accuracy?
> ...


 
  
 First we have to agree with the general concept that more data equals more accuracy.
  
 Then we have to deal with the concept that there might be other ways of perceiving sound besides with our ears.
  
 ( ••  In Audio Critic's last post, IIRC, he mentioned he hadn't heard speakers that were able to fully reproduce a live venue and in Nick's post, IIRC, somatic perception/reaction to ultra high frequencies only occurred with speakers and not headphones.  Add to that, the sub bass that can be felt but not heard •• )


----------



## StanD

upstateguy said:


> First we have to agree with the general concept that more data equals more accuracy.
> 
> Then we have to deal with the concept that there might be other ways of perceiving sound besides with our ears.
> 
> ( ••  In Audio Critic's last post, IIRC, he mentioned he hadn't heard speakers that were able to fully reproduce a live venue and in Nick's post, IIRC, somatic perception/reaction to ultra high frequencies only occurred with speakers and not headphones.  Add to that, the sub bass that can be felt but not heard •• )


 
 If there is no benefit then it's a waste of resources.


----------



## interpolate

In terms of what really matters is the resolution you capture audio data at and how you record the information. Our ears, brain and other senses are incredibly dynamic. This is where the marketing starts to take over and with big numbers try to impress and otherwise convince now knowledgeable people.
  
 The science behind this must mean something otherwise companies like Naxos wouldn't be wasting their time on 352KHz/32-bit DXD recording techniques.
  
 The CD playback can only be as good as the CD Player, the DAC it uses and any other cable/speaker transport it may use.  Since this is a headphone forum, I suppose it's down to the headphones you have, the amplifier and all the other usual factors that go with it.
  
 What's wasted on some is appreciated by others.


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## nick_charles

upstateguy said:


> First we have to agree with the general concept that more data equals more accuracy.
> 
> Then we have to deal with the concept that there might be other ways of perceiving sound besides with our ears.
> 
> ( ••  In Audio Critic's last post, IIRC, he mentioned he hadn't heard speakers that were able to fully reproduce a live venue and in Nick's post, IIRC, somatic perception/reaction to ultra high frequencies only occurred with speakers and not headphones.  Add to that, the sub bass that can be felt but not heard •• )


 
  
  
 The Oohashi results have never been verified/replicated and Oohashi's follow-up lacks enough details to assess it accurately, however with direct bone conduction you can get detection of frequencies not normally audible...


----------



## StanD

interpolate said:


> In terms of what really matters is the resolution you capture audio data at and how you record the information. Our ears, brain and other senses are incredibly dynamic. This is where the marketing starts to take over and with big numbers try to impress and otherwise convince now knowledgeable people.
> 
> The science behind this must mean something otherwise companies like Naxos wouldn't be wasting their time on 352KHz/32-bit DXD recording techniques.
> 
> ...


 
 A true 32 bit DR playback that effectively uses all 32 bits,considering the ambient noise level and hearing all bits would cause a human irreparable harm.


----------



## maverickronin

interpolate said:


> The science behind this must mean something otherwise companies like Naxos wouldn't be wasting their time on 352KHz/32-bit DXD recording techniques.


 
  

 So yeah...assuming it was even possible to get  32 ENOB from a DAC with some kind of sci-fi magi-tech...
  
 Dynamic range = 32 bits * 6dB + 35dB background in a quite room = 227dB SPL...
  
Which is higher than the maximum limit before it turns from "sound" into a flatout shockwave!
  
 Which thread were we talking about the sonic weapons in again?


----------



## interpolate

I think the main reason they stick with groups of 4-bit is because it's easier to join double word-byte values than singular length bit.
  
 Take for example, 32-bit which will be 8-bit float for your decimal values, 24-bit for the other audio and usually up to 20-bit of audible data leaving 4-bit for the any low-level noise. The absolute figures look good on paper however it is a range like an Array definition in programming where all objects must be occupied however it's not always neccesary for them to contain a value other than 0 or null.
  
 There's plenty information on how DSP, DAC and other digital conversion processes are used in different combinations. 32-bit of ambient noise wouldn't be really audible noise although it will help with noise shaping odd harmonics like 12784.343Hz rather than fixed value frequencies.
  
 Although this has always been possible for a long time, it probably hasn't quite as accurate and more squared off than analogue sounds in comparison. Vinyl for the moment is the king of dynamics however with DSD and analogue amplification it may be a different story.


----------



## StanD

maverickronin said:


> So yeah...assuming it was even possible to get  32 ENOB from a DAC with some kind of sci-fi magi-tech...
> 
> Dynamic range = 32 bits * 6dB + 35dB background in a quite room = 227dB SPL...
> 
> ...


 
 If AnalogSurvivor can survive that, my hat's off to him.


----------



## StanD

interpolate said:


> I think the main reason they stick with groups of 4-bit is because it's easier to join double word-byte values than singular length bit.
> 
> Take for example, 32-bit which will be 8-bit float for your decimal values, 24-bit for the other audio and usually up to 20-bit of audible data leaving 4-bit for the any low-level noise. The absolute figures look good on paper however it is a range like an Array definition in programming where all objects must be occupied however it's not always neccesary for them to contain a value other than 0 or null.
> 
> ...


 
 I prefer leather to vinyl.


----------



## interpolate

Kinky.....I prefer pleather.


----------



## nick_charles

interpolate said:


> In terms of what really matters is the resolution you capture audio data at and how you record the information. Our ears, brain and other senses are incredibly dynamic. This is where the marketing starts to take over and with big numbers try to impress and otherwise convince now knowledgeable people.
> 
> The science behind this must mean something otherwise companies like Naxos wouldn't be wasting their time on 352KHz/32-bit DXD recording techniques.
> 
> ...


 
  
 You mention marketing, perhaps Naxos' marketing department are motivated by the fact that they can charge more for ultra high res recording a la Neil Young and Pono, Naxos has produced CD and BluRay audio versions of Prokofiev: Symphony No. 3 ( Alsop, São Paulo Symphony )  where the CD version is a *lot* cheaper. If enough people want it a company might make it to fill that demand regardless of whether there is any meaningful difference. I have a lot of respect for Naxos as they made Classical music affordable for me back in the late 80s when the bigger labels charged a lot for classical. Still, I don't see them listing any meaningful listening tests between the different formats...but it would be unfair to single out Naxos here, Sony has never done any credible controlled listening tests between CD and SACD, nor have the developers of high res PCM systems. 
  
 If you do a search you just get the same old dog and pony shows that go all the way back to Edison...


----------



## interpolate

Too much money spent on R & D and not enough output. I refused to buy downloads for years in preference to CD. People I know are happy with Torrent copies of albums, however the audio snob in me always craves a sound which leaves me thinking "Wow" rather than "it'll do".


----------



## interpolate

Naxos is a good source for classical. Classical to me is a bit of a mystery. I mean how much does it take to produce a performance of a dead persons music.


----------



## RRod

interpolate said:


> There's plenty information on how DSP, DAC and other digital conversion processes are used in different combinations. 32-bit of ambient noise wouldn't be really audible noise although it will help with noise shaping odd harmonics like 12784.343Hz rather than fixed value frequencies.


 
  
 Is this the noise shaping I'm thinking of, because there I thought you want a higher sampling rate (not bit depth) so that you have more frequency range in which to concentrate originally broadband noise. After all, DSD uses just 1 bit and depends upon noise shaping to get dynamic range.
  


interpolate said:


> Naxos is a good source for classical. Classical to me is a bit of a mystery. I mean how much does it take to produce a performance of a dead persons music.


 
  
 Classical is the Linux of music: everyone does his/her one thing really well.


----------



## interpolate

stand said:


> If AnalogSurvivor can survive that, my hat's off to him.


 
  
 I was just thinking about that, if the measurement of noise is in a room larger than 1 metre then surely the sound will diffused by the different acoustic properties. And if indeed we can only hear around 20-bit of that information on a normal basis, the extra headroom won't be really experience in the remaining 12-bit. Alas it should be more coherent in the initial 20-bit of audio data. If you think about it, we only hear would we need to and not what everything we can. It's a waste of time really although it keeps someone in a job.


----------



## interpolate

Sampling frequency is different from bit-depth as we know. The higher frequency range and deeper the bit depth - the more noise can pushed to the nether regions in theory at least.
  
 For example, if you record a violin in 16-bit. It will sound good. In 24-bit, it might sound similar however it will have less audible noise and more character due to the reduced quantisation added to final recording when digitised.
  
  
 I'm a hobbuist/muso who likes tech stuff, so I can give you the reasons why I think it's beneficial. For the end user (the listener) these things are less important however it does mean when buy that 24/96 FLAC, DSD or Bluray Audio disc then the quality of mastering should be surplus to requirement alas noticeable when comparing an ordinary CD or MP3 download.


----------



## nick_charles

interpolate said:


> Naxos is a good source for classical. Classical to me is a bit of a mystery. I mean how much does it take to produce a performance of a dead persons music.


 
  
 If you are doing it from start to finish you need a ****ing big orchestra for a lot of stuff such as Mahler symphonies, a full size orchestra can easily be over 100 players, the base salary for a grunt player in the CSO is over $144K(2012)  add in a star conductor, principals, facilities, rehearsal time, recording time and that just gets you to the master tape stage... not cheap


----------



## interpolate

I guess not...and I sort of knew it...
  
 My attempt at irony was misread.


----------



## StanD

interpolate said:


> Sampling frequency is different from bit-depth as we know. The higher frequency range and deeper the bit depth - the more noise can pushed to the nether regions in theory at least.
> 
> For example, if you record a violin in 16-bit. It will sound good. In 24-bit, it might sound similar however it will have less audible noise and more character due to the reduced quantisation added to final recording when digitised.
> 
> ...


 
 I'll bet that there is more on site ambient noise than the SNR of good 16 Bit hardware, So I think higher resolution does nothing useful in this regard. One might have to be sure the violinist neither breathe or fart.


----------



## arnyk

upstateguy said:


> First we have to agree with the general concept that more data equals more accuracy.
> 
> Then we have to deal with the concept that there might be other ways of perceiving sound besides with our ears.
> 
> ( ••  In Audio Critic's last post, IIRC, he mentioned he hadn't heard speakers that were able to fully reproduce a live venue and in Nick's post, IIRC, somatic perception/reaction to ultra high frequencies only occurred with speakers and not headphones.  Add to that, the sub bass that can be felt but not heard •• )


 
  
 More data can equal more numerical accuracy, but what about audible accuracy?
  
 Ever hear of the law of diminishing returns?
  
 Ever hear about the concept of the threshold of audibility?
  
 These are all proven scientific concepts.
  
 As far as perceiving sound in other ways than just hearing, that is old news to those of us who have for example felt bass pounding into our chests.
  
 Do you think that an ABX test is just limited to perceiving sound in a narrow frequency band?  Isn't it true that if you are comparing sound A to sound B that any listener would use every perceptual means available to them?


----------



## RRod

interpolate said:


> Sampling frequency is different from bit-depth as we know. The higher frequency range and deeper the bit depth - the more noise can pushed to the nether regions in theory at least.
> 
> For example, if you record a violin in 16-bit. It will sound good. In 24-bit, it might sound similar however it will have less audible noise and more character due to the reduced quantisation added to final recording when digitised.
> 
> ...


 
  
  
 You seem to be calling upon the argument I've seen that since 24bits rounds more closely to the actual value, it MUST have less audible noise, but that's only true if the distortion due to the truncation (if we're not dithering) is actually of enough magnitude to punch through the musical material. And if that is the case, I'd say someone didn't record the violin correctly.


----------



## interpolate

24-bit (binary terms) is purely mathematical. In theory it should have less noise as the signal to noise ratio produced by the electronics is pushed further down in terms of dynamics. 
  
 You can use the same equipment to record from the same source, the noise profile should be the same however it only sample the first 16-bit and everything else will be binned in the signal captured. Any decimal values will be made absolute and squared with quantisation.
  
 The violin example is a good one because the extra resonance is captured particularly with higher sampling frequencies and larger bit-depths. The returning factor for end listener is more pleasurable experience particular if you don't have to reduce or otherwise alter the content distributed.
  
 However my point is without dithering added to the mastering path and kept from any other quantisation, something should sound as good as the time it was recorded in theory,,,,
  
 I wouldn't say I am 100% qualified to give an Oxford Dictionary reference on the subject although I do and have had read and studied a lot of the scientific stuff that makes some of this stuff work. 
  
 Also did you know, the lower harmonics of a turntable can be as low as 10Hz.


----------



## interpolate

Nota Bene:
  
 I realise not everything I believe or state is always correct. However I do know I am Scottish, and you cannae take that awa' fae meh aye.


----------



## RRod

interpolate said:


> 24-bit (binary terms) is purely mathematical. In theory it should have less noise as the signal to noise ratio produced by the electronics is pushed further down in terms of dynamics.
> 
> You can use the same equipment to record from the same source, the noise profile should be the same however it only sample the first 16-bit and everything else will be binned in the signal captured. Any decimal values will be made absolute and squared with quantisation.
> 
> ...


 
  
 24 is definitely theoretical since the best specs I've seen for an ADC/DAC are around 21 bits. I can't see how the violin has the dynamic range to tax 16-bits, so I'd disagree on that point. Frequency extension is also something I'm not too convinced by either, since I've downsampled violin/string quartet material to sub-Redbook samples/bit specs and not heard any problems.
  
 I've got a CD somewhere that has a recording of a shuttle lift-off that has all kinds of stuff from 5-10Hz.


----------



## interpolate

I use a software sampler where the strings were sampled at 24-bit/88.2Khz and reduced to 24-bit/44.1Khz (I have the 16-bit version). Am I missing out on anything significant...maybe although perhaps nothing major as long as the sampling frequencies are the same.


----------



## StanD

interpolate said:


> I use a software sampler where the strings were sampled at 24-bit/88.2Khz and reduced to 24-bit/44.1Khz (I have the 16-bit version). Am I missing out on anything significant...maybe although perhaps nothing major as long as the sampling frequencies are the same.


 
 16 bits should be fine for playback. If there are 32 bit computations done in the sampler for processing sounds then the higher resolution might be useful for numerical precision as it related to the final product of the processing, just the same as in a recodring studio. Then again the sampler might scale up the 16 bit source before processing. Who nows what goes on inside the gadget.


----------



## interpolate

It's a 64-bit sample engine.


----------



## StanD

interpolate said:


> It's a 64-bit sample engine.


 
 That doesn't say how it processes samples of lesser bit depths or different sample rates.


----------



## interpolate

Quite. I never said it did. The 64-bit is merely to buck the ram cap of 4gb per instance. Anyway lets get back on topic.


----------



## old tech

interpolate said:


> I think the main reason they stick with groups of 4-bit is because it's easier to join double word-byte values than singular length bit.
> 
> 
> 
> ...



 


Vinyl is not the king of dynamics. It has a much lower dynamic range than 16bit digital. Whether the mastering takes advantage of the superior digital is another story. Can you provide just one objective measure which shows vinyl to be equal, let alone exceed, 16bit?


----------



## old tech

nick_charles said:


> The Oohashi results have never been verified/replicated and Oohashi's follow-up lacks enough details to assess it accurately, however with direct bone conduction you can get detection of frequencies not normally audible...



 


Professor Oohashi has accepted the peer reviews which invalidated his study. The Meyer and Moran study on the other hand has after eight years stood the test of time. As in the visual sphere, we cannot hear (see) frequencies outside the established hearing (seeing) range of humans.


----------



## old tech

Also did you know, the lower harmonics of a turntable can be as low as 10Hz.

 


Maybe so, but that 10hz is hardly going to be a clear, well defined note. It is not possible for any turntable to reproduce frequencies below 100hz that are accurate. The vinyl medium and its ability to store this information in the grooves is just one limiting factor.


----------



## Gr8Desire

My _remastered_ 176.4/24 recording of the 1959 album *Time Out by Dave Brubeck* is the best sounding music I own.  Much better than my existing 44/16 recording of the same. Time Out was the first hi-res album I bought.  I was impressed.

 Because of this album, I bought a few more.  All genres.  Some ADD like the Brubeck recording (including other Brubeck recordings) and some DDD.  None sound remarkably different than the equivalent 44/16 recording.  

 I have good DACs that handle at least 176.4/24.  And average DACs that work fine at 44/16.  And then there is my iPhone 6 DAC.

 Everything except this one Brubeck album sound pretty much the same.  *What gives?*

 = = = 
  
*​*I have down sampled 176.4/24/24 to 44/16 and up sampled to 384/32.   Still, nothing changes much.

*My decidedly non-audiophile conclusion:*

 The Brubeck recording was remastered extremely well.  Doesn't mater how I play it.  The sound is good.  

 Other hi-res material I have is the opposite.  It doesn't sound much different at any sample / bit rate.  I figure: The re-mastering (if any) was mediocre and so was the final result. 

_Maybe people are focusing on the wrong facets of hi-res audio?_


----------



## castleofargh

we can remove the "maybe" and the question mark.


----------



## StanD

old tech said:


> Also did you know, the lower harmonics of a turntable can be as low as 10Hz.


 
  
  

 Maybe so, but that 10hz is hardly going to be a clear, well defined note. It is not possible for any turntable to reproduce frequencies below 100hz that are accurate. The vinyl medium and its ability to store this information in the grooves is just one limiting factor.
  
 Lower Harmonics? A Harmonic is a whole numbered multiple of a lower fundamental frequency.If so what is the fundamental frequency that you are referring to?


----------



## old tech

stand said:


> Maybe so, but that 10hz is hardly going to be a clear, well defined note. It is not possible for any turntable to reproduce frequencies below 100hz that are accurate. The vinyl medium and its ability to store this information in the grooves is just one limiting factor.
> 
> Lower Harmonics? A Harmonic is a whole numbered multiple of a lower fundamental frequency.If so what is the fundamental frequency that you are referring to?



 

Sorry, it was not me saying the10hzof lower harmonic - I quoted Interpolate but for some reason the quote box did not appear around it.


----------



## MacacoDoSom

stand said:


> maverickronin said:
> 
> 
> > So yeah...assuming it was even possible to get  32 ENOB from a DAC with some kind of sci-fi magi-tech...
> ...


 

 There seems to be some confusion between dB SPL and dB FS not to mention the background noise that doesn't exist with headphones, so, you'll have 192 dB DR... witch  is nothing for a guy that can hear 50kHz while listening a XVII century harpsichord in a chapel with no background noise.... and by the way, what is that thing... background noise...? isn't it part of the music?
 and ...can tinnitus be considered as background noise? or is it a constant that can be disregarded?
 I want to listen to the CD MATS test files *AnalogSurvivor*... can I expect them by Christmas?


----------



## StanD

macacodosom said:


> There seems to be some confusion between dB SPL and dB FS not to mention the background noise that doesn't exist with headphones, so, you'll have 192 dB DR... witch  is nothing for a guy that can hear 50kHz while listening a XVII century harpsichord in a chapel with no background noise.... and by the way, what is that thing... background noise...? isn't it part of the music?
> and ...can tinnitus be considered as background noise? or is it a constant that can be disregarded?
> I want to listen to the CD MATS test files *AnalogSurvivor*... can I expect them by Christmas?


 
 You'll play that harpsichord for WhaleBats on a CD Mat before AanlogSurvivor delivers.


----------



## MacacoDoSom

macacodosom said:


> stand said:
> 
> 
> > maverickronin said:
> ...


 
  


stand said:


> macacodosom said:
> 
> 
> > There seems to be some confusion between dB SPL and dB FS not to mention the background noise that doesn't exist with headphones, so, you'll have 192 dB DR... witch  is nothing for a guy that can hear 50kHz while listening a XVII century harpsichord in a chapel with no background noise.... and by the way, what is that thing... background noise...? isn't it part of the music?
> ...


 

 ...well, I was thinking about Christmas 2116...maybe by then...
  
 and I really like that new species... the WhaleBats, they seem to be in expansion populating this (and not only) forum... if not the earth...
 at least they seem to be very happy, contrary to the mere humans that are always being confronted with their claims and feeling inferior,  if not completely deaf... it would be nice if we could hear as they do... or, maybe not...
 there is a emerging new industry dedicated to products for the WhaleBats!


----------



## interpolate

10Hz is the about where any motor resonance from a belt drive would be. 7


----------



## StanD

interpolate said:


> 10Hz is the about where any motor resonance from a belt drive would be. 7


 
 Then it's not harmonics.


----------



## interpolate

No it's not alas sometimes it can crossover into badly grounded pick-ups.


----------



## StanD

interpolate said:


> No it's not alas sometimes it can crossover into badly grounded pick-ups.


 
 You don't have to ground an F-150.


----------



## interpolate

Do you mean the Ford truck or something else like my chances of winning the national lottery?


----------



## StanD

interpolate said:


> Do you mean the Ford truck or something else like my chances of winning the national lottery?


 
 Well we were talking about Pickups so don't expect a Ferrari. AS will throw in one of his CD Mats as an extra prize.


----------



## interpolate

Nice. About time I got my just deserts.


----------



## old tech

Hey, if you guys like a chuckle have a read of the comments by Fremer and some of the replies in the comments section of this you tube page.
https://www.youtube.com/watch?v=4eC6L3_k_48


----------



## Gr8Desire

_Also did you know, the lower harmonics of a turntable can be as low as 10Hz?_
  
  

 Also, did you know Einstein postulated that time slows down when traveling at the speed light?

 Today you have about as much chance of traveling at the speed of light as accurately reproducing a 10 Hz tone from vinyl.  


*Down with science!*  There. _That's better._


----------



## interpolate

http://www.instantrimshot.com


----------



## interpolate

Although to be serious for one moment.
  
 http://www.hi-fiworld.co.uk/vinyl-lp/70-tests/105-turntable-tests.html
  
 See I am not talking total nonsense created by some wild yet vivid dream.


----------



## Gr8Desire

interpolate said:


> Although to be serious for one moment.
> 
> http://www.hi-fiworld.co.uk/vinyl-lp/70-tests/105-turntable-tests.html
> 
> See I am not talking total nonsense created by some wild yet vivid dream.


 
  
 The problem with these measurements:  Back in the day, no one ever mastered content to support playback at these theoretical limits. Content below 100Hz was attenuated to avoid having the stylus jump from its track. Content was compressed to raise quiet passages above a high noise floor. Finally, all music was heavily EQ'd to try and create decent playback by archaic components. The Phil Spector "Wall of Sound" is the result of proprietary EQ and volume changes that simply made vinyl sound better on mediocre equipment.  

 Nothing was real or accurate because vinyl was such a limited playback medium. Now that we finally have accurate recording and playback technology, some audiophiles say we must eschew it in favour of content and playback devices that were never very good in their day. A few dubious improvements in turntables can't change the physics - _except in the heads of those who want to believe_. 

 >48K sampling and >16-bit dynamic range _encoding_ has its place: Mastering of content can benefit from higher bits rates until the final mix down. Since most only play the final mix down, higher bit rates are superfluous - _except in the heads of those who want to believe _(see all threads with Pono in subject line for more)

 So it is a wild yet vivid dream.  And it's a dream that never even happened.


----------



## interpolate

That's me told then....

But seriously, some good points made.


----------



## arnyk

interpolate said:


> 10Hz is the about where any motor resonance from a belt drive would be. 7


 
  
  
 Resonances in LP players in the vicinity of 10 Hz are due to the mass of the tone arm acting on the compliance of the needle suspension in the cartridge.


----------



## miceblue

ffbookman said:


> You guys are cute.
> 
> Here's my proof ----  1400k > 320k.  5800k > 1400k.   More data = more sound = more accuracy.   Prove that wrong.
> 
> ...



So you're saying you can hear a difference between this 24/96 track and a downsampled 16/44.1 version of it? I'd be impressed if you could.
https://dl.dropboxusercontent.com/u/2816447/2496vs1644.zip
53 MB

Or 24/96 vs 16/96?
https://dl.dropboxusercontent.com/u/2816447/2496vs1696.zip
43 MB


Be sure to post an ABX log with at least 10 trials too.


----------



## RRod

FFB rejects ABX as a viable test method for detecting differences between hi-res and Redbook format.


----------



## upstateguy

miceblue said:


> ffbookman said:
> 
> 
> > You guys are cute.
> ...


 
  
 Thanks for the links, I'm going to give them a listen and have a look at the wave forms.
  
 And please link me to where you posted your 10 ABX trials.  I thought I saw them somewhere but don't remember where.


----------



## icebear

gr8desire said:


> My _remastered_ 176.4/24 recording of the 1959 album *Time Out by Dave Brubeck* is the best sounding music I own.  Much better than my existing 44/16 recording of the same. Time Out was the first hi-res album I bought.  I was impressed.
> 
> Because of this album, I bought a few more.  All genres.  Some ADD like the Brubeck recording (including other Brubeck recordings) and some DDD.  None sound remarkably different than the equivalent 44/16 recording.
> 
> ...


 
  
 This one should be permanently pinned


----------



## StanD

rrod said:


> FFB rejects ABX as a viable test method for detecting differences between hi-res and Redbook format.


 
 Truth be damned, I think he prefers anecdotes that support his speculations.


----------



## miceblue

rrod said:


> FFB rejects ABX as a viable test method for detecting differences between hi-res and Redbook format.



Ah, I see. Well then, since this is the Internet, you can believe anyone for anything. I can pass the ABX test really easily, so easily that I don't need to provide an ABX log because it's just a waste of my time. It's okay, I can do it absolutely for certain. I'm an Expert; you cannot deny my claims. I have been working in the industry for over 30 years and you pathetic, think-you-know-all plebeians know nothing about 24-bit audio and high-resolution music. My brain, ears, and ego have transcended all.

[video]https://www.youtube.com/watch?v=BKorP55Aqvg[/video]








upstateguy said:


> Thanks for the links, I'm going to give them a listen and have a look at the wave forms.
> 
> And please link me to where you posted your 10 ABX trials. I thought I saw them somewhere but don't remember where.



I had logs for a different test. The Foobar ABX plugin also got updated recently, so it's a lot more accurate with a checksum and it makes it harder to make up data.
http://www.foobar2000.org/abx/signaturecheck


Spoiler: abx log



Setup: MacBook, USB A-USB B printer cable, LH Labs Linear Power Supply 4, LightSpeed 1G USB cable to LH Labs Pulse X Infinity, high gain, Femto Time Mode digital filter, XLR balanced headphone out, HIFIMAN HE1000 [beta]. I was using the original 24/96 file and the original 16/44.1 downsampled file, so the DAC did have a delay for sample rate switching. I wanted to do this in case someone made the argument that an upsampled 16/44.1 file to 24/96 somehow makes the ABX test invalid. Foobar's output was via ASIO, minimum latency, small 256 sample buffer size.


foo_abx 2.0 report
foobar2000 v1.3.8
2015-07-13 11:31:47

File A: A.flac
SHA1: 8ce7cdf117f2f7a0a4c934dcd04724c593842218
File B: B-1644_Original.flac
SHA1: a82d0e8be147c62b4532a1b9316795d13c7b16e8

Output:
ASIO : foo_dsd_asio
Crossfading: NO

11:31:47 : Test started.
11:33:54 : 00/01
11:34:50 : 00/02
11:35:39 : 00/03
11:36:03 : 01/04
11:36:41 : 02/05
11:36:58 : 02/06
11:37:10 : 02/07
11:37:38 : 02/08
11:38:02 : 02/09
11:38:47 : 03/10
11:39:34 : 04/11
11:39:47 : 05/12
11:39:47 : Test finished.

 ---------- 
Total: 5/12
Probability that you were guessing: 80.6%

 -- signature -- 
b6ec774cd9e35aef1a5436dac9938588881fcbbf


----------



## upstateguy

icebear said:


> gr8desire said:
> 
> 
> > My _remastered_ 176.4/24 recording of the 1959 album *Time Out by Dave Brubeck* is the best sounding music I own.  Much better than my existing 44/16 recording of the same. Time Out was the first hi-res album I bought.  I was impressed.
> ...


 

 Thanks for the reminder....  it is a great recording.


----------



## arnyk

rrod said:


> FFB rejects ABX as a viable test method for detecting differences between hi-res and Redbook format.


 
  
 It's all part of his personal little logic-tight box.


----------



## Steve Eddy

arnyk said:


> It's all part of his personal little logic-tight box.




Some people's brains just aren't wired for terribly rational thought. In that respect you can't really blame them. It's just how their DNA wired them up. Doesn't make it any less frustrating dealing with them though.

se


----------



## arnyk

steve eddy said:


> Some people's brains just aren't wired for terribly rational thought. In that respect you can't really blame them. It's just how their DNA wired them up. Doesn't make it any less frustrating dealing with them though.
> 
> se


 
  
 I know exactly what you are talking about Steve, but just because its part of nature doesn't mean that people are absolved of the responsibility to know themselves and their personal limitations and act accordingly.
  
 I can't throw a baseball worth a darn (never have) so I don't present myself as being a good pitcher, always defer to those who have better natural ability (just  about everybody), and don't try to lecture people about how to throw a ball.


----------



## Steve Eddy

arnyk said:


> I know exactly what you are talking about Steve, but just because its part of nature doesn't mean that people are absolved of the responsibility to know themselves and their personal limitations and act accordingly.




Do you realize what you're saying here? 

Think about it for a moment.

se


----------



## icebear

ROFL ... that door was w i i i i i d e open


----------



## arnyk

steve eddy said:


> Do you realize what you're saying here?
> 
> Think about it for a moment.
> 
> se


 
  
  
 No Steve I never think one second about what I post here. It makes my posts fit in with yours that show zero thought and insight, just ongoing hatred. What do you think?


----------



## Steve Eddy

icebear said:


> ROFL ... that door was w i i i i i d e open




And sadly, I think I'm going to have to add Arny to the list of those who just aren't wired for terribly rational thought. The irony is downright painful.

se


----------



## arnyk

steve eddy said:


> And sadly, I think I'm going to have to add Arny to the list of those who just aren't wired for terribly rational thought.
> 
> se


 
  
 Crocodile tears become you, Steve. Or should I say you have become them? 
  
 It is true that I'm not wired for terribly rational thought. I'm wired for really pretty good rational thought. 
  
 You want to talk about opening doors? In the interest of peace I have granted you mercy. No matter what you do, no matter what insult you contrive, no matter how many of your friends you try to get to pile up on me, that will continue. Enjoy!


----------



## Steve Eddy

arnyk said:


> It is true that I'm not wired for terribly rational thought. I'm wired for really pretty good rational thought.




You just pulled down your pants and have demonstrated to everyone reading this that obviously you are not.




> You want to talk about opening doors? In the interest of peace I have granted you mercy.





[VIDEO]http://youtu.be/zKhEw7nD9C4[/VIDEO]


se


----------



## interpolate

I know a song that'll you get on your nerves.


----------



## Gr8Desire

arnyk said:


> I know exactly what you are talking about Steve, but just because its part of nature doesn't mean that people are absolved of the responsibility to know themselves and their personal limitations and act accordingly.
> 
> I can't throw a baseball worth a darn (never have) so I don't present myself as being a good pitcher, always defer to those who have better natural ability (just  about everybody), and don't try to lecture people about how to throw a ball.


 

 If more audiophiles played baseball...
  

I dipped my baseball in helium. Now it goes farther (I hasten to use the verb _further, _but that distinction will only encourage the fringe elements).  
I changed to oxygen-free stitching on my baseball. It now spins with less obfuscation.
I extended my glove by 8 fingers. It is now more digits-al.
I upgraded to vinyl shoes. They have more natural curves and reduce ringing by 20%.
Added unidirectional laces and increased my dynamic static potential by a factor of 2.


----------



## bfreedma

gr8desire said:


> If more audiophiles played baseball...
> 
> 
> I dipped my baseball in helium. Now it goes farther (I hasten to use the verb _further, _but that distinction will only encourage the fringe elements).
> ...


 
  
 Ironically, you might be surprised at how many takers your would get.  Superstition and belief in unproven magical abilities is as prevalent in professional sports as it is in audio.
  
 Copper bands and titanium performance necklaces anyone?.....


----------



## interpolate

I have Titanium watches and Glasses. Also Tommy Hilfiger spares made from industry glass plastic frames...
  
 The performance from them is insurmountable.


----------



## interpolate

The Monty Python video is great.


----------



## castleofargh

the holy grail is in the @castleofargh


----------



## interpolate

Ah so not the cave of the terrible beast then.....right.....


----------



## Steve Eddy

castleofargh said:


> the holy grail is in the @castleofargh




Was wondering when you'd chime in. 

se


----------



## FFBookman

ABX tests test *memory* only.  
  
 Memory recall of music is very different than initial reception of music.
  
 Since we cannot listen to and compare multiple audio sources in parallel we have to rely on memory to perform this test.
  
 But it isn't a memory test, it's a sound quality test.
  
 Here is your fatal flaw in ABX tests. Not the only fatal flaw either.


----------



## maverickronin

ffbookman said:


> Memory recall of music is very different than initial reception of music.


 
  
 What on Earth does that even mean?


----------



## Gr8Desire

ffbookman said:


> ABX tests test *memory* only.
> 
> Memory recall of music is very different than initial reception of music.
> 
> ...


 


 A common goal for ABX testing is: *Can the listener consistently distinguish a difference?*

 Audiophiles claim they need to listen longer to decide on a preference (or what you call a _sound quality test_) .  

 ABX testing methodology says " Fair enough but 1) first tell me which samples are different and then 2) we'll let you listen longer to tell us the one you prefer".

 When it comes to wires, extreme sample rates, insane dynamic range and fairy dust, audiophiles have trouble getting past step 1. 

 You can't prefer something that you don't hear as different.


----------



## DreamKing

ffbookman said:


> ABX tests test *memory* only.
> 
> Memory recall of music is very different than initial reception of music.
> 
> ...


 
  
 You don't know what an ABX test is, with the proper equipment switching should be quasi-instantaneous. But if you think your brain and ears are objective measuring tools, your sense of reality is already distorted.


----------



## StanD

ffbookman said:


> ABX tests test *memory* only.
> 
> Memory recall of music is very different than initial reception of music.
> 
> ...


 
 And done properly within the constraints of Echoic Memory one can do well. Reception? 
  
  You are funny.


----------



## jcx

gr8desire said:


> ...You can't prefer something that you don't hear as different.


 
 actually that being wrong in practice is the main problem - you can prefer the story that goes along with - sighted listening definitely gives different neuron firing patterns in your brain even if the sound waves reaching your ear are the same
  
 logic isn't our most common decision making strategy


----------



## StanD

jcx said:


> actually that being wrong in practice is the main problem - you can prefer the story that goes along with - sighted listening definitely gives different neuron firing patterns in your brain even if the sound waves reaching your ear are the same
> 
> logic isn't our most common decision making strategy


 
 Call it Fantasy Audio.


----------



## castleofargh

ffbookman said:


> ABX tests test *memory* only.
> 
> Memory recall of music is very different than initial reception of music.
> 
> ...


 
 judges ABX. doesn't know what it is. 
	

	
	
		
		

		
		
	


	



  
*abx doesn't test memory, it's the opposite.* it tries to limit the impact of memory by offering instant switching.
 and it isn't a memory test, just like it isn't a sound quality test. it's a test to tell if people are repeatedly able to notice differences. not to test which sounds better or is better.
  
  
 for somebody so concerned about fatal flaws, you write quite a load a BS. again.


----------



## Gr8Desire

jcx said:


> actually that being wrong in practice is the main problem - you can prefer the story that goes along with - sighted listening definitely gives different neuron firing patterns in your brain even if the sound waves reaching your ear are the same
> 
> logic isn't our most common decision making strategy


 
  

 Not being able _distinguish a difference_ is an outcome from most types of AB testing.

_Hearing differences_ based on bias is an outcome from most types of audiophile testing.  

_Logic is not possible when one is uninformed.  Fantasy is._


----------



## arnyk

ffbookman said:


> ABX tests test *memory* only.


 
  
 Obviously you have never actually personally done an ABX and have no idea what they are because ABX tests are not just tests of memory.
  
 ABX tests can test the whole experience of listening to music. If you don't know that, you know nothing about ABX.


----------



## vcoheda

comment on ABX test:
  
 "In summery ABX double-blind testing does not prove that everything sounds the same as real sonic differences are easily heard in casual listening. No, instead what ABX double-blind testing proves is that human subjects do not have the ability to compare three sonic events sequentially with any statistically significance, revealing a deficiency in short-term sonic memories of our species."
  
 http://www.positive-feedback.com/Issue56/abx.htm


----------



## maverickronin

If his conclusion is that ABX tests prove that short term echoic memory sucks where exactly to he get off assuming that our long term echoic memory is any better?


----------



## RRod

vcoheda said:


> comment on ABX test:
> 
> "In summery ABX double-blind testing does not prove that everything sounds the same as real sonic differences are easily heard in casual listening. No, instead what ABX double-blind testing proves is that human subjects do not have the ability to compare three sonic events sequentially with any statistically significance, revealing a deficiency in short-term sonic memories of our species."
> 
> http://www.positive-feedback.com/Issue56/abx.htm
 
  
 As usual, an article calling science into question calls upon no science.
  
 You can take any favorite track and find some downsampling/quantization that will cause audible differences statistically detectable via ABX. As you increase the specs back towards Redbook, it will become gradually harder to detect differences. That's a nice, coherent testing methodology.


----------



## jcx

I quickly verified phase audibility with foobar ABX plugin - phase audibility seems obscure enough that many audio pros dismiss it altogether and may well be more difficult to hear with speakers in a room than with headphones
  
 Ethan Winer - a musician, producer and audio writer was initially in the "phase denier" camp and when badgered to ABX did hear the difference in ABX, verified, duplicated the files in his pro audio workstation


----------



## RRod

jcx said:


> I quickly verified phase audibility with foobar ABX plugin - it seems obscure enough that many audio pros dismiss it altogether and may well be more difficult to hear with speakers in a room than with headphones
> 
> Ethan Winer - a musician, producer and audio writer was initially in the "phase denier" camp and when badgered to ABX did hear the difference


 
  
 What exactly are we talking about as "phase denial"?


----------



## jcx

many don't believe that the polarity/"phase" of a musical signal makes an audible difference
  
 the easiest to think about for most: a mic could be placed on either side of a drumhead, receive a positive or negative initial going pressure from the strike - many pros will not take care to capture the "audience perspective" polarity or go to any effort to preserve signal polarity in the processing chain
  
  
 I tested continuous tone bursts with added 2nd harmonic - depending on the relative phase of the added 2nd the waveform is more peaked or rounded in one polarity or the other; peaking at positive V for instance - can also be done as sawtooths where the leading edge is the steeper or slower/positive or negative
  
 in the continuous tone bursts there is a perception of slight pitch change even though the frequency is absolutely unchanged
  
  
 the issue popped up a few pages previous, continued for quite a while:
  
http://www.head-fi.org/t/486598/testing-audiophile-claims-and-myths/2370#post_10323166


----------



## castleofargh

vcoheda said:


> comment on ABX test:
> 
> "In summery ABX double-blind testing does not prove that everything sounds the same as real sonic differences are easily heard in casual listening. No, instead what ABX double-blind testing proves is that human subjects do not have the ability to compare three sonic events sequentially with any statistically significance, revealing a deficiency in short-term sonic memories of our species."
> 
> http://www.positive-feedback.com/Issue56/abx.htm
 
 from the woman who only swears by subjectivism and doesn't mind calling science, the enemy of music. in a world where everything audio has been built by an engineer, every sound recorded by another engineer, just one of the many many things that go to show how weak is her logic.
"real sonic differences are easily heard in casual listening." yeah right, and blind test are just there to annoy people because bias and placebo don't exist. and the laser cats are dancing on the rainbow and unicorns sing the song of the SACD. ignorance, ignorance, ignorance. I would really enjoy seeing evidence of that idiotic claim. after all I've only seen tens of examples in music(or outside music) demonstrating the exact opposite and how people are partial to bias in casual listening, and how we could make people believe anything with the right conditioning prior to the "casual listening" session.
how ignorant must one be of the human being, to pretend that abx is useless and at the same time, neglects bias, placebo, and all forms of preconceptions as if they were a non issue. that's really missing the beams in her own eyes.
  
 do you actually endorse that quote or was it a troll joke?


----------



## castleofargh

jcx said:


> many don't believe that the polarity/"phase" of a musical signal makes an audible difference
> 
> the easiest to think about for most: a mic could be placed on either side of a drum, receive a positive or negative initial going pressure from the strike - many pros will not take care to capture the "audience perspective" polarity or go to any effort to preserve signal polarity in the processing chain
> 
> ...


 
  it's not unusual for IEMs to have inverted polarity. don't know if headphones are the same or if they pay more attention?
 and with the most common IEM cables, hard to know when it's plugged-in right.


----------



## GrindingThud

Link to a test I did with absolute phase a while back. With some material, I could hear a difference: http://www.head-fi.org/t/715199/absolute-phase-the-next-frontier-yeah-right#post_10473704
In terms of music enjoyment....either was fine, nor could I tell what one was "correct" only that they were different.


----------



## arnyk

vcoheda said:


> comment on ABX test:
> 
> "In summery ABX double-blind testing does not prove that everything sounds the same as real sonic differences are easily heard in casual listening. No, instead what ABX double-blind testing proves is that human subjects do not have the ability to compare three sonic events sequentially with any statistically significance, revealing a deficiency in short-term sonic memories of our species."
> 
> http://www.positive-feedback.com/Issue56/abx.htm
 
  
 A good example of how thoroughly the writer simply does not get science, logic, reason or just rhetoric.
  
 No ABX test can prove that everything sounds the same, or that any recorded sound is indistinguishable from the corresponding natural sound.
  
 For one thing tests don't prove anything - they just provide evidence that can be used to form a logical proof.
  
 An ABX test that attempted to prove that two sounds sounded the same would be a test with a negative hypothesis, and everybody who knows scratch about science, logic, reason or just rhetoric knows that negative hypothesis are difficult or impossible to prove.


----------



## miceblue

So let's just assume ABX tests are super bad for everything, as *FFBookman* suggests. What other alternative is there? People opposed to ABX tests never mention any other method....


Anyway, I just laughed out loud at this.
Michael Jackson's "Xscape" album
CD master:
http://dr.loudness-war.info/album/view/84564
DR Min: 9
DR Max: 12
DR Avg: 10

HD Tracks master:
http://dr.loudness-war.info/album/view/64734
DR Min: 3
DR Max: 9
DR Avg: 6

Mastered for iTunes master:
http://dr.loudness-war.info/album/view/64388
DR Min: 5
DR Max: 9
DR Avg: 7



What did they do for those other masters?


----------



## StanD

miceblue said:


> So let's just assume ABX tests are super bad for everything, as *FFBookman* suggests. What other alternative is there? People opposed to ABX tests never mention any other method....


 
 Fantasy Audio.


----------



## RRod

miceblue said:


> What did they do for those other masters?


 
  
 When in doubt, compress!


----------



## RRod

jcx said:


> many don't believe that the polarity/"phase" of a musical signal makes an audible difference
> 
> ...
> 
> ...


 
  
 Thanks; back when bigshot was still around 
	

	
	
		
		

		
		
	


	




, and I see Winer was on here too at one point.


----------



## interpolate

I find people it funny how much time and attention spent on disproving things rather than adding to or improving the understanding of it. In the first case it's the absolute opposite of the scientific benefits of the research in the first place.


----------



## arnyk

interpolate said:


> I find people it funny how much time and attention spent on disproving things rather than adding to or improving the understanding of it. In the first case it's the absolute opposite of the scientific benefits of the research in the first place.


 
  
 In many ways, proving and disproving are two sides of the same coin.
  
 It is important to disprove false claims and support correct ones. Many times, doing one implies the other.


----------



## interpolate

Quod Erat Demonstrandum.


----------



## miceblue

interpolate said:


> I find people it funny how much time and attention spent on disproving things rather than adding to or improving the understanding of it. In the first case it's the absolute opposite of the scientific benefits of the research in the first place.



The 24-bit audio myth helped me understand what 24-bit audio actually means rather than just being an uninformed consumer and falling for the "HD audio gives you xxx times more information than a CD!" marketing. 

I have yet to see someone prove that 24-bit audio provides any audible benefits to the end-listener. And no, 999999 kbps does not mean it sounds better to our human's finite resolution of hearing than 1000 kbps. Prove it if you're so certain about that because there is overwhelming evidence against that claim.

The masters make far more of a difference in sound than resolution, far more, and I can pass ABX tests between many masters, at the same resolution.


----------



## interpolate

Yes that's true However i consider sampling frequency to be separate from the audible range. If something is recorded high and kept at the same rate then there is more chance of it working as a studio master than at a dithered and resampled size. How much benefit can be acheived is another story. As for bit-depth that really only counts during the recording process not afterwards really.

I have 2 albums which are 24 96 and to be honest, they sound really good although if i ran pow\r 2 dither on them and resampled to 44.1k i probably wouldn't notice much difference just from listening.


----------



## arnyk

interpolate said:


> Quod Erat Demonstrandum.


 
  
  
 That would be excellent advice, provided the means of demonstration is itself not full of falsehoods. Casual sighted audiophile evaluations are the usual demonstrations offered but they fail to be reasonable due to their vast array of inherent failings and sources of falsehoods.


----------



## arnyk

interpolate said:


> I have 2 albums which are 24 96 and to be honest, they sound really good although if i ran pow\r 2 dither on them and resampled to 44.1k i probably wouldn't notice much difference just from listening.


 
  
 That appears to be an unbelievably broad interpretation of highly limited evidence.
  
 I've had excellent resources at my disposal for making 24/96 recordings since about Y2k, and even more excellent resources for making 24/192 recordings since a few years after that.
  
 I've made some very high resolution recordings and along with other volunteers that I have gathered along the way done extensive listening tests to find if there were any relevant audible differences.  When the listening tests are done right, there are no audible differences.


----------



## interpolate

Nobody said there is an audible difference although the data from higher resolution recordings will be different.


----------



## StanD

interpolate said:


> Nobody said there is an audible difference although the data from higher resolution recordings will be different.


 
 But not different to a human being's senses, so there's nothing to get excited about.


----------



## arnyk

interpolate said:


> Nobody said there is an audible difference although the data from higher resolution recordings will be different.


 
  
  
 Please explain that in the light of this post, for example:
  
 "Wouldn't that be good - a portable SACD player? Bloody expensive yet pristine audio at the same time. I really struggle to find DSD downloads which aren't classical because it's either some classical/jazz thing or it isn't licensed. At least here in the UK, it seems to be very limited at the moment.   I see that Sony are now putting a proper effort into promoting higher-resolution music even if the advertising is misleading to a certain extent. Companies like A&K, FiiO.."


----------



## interpolate

Well doesn't SACD sound better than CD?  If it didn't then everyone who spent hundred and thousands on equipment must be ill-judging imbeciles then. Sony's approach to promoting hi-fi is more gimmicky and non-techy which is fine for some. I suppose my point is you get out what you put in or very much approximately a good proportion of what you do. 
  
 I'm going try recording some vinyl at the weekend into my laptop through a Focusrite Scarlett 2i4 and Phono amp. It's a vinyl version of the KLF - White Room album. Not to make any real comparisons however I don't see it available to buy except from sites like Discogs.com.


----------



## arnyk

interpolate said:


> Well doesn't SACD sound better than CD?
> 
> 
> If it didn't then everyone who spent hundred and thousands on equipment must be ill-judging imbeciles then. Sony's approach to promoting hi-fi is more gimmicky and non-techy which is fine for some. I suppose my point is you get out what you put in or very much approximately a good proportion of what you do.


 
  
 No SACD doesn't necessarily sound better than CD, and here is proof: About half of the SACDs sold before 2006 were mastered from sources that were CD quality _*or worse*_. Nobody noticed until some techies did FFT analysis of the SACDs and found out the truth.


----------



## nick_charles

interpolate said:


> Well doesn't SACD sound better than CD?  If it didn't then everyone who spent hundred and thousands on equipment must be ill-judging imbeciles then. Sony's approach to promoting hi-fi is more gimmicky and non-techy which is fine for some. I suppose my point is you get out what you put in or very much approximately a good proportion of what you do.


 
  
 If an SACD of a specific "Album" sounds "better" than a specific CD version of the same then the most likely reason by a sizable margin would be differences in mastering, closely followed by differences in level matching and then poorly controlled listening tests. To date there is genuinely almost zero evidence for the potential technical advantages of SACD being audible in normal listening...


----------



## interpolate

OK fair enough, well until I get a SACD player I will remain an ignorant outsider.


----------



## interpolate

It makes me want to throw all my high-res gear in the bucket


----------



## arnyk

interpolate said:


> OK fair enough, well until I get a SACD player I will remain an ignorant outsider.


 
  
 I'm under the impression that a great deal of so-called hi-rez tracks are downloaded these days.


----------



## StanD

interpolate said:


> It makes me want to throw all my high-res gear in the bucket


 
 I'd recommend the FS forum, plenty of audiophiles ready for you to unload the kit on.


----------



## interpolate

Yeh maybe but I'm keeping the Unicorn and Fairy Dust.


----------



## miceblue

SACD's masters tend to sound better than the CD counterpart, but not always. For the vast majority of those masters, they're probably something like 24/96, or lower, converted to 1 bit/2.8 MHz.


----------



## interpolate

I converted some WAV 24/96 to DSD format (DSF) and you know the difference is indecipherable other than the processing power to output a signal. DoP tends to settle at 174K for some reason with jRiver Media Player.


----------



## miceblue

Decided to try this out. I just got a bunch of Michael Jackson's albums on CD and I wanted to see how they sounded differently from the HD masters.

"Butterflies" from _Invincible_
24/96 HD


CD


24/96 converted to 24/44.1 (-r 44100 -o "filename".flac via SoX)




Spoiler: abx log



Pretty close. I think I could pass it if I did it again. The differences are not nearly as large as other HD vs CD masters I've heard.

foo_abx 2.0.1 report
foobar2000 v1.3.8
2015-07-21 22:59:55

File A: 07 Butterflies-CD.flac
SHA1: bdae6fd77dc7540ec35d1e48b68565b10848e54f
Gain adjustment: -9.85 dB
File B: 07 Butterflies-HD-441.flac
SHA1: 790dbd114cffe43520335ce39cff258b0fa27ba5
Gain adjustment: -9.09 dB

Output:
DS : Geek Pulse X Infinity 1V5 Output, 16-bit
Crossfading: NO

22:59:55 : Test started.
23:02:31 : 00/01
23:03:22 : 01/02
23:03:50 : 02/03
23:05:03 : 03/04
23:05:21 : 04/05
23:05:35 : 04/06
23:05:48 : 05/07
23:06:24 : 06/08
23:06:35 : 07/09
23:06:48 : 08/10
23:06:48 : Test finished.

 ---------- 
Total: 8/10
Probability that you were guessing: 5.5%

 -- signature -- 
8ce3f60e31b3ca2e52382cd507aa34b932f28ba5



24/96 converted to 16/44.1 (-r 44100 -b 16 -o "filename".flac via SoX)




Spoiler: HD to 16-44.1 vs 24-44.1 abx log



Myth debunked; 24-bit audio makes no audible difference; 928 kbps average sounded just as good as 1632 kbps average. I was just guessing for these tests since I couldn't even tell A apart from B.

foo_abx 2.0.1 report
foobar2000 v1.3.8
2015-07-21 23:30:27

File A: 07 Butterflies-HD-CD.flac
SHA1: 00557c6ebe4256509eaab281d5af5b8d7454f238
File B: 07 Butterflies-HD-441.flac
SHA1: 790dbd114cffe43520335ce39cff258b0fa27ba5

Output:
DS : Geek Pulse X Infinity 1V5 Output, 16-bit
Crossfading: NO

23:30:27 : Test started.
23:33:26 : 00/01
23:37:31 : 00/02
23:38:06 : 00/03
23:38:45 : 01/04
23:39:13 : 02/05
23:39:34 : 02/06
23:40:07 : 03/07
23:40:28 : 03/08
23:40:38 : 03/09
23:40:45 : 03/10
23:40:45 : Test finished.

 ---------- 
Total: 3/10
Probability that you were guessing: 94.5%

 -- signature -- 
e6a89011212e5129d89489bc68e6189151d7926b








"Smooth Criminal" from _Bad_ (HD) and _Bad [2001 Special Edition]_ (CD)
24/48 HD



CD


24/48 converted to 24/44.1 (-r 44100 -o "filename".flac via SoX)




Spoiler: abx log



Flawed. The timing between the two files is different, as evident by the spectrograms too, and it's easy to pick them out.

foo_abx 2.0.1 report
foobar2000 v1.3.8
2015-07-21 23:19:34

File A: 10 Smooth Criminal-CD.flac
SHA1: 6f6dc6adb56a17fdb1672643bf0be14e2778e004
Gain adjustment: -8.76 dB
File B: 10 Smooth Criminal-HD-441.flac
SHA1: 9d8e137cbe0d27369d0442125aea611d471e5ace
Gain adjustment: -9.02 dB

Output:
DS : Geek Pulse X Infinity 1V5 Output, 16-bit
Crossfading: NO

23:19:34 : Test started.
23:22:31 : 01/01
23:22:42 : 02/02
23:22:53 : 03/03
23:23:06 : 04/04
23:23:12 : 05/05
23:23:18 : 06/06
23:23:26 : 07/07
23:23:31 : 08/08
23:23:39 : 09/09
23:24:35 : 10/10
23:24:35 : Test finished.

 ---------- 
Total: 10/10
Probability that you were guessing: 0.1%

 -- signature -- 
b4208571ae58ca867651b80e7cdbf9f99c1ddc58


----------



## RRod

miceblue said:


> Decided to try this out. I just got a bunch of Michael Jackson's albums on CD and I wanted to see how they sounded differently from the HD masters.


 
  
 Something like DiffMaker should be able to align the files or even compensate for minor time expansion/contraction, if you wanted to have another go at the last comparison. For the 8/10 case, it would be interesting to see the null/difference file to see what the mastering differences are.


----------



## interpolate

I should be recording some vinyl soon,so I may upload a blind test for you people. No graphs, just ears.
  
 Providing I can get the thing rigged up.
  
 It will be The KLF - White Room recorded at 24/48, so I can save space and processing time. Some other vinyl like older Pink Floyd, Jazz maybe..who knows. There's a lot to choose from.


----------



## Gr8Desire

interpolate said:


> I should be recording some vinyl soon,so I may upload a blind test for you people. No graphs, just ears.
> 
> Providing I can get the thing rigged up.
> 
> It will be The KLF - White Room recorded at 24/48, so I can save space and processing time. Some other vinyl like older Pink Floyd, Jazz maybe..who knows. There's a lot to choose from.


 
  
 This _proof_ will be amusing to some of us.  When we did recordings in the 1980's we learned a thing or two about analog recordings.  We did digital recording as early as 1982 - and universally liked the results - but didn't release anything mainly due to obscene cost of multitrack digital rigs. So multitrack-tape-to-vinyl remained the default until about 1990. 

 Here are some of the realities that seem to have been lost in the intervening years:
  

The dynamic range of vinyl was anemic.  A high noise floor combined with the fact that the stylus would not stay in the grove at high volumes meant that we compressed EVERYTHING more than anyone wanted. You absolutely needed something like dbX expansion to restore proper sound levels at playback. I don't see much expansion these days. But you audiophiles don't even use EQ which is funny to me since EVERY recording had extensive amounts of EQ to deal with limitations of vinyl and acoustic recording issues.  
  
Tape was the best medium for playback. Whenever possible we would scoop an original recording on reel-to-reel. We'd take it home and know we had a recording superior to vinyl.  None of us wanted vinyl except for maybe drink coasters.
  
We knew tape degraded over time. Not only did the mylar substrate get brittle and break, you were always getting magnetic remanence decay from recorded tapes sitting in a roll. This means even 'remastered' recordings are now suspect.  That's too bad.  If we used more digital equipment in the 1980's, we would have been better off. 
  
  
 Enjoy your vinyl - _we never did._


----------



## Gorquin

I'm not an engineer, HiFi buff, nor do I have a multi-thousand dollar system but, IMO, a LOT of what sounds good on Vinyl, Tape, CD or otherwise is because of what goes on in the initial recording process.
  
 Like the music or not Steely Dan's "Asia" was recorded in 1977 and it still "sounds" good on vinyl or CD. It's a great recording. 
  
 I haven't read the article yet but I have read others that stated 24/96 was the optimum because the 24 bit gave one more head room for recording transients and the 96Khz sampling rate cut down on the duration of minor imperfections where as above that, at say 192Khz, the listen couldn't perceive any difference in SQ.
  
 I'll give the article a read.


----------



## interpolate

I'm not trying to prove anything. Sod it, i'll just record the vinyl for my own personal pleasure then. It's because it's there really no other reason.


----------



## Gorquin

I read the article. Like I said, I'm not an engineer but I was a audio tech and business owner. The article makes a lot of sense. So, as stated or implied in other articles as well. Both Dither and component quality in recording may have more of an impact on SQ than higher bit or sampling rates.


----------



## Gr8Desire

gorquin said:


> I read the article. Like I said, I'm not an engineer but I was a audio tech and business owner. The article makes a lot of sense. So, as stated or implied in other articles as well. Both Dither and component quality in recording may have more of an impact on SQ than higher bit or sampling rates.


 

 FWIW: Dithering with respect to different time bases is not an issue if the source sample rate is an integral multiple or divisor of the playback rate. For example: Source @ 176.4 will downsample fine to 88.2 or  44.1 without dithering. The interpolation of missing bits is not destructive. Look for these multiples if you are interested in hi-res content. Higher sample rates don't always mean better sound if your DAC does not directly support a sample rate or an integral divisor.  For example: iPhone DACs are decent but iOS do not support much beyond 44.1/16 - yet.  96KHz and 192 KHz content will not generally downsample well as 88.2 KHz and 176.4 KHz.  

 Also, variation in bit depth never causes dithering. It can result in minor dynamic compression depending the loudness range being represented in source and playback. 

 Here is a good summary of the issues (Hint: Don't read if you believe in magic):

 http://bitperfectsound.blogspot.ca/2015/05/how-does-sample-rate-conversion-work.html


----------



## arnyk

gr8desire said:


> FWIW: Dithering with respect to different time bases is not an issue if the source sample rate is an integral multiple or divisor of the playback rate. For example: Source @ 176.4 will downsample fine to 88.2 or  44.1 without dithering. The interpolation of missing bits is not destructive. Look for these multiples if you are interested in hi-res content. Higher sample rates don't always mean better sound if your DAC does not directly support a sample rate or an integral divisor.  For example: iPhone DACs are decent but iOS do not support much beyond 44.1/16 - yet.  96KHz and 192 KHz content will not generally downsample well as 88.2 KHz and 176.4 KHz.
> 
> Also, variation in bit depth never causes dithering. It can result in minor dynamic compression depending the loudness range being represented in source and playback.
> 
> ...


 
  
 If the data word length doesn't decrease, the existing dither suffices even though the sample rate changes.
  
 The purpose of dither is to randomize the quantization error, and while changing the sample rate changes the bandwidth of the quantization error, it doesn't change its randomness.


----------



## interpolate

@arnyk If I could only borrow your brain one night for a pub quiz.


----------



## Gorquin

Thanks for the article. It does make sense.


----------



## interpolate

> This _proof_ will be amusing to some of us.  When we did recordings in the 1980's we learned a thing or two about analog recordings.  We did digital recording as early as 1982 - and universally liked the results - but didn't release anything mainly due to obscene cost of multitrack digital rigs. So multitrack-tape-to-vinyl remained the default until about 1990.


 
  
 You know what, your comment is a bit hollow, self-assured and a little aloof. This is the 2010's now and equipment with their other related technologies has moved on. Even prosumer equipment is far more suited to studio tasks, so you really need pull your head in a bit. Not everything you witnessed and believe is always the emperical answer and the same null. The same goes for me and whether you are a top sound engineer or 10 album producer doesn't matter to me.


----------



## hogger129

I would only say anything past CD quality (16-bits, 44.1khz, 1411.2kbps) is pointless since you won't hear a difference.  Some people say they do, but all the ones I've talked to will not perform a blind test and prove it.  I've heard something about filtering that can be heard with 44.1khz sampling rates, but I can't hear it.  Can you guys?  What's the point of 48khz if the ceiling of human hearing is around 20khz.


----------



## interpolate

48Khz is the frequency used to synchronise movie and other MTC commands in a lot of older MIDI equipment and other platforms. So whilst it has no direct correlation on our hearing it does have mathematical benefits.
  
 People get to obsessed with small details; what happened to just enjoying music?


----------



## StanD

interpolate said:


> 48Khz is the frequency used to synchronise movie and other MTC commands in a lot of older MIDI equipment and other platforms. So whilst it has no direct correlation on our hearing it does have mathematical benefits.
> 
> People get to obsessed with small details; what happened to just enjoying music?


 
 It appears for may that enjoying music takes back sea to shooting some BS with the flash mob of audiophiles and getting some attention. Why let the truth get in the way with BS? Hence Hires, DSD, etc.


----------



## fiascogarcia

stand said:


> It appears for may that enjoying music takes back sea to shooting some BS with the flash mob of audiophiles and getting some attention. Why let the truth get in the way with BS? Hence Hires, DSD, etc.


 

 Agreed!


----------



## Gr8Desire

interpolate said:


> You know what, your comment is a bit hollow, self-assured and a little aloof. This is the 2010's now and equipment with their other related technologies has moved on. Even prosumer equipment is far more suited to studio tasks, so you really need pull your head in a bit. Not everything you witnessed and believe is always the emperical answer and the same null. The same goes for me and whether you are a top sound engineer or 10 album producer doesn't matter to me.


 
  
_Did you get up on the wrong side of the bed when you felt compelled to reply to an obscure 3 week old posting? _

 I gave specific examples where vinyl recording technology was - and still is - lacking. You conveniently snipped that part.

 If you had understood what you read, you might have realized that I was indeed implying that the recording equipment I use today is many times better than anything we used 35 years ago. _Back in the day:_ We just thought very little of vinyl.  The fact that some of you have now chosen vinyl is certainly interesting. I can't wait to see what happens when you _discover_ analog tape.

 If I didn't think you were an adolescent troll looking for attention, I would probably add a few more examples.  For now, I will simply admit you are completely right and that I expect to learn a great deal from you.


----------



## jonstatt

gr8desire said:


> Enjoy your vinyl - _we never did._


 
  
  
 Who is we?! I think analogue focused mastering engineers like Steve Hoffman would strongly disagree with that statement. In fact I posted a link earlier in this thread where Steve mastered as an experiment the same source tape to vinyl and CD and SACD and found a strong preference for the vinyl version being closest to the original master. But while there will always be some who dismiss this, Steve is not alone in a preference for maintaining a pure analogue path from source to end product media. Of course Analogue Productions tend to specialise in this and used the well known skills of mastering engineers like the late Doug Sax.
  
 However, whether you agree or disagree, hear or don't hear the benefits of a pure analogue end to end path.....there is also the fact that the dynamic range capability of a CD is totally irrelevant for (at a guess) at least 90% of modern music releases. This is because of the loudness wars and the fact that a lot of modern music is squashed into 5 or 6dB of dynamic range. Even cheap turntable cartridge set-ups can cope with more dynamic range than that! In many instances the output is brickwall limited (hard limiter to keep the output near 0dB output on a CD). Of course this cannot be done with vinyl, so there are plenty examples of vinyl versions where the music has a bit more room to breath. In fact it is because the format is more limited, that perversely it seems to gain from some of the severe ruination that takes place on the digital media versions (as a result a lot of modern releases I find exhausting to listen to). Of course there are also examples of vinyl versions where it is the same as the CD just recorded quieter!! 
  
 While I do remain totally open minded on whether there is any sonic benefit to hi-res digital media versions vs CD, it is pretty clear those benefits don't come from the dynamic range capability as no music even uses the max DR a CD is already capable of.


----------



## interpolate

Well that got a reaction.....
  
 I'm just sick of people mixing up sample rate and audible frequency ranges and adding ABX test graphs. Of course, we can't hear anything beyond the average range. Unless you are Superman or some fictional Marvel hero.
  
 When I listened to vinyl the overall sound just never sounded so clinical. Hi-Res was meant to bridge this gap  however it's caused hours and pages of needless arguments about inaudible infrasonic sound.


----------



## interpolate

Digital is great really..binary logic and of that. However some randomness makes life more interesting, no?


----------



## interpolate

@Gr8Desire I don't think I was quoting you.


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## icebear

As a replay medium the redbook format offers totally sufficient potential.
  
 Take a look or better listen at some JVC xrcd releases e.g. Art Pepper "Live in Tokyo '79" or Sarah Vaughan "How long has this been going on". To me this is absolutley as close to live as it gets. Yes, sure there is totally crappy stuff out there on CD but that is someone's particular responsibility because the producer or whoever had the say, decided it that way. It is NOT the fault of the format itself.
  
 Using higher rez equipment for recording, mixing and mastering and then converting the end result to 16/44 makes total sense. When you work on a picture in LR or Photoshop for best results you start with the original DNG file of max. resolution and bit depth, not with an already compressed JPG. You will always loose some information along the way, so start with the best you can possibly get. If all the care has been taken in all steps along the way then surprisingly little is lost ...JVC K2HD examples.
  
 As a 35 year old format (CD) of course marketing want to get rid of it and instill the latest and greatest high rez as waaay better, just to sell something new. They have sold quite a few DACs, didn't they 
	

	
	
		
		

		
			





?  And of course all examples picked will show that it sounds better in presentations. Getting the playing field level and down sampling the identical high rez file to lower rez formats doesn't lead to easy to pick differences, how come 
	

	
	
		
		

		
		
	


	




?
  
 Nevertheless I immensely enjoy cheap classic and jazz SACD's that you can pick up for small money as the format is "dead" in terms of consumer perception. So nobody wants them anymore. Of course the remastering was done with great care because at the time they wanted the new format to sound better. Analogue Production or MOFI releases are a real joy. Yes, I am old fashioned, I like to have a disc that I can listen too when my internet service is not working


----------



## Gr8Desire

interpolate said:


> @Gr8Desire I don't think I was quoting you.


 

 Don't mention it.

 I wouldn't give you even a moment of consideration.


----------



## interpolate

@Gr8Desire Very droll and not very necessary. At least I know where we stand now.


----------



## kar13

I dont want stir the hornet's nest, but can't resist the action..
	

	
	
		
		

		
		
	


	



 despite all the arguments against High res audio, at the end of the day only
 A/B ing the recordings matters.Listen to Take five(Dave Bruebeck) in 24/96 and 16 bit flac.
 The noise floor is drastically less and the dynamic range pounces on you like a tiger.I have A/B ed these 2 tracks on systems without any tube amp or desktop amps or dacs and still could identify the versions without guessing, 100% of the times adjusted for the individual gains.
 Thats why listening experience is and will always remain the king
	

	
	
		
		

		
		
	


	




 and rest of any arguments based on engineering is feckless.
 On other hand badly mastered recordings like LP of Aerosmith's Sweet emotion have much of distortion and noise) which is also audible.
 Thus, even though I agree with them logically and technically, I dont quite agree with the article based on my experience.


----------



## earthpeople

kar13 said:


> I dont want stir the hornet's nest, but can't resist the action..
> 
> 
> 
> ...


 
  
 What about when you take your 24/96 and convert it to 16/44? Make sure your lower res copy is created from your higher res version and compare those two.


----------



## icebear

kar13 said:


> ... A/B ing the recordings matters.*Listen to Take five(Dave Bruebeck) in 24/96 and 16 bit flac.*
> The noise floor is drastically less and the dynamic range pounces on you like a tiger...


 
 Do you have detailed information about the origin of both versions that sound so obviously different ?
 Maybe the 16 bit flac is a CD release version from the 80's and the 24/96 is a less than 5 year old remaster?
 Just guessing of course


----------



## cjl

Quote:


kar13 said:


> I dont want stir the hornet's nest, but can't resist the action..
> 
> 
> 
> ...


  
 If the dynamic range is different between the two versions, the master is different, and you aren't really just comparing the formats. Also, if you can hear the noise floor on the 16 bit recording, it's either noise from the original recording (tape hiss?), or it was mastered at far too low a level (like -40 or -50dBFS). It's also possible that your playback system has appallingly bad noise when playing 16 bit. For a fair comparison, take the 24 bit version and downconvert it to 16/44, then upconvert it back to 24/96, then compare those two.


----------



## KeithEmo

earthpeople said:


> What about when you take your 24/96 and convert it to 16/44? Make sure your lower res copy is created from your higher res version and compare those two.


 
  
 Unfortunately, whenever you convert a file, there is filtering involved, which introduces some (very slight) changes to the sound. If you don't hear a difference between the 24/96k original and the converted 16/44 version, then this will prove that neither difference is significant (in your test rig, with your sample content, and through your ears). But, if you do hear a slight difference, you won't know for sure how much of it is due to the difference in sample rate or bit depth, and how much is due to slight alterations introduced in the conversion process itself.
  
 It's also fair to remind everyone that we are talking about subtle differences here... subtle to the point where they may only be audible at all with certain source material, or with certain speakers or headphones, and maybe not all of us can hear them at all... however that doesn't rule out the fact that they may be audible and significant under some circumstances. (Either way, they may well be less major than differences between speakers or headphones.)
  
 Another post suggested that the differences may be due to other things - perhaps because a different master was used. While this is certainly true, I think its importance _to end users_ is being overemphasized. It may matter a great deal _to a music producer, or a streaming service_, whether a high-def download really sounds better _because_ it's high-def or because it was remastered. However, to the person buying and listening to it, all that matters is that the "high-def remaster" does in fact sound better. Most of the current crop of high-def reissues have been remastered, often in a way that is significantly better than the original CD version... in which case it's worth buying because of the better remastering. (I might even suggest that, once we've established that the 24/192 remaster of Album X sounds better than the CD version, we can also assume that the "master copy" of the remaster was in fact done at 24/192. Therefore, even if the sample rate itself doesn't make a significant difference, the 24/192 version will be "a copy of the master" while the 16/44 "CD quality" version will have been converted form that new master - and so will be "one generation out - and possibly slightly different". In that situation, even if we were to agree that the fact that one is at 24/192 didn't actually matter, it would still make sense to buy the 24/192 version that was a direct 1:1 copy of the new master, rather than the 16/44 version which had been converted from it.)
  
 If you accept that many remasters sound better than the original - for whatever reason - then it simply doesn't make sense to agonize over the differences between a 16/44 version and a 24/192k version (the price difference is usually small, and the cost of storage has gotten so low that the size of the file itself simply doesn't matter all that much). A lot of people bought a lot of music in AAC or MP3 format, only to find out later that the difference was obvious on "their new stereo", and end up being disappointed, or end up spending a lot of money buying their collection all over again. I'd rather spend a few $$$ more and buy the best quality version that's available when I make my purchase, and so minimize the risk of having to buy it again later.
  
 (I think it's kind of cool to be able to buy an actual copy of the 24/192k remaster; I used to just hate buying a vinyl album... and knowing that what I had in my hand wasn't nearly as good as the original version that was recorded on the master tape. Even if, in a particular case, I don't notice a difference, It makes me feel better to know that there isn't a better quality copy out there "that I'm missing".)


----------



## Gr8Desire

kar13 said:


> I dont want stir the hornet's nest, but can't resist the action..
> 
> 
> 
> ...


 

 You are not stirring up a hornet's nest. You have made a terrific observation.  Unfortunately, you also made a critical mistake.

96/24 is NOT an integral multiple of the sampling rate of 44.1/16.  As such, you get dithering artifacts inserted when the content is down-sampled for 44.1 MHz playback.
  
 Find an *88.2*/24 recording and it will sound much better at 44.1 MHz because no dithering artifacts will be added.

 Dithering is affecting the dynamic range and - as you say - the 'impact' at 44.1 MHz playback is also diminished. _I am not surprised. _Not only is the noise floor potentially higher, but you have more distortion across the entire audio spectrum. 24 bits of dynamic range details could help marginally but it can't fix timebase resampling artifacts.

 Your comment is _even more interesting_ because I think the Hi-Res version *Brubeck's Timeout is **the BEST album I own!!!*  It was remastered in 2013 to *176.4*/24. It sounds just as good at *44.1*/16 as *176.4*/24 because the sampling timebase is a proper integer multiple. Moreover, I contend: It is not the sampling rate or bit depth that makes this recording stand out. It is the 2013 remix that did the job.


----------



## icebear

> Originally Posted by *KeithEmo* /img/forum/go_quote.gif
> ....
> Therefore, even if the sample rate itself doesn't make a significant difference, the 24/192 version will be "a copy of the master" while the 16/44 "CD quality" version will have been converted form that new master - and so will be "one generation out - and possibly slightly different". In that situation, even if we were to agree that the fact that one is at 24/192 didn't actually matter, *it would still make sense to buy the 24/192 version that was a direct 1:1 copy of the new master, rather than the 16/44 version which had been converted from it.)...*


 
 Usually most of your walls of text 
	

	
	
		
		

		
		
	


	




 do make some sense...
 BUT arguing about differences of digital formatting/resampling or generations of digital copies that originate from the exact same source file ... nope, lost me there 
	

	
	
		
		

		
		
	


	




.


----------



## kraken2109

kar13 said:


> I dont want stir the hornet's nest, but can't resist the action..
> 
> 
> 
> ...


 

 You're probably A/B testing different masters.


----------



## interpolate

I bought Mark Ronson - Uptown Funk in 88.2K/24 FLAC format it's nice and clean, loud without any mid-frequency issue. Whether the 44.1K CD version sounds the same is irrelevant to me.


----------



## KeithEmo

icebear said:


> Usually most of your walls of text
> 
> 
> 
> ...


 
  
 It's really pretty simple.
  
 We're NOT talking about generations in terms of copies not being the same as originals. Of course, since it's just bits we're talking about, a copy will be identical to the original (if you don't do something wrong). The same is true if you convert from one lossless format to another - no change. However, when you convert a digital audio file to a different sample rate, new numbers must be calculated, and part of the process involves digital filtering. It is _NOT_ a totally lossless conversion. Therefore, whenever you convert one sample rate to another, you slightly alter the content. (For example, if you convert a 16/44k file to 16/96k, then convert the 16/96k file back to 16/44k, you might expect to end up with exactly what you started with, but you won't.) Some conversion programs alter the signal more than others, and some offer several different options in terns of filtering, but none is "absolutely perfect". Therefore, if you have a file that was actually mastered at 24/192k, and you buy a copy at 24/192k, your copy should be identical to the original. However, if you convert it to 16/44k, even excluding any possible difference because of the different sample rate, you will also have differences due to the artifacts of the conversion process. (And so, if what you're buying was mastered at 24/192k, a 24/192k copy should be identical to the original, but, even excluding any possible audible difference because of the different sample rate itself, the 16/44k version will have been altered slightly by the conversion process itself.)
  
 An easy to demonstrate this for yourself is to start with a 24/96k "master file" and convert it to 16/44 using several different "high end" programs; you will find that the results are similar, but not identical (and sometimes the differences are audible).


----------



## icebear

Computer audio has one significant disadvantage : your have to use a computer 
	

	
	
		
		

		
		
	


	



 If you are a PC wizard or like to deal with all the necessary trouble shooting to get things running smoothly, then be my guest. I like to listen to music. I press play on my old fashioned disc player and enjoy 16/44 and some SACD's. If people feel the high rez formats are the best thing since the invention of electricity, then again be my guest.
  
 I'm also into photography a little bit and you have a similar phenomenon there, the pixel and the ISO race. More and more MP, flagships are around 50MP and ISO up to 125,000 and no new camera has less than 20MP and highest ISO below 12,500. And does it really matter? If you blow up the files big enough, yeah you can proof a there is a difference, as always "moar is bettar" ... on the spec sheet
	

	
	
		
		

		
		
	


	




. In real world application or viewing size, a little more than 10MP give excellent results. And yeah you can take pictures in complete darkness, only choosing the subject and framing might be an issue.
	

	
	
		
		

		
		
	


	



 Why have the TV's grown to a size of half the living room wall? Because with a reg. 46 inch screen you won't recognize any difference. Of course 4k resolution makes it possible to identify the sort of bug on the leaf of grass on a football field. Does it make the game more interesting?
  
 Back to audio ... there is no equivalent option to increase the screen size. Just positioning the speakers further apart is not really going to help 
	

	
	
		
		

		
		
	


	




 And since this is head-fi we're screwed anyway. With all the splitting hairs over formats - I just have to listen to some Mercury living presence recording, made with 3 microphones and 35mm magnetic tape more than 50 years ago and I realize that the ability of the recording engineer to listen to the space of the concert hall, pick the right microphones for a specific task and position them in the best possible location - THIS will make a fascinating recording and an artistic and audiophile document.
  
 And btw I think for a processing chip in a computer it doesn't really make a difference if the multiplication / division is by 2.000000000 , 4.000000000, 8.000000000 or by 1.89512345678. It's only humans who like even numbers as our brain handles them easier. I'm certain a chip doesn't care. Yes, theoretically there might be differences in the 10th digit when you shovel the numbers back and forth. On a 80 inch screen you might see it, on a 46 inch, who cares? Ooops wrong medium again


----------



## DreamKing

"Computer audio" isn't any harder than pressing play for music. It's a weak task. It does essentially the same thing as your disc player, which is also a computer funnily enough. Press play and enjoy anything computer made (which accounts for nearly all of the music industry's output). It's simply audio.


----------



## arnyk

kar13 said:


> I dont want stir the hornet's nest, but can't resist the action..
> 
> 
> 
> ...


 
  
 Please do tell about these alleged listening experiences. Were they carefully controlled comparisons with adequate quality controls or were they the usual audiophile casual sighted evaluations?
  
 The usual casual sighted audiophile evaluations are utter junk when it comes to reliable tests of sound quality.
  
 They usually suffer from one or more of the following fatal problems, usually all of them. Any of these problems is fatal to the credibility of the evaluation for the reasons stated:
  
 (1) Audiophile Casual Sighted evaluations are not reliable evidence because they are not even tests. That is, they do not involve comparison to a fixed, reliable standard.
  
 (2) Audiophile Casual Sighted Evaluations are not  reliable evidence because they involve excessively long switchover times, which makes them highly susceptible to false negatives because they desensitize the listeners.
  
 (3) Audiophile Casual Sighted Evaluations are not reliable evidence because the do not involve proper level matching, which makes them highly susceptible to false positives because people report the level mismatches as sonic differences.
  
 (4) Audiophile Casual Sighted Evaluations are  reliable evidence because they do not involve listening to the identical same piece of music or drama within a few milliseconds, creating false positives because people report the mismatched music as sonic differences in the equipment.
  
 (5) Audiophile Sighted Casual Evaluations are not  reliable evidence because they constantly reveal the true identity of the UUTs to the listener, creating false positives because people report their prejudices and preconceived notions as sonic properties of the equipment.


----------



## dazzerfong

icebear said:


> Computer audio has one significant disadvantage : your have to use a computer
> 
> 
> 
> ...


 
 But unlike audio, taking pictures with high pixel counts may be useful when you want to crop pictures. Having more ISO is better _provided that the noise level is below acceptable levels._ The photography argument doesn't really work with audio TBH.......................................


----------



## icebear

dazzerfong said:


> But unlike audio, taking pictures with *high pixel counts may be useful when you want to crop pictures. *Having more ISO is better _provided that the noise level is below acceptable levels._ The photography argument doesn't really work with audio TBH.......................................


 
  
 ... see great specs work, there is always some reason that this necessary, at least in some cases 
	

	
	
		
		

		
		
	


	




.
 It is just an example of bigger numbers are used to sell something.
 Marketing convinces most consumers that this is sooo much better and a lot of folks buy first the argument and then the product.


----------



## kar13

earthpeople said:


> What about when you take your 24/96 and convert it to 16/44? Make sure your lower res copy is created from your higher res version and compare those two.


 

 I have tried and haven't found any difference between 16/44 cd and 16/44 from 24/96.
  
 But i have a question..
 is the effect of different downsampling algorithms used noticeable to a discerning ear..because of dithering?


----------



## StanD

kar13 said:


> I have tried and haven't found any difference between 16/44 cd and 16/44 from 24/96.
> 
> But i have a question..
> is the effect of different downsampling algorithms used noticeable to a discerning ear..because of dithering?


 
 Unlikely due to its insignificant values  it is done by adding noise of a level less than the LSB (Least Significant Bit) before rounding to 16 bits.


----------



## castleofargh

kar13 said:


> I have tried and haven't found any difference between 16/44 cd and 16/44 from 24/96.
> 
> But i have a question..
> is the effect of different downsampling algorithms used noticeable to a discerning ear..because of dithering?


 

 some will tell you yes, some will tell you no. as always it's best to try abx yourself and make your own opinion.
 I have a very hard time discerning stuff -60db under most playing musics, so I had no trouble making my opinion on dither. but that's me.


----------



## kar13

icebear said:


> Do you have detailed information about the origin of both versions that sound so obviously different ?
> Maybe the 16 bit flac is a CD release version from the 80's and the 24/96 is a less than 5 year old remaster?
> Just guessing of course


 

 24/96 is a vinyl lp release from bruebeck's greatest hits..and the 16/44 is from cd version. vinyl is cbs records 1967 release.afaik the vinyl is not a remastered version.


cjl said:


> If the dynamic range is different between the two versions, the master is different, and you aren't really just comparing the formats. Also, if you can hear the noise floor on the 16 bit recording, it's either noise from the original recording (tape hiss?), or it was mastered at far too low a level (like -40 or -50dBFS). It's also possible that your playback system has appallingly bad noise when playing 16 bit. For a fair comparison, take the 24 bit version and downconvert it to 16/44, then upconvert it back to 24/96, then compare those two.


 
  
 I have tried it on pc's with a dac and hence Im sure its not the noise in my system and that is why i dared to post such controversial observations as it baffles me..
  


keithemo said:


> It's really pretty simple.
> 
> We're NOT talking about generations in terms of copies not being the same as originals. Of course, since it's just bits we're talking about, a copy will be identical to the original (if you don't do something wrong). The same is true if you convert from one lossless format to another - no change. However, when you convert a digital audio file to a different sample rate, new numbers must be calculated, and part of the process involves digital filtering. It is _NOT_ a totally lossless conversion. Therefore, whenever you convert one sample rate to another, you slightly alter the content. (For example, if you convert a 16/44k file to 16/96k, then convert the 16/96k file back to 16/44k, you might expect to end up with exactly what you started with, but you won't.) Some conversion programs alter the signal more than others, and some offer several different options in terns of filtering, but none is "absolutely perfect". Therefore, if you have a file that was actually mastered at 24/192k, and you buy a copy at 24/192k, your copy should be identical to the original. However, if you convert it to 16/44k, even excluding any possible difference because of the different sample rate, you will also have differences due to the artifacts of the conversion process. (And so, if what you're buying was mastered at 24/192k, a 24/192k copy should be identical to the original, but, even excluding any possible audible difference because of the different sample rate itself, the 16/44k version will have been altered slightly by the conversion process itself.)
> 
> An easy to demonstrate this for yourself is to start with a 24/96k "master file" and convert it to 16/44 using several different "high end" programs; you will find that the results are similar, but not identical (and sometimes the differences are audible).


 

 thanks for clarifying my doubt in my previous post!


arnyk said:


> Please do tell about these alleged listening experiences. Were they carefully controlled comparisons with adequate quality controls or were they the usual audiophile casual sighted evaluations?
> 
> The usual casual sighted audiophile evaluations are utter junk when it comes to reliable tests of sound quality.
> 
> ...


 

 I use iems on any system where a/b ing is done.
 I mentioned in my post that (1), (3) are already taken care of.
 (2)Over many comparison tests that i have done and continue to do this effect is of course taken care of.
 (4) is overcome as the 2 versions are ripped in random sequences by software and usually burned to a cd.
 The environment is ensured to be quiet, and although I do understand your skepticism,i am not enforcing you to believe in my experience.
 If there is anything else I can do, to prove this as false positive please tell me about it.
  
 I would like to point out that this happens only with Take five and its gotten to the point where I look for sonic cues and differences and can distinguish almost effortlessly..
 I have done similar A/b ing with tracks of edm electronic, and rock genres and have been unsuccessful at similar identification.


interpolate said:


> I bought Mark Ronson - Uptown Funk in 88.2K/24 FLAC format it's nice and clean, loud without any mid-frequency issue. Whether the 44.1K CD version sounds the same is irrelevant to me.


 

 +1 enjoyment of music always rules..


----------



## interpolate

There is a reason most modern mastering limiters support a 60dB dynamic range. The average RMS peak determines how loud the actual recording is and how it's been compressed.


----------



## kar13

castleofargh said:


> some will tell you yes, some will tell you no. as always it's best to try abx yourself and make your own opinion.
> I have a very hard time discerning stuff -60db under most playing musics, so I had no trouble making my opinion on dither. but that's me.


 

 i will have to take this as my next experiment..thanks for your input!


----------



## old tech

Bear in mind this peer reviewed paper of some 10 years standing has not been refuted...

http://drewdaniels.com/audible.pdf


----------



## arnyk

old tech said:


> Bear in mind this peer reviewed paper of some 10 years standing has not been refuted...
> 
> http://drewdaniels.com/audible.pdf


 
  
 Moran, one of the two authors emailed me earlier this year pointing out a recent AES conference paper that some thought constituted a refutation:
  
http://www.realhd-audio.com/?p=3627
  
Mark Waldrep (well known High Rez advocate) wrote:
  
 "I finished the afternoon by attending a few paper sessions. The first was titled, “The Audibility of Typical Digital Audio Filters in a High-Fidelity Playback System”. Although it may not be obvious from the title of the paper, this is the first AES publication that refutes the Meyer/Moran research that has been so often quoted as “proof” that CD specification PCM audio is enough for music reproduction (Meyer and Moran’s research has been widely discredited including by myself because of the lack of real high-resolution content used during the study)."
  
However, that was before the AES community commented on the alleged refutation:
  
https://secure.aes.org/forum/pubs/conventions/?ID=416
  
For example: 
  
"
 The conclusions in the abstract and in the introduction differ markedly from those offered at the end of the paper. They are also not adequately supported by the research presented in the paper.
 The introduction contains a lengthy series of speculations which seem to be preoccupied with casting doubt on some past research, yet the paper essentially fails to substantiate the speculations. This comes dangerously close to being unfair.
 "
 Any lack of real high resolution content used in the M&M study is the fault of the promoters of the SACD and DVD-A formats, as it was recordings that were advertised as being high resolution audio that formed the basis of the Meyer and Moran study.
  
 After the publication of the Meyer and Moran study it became known that about half or more of the SACD and DVD-A formatted recordings were sourced from low resolution masters, and once that resolution is gone, just putting it on media that is capable of high resolution performance does not correct the situation at all.


----------



## arnyk

> > I use iems on any system where a/b ing is done.
> > I mentioned in my post that (1), (3) are already taken care of.
> > (2)Over many comparison tests that i have done and continue to do this effect is of course taken care of.
> > (4) is overcome as the 2 versions are ripped in random sequences by software and usually burned to a cd.
> ...


 
  
  
  
  
 I don't see where in the above any of the points I raised were even touched on, so the entire above makes no sense to me.  Perhaps a little fact-based discussion of the points raised rather than the wave of a hand and flat dismissal would help?
  
 For example, using IEMs addresses zero of the points I raised.  They are fine, I use headphones, earphones and loudspeakers about equally often, but so what?
  
 One of the points - the first point about audiophile evaluations not even being tests, can't be "taken care of" because it is an inherent failing.
  
 The casual apparent  attempt at sort-of randomization shouldn't impress anybody who even took just first year statistics.


----------



## old tech

arnyk said:


> Moran, one of the two authors emailed me earlier this year pointing out a recent AES conference paper that some thought constituted a refutation:
> 
> http://www.realhd-audio.com/?p=3627
> 
> ...



 


The irony is that the so called flaw is that none of the listeners could tell which of the SACDs or DVD-As were sourced from the lower resolution masters. If anything, this strengthened the Moran and Myer findings.


----------



## StanD

old tech said:


> arnyk said:
> 
> 
> > Moran, one of the two authors emailed me earlier this year pointing out a recent AES conference paper that some thought constituted a refutation:
> ...


 
 I doubt that's going to fly since the common source master's resolution was already limited. To be valid I would think that all test materials be made from the same high resolution source.


----------



## arnyk

stand said:


> I doubt that's going to fly since the common source master's resolution was already limited. To be valid I would think that all test materials be made from the same high resolution source.


 
  
 I totally agree with both points.  
  
 The use of resolution-limited sources invalidates the experiment. For the test to be valid as proposed, high resolution sources had to be used.
  
 There are a number of valid defenses of Meyer & Moran:
  
 (1) Their mistake was to take vendor claims at face value. They and the rest of the audio world were hoodwinked by the publicity and marketing of the SACD and DVD-A media. 
  
 (2) In fact nobody seems to have noticed the absence of actual high resolution source material for 5-6 years after the introduction of the SACD and DVD-A.
  
 (3) The people who finally brought this issue to light seem to have based their knowledge on technical tests, not listening tests.
  
 Putting these facts together, we find evidence of a large, unintentional DBT providing solid evidence that there are no audible benefits to high resolution media because the legacy media format (audio CD) is entirely adequate for accurate distribution of music as it is recorded to this day. This conclusion is also supported by all extant formal testing to this day.


----------



## old tech

Hey Arnyk

Why not ask Moran to redo the study and settle this once and for all? Apart from the vendor supplied music issue, the time that has elapsed since that paper (particularly with all the marketing claims around downloadable 24/96 and 24/192 files, it would be very prescient.


----------



## arnyk

old tech said:


> Hey Arnyk
> 
> Why not ask Moran to redo the study and settle this once and for all? Apart from the vendor supplied music issue, the time that has elapsed since that paper (particularly with all the marketing claims around downloadable 24/96 and 24/192 files, it would be very prescient.


 
  
  
 I actually haven't heard much of anything from those guys since. Brad Meyer's public email address is: EBradMeyer@att.net .


----------



## RRod

old tech said:


> Hey Arnyk
> 
> Why not ask Moran to redo the study and settle this once and for all? Apart from the vendor supplied music issue, the time that has elapsed since that paper (particularly with all the marketing claims around downloadable 24/96 and 24/192 files, it would be very prescient.


 
  
 I would surprise me if any hi-res vendor would now willingly donate material to such a test, especially if results were again destined for a professional publication. Too much to lose; nothing to gain.


----------



## old tech

rrod said:


> I would surprise me if any hi-res vendor would now willingly donate material to such a test, especially if results were again destined for a professional publication. Too much to lose; nothing to gain.



 

But why is a vendor required? Could they not just download commercially available hi res files, purchase some DVD A's and SACD's and do the test themselves?


----------



## arnyk

rrod said:


> I would surprise me if any hi-res vendor would now willingly donate material to such a test, especially if results were again destined for a professional publication. Too much to lose; nothing to gain.


 
  
  
 The last instance of such a thing that I know of involved a listening test that was conducted on the AVS forum by a representative of a high def recording vendor: AIX.
  
 The best test files he could provide included a relatively large sliding inter-recording time delay that was easy enough to ABX all by itself. He was made aware of this promply but when I downloaded the latest versions of the files some months later the delay was still there and I could ABX it  quite quickly 16/16.


----------



## arnyk

old tech said:


> rrod said:
> 
> 
> > I would surprise me if any hi-res vendor would now willingly donate material to such a test, especially if results were again destined for a professional publication. Too much to lose; nothing to gain.
> ...


 
  
 Its been done many times with negative results.


----------



## kar13

arnyk said:


> I don't see where in the above any of the points I raised were even touched on, so the entire above makes no sense to me.  Perhaps a little fact-based discussion of the points raised rather than the wave of a hand and flat dismissal would help?
> 
> For example, using IEMs addresses zero of the points I raised.  They are fine, I use headphones, earphones and loudspeakers about equally often, but so what?
> 
> ...


 
 Please make an effort to understand, instead of accusing somebody with flat dismissal or wave of hand or whatnot.Or maybe I'm not understanding your rhetoric, as there is a miscommunication.
  
 iem's matter, because of the passive noise reduction, which helps in concentrating on the tracks, which implies an unmeasured improvement in listening, no matter how small.
  
 Even though this is going to be a regurgitation of precautions which were diligently carried out, will attempt it one last time.
 If you still feel the same, please tell me the manner in which, you would perform another test to obviate the very issues you raised.I would be glad to do that.
  
 (2) was taken care of by the tracks being burned in random manner, as you were unable to glean from me having mentioned it before..
  
 (3)I had mentioned in my original post that level matching had been done!I don't change the loudness level in the middle of a listening session, using loudness meter and gain adjusted accordingly.
  
 (4) was again taken care as I have listened to the same duration of interval especially with the 2 different versions of Take Five.Not just the different versions of the tracks, but also their bits were burned to create a wav file which was then used for listening purpose.It was also burned to a
  
 Your 1st point subsumes 5th point, you made.
 I have no vested interests as I don't own any expensive equipment, and no known bias/prejudice resulting from any psychological/physical reasons, afaik.If you are kind enough to point them out, i will make another A/B test to try and include it.Otherwise, you can happily assume that my results are a biased/subjective.
  
 But, that was kind of my point with my original post that experience is the only test.


----------



## arnyk

> > Originally Posted by *kar13* /img/forum/go_quote.gif
> > iem's matter, because of the passive noise reduction, which helps in concentrating on the tracks, which implies an unmeasured improvement in listening, no matter how small.
> 
> 
> ...


----------



## FFBookman

kar13 said:


> I have tried and haven't found any difference between 16/44 cd and 16/44 from 24/96.
> 
> But i have a question..
> is the effect of different downsampling algorithms used noticeable to a discerning ear..because of dithering?


 

 You are correct that every dithering algorithm sounds different. At least every one I've heard (about 6 total) sound different on my rig.  
  
 This was tested when reviewing music I mixed at 24bit that was returned to me with various dithers applied when downsampling to 16bit. My engineer wanted me to blindly pick my favorite.
  
 Most of the people arguing about 16bit being max resolution have never tested different dithers on their own mixes, so they are not intimately familiar with how it sounds.
  
 I can't say I have a favorite, or can really describe the differences better than saying the size, shape, and timbre of the overall room changes. When you spend many hours mixing a piece of music at 24bit you know exactly where each instrument sits and how the various voices blend together on the EQ.
  
 The dithers that have 'noise shaping' impart an artificial boosted sound to my ears. I think when targeting mp3 format the stuff is used to brighten and thin the sound a bit before lossy time.
  
 That said - dither is critical. Turning it off and leaving downsample artifacts is a bad idea.
  
 It seems to me that if CD's would have been 20bit/48k we wouldn't even have 'hi-res' now, because that's probably enough bandwidth for no-one to want more. I think you can push about 3000k true bitrate though that format, which is enough for most commercially recorded music.


----------



## FFBookman

btw Kar13 - be prepared for every angle of absurdity telling you that you can't hear, don't understand, and are just a flat out idiot for even thinking that we have any better digital audio than 1978.
  
 they scream ABX tests but they can't point to one ABX test that shows any bit of pre-recorded music is better than any other.  the test is not used for that but that doesn't stop some from claiming everything is garbage unless it passes an ABX test.
  
 most of us can't pass ABX tests consistently even with major changes in quality (like 8bit files). this is because of flaws in the test make it completely inappropriate for this use. it will only result in confusion, which is why the side of "there is no such thing as quality" use them.
  
  
 Hey guess what Arnyk -- 50% of people can't reliably pick out the differences between a $10,000 mic and a $10 mic when listening to some random person talking through their computer.  Does this make the two mics the same?  Does this make Aretha Franklin want to sing through the $10 mic?  Will it sound the same when Aretha sings through it in a recording studio?  Who even cares what mic she uses besides Aretha Franklin and her producer?  Do artists you love dissapoint you with this audio snake oil of using quality instruments and microphones?
  
 The point is this is an artistic discussion. If you bring horrible content or measuring equipment only to the debate you won't get far.  You've got to account for the artistry of the creation and production of the music, and let the listener make the choice. What I really hate about some on this board is how they take this stance that you are just an ignorant fool if you want better quality.


----------



## arnyk

kar13 said:


> I have tried and haven't found any difference between 16/44 cd and 16/44 from 24/96.


  
 That is known as evidence. It is evidence that answers the question below.
  
  


kar13 said:


> But i have a question..
> is the effect of different downsampling algorithms used noticeable to a discerning ear..because of dithering?


 
  
 A conversion from 24/96 to 44/16 involves downsampling, digital filtering and dither.
  
 If you haven't found any difference between 16/44 cd and 16/44 from 24/96 then you have combined downsampling, digital filtering and dither and found no difference.
  
 Why would all three cause no audible difference, while just one of them cause an audible difference?
  
 They are all different, independent and don't cancel or augment each other.


----------



## RRod

ffbookman said:


> most of us can't pass ABX tests consistently even with major changes in quality (like 8bit files). this is because of flaws in the test make it completely inappropriate for this use. it will only result in confusion, which is why the side of "there is no such thing as quality" use them.


 
  
 I can consistently pass ABX on 8bit files, if the original file is of high enough dynamic range, which is the exactly what the bit depth affects. The second I get to some über-compressed metal track, suddenly the loss of bits becomes inaudible. For any given track in my collection, there is some bit/sample rate spec at which I can finally pass an ABX.


----------



## interpolate

I can't help thinking people are worrying about things that yet to happen or even matter.


----------



## castleofargh

when the problem is about an relatively effective test contradicting another relatively effective test, it's ok to argue and often hard to know who's right. but when it's a guy with his delusions vs what most human testing is telling us(even our own controlled testing). it's not a matter of opinions anymore.
  
  
 bookman makes it clear that he knows his stuff and that we're all just too ignorant to recognize the obvious. if only we had some other guys working in audio who could also tell us all about the obvious differences of dither and all things occurring below -90db, or how ABX is limited. if only hydrogen audio existed, if only Ethan Winer had made a video an some conferences talking about his own experiments on the matter and explained it all in great details, if only we had maybe someone on this very topic who is at the origin of ABX, to tell us about the pros and cons of such a test.
  
 too bad none of those guys exist or talk on the net, so obviously we should keep on listening to bookman telling his stories without any form of evidence of all the times where his ego TKOed reality in a humiliating way. or tell Arny about how abx isn't objective enough but his ears are.


----------



## Ruben123

So what exactly is the problem with abx? That is doesn't give the results some want to see? If you can hear 6 different sorts of downscaled dither music samples... Why can't you hear that blind? And IF you can hear them blind, then that's sort of the same as abx. Mr Bookman.


----------



## dazzerfong

ruben123 said:


> So what exactly is the problem with abx? That is doesn't give the results some want to see? If you can hear 6 different sorts of downscaled dither music samples... Why can't you hear that blind? And IF you can hear them blind, then that's sort of the same as abx. Mr Bookman.


 

 What's wrong with ABX is that it doesn't fit with his agenda.​ Ergo, in his mind, it's pointless.


----------



## StanD

I see you guys have thrown the book at bookman.


----------



## castleofargh

stand said:


> I see you guys have thrown the book at bookman.


 

 I've tried the book, the baby, the bathwater, the bathtub, I even tried being nice once(I didn't like it), nothing seems to work. no way to get any evidence, no way to have him be as critical and skeptic of himself as he can be of abx, mp3 or everything else..


----------



## Gr8Desire

ffbookman said:


> You are correct that every dithering algorithm sounds different. At least every one I've heard (about 6 total) sound different on my rig.
> 
> This was tested when reviewing music I mixed at 24bit that was returned to me with various dithers applied when downsampling to 16bit. My engineer wanted me to blindly pick my favorite.
> 
> ...


 
  
 Every dithering algorithm may indeed be different. *Applying the same dithering algorithm to the same source twice will generally give different results.*  
  
 Why? Dithering i used to insert _random noise_ in the absence of content.  
  
*Contrary to what several people have suggested: Changing bit depth does not induce dithering.* Whether up sampling or downsampling, you simply interpolate or remove content to get a clean result.
  
 Changing sample rates will often require dithering to estimate missing data points in the new time base. *Convert to an integral sampling rate, and you will not get dithering.* 44/88/176K are all compatible sample rates that will not induce dithering when up or downsampled. In all cases, interpolation or truncation is used to do a conversion. 48/96/384K are also compatible sample rate groups. You simply should not mix groups. Playing 44K content at 384K may seem like a good idea. It's not. Playing 96K content at 44K is also bad. 176K content played at 44K will sound perfect.  Same goes for 48K content played at 96K.

 Not rocket science.


----------



## RRod

gr8desire said:


> Every dithering algorithm may indeed be different. *Applying the same dithering algorithm to the same source twice will generally give different results.*
> 
> Why? Dithering i used to insert _random noise_ in the absence of content.
> 
> *Contrary to what several people have suggested: Changing bit depth does not induce dithering.* Whether up sampling or downsampling, you simply interpolate or remove content to get a clean result.


 
  
 What do you mean "induce" dithering. You either add dither or you don't. The point of it is to randomize the errors due to quantization, substituting broadband noise for the truncation distortion due to the rounding. Quantization intimately relates to the bit depth, not the sample rate, as the number of bits determines the numerical set onto which we must map the samples.


----------



## Gr8Desire

rrod said:


> What do you mean "induce" dithering. You either add dither or you don't. The point of it is to randomize the errors due to quantization, substituting broadband noise for the truncation distortion due to the rounding. Quantization intimately relates to the bit depth, not the sample rate, as the number of bits determines the numerical set onto which we must map the samples.


 

 I am using the term from my experience in writing related algorithms.  _Inducing dither _is correct because that's what the algorithm does. 

 When you change timebases you have to estimate the quantized values for the new timebase. Dithering (which is the same term used in image processing where I have more experience) is induced into the result to help "guess" at missing data points (or as you might say: quantized values in a different non-integral timebase).  

 FWIW: Audiophiles have little actual experience, so they get these simple ideas things wrong.

 And as I said before, you don't dither when changing with bit depths. No need to. Any missing values can be accurately determined by interpolation when up sampling or just removing data points when down sampling.


----------



## icebear

castleofargh said:


> I've tried* the book, the baby, the bathwater, the bathtub,* I even tried being nice once(I didn't like it), nothing seems to work. ...


 
 ... there is still the kitchen sink, you have forgotten that might finally work


----------



## icebear

2^16=            65,536
 2^24=     16,777,216
 2^32=4,294,967,296
  
 2^8= 256
  
 With 16 bits data can represent 65,536 different volume levels and the signal is sampled 44100 times per second. The theoretical dynamic range capacity is way enough to accurately capture actual music. A couple of examples exist that you can cut down the word length to 8 bits where quality loss gets pretty obvious but stil at 10 or 12bits, depending on the music type it's rather difficult to tell degradation right away.
  
 I am not sure which marketing genius used the jagged steps picture for the first time in the early 80's, most likely someone of Philips or Sony. There are no steps, there are isolated points that get connected when the signal is coverted to a continous analog sine type wave, all "steps" are GONE. And 44100 data points per second ... they are already pretty tightly packed next to each other
	

	
	
		
		

		
		
	


	




.
  
 Enjoy the music and don't worry about the numbers folks. If it sound right to your ears, if the music involves you emotionally and is fun to listen to and you forget about the equipment, then it's good enough. No matter what numbers.


----------



## cjl

gr8desire said:


> I am using the term from my experience in writing related algorithms.  _Inducing dither _is correct because that's what the algorithm does.
> 
> When you change timebases you have to estimate the quantized values for the new timebase. Dithering (which is the same term used in image processing where I have more experience) is induced into the result to help "guess" at missing data points (or as you might say: quantized values in a different non-integral timebase).
> 
> ...


 
 Dither can also be used for reduction in bit depth though (and it frequently is). If you simply truncate or round each value, you get quantization noise, which is correlated with the signal and generally undesirable (though inaudible for 16 bit, at least in every case I can think of). If you add dither when reducing the bit rate, quantization noise can be eliminated in favor of a broadband noise completely decorrelated from the signal, and of any character you want (usually shaped to minimize audibility).


----------



## RRod

gr8desire said:


> I am using the term from my experience in writing related algorithms.  _Inducing dither _is correct because that's what the algorithm does.
> 
> When you change timebases you have to estimate the quantized values for the new timebase. Dithering (which is the same term used in image processing where I have more experience) is induced into the result to help "guess" at missing data points (or as you might say: quantized values in a different non-integral timebase).
> 
> ...


 
  
 What do you mean "removing data points" when changing bit depths. If you change from 16/44.1 to 8/44.1, you have just as many samples, but the values of those samples must be mapped to fewer integers.
  
 I see what you mean about sample rate conversion, but the dithering there still seems ancillary to doing calculations at higher bit depth and then bringing things back down.


----------



## RRod

cjl said:


> Dither can also be used for reduction in bit depth though (and it frequently is). If you simply truncate or round each value, you get quantization noise, which is correlated with the signal and generally undesirable (though inaudible for 16 bit, at least in every case I can think of). If you add dither when reducing the bit rate, quantization noise can be eliminated in favor of a broadband noise completely decorrelated from the signal, and of any character you want (usually shaped to minimize audibility).


 
  
 And when people talk about dither (and "hearing it") across the boards here, they mean exactly this. The feel that the use of it when quantizing from 24 or 32-bit down to 16-bit produces audible artifacts that both a) change in quality when different dithering algorithms are used and b) make 16-bit delivery sound "clearly" inferior to 24-bit. They would say this even for 16/44.1 vs. 24/44.1 where no sample rate conversion is needed. Now whether we're all using the term incorrectly, I'm down for discussing that.


----------



## jcx

any example of people doing this with rational gain structure, better perceptual noise shaped dithers - not just turning up the volume when the music is low to hear the dither?


----------



## castleofargh

just to be clear on something, are all the "I'm an idiot" or "I have no idea what I'm doing" situations, a part of what we have to consider as the norm? or can we just presuppose that when we talk about a case, we talk about the people using the devices and software as they should be paired and used?
  
 because if we take in the missuses as if they're standard uses, then I have to say I believe in cable sound, I believe that amps all can sound different. and of course after using -40db gain on my tracks and using windows volume at 2% and compensating with the amp maxed out, I believe that different dither choices might just become audible among other things.


----------



## interpolate

8-bit is going sound like a video game or a glitch techno drumbeat. It in theory does not have a large storage array, so it's kind of pointless comparing 16-bit to much lower bit-depths don't you think. 
  
 Dither algorithms are designed to make use of the frequency attenuation as we all know. In a way, everybody has become very complacent and regards that everyone should be knowledgeable about everything under the sun. The only idiots are those are presume this and denounce others for not.


----------



## RRod

interpolate said:


> 8-bit is going sound like a video game or a glitch techno drumbeat. It in theory does not have a large storage array, so it's kind of pointless comparing 16-bit to much lower bit-depths don't you think.
> 
> Dither algorithms are designed to make use of the frequency attenuation as we all know. In a way, everybody has become very complacent and regards that everyone should be knowledgeable about everything under the sun. The only idiots are those are presume this and denounce others for not.


 
  
 Here's a 5sec clip of 8bit versus 16bit, from the kind of track where you can get away with it. The 8bit version was made by truncation (no dither), and then zeros were padded back on to put it back into 16/44.1 format. Enjoy:
 https://drive.google.com/file/d/0BwmVtb5IwniEYUdRc0R6WjBlX00/view?usp=sharing


----------



## castleofargh

interpolate said:


> 8-bit is going sound like a video game or a glitch techno drumbeat. It in theory does not have a large storage array, so it's kind of pointless comparing 16-bit to much lower bit-depths don't you think.
> 
> Dither algorithms are designed to make use of the frequency attenuation as we all know. In a way, everybody has become very complacent and regards that everyone should be knowledgeable about everything under the sun. The only idiots are those are presume this and denounce others for not.


 
 notice that I'm wording 2 options with the use of "or" in the middle. I'm not saying everybody is an idiot, most just don't know enough to understand that they don't know and fall into my second option. like you when you mistake 8bit music from old video games for the actual sound of 8bit music 
	

	
	
		
		

		
		
	


	



  
 and I still believe my question to be legitimate. are we talking about correct use of thing? or do we need to talk audio like we talk micro waves and put warnings all over it to tell people not to dry their pets with it? and take the exploding yorkshire as stuff that happen to some people? or can we limit our talks to the proper use of a microwave? that really leads to a very different debate IMO.
 and I never ever thought that someone not knowing was similar to someone being an idiot. never ever. we all start somewhere with no information and go somewhere. still idiots do exist!


----------



## interpolate

Yeh OK. But what does that prove?
  
 8-bit audio was the norm for years in computing however up until 16-bit, the only real choice was analogue.


----------



## RRod

interpolate said:


> Yeh OK. But what does that prove?
> 
> 8-bit audio was the norm for years in computing however up until 16-bit, the only real choice was analogue.


 
  
 Just proves that 8bit doesn't have to sound like a video game. But since there are genres that are not sludge metal, you need more bits to get the dynamic range you need. I've yet to find anything that really needs more than 14, at least listening on my HD800 setup. With really well sealed IEMs, maybe I'd find something that needs a 15th bit.


----------



## interpolate

OK. Granted. 
  
 I'm much more a producer than a listener, so do have an interest in such things although with modern audio anything below 16-bit isn't really worth caring about. In Low-Frequency filters and bit-depth decimation it becomes more useful for effects and in some cases DSP effects where inaudible audio is enhanced at a fundamental level.


----------



## RRod

interpolate said:


> OK. Granted.
> 
> I'm much more a producer than a listener, so do have an interest in such things although with modern audio anything below 16-bit isn't really worth caring about. In Low-Frequency filters and bit-depth decimation it becomes more useful for effects and in some cases DSP effects where inaudible audio is enhanced at a fundamental level.


 
  
 Well that just speaks to how the system is (and should be set up): the people making the stuff use the really good gear and higher specs, since it gives more leeway for whatever problem or desire might come up. Then us at the end of the chain get to use cheaper stuff because they did their job right. It's a wonderful setup. Now there's nothing fundamentally wrong with saying "let's just keep things at 24/192 since that's what the master is on," but that's not an argument about audibility, but rather convenience. The flip side is that everyone and his mother can get access to cheap, good hardware at 16/44.1 and 48 without any effort, so what's so wrong with keeping things there for delivery?


----------



## interpolate

Well my argument for keeping everything at the same bit depth/sample rate would be exact imaging as opposed to resampled/bit-shifted. This not only ascertains an accurate representation of the original master it means no alteration in sound was made to get there other than codec conversions. I think it's all subjective. I bought Janis Joplin - Pearl 24/96 FLAC. Whilst a 16/48 would do the same job, it sounds pristine..can't fault it.


----------



## RRod

interpolate said:


> Well my argument for keeping everything at the same bit depth/sample rate would be exact imaging as opposed to resampled/bit-shifted. This not only ascertains an accurate representation of the original master it means no alteration in sound was made to get there other than codec conversions. I think it's all subjective. I bought Janis Joplin - Pearl 24/96 FLAC. Whilst a 16/48 would do the same job, it sounds pristine..can't fault it.


 
  
 Other than all the alterations in the sound that happened during mixing/editing. I guess my perspective is that it's odd for people to have no qualms about applying all kinds of corrections, filters, and effects to audio but then scoff at the idea of doing a final conversion to 16/44.1, which most definitely produces fewer audible changes (none, some of us would argue) than the other stuff. I will be honest and say I'm fine with the current situation, as handling hi-res isn't any kind of big deal. It's the principle, though, especially when it comes to companies charging for the stuff.


----------



## interpolate

I kind of agree, they are charging for an extra sound benefit most people won't be able to distinguish. Personally I always mixdown to 24/48Khz until the final mixdown for CD.  Having FLAC, APE and other hi-res formats opens more potential. Record companies are charging some based on hype, placebos and unwarranted claims.
  
 Despite what others say, DSD to me sounds pretty good although again the source material has be sampled at output rate (through analogue conversion) rather than upsampled to fill all of the 1&0's.
  
 I will buy SACD player which will also act as a high-end CD player at some point.


----------



## RRod

interpolate said:


> I kind of agree, they are charging for an extra sound benefit most people won't be able to distinguish. Personally I always mixdown to 24/48Khz until the final mixdown for CD.  Having FLAC, APE and other hi-res formats opens more potential. Record companies are charging some based on hype, placebos and unwarranted claims.
> 
> Despite what others say, DSD to me sounds pretty good although again the source material has be sampled at output rate (through analogue conversion) rather than upsampled to fill all of the 1&0's.
> 
> I will buy SACD player which will also act as a high-end CD player at some point.


 
  
 Well SACD at least tends to have 5.1 mixes, which is some extra value regardless of the audible benefits of DSD. There's a label I've been into recently that to me does it right. Their albums tend to be about $20, but for that you get a Redbook CD and a Blu-ray audio disc with 2.0 and 5.1 mixes @ 24/192, and a 7.1 mix @ 24/96.


----------



## interpolate

I think potentially the problem is the misunderstanding between bit depth and sample rate.,
  
 Bit-depth as far as I understand, determines how much dynamics you have. So a vinyl player with 60dB SNR is about 10-bit sound and 8-bit in theory only offering 48dB which should be enough for the top end. However it does mean there is potentially missing information filled up as noise.
  
 The less digital information that can be stored in an array, the more truncation and jitter transfer errors will occur. So what is the aim analogue accuracy or digital precision? What is the highest elite sound people really desire. Or is it just one of these things desire yet will never achieve without mortgaging the house.


----------



## RRod

interpolate said:


> I think potentially the problem is the misunderstanding between bit depth and sample rate.,
> 
> Bit-depth as far as I understand, determines how much dynamics you have. So a vinyl player with 60dB SNR is about 10-bit sound and 8-bit in theory only offering 48dB which should be enough for the top end. However it does mean there is potentially missing information filled up as noise.
> 
> The less digital information that can be stored in an array, the more truncation and jitter transfer errors will occur. So what is the aim analogue accuracy or digital precision? What is the highest elite sound people really desire. Or is it just one of these things desire yet will never achieve without mortgaging the house.


 
  
 Well extra sample rate buys you space for doing things like noise shaping as well (look at the content of an SACD much above 20kHz), so sample rate can technically buy you dynamic range as well. But yes, at the standard 6dB/bit calculation, 8bit gets you 48dB, but there are tracks that really have no dynamic range to speak of, so they can work at that depth.
  
 The thing about digital accuracy is that digital should be entirely capable of being accurate if things have been bandlimited properly and you have enough bits, which should be true in any good chain. The question is where you set the limit. For frequency extension, based on testing single tones, 22kHz is enough. Hi-res advocates say more is needed, but they can't possibly be basing that on what humans can hear as single tones. Various theories are out there (non-aural sensation, non-linearity in the ear, etc.), but any evidence that comes out on both sides is readily rejected by the opposing side. As a little person, I figure all I can do is listen to music, do my own tests, and read published papers to judge where the science actually is. Currently, this has me at 16/44.1 being plenty enough for end-user delivery.


----------



## sonitus mirus

For media playback, it seems to me that Red Book already greatly exceeds the comfortable dynamic range for human hearing.   Other than my "Whispers and Air Raid Sirens Volume 2" CD, I think we have already achieved all that we can hope to with dynamic range unless some major breakthrough is discovered...that doesn't come from Meridian and their business partners.


----------



## KeithEmo

interpolate said:


> I kind of agree, they are charging for an extra sound benefit most people won't be able to distinguish. Personally I always mixdown to 24/48Khz until the final mixdown for CD.  Having FLAC, APE and other hi-res formats opens more potential. Record companies are charging some based on hype, placebos and unwarranted claims.
> 
> Despite what others say, DSD to me sounds pretty good although again the source material has be sampled at output rate (through analogue conversion) rather than upsampled to fill all of the 1&0's.
> 
> I will buy SACD player which will also act as a high-end CD player at some point.


 
  
 I think this discussion depends to a major extent on which side of the table you're coming from.
  
 Purely as a customer, I'm inclined to believe that, _IF THERE IS A DIFFERENCE_, the higher resolution version will be better. Assuming the company selling the product isn't "playing games", if they're offering a 24/192k version and a 16/44k version, they will have created the master at 24/192k first, then down-sampled it to create the 16/44k version. So, _IF_ the conversion process is audible, then the 24/192k version will be closer to the master and, if the conversion really is totally inaudible, then they will be equal. Likewise, _IF_ the company has deliberately chosen to make the two versions sound different, they will be most likely to have deliberately reduced the quality on the 16/44k version. (Either to deliberately ensure that their more expensive "premium" version sounds better, or to deliberately reduce the details audible in the "non-audiophile" version because they believe that's what their audience expects.) All of these factors led me to believe that the 24/192k version will be either audibly the same as, or audibly better than, the 16/44k version. And so, by buying the 24/192k version, I can be most sure of getting the best version (or, at the very least, not a deliberately or accidentally reduced version).
  
 I do _NOT_ find the various claims about how high-resolution and better frequency response can actually have a _negative_ effect on sound quality to be at all credible with modern equipment.
  
 Now, as a seller of music, I think the current reality is simply that high-resolution is an excellent marketing strategy. If my customers are willing to buy a "24/192k remaster" just because it's 24/192k, or if they're willing to pay a few dollars more for it than the "regular" version, then that's all the motivation I require to offer it for sale. There's no motivation for me to perform testing to prove that it's better, and certainly no motivation for me to do testing that might prove the opposite; and, obviously, even a test result that proved that the high-res version was "a tiny bit better, but most people don't notice the difference" would be bad for business.
  
 And, finally, as a seller of audio _equipment_, I can use the fact that my products support the higher sample rate as a positive selling point, and as a reason why my customers should upgrade their older equipment to "current high-res products". (And, if anything, I might have more motivation to do so, since, presumably, people will buy music either way, but they may not choose to upgrade hardware without a "solid reason".) However, the reality is that _MOST_ current DACs support sample rates up to and including 24/192k so, while it might be considered to be an important feature, it's hardly a "product differentiator".
  
 My overall conclusion would be that "enough people want high-resolution music" that it makes sense to both produce and to sell it... and that technical "proof positive" that the difference is audible is pretty much unnecessary at this point. It simply seems unlikely whether a scientific study that provides conclusive proof either way will have much effect on sales - and it is sales that "drive the need to produce the product". (You can make your own judgement as to whether "there's simply no point in fighting the amount of advertising money that's been spent to sell high-res to the public" or "the horse is already out of the barn and it's way too late to try and get him back inside - or to worry about how he escaped".) Bear in mind that storage space has gotten very cheap, which means that 24/192k files don't cost much more to store or download than 16/44k ones. And also remember that, as downloads have overtaken CDs as the most popular medium, and download bandwidth has also gotten cheaper, the fact that 16/44k is "the CD standard" has become less significant (more people are buying downloads than CDs, and the cost of downloading a 24/192k album is not significantly more than the cost of downloading a 16/44k album).
  
 Also, as you've noted, simply offering a few more formats, and so more options, also tends to be good for business.


----------



## FFBookman

rrod said:


> Other than all the alterations in the sound that happened during mixing/editing. I guess my perspective is that it's odd for people to have no qualms about applying all kinds of corrections, filters, and effects to audio but then scoff at the idea of doing a final conversion to 16/44.1, which most definitely produces fewer audible changes (none, some of us would argue) than the other stuff. I will be honest and say I'm fine with the current situation, as handling hi-res isn't any kind of big deal. It's the principle, though, especially when it comes to companies charging for the stuff.


 

 It's because everything else in the creation and mixing phase _adds_ to the total sound quality. Downsample+dither is the first and only forced _step back_ in quality in the modern recording environment. I mean the last 20 years. Artists/producers ask why, were told "so it fits on a CD" and are now told "that's what online store stocks" and it's basic commerce at that point. How far can we degrade it before people stop paying? Hmm.....
  
 This downgrade to a smaller file size (bandwidth restriction) is not described accurately as being necessary for conveniently timed and priced transmission, storage, and playback of the file. That's the basics and the whole story.
  
 All the ABX gar'baaage, all the name calling, all the psychoacoustic research cited, could care less about what _music sounds best_. That can only be determined when the barriers to proper rendering at large are removed.
  
  
 That's why products like Pono are important, they bring quality that only few knew existed to the masses, and then they can decide if they hear or care about the quality differences.


----------



## FFBookman

interpolate said:


> I think potentially the problem is the misunderstanding between bit depth and sample rate.,
> 
> Bit-depth as far as I understand, determines how much dynamics you have. So a vinyl player with 60dB SNR is about 10-bit sound and 8-bit in theory only offering 48dB which should be enough for the top end. However it does mean there is potentially missing information filled up as noise.
> 
> The less digital information that can be stored in an array, the more truncation and jitter transfer errors will occur. So what is the aim analogue accuracy or digital precision? What is the highest elite sound people really desire. Or is it just one of these things desire yet will never achieve without mortgaging the house.


 
  
 I can't prove it but I think there's something _there_ in the sound field _air_ that gets digitally reduced along with the things we can measure. It might even get scrambled on the way in, but I haven't recorded enough with a purely analog signal chain to test that theory. I just think the entire ADC/DAC process is ripe with error and compromise and we are left to accept them since we cannot easily swap components inside those digital processes.
  
 As far as upper limits of resolution, stereo PCM 24/192 has plenty of headroom in all directions to my ears.  I have not heard DSD or MQA yet but I've been in many recording environments and I can't see going beyond 24/192 for a few more decades at least. They'd really need some new inventions with speakers to need more bandwidth than that.


----------



## RRod

ffbookman said:


> It's because everything else in the creation and mixing phase _adds_ to the total sound quality. Downsample+dither is the first and only forced _step back_ in quality in the modern recording environment. I mean the last 20 years. Artists/producers ask why, were told "so it fits on a CD" and are now told "that's what online store stocks" and it's basic commerce at that point. How far can we degrade it before people stop paying? Hmm.....
> 
> This downgrade to a smaller file size (bandwidth restriction) is not described accurately as being necessary for conveniently timed and priced transmission, storage, and playback of the file. That's the basics and the whole story.
> 
> ...


 
  
 To me as a consumer, there are no guarantees that everything you might do to a mix will be an improvement; see the whole loudness war as an example. The barriers to proper rendering should be at the far ends of the chain: the venue and the mics, the speakers and the listening room. ADCs and DACs do their job and do it well, and DAWs will do exactly as told including turning a mix into utter crud. And while the Pono player is certainly good hardware, the whole Pono setup *assumes* that people can hear the differences, and charge for it accordingly, esp. in the store. Your statement is an example: "quality that only few knew." That assumes nothing before was available at < $400 that had quality sound, which is just plain untrue.


----------



## FFBookman

sonitus mirus said:


> For media playback, it seems to me that Red Book already greatly exceeds the comfortable dynamic range for human hearing.   Other than my "Whispers and Air Raid Sirens Volume 2" CD, I think we have already achieved all that we can hope to with dynamic range unless some major breakthrough is discovered...that doesn't come from Meridian and their business partners.


 
  
 Dynamic Range is just another attempt at finding a singular measure for quality. Then there's that word 'comfortable'. I don't listen to air sirens, I listed to instruments, and it makes me comfortable to hear them as accurately as possible, in a position in the room.
  
 Too bad you've never heard better than CD playback. I'm not snarking, I kind of feel bad for you, to think that in 1978 they designed the most perfect digital audio system as to never need a basic file format upgrade. That's mind blowing to me. I bet every other digital anything in your life has been upgraded multiple times in that timeframe. Wait - most of them didn't even exist in 1980.
  
 It's like digital audio v 1.0 and you are still believing it can never be improved upon.
  
 I know that ADC and DAC chips have improved since 1980 but it's more than that. People have been working at qualities above 16/44 literally since the CD shipped to the public. You can argue that the general public doesn't _need_ it, but to think that it doesn't exist is just crazy to me. Why record at 24/88 to ship at 16/44?


----------



## FFBookman

rrod said:


> To me as a consumer, there are no guarantees that everything you might do to a mix will be an improvement; see the whole loudness war as an example. The barriers to proper rendering should be at the far ends of the chain: the venue and the mics, the speakers and the listening room. ADCs and DACs do their job and do it well, and DAWs will do exactly as told including turning a mix into utter crud. And while the Pono player is certainly good hardware, the whole Pono setup *assumes* that people can hear the differences, and charge for it accordingly, esp. in the store. Your statement is an example: "quality that only few knew." That assumes nothing before was available at < $400 that had quality sound, which is just plain untrue.


 

 True there was stuff around to play hi-def but very few all-in-one iPod style devices to play it properly. By around 2013  I knew of A&K selling one for $1400 and Fiio had one for $500. There have been external DAC's and amps forever. 
  
 What hooked me on the pono pitch was the push for provenance and open standards/no DRM on the music. Buy from us, trust it's the best version available, and it comes with no security or catch. That's nice and they are living up to their promise, so far.
  
 The player being a bad little mother is just icing on the cake. Sure you can piece together a few things that will sound as good as Ponoplayer, but it won't be as small, rugged, simple, and come in under $500.
  
 Neil's marketing was key because it cracked through the "lossy is fine, music is fine, buy new headphones" line that Apple and others have been pounding us with. They haven't talked quality in 15+ years now. Those bastards injest 24bit files and then lossy them to 256 AAC to sell and stream them. They have masters they won't even sell!


----------



## FFBookman

rrod said:


> To me as a consumer, there are no guarantees that everything you might do to a mix will be an improvement; see the whole loudness war as an example. The barriers to proper rendering should be at the far ends of the chain: the venue and the mics, the speakers and the listening room. ADCs and DACs do their job and do it well, and DAWs will do exactly as told including turning a mix into utter crud. And while the Pono player is certainly good hardware, the whole Pono setup *assumes* that people can hear the differences, and charge for it accordingly, esp. in the store. Your statement is an example: "quality that only few knew." That assumes nothing before was available at < $400 that had quality sound, which is just plain untrue.


 

 To address the rest of your point -- if you are buying something from an artist, that artist has a producer, there's a mixing engineer or two, maybe a label person, and the team of them decide how it's going to sound.  You as a consumer just have to accept that they created a piece for your listening pleasure.
  
 Loudness war is a combination of many things, it can't be blamed on one entity, one stage, one format, even one decade. Its been a continual thing since the beginning of recorded music. You want yours sounding bigger, louder, more current, and more hip than the others. This goes all the way back.
  
 Today's loudness wars are the result of infinite digital duplication on the production end colliding with a massively restricted output format that isplayed on $4 audio signal chains into $5 headphones.
  
 There's a lot wrong with modern music. And to me, it all started started heading off the rails when 16/44 shipped as the first digital audio standard.


----------



## castleofargh

ffbookman said:


> True there was stuff around to play hi-def but very few all-in-one iPod style devices to play it properly. By around 2013  I knew of A&K selling one for $1400 and Fiio had one for $500. There have been external DAC's and amps forever.
> 
> What hooked me on the pono pitch was the push for provenance and open standards/no DRM on the music. Buy from us, trust it's the best version available, and it comes with no security or catch. That's nice and they are living up to their promise, so far.
> 
> ...


 
 I'm loving it. you're blind when it goes as you like, and super picky when it doesn't. the typical 2 tiers reality.
 dither at -96db, oh boy that ruins your sondstage and whatever. but pono is great? \o/ I can't stop laughing.
  
 how about 5ohm output that changes the frequency response of most high end multidriver IEMs? and that's single ended.
 how about the same impedance that varies with the voltage output because the ayre guy decided 30years ago that feedback was bad when everybody uses it to stabilize frequency response, reduce impedance output, and improve overall distortion levels. but you wouldn't care about distortions and FR changes right?  those are not important stuff, not like dither and signal with no music content down below -96db. oh irony.
 and the fact that pono is oriented for highres, well better tell people to use highres, the thing sucks just plain bad at 44khz. no good in FR,  rolls off already 0.5db @10khz. THD is almost as high as the stuff masked on a mp3
	

	
	
		
		

		
			





. IMD is pretty bad too. that's how great it it for 500$. you have the signal resolution of a little better than mp3.
  
 what your post tells us is that you're talking about stuff happening behind the mountain but you miss the bird that landed on your nose. in relative magnitude that's about it. you complain about stuff down at -96db and don't notice the stuff happening at around -70db. (I fully believe most people wouldn't notice either, but if one should be noticed, I wouldn't bet on the -96 signal. just saying).
  
 you the 2 is more than 1 argument guy, how about that? complaining about signals( the pono goes almost 1V so I'll take that to give a scale) 96db below that's 0.000016v(people plz tell me if I mess up). the distortions of the pono up to -70db(a little higher in fact and that's with almost ideal load...) that means errors in the signal of about 0.000316v that's 19 times bigger, I know you like it when it's about numbers being bigger. you've told us so many times. of course people will say and will be right, that 2nd order HD are unlikely to be noticed. but so is dither. and you've got no excuse for IMD not being audible or being euphonic.
  
 for once in your life how about taking off your blinders and accept that you've been talking nonsense all this time?
 you can see it all there http://www.stereophile.com/content/pono-ponoplayer-portable-music-player-measurements. and again the review is suspiciously nice, I would kill to see measurements into a 16ohm load. highres DAP would surely take a very special definition at that moment.
  
 but we're not done, now the pono files, did you read archimago's take on the "only the best" marketing stunt?  let me give you the link http://archimago.blogspot.fr/2015/01/last-words-on-pono-mastering-analysis.html
 they're just like any other highres provider, if the guy giving them the masters says it's good, they will take it.
  
  
 and people say to buy a better headphone because most headphones have distortions in the 1% area, even those really low like some ortho/planar stuff, they are still in the 0.1% zone for distortions. that's -60db, the limit of signal fidelity of a MP3@320. seeing how you hate mp3s, headphones should scare the hell out of your with the same magnitude of signal getting corrupted.
 but too bad speakers are usually worst+the room. feel silly for sprouting nonsense about stuff at -96db you pretend to hear yet?
  
 I'd say good luck explaining all this, but we both know that you just ignore anything that proves your wrong. that's how you have been able to keep pretending all this time.


----------



## RRod

ffbookman said:


> True there was stuff around to play hi-def but very few all-in-one iPod style devices to play it properly. By around 2013  I knew of A&K selling one for $1400 and Fiio had one for $500. There have been external DAC's and amps forever.
> 
> What hooked me on the pono pitch was the push for provenance and open standards/no DRM on the music. Buy from us, trust it's the best version available, and it comes with no security or catch. That's nice and they are living up to their promise, so far.
> 
> ...


 
  
  


ffbookman said:


> To address the rest of your point -- if you are buying something from an artist, that artist has a producer, there's a mixing engineer or two, maybe a label person, and the team of them decide how it's going to sound.  You as a consumer just have to accept that they created a piece for your listening pleasure.
> 
> Loudness war is a combination of many things, it can't be blamed on one entity, one stage, one format, even one decade. Its been a continual thing since the beginning of recorded music. You want yours sounding bigger, louder, more current, and more hip than the others. This goes all the way back.
> 
> ...


 
  
 But once again we're conflating things like provenance with the final delivery format.  My argument is that taking a well-done hi-res remastering of a previously badly mastered album and converting it to 16/44.1 (and nothing else) will result in an audibly indistinguishable product from the hi-res master, that will similarly sound better than the previously badly-mastered version.  In fact I have heard hi-res remasters that sound worse than Redbook.  That badly mastered Redbook releases exist is not the fault of Redbook; it's the fault of the people who use the format to do things like brickwalling.
  
 On that point, you can see why I, as the consumer, do not simply accept that the whole production team is creating a track for my pleasure, because what I consider pleasureful might not match what they or other listeners think. After all, it's not like bands haven't been signing off on loudness war nonsense.
  
 I also cannot see how someone thinks that Redbook was the beginning-of-the-end, given that I have releases from the early- and mid-1980s through today that sound superb.  In classical, at least, I've never had any issues. Now that I've been able to get hi-res classical, listening to difference files has confirmed that there ain't nothing in there that Redbook couldn't give me as a delivery format.
  
 As far as Apple, well they have their agenda. But again, any lack of transparency in what masters they are using shouldn't be conflated with the abilities of AAC.
  
 Anyway, off for adventures away from internets. Other people can chime in.


----------



## icebear

ffbookman said:


> *It's because everything else in the creation and mixing phase adds to the total sound quality. *Downsample+dither is the first and only forced _step back_ in quality in the modern recording environment. I mean the last 20 years. Artists/producers ask why, were told "so it fits on a CD" and are now told "that's what online store stocks" and it's basic commerce at that point. How far can we degrade it before people stop paying? Hmm.....


 
  
 The less processed and simpler a recording and mixing process is, the better. There are a few direct to dics examples of live sessions. The recordings sound spectacularly live. The more processing post recording, the more harm will done to the ambient information, the sound of the room.
  


ffbookman said:


> I can't prove it but I think there's something _there_ in the sound field _air_ that gets digitally reduced along with the things we can measure. It might even get scrambled on the way in, but *I haven't recorded enough with a purely analog signal chain to test that theory. *I just think the entire ADC/DAC process is ripe with error and compromise and we are left to accept them since we cannot easily swap components inside those digital processes.
> ...


 
 How about comparing your recordings to the live session you were attending?
 Does your recording sound like the live event when the musician were creating the sound?
 But darn it that would be close to ABX testing and that only works visually and not audible. It looks like you are in jam there 
	

	
	
		
		

		
			




  


ffbookman said:


> Dynamic Range is just another attempt at finding a singular measure for quality. Then there's that word 'comfortable'. I don't listen to air sirens, I listed to instruments, and it makes me comfortable to hear them as accurately as possible, in a position in the room.
> 
> *Too bad you've never heard better than CD playback. [1]* I'm not snarking, I kind of feel bad for you, to think that in 1978 they designed the most perfect digital audio system as to never need a basic file format upgrade. That's mind blowing to me. I bet every other digital anything in your life has been upgraded multiple times in that timeframe. Wait - most of them didn't even exist in 1980.
> 
> ...


 
  
 [1] What live concerts have you attended lately that have exceeded a dynamic range of 96db?
 [2] nobody said(wrote) that


----------



## dprimary

icebear said:


> The less processed and simpler a recording and mixing process is, the better. There are a few direct to dics examples of live sessions. The recordings sound spectacularly live. The more processing post recording, the more harm will done to the ambient information, the sound of the room.
> 
> How about comparing your recordings to the live session you were attending?
> Does your recording sound like the live event when the musician were creating the sound?
> ...


 
 I agree, anything you do beyond mic pre amp direct to the recording is reducing sound quality. Any processing and mixing you do is impacting the recording in a negative manner. You cannot undo it. (well in a DAW  you might be able to undo if it was non destructive)
  
 No analog recording medium sounds anything like what went in to. I started in analog even the best Studer and Ampex tape machines never played back anything close to the signal fed into them. Sometimes I liked what played back most of the time it was frustration of what was lost. By the mid early to mid 90's digital was able to deliver back what was fed into the recorder.


----------



## arnyk

keithemo said:


> I'm inclined to believe that, _IF THERE IS A DIFFERENCE_, the higher resolution version will be better


 
  
 That argument seems at least as valid any Creationist argument against Evolution.
  
 Anybody who says such a thing in public would seem to qualify themselves for maximum scrutiny of any evidence that they bring forth because they seem biased well beyond reasonable doubt.
  
 It also seems unreasonable because to be true there would have to be some kind of magic halo surrounding all high resolution recordings that magically eliminates any of the usual production mistakes that can affect any recording in any format on any media.


----------



## icebear

I am certain that there is indeed a difference BUT
  
 1. The difference doesn't really matter in real world listening
  ( about as much as one care can drive 160mph and the other can drive 200mph top speed and both are stuck in traffic next to each other
	

	
	
		
		

		
		
	


	




 )
  
 2. For most users (all not on audiphile forums) and most equipment any possible positive difference is completely lost.
 They don't really pay attention to the minute details, they go about their business and daily routines. They drive in their cars or subway.
 Heck they don't even care about mp3 or compression, they just enjoy the music as soundtrack for their day.


----------



## Thad-E-Ginathom

ruben123 said:


> So what exactly is the problem with abx? That is doesn't give the results some want to see? If you can hear 6 different sorts of downscaled dither music samples... Why can't you hear that blind? And IF you can hear them blind, then that's sort of the same as abx. Mr Bookman.


 
  
 The problem with ABX is that a person has _only_ their ears to rely on. There is no help from brand name, price tag, visual appearance, etc, etc, etc.
  
 The blind tester is _forced_ to trust their ears. The more people scream that that is what we _should_ do, the less is it what they _do_ do.


----------



## FFBookman

thad-e-ginathom said:


> The problem with ABX is that a person has _only_ their ears to rely on. There is no help from brand name, price tag, visual appearance, etc, etc, etc.
> 
> The blind tester is _forced_ to trust their ears. The more people scream that that is what we _should_ do, the less is it what they _do_ do.


 

 No, the person only has their ears and their memory to rely on.  If they only had their ears they would need to play both sounds at the same time.
  
 Guess what - their memory fails them, and their ears can't play detective with sound quality unless they are intimately familiar with that content.  So they guess, they follow agendas, and most importantly, they grow bored with this stupid test and fire off bad data.
  
 ABX tests are garbage for music testing. They are completely counter to how we enjoy music. Until you accept this could be the case your arguments against quality mean nothing to me.
  
 You can't hide behind a flawed test that reports that nothing can be reliably detected. It nulls everything. No other test could be as reliably unreliable as ABX, always showing statistical garbage. 
  
 Show me the ABX tests that report that 16bit well mastered audio sounds better than 8bit audio. Show me the ABX tests that show a properly wired and operated playback system sounds better than a malfunctioning one. Show me an ABX test that actually confirms anyone can hear anything on a mass scale.


----------



## FFBookman

icebear said:


> I am certain that there is indeed a difference BUT
> 
> 1. The difference doesn't really matter in real world listening
> ( about as much as one care can drive 160mph and the other can drive 200mph top speed and both are stuck in traffic next to each other
> ...


 

 We are making some progress here!   
  
 I think 10 years ago #2 was correct. But with cheap DAP's with good DAC's out now, and better DAC's getting around, and better headphones getting around, this doesn't have to be the case anymore.
  
 Also, I think #2 was sold to them with marketing and convenience, and people are noticing that the music is not as fulfilling as it once was.  They can't always describe that 10% doesn't taste as good or give them as much nourishment as 100%, but they feel it and they know it.
  
 When one of those types here's my Pono, especially playing 24bit files, they smile, they understand, it's kind of obvious. They were lied to to believe that a $4 chip in a phone playing 10% files was about as good as they could get these days.
  
 #1 - it doesn't matter for new music, music made since the MP3 format took over. If you listen to music made in the last 15 years you don't need hi-resolution, you are right about that. That music was created and produced for low-resolution consumption. But if you listen to real instruments played by real people, mixed by humans not machines, the resolution matters.  But most of the worlds finest recorded music has not been made between 2000-2015, so "doesn't really matter" is not true.


----------



## FFBookman

dprimary said:


> I agree, anything you do beyond mic pre amp direct to the recording is reducing sound quality. Any processing and mixing you do is impacting the recording in a negative manner. You cannot undo it. (well in a DAW  you might be able to undo if it was non destructive)
> 
> No analog recording medium sounds anything like what went in to. I started in analog even the best Studer and Ampex tape machines never played back anything close to the signal fed into them. Sometimes I liked what played back most of the time it was frustration of what was lost. By the mid early to mid 90's digital was able to deliver back what was fed into the recorder.


 

 Oh I have to disagree here - a mix engineer and musician work to improve the sound coming from the instruments, usually so they mix together in a pleasing and artistic way.  A mix is a piece of art designed to enhance the material, not degrade.
  
 Much of this is style, but I don't consider an electric guitar degraded after going through the pedal board and amp. I don't consider a snare degraded after it runs through a plate reverb. And I definitely don't consider anything degraded after running through a nice Focusrite EQ.
  
 Overall - arranging the song, building the parts, layering the voices, and putting a mix on a song should not degrade it at all, it should improve every last bit of it as an overall work of art. If it degrades it, just record the band live and call it a day.
  
  
 As far as comparing to live - it is different. Standing in front of the stage and listening to a rock band with 3-part harmonies is thrilling. But I can't hear all the separation and full spectrum of sound in most live environments. I love live the best, in person. If i'm going to listen to a pre-recorded, mixed piece of music the studio version will provide me with the best overall sound quality.


----------



## KeithEmo

rrod said:


> But once again we're conflating things like provenance with the final delivery format.  My argument is that taking a well-done hi-res remastering of a previously badly mastered album and converting it to 16/44.1 (and nothing else) will result in an audibly indistinguishable product from the hi-res master, that will similarly sound better than the previously badly-mastered version.  In fact I have heard hi-res remasters that sound worse than Redbook.  That badly mastered Redbook releases exist is not the fault of Redbook; it's the fault of the people who use the format to do things like brickwalling.
> 
> On that point, you can see why I, as the consumer, do not simply accept that the whole production team is creating a track for my pleasure, because what I consider pleasureful might not match what they or other listeners think. After all, it's not like bands haven't been signing off on loudness war nonsense.
> 
> ...


 
  
 Setting aside discussions of _MINOR_ differences between different sample rates which might or might not be audible, I've also got several Red Book CDs that, to put it bluntly, sound so good that I can't imagine how they could sound better..... Certainly, with nothing to compare them to, I can't even suggest how they could have been improved. Sadly, this just serves to highlight how not-good many CDs sound. (But, as you say, this proves that you obviously can't blame how bad most CDs sound on the format - the existence of even a few excellent quality ones proves that the format is capable of sounding very very good.)


----------



## FFBookman

Please stop changing the subject to "loudness wars". They are a combination of many things, but they are not the format or resolution of the file.  That's a different debate.
  
 "Loudness wars" is a combination of several business decisions.
  
 "Loudness wars" have really escalated in the last 25 years because
  
 a) The CD format dropped the noise floor, removed the rumble from sub-frequencies, removed dust and dirt from the high frequencies, and allowed for faster dynamics than a needle in a groove could provide. Producers took advantage of these facts of the format to get louder, brighter, with longer run times.
  
 b) The DAW allowed for unlimited instances of compression across all bands of frequencies, unlimited automation of every parameter, unlimited routing, unlimited replication and unlimited tracks in a production. Producers took advantage of these facts of the production environment and applied more and more effects, more and more compression, more and more automation.  Dub step as we know it wasn't possible 30 years ago because the computers weren't fast enough to assist like that.
  
 c) The MP3 format sucks all the natural air and space out of the mix, thins out the soundstage, the depth, and obliterates things like long decays and delay timing that make acoustic instruments like drums, strings, guitar, and voices, leaving them sounding very unnatural.  Producers took advantage of these facts of the format to get more robotic, autotuned, tweaked out movement, "the drop", avoid real cymbals, hi-hats, real drums, real guitar, and use any and all the tricks you can program to get the speakers moving and popping and sound bigger than they are.
  
 The format and the production environment does drive these choices. But to say "we don't need higher resolution because of loudness wars" is simply not true.  Higher resolution would lay the groundwork to step back from the precipice we are currently at where sound quality is beat to a pulp.


----------



## Thad-E-Ginathom

ffbookman said:


> No, the person only has their ears and their memory to rely on.  If they only had their ears they would need to play both sounds at the same time.
> 
> Guess what - their memory fails them, and their ears can't play detective with sound quality unless they are intimately familiar with that content.  So they guess, they follow agendas, and most importantly, they grow bored with this stupid test and fire off bad data.
> 
> ...


 
  
 Show me _your_ understanding (not your biases and misunderstandings) of any sort of blind test.
  
 No? I thought not.


----------



## FFBookman

thad-e-ginathom said:


> Show me _your_ understanding (not your biases and misunderstandings) of any sort of blind test.
> 
> No? I thought not.


 

 Huh?   Read my posts, I don't need to hit you people with links, there's no test, I"m not your professor. This is a discussion, no one owns fact.
  
 I understand ABX listening tests, I've participated, and I'm designing a better test to judge musical enjoyment.
  
 I got tired of complaining and no one doing anything about it, so I'm doing it myself. It might take me a bit but I'll get there.
  
  
  
 Question about the psyche of you folks -- is there any other digital standard you argue against improving?  Did you avoid HD TV when it came out? I didn't get one until 2012 myself.  
  
 Do you take low resolution digital pictures and post them to photo sites? Or do you take them full quality then reduce them to 20% JPG and throw out the original?
  
 Do you swear there is no better gaming technology than Nintendo?
  
 Do you work happily on a 286 or other computer from the 80's?
  
 Do you use dial-up?
  
 I'm the Mr. Vintage in my circles, so you types really fascinate me. Is there any other senses in the human body you think can be recreated with 256k of data?


----------



## KeithEmo

arnyk said:


> That argument seems at least as valid any Creationist argument against Evolution.
> 
> Anybody who says such a thing in public would seem to qualify themselves for maximum scrutiny of any evidence that they bring forth because they seem biased well beyond reasonable doubt.
> 
> It also seems unreasonable because to be true there would have to be some kind of magic halo surrounding all high resolution recordings that magically eliminates any of the usual production mistakes that can affect any recording in any format on any media.


 
  
 I disagree entirely - and, in fact, I would assume the opposite. If a particular production company is going to offer both "standard res" and "high-res" versions of a given disc or song for sale, and is going to charge more for the high-res copy, then I would most certainly expect them to do their best to ensure that "the more expensive premium version" would sound "better" than "the economy version" - whether that simply means being more careful in production - or even means deliberately "degrading" the quality of the cheaper copy. (Just as I would expect a given car company's $120k model to be "better" than their $20k model - even if they have to deliberately compromise the performance of the cheaper model to maintain that relationship.) If there are any ways in which they can improve the quality, then the premium version will definitely get them first and, if there are any corners to be cut, then the economy version will see them cut first.
  
 I wasn't specifically suggesting that they would be different - but,_ IF _they are different, it would be a foolish company who actually offered a "regular copy" that sounded _better_ than their high-res version.


----------



## cjl

ffbookman said:


> No, the person only has their ears and their memory to rely on.  If they only had their ears they would need to play both sounds at the same time.


 
 So when you're comparing things in a sighted evaluation, you play two different things simultaneously? Otherwise, I fail to see how this is any different than any other evaluation of two different sounds.


----------



## Thad-E-Ginathom

ffbookman said:


> Huh?   Read my posts, I don't need to hit you people with links, there's no test, I"m not your professor. This is a discussion, no one owns fact.
> 
> I understand ABX listening tests, I've participated, and I'm designing a better test to judge musical enjoyment.


 
   
 I didn't think that ABX tests were to test "musical enjoyment." It seems to me that they set out to answer questions such as "A and B sound different to me: am I imagining it, or is it it for real?"
  
 Having established difference, preference and enjoyment are different matters, but _what if, _having decided that you prefer A to B, you then cannot tell the difference in a blind test? It would be a strange (and possibly dysfunctional?) brain that did not have its perceptions altered by bias and expectation. It is the way our brains work.
  
 Quote:


> I got tired of complaining and no one doing anything about it, so I'm doing it myself. It might take me a bit but I'll get there.


 
 Well, good! The more the merrier.
   


> > Question about the psyche of you folks -- is there any other digital standard you argue against improving?  Did you avoid HD TV when it came out? I didn't get one until 2012 myself.
> >
> > Do you take low resolution digital pictures and post them to photo sites? Or do you take them full quality then reduce them to 20% JPG and throw out the original?
> >
> ...


----------



## castleofargh

thad-e-ginathom said:


> ffbookman said:
> 
> 
> > Huh?   Read my posts, I don't need to hit you people with links, there's no test, I"m not your professor. This is a discussion, no one owns fact.
> ...


 
 ask him to give back the link with his ideas about a proper test. if he didn't edit it, it's hilarious.


----------



## arnyk

ffbookman said:


> ABX tests are garbage for music testing. They are completely counter to how we enjoy music.


  
  
 Assertion without evidence. Do you understand what a logical argument is?


----------



## arnyk

keithemo said:


> I disagree entirely - and, in fact, I would assume the opposite. If a particular production company is going to offer both "standard res" and "high-res" versions of a given disc or song for sale, and is going to charge more for the high-res copy, then I would most certainly expect them to do their best to ensure that "the more expensive premium version" would sound "better" than "the economy version" - whether that simply means being more careful in production - or even means deliberately "degrading" the quality of the cheaper copy. (Just as I would expect a given car company's $120k model to be "better" than their $20k model - even if they have to deliberately compromise the performance of the cheaper model to maintain that relationship.) If there are any ways in which they can improve the quality, then the premium version will definitely get them first and, if there are any corners to be cut, then the economy version will see them cut first.


 
 If this is true why is it that something like half of the SACDs and DVD-As released in the first 5 years after the formats were introduced were based on low resolution masters? The basic argument seems to be a tired reiteration of the worn out false idea that you always get what you pay for and the most expensive alernative is the best alternative. Do you really go into every store and buy the most expensive product so that you are sure to get the best?


----------



## interpolate

I'm a great believer in scientific prognosis however not of marketing projectile claims by advertisements. Didn't they say CD was indestructible?


----------



## castleofargh

interpolate said:


> I'm a great believer in scientific prognosis however not of marketing projectile claims by advertisements. Didn't they say CD was indestructible?


 

 very fragile as a freesbie, you need to take out at least 30CDs if you want to get any fun.


----------



## KeithEmo

arnyk said:


> If this is true why is it that something like half of the SACDs and DVD-As released in the first 5 years after the formats were introduced were based on low resolution masters?The basic argument seems to be a tired reiteration of the worn out false idea that you always get what you pay for and the most expensive alernative is the best alternative.Do you really go into every store and buy the most expensive product so that you are sure to get the best?


 
  
 The problem here is that you're trying to read each of my "claims" in isolation - when, in fact, they all contribute.
  
 Claim #1: If someone is selling multiple versions of the same product, similar but at different price points, it will be easier to sell the more expensive product if you can demonstrate some obvious superiority. (So, if you plan to charge more for the high-res download, then it does make sense to try and arrange things so it does sound noticeably better.)
  
 Claim #2: Because many people, and Americans in particular, have a tendency to believe that "they get what they pay for", they tend to believe that more expensive products are better. (So, all else being equal, and no actual difference being there, they will tend to imagine that the more expensive one sounds - or tastes - better.)
  
 Claim #3: Expectations in general affect what people experience. One such expectation is that more expensive products are better, but there are many others. In your specific example, lots of money and advertising was spent to convince people that SACDs and DVD-A discs are in fact "better" - which sets an expectation that they are superior. This would still have had some effect if the prices were the same. However, by also pricing them a bit higher, the two "expectation modifiers" work together.....  and people think: "It's the more expensive premium product _AND_ I've heard they sound really good" - which gives them a stronger bias to expect a difference than either individual thing by itself.
  
 There's an excellent book called "Influence" by Cialdini that goes into a lot of detail about all the different ways of influencing people - which talks about #2 and #3. There is one well known case where a liquor manufacturer introduced a new "top shelf" brand at the same reasonable prices as their "normal" products - and was greeted with rather poor sales. The solution was to simply raise the price by about 50%, while leaving the product untouched - and the result was that sales improved dramatically. (People were simply unwilling to believe that it was 'a top shelf product" because the price wasn't "top shelf".
  
 My Claim #1 doesn't really fall into the category of "influence" since I'm suggesting that a manufacturer might deliberately reduce the quality of the cheaper product to ensure that the more expensive product is better, or that they might be willing to simply sell very similar products, and use various methods to manipulate people into perceiving that the more expensive product was better. Of course, real life isn't always so simple, so they may do a little of both.
  
 To answer your question, I don't generally buy "the most expensive product"; however, if a company offers a line of several products at different price points, I tend to avoid the lowest priced one - because I do tend to be suspicious that they may have deliberately omitted important features to make that product distasteful enough to convince people to buy a more expensive one. ("Yes, we said we would sell you one for $10, and we wouldn't lie, but that $10 one is really cheesy - you really want at least the $20 one.") It's a pretty standard strategy to offer a very inexpensive alternative that you know is inadequate so you can claim the really low price, but not sell that low-profit unit to many customers.  
  
 In general, what I've found is that, when a product line contains several similar products, the very lowest priced one is inadequate, the highest priced model includes lots of extra luxury features I really don't need, and, somewhere in the middle of the line, is the product that isn't missing any important features, but also doesn't have too many expensive luxury extras.


----------



## arnyk

keithemo said:


> Claim #1: If someone is selling multiple versions of the same product, similar but at different price points, it will be easier to sell the more expensive product if you can demonstrate some obvious superiority. (So, if you plan to charge more for the high-res download, then it does make sense to try and arrange things so it does sound noticeably better.)
> 
> Claim #2: Because many people, and Americans in particular, have a tendency to believe that "they get what they pay for", they tend to believe that more expensive products are better. (So, all else being equal, and no actual difference being there, they will tend to imagine that the more expensive one sounds - or tastes - better.)
> 
> Claim #3: Expectations in general affect what people experience. One such expectation is that more expensive products are better, but there are many others. In your specific example, lots of money and advertising was spent to convince people that SACDs and DVD-A discs are in fact "better" - which sets an expectation that they are superior. This would still have had some effect if the prices were the same. However, by also pricing them a bit higher, the two "expectation modifiers" work together.....  and people think: "It's the more expensive premium product _AND_ I've heard they sound really good" - which gives them a stronger bias to expect a difference than either individual thing by itself.


 
 How does all of the evidence-free speculation above square with posting on a forum titled "Sound Science"


----------



## dprimary

ffbookman said:


> Oh I have to disagree here - a mix engineer and musician work to improve the sound coming from the instruments, usually so they mix together in a pleasing and artistic way.  A mix is a piece of art designed to enhance the material, not degrade.
> 
> Much of this is style, but I don't consider an electric guitar degraded after going through the pedal board and amp. I don't consider a snare degraded after it runs through a plate reverb. And I definitely don't consider anything degraded after running through a nice Focusrite EQ.
> 
> ...


 
 You are talking about production not audio quality. You are building up the mix in a pop production. You can polish the audio but you can't get more then what you started with. 
  
 Considering the amount of times I have unplugged a guitar from a chain of pedals and directly into the amp after a guitarist complains about they can't get a good tone, and then they are amazed at the rich tone that was always there. 
  
 Compared to the right microphone in the right place in a nice room a snare with plate reverb is second best. How many plate reverbs are still working? All EQ's have side effects, often the side effects are greater then what you are trying to correct. Many of the recordings of the last 10 years would sound much better if anyone tried the bypass button.
  
 If the music is classical or jazz there is no mixing, overdubs, or effects. Microphone selection , preamps and placement is everything.
  
 The change in sound quality is greater with a few inches of mic placement, then the difference between between 16/44.1 and 24/88.2 has ever been.


----------



## Thad-E-Ginathom

keithemo said:


> In general, what I've found is that, when a product line contains several similar products, the very lowest priced one is inadequate, the highest priced model includes lots of extra luxury features I really don't need, and, somewhere in the middle of the line, is the product that isn't missing any important features, but also doesn't have too many expensive luxury extras.


 
  
 In other words, you have more or less the same expectations and biases that most of us have (err, more or less) which is why the marketing guys have such ranges, with different price points, available.
  
 I'm not accusing you of being a marketing man, or anything really rude like that (  ), but you manufacture/sell stuff, right? You must be aware of at least some of the psychology of buying and selling stuff?
  
 I try, as a buyer, to be aware, but that still, often, does not change my behaviour. I _want_ to buy a more expensive model when I go shopping. I can even get really pissed off if the salesman is an honest guy listing all the reasons why the cheaper one is just fine for me!


----------



## arnyk

keithemo said:


> Claim #1: If someone is selling multiple versions of the same product, similar but at different price points, it will be easier to sell the more expensive product if you can demonstrate some obvious superiority. (So, if you plan to charge more for the high-res download, then it does make sense to try and arrange things so it does sound noticeably better.)


 
  
 The problem here is that we're talking audio where sound quality is usually determined by sighted evaluations that are inherently debilitatingly flawed as I have explained here without credible rebuttal many, many times.
  
 In the minds of many audiophiles the demonstrations of obvious superiority is any reviewer's claim, any blogger's claim, any audiophile's claim no matter how inherently flawed and therefore irrelevant.
  
 One of the best large-scale examples of this was given during the first 5-7 years after the introduction of SACD and DVD-A when audiophiles almost universally praised it, but in fact about 50% of all recordings were based on low resolution masters. 
  
 The recordings were low resolution, but the public was told that they were high resolution and sold them at the higher price point. The "Obvious superiority" did not in fact exist.
  
 This is obviously fraud. Do you support massive fraud like this?:


----------



## FFBookman

dprimary said:


> You are talking about production not audio quality. You are building up the mix in a pop production. You can polish the audio but you can't get more then what you started with.
> 
> Considering the amount of times I have unplugged a guitar from a chain of pedals and directly into the amp after a guitarist complains about they can't get a good tone, and then they are amazed at the rich tone that was always there.
> 
> ...


 
 I'm not disagreeing with your overall point, I totally agree.  There's all sorts of way to enhance, degrade, or modify the audio.  Of course performance is more important than any format or gear down the line. 
  
 But I'm talking about distribution formats specifically, and I'm using production stories to give an example of what type of "polish" is lost when you degrade the audio format just for commercial purposes.
  
 Most of the stuff removed from an MP3 is the "polish" of the instruments, this provides the depth, the placement, the timbre of the instrument. When that is attacked through lossy compression, AFTER being degraded with downsampling and dithering there is a noticeable drop in overall data in the sound.  This 2-step degradation is what I am working to remove from existance.
  
 Of course, Apple tried to fix it and got it wrong. They pushed for 24bit masters for everyone (mastered for iTunes), got rid of the 16/44 standard, just so they can make better sounding 256k versions in their AAC format. It's asinine.
  
 Apple:
_"Give us the highest quality possible, so we can make tiny versions to sell at full price."_
  
 Plus the MP3 model sets the price standard and that's not even treated uniformly, because people don't understand the math: 
  
 Apple sells 10 songs @ 256k for $10 
 and
 Ponomusic sells the same 10 songs @ 3800k for $18
  
 which is the better value? (assuming the masters are >= 3800k)


----------



## sonitus mirus

ffbookman said:


> Apple sells 10 songs @ 256k for $10
> and
> Ponomusic sells the same 10 songs @ 3800k for $18
> 
> which is the better value? (assuming the masters are >= 3800k)


 
  
 That is easy.  For me, 10 songs at $10 is a bargain compared to 10 songs at $18.


----------



## Thad-E-Ginathom

> Originally Posted by *FFBookman* /img/forum/go_quote.gif
> 
> ...
> 
> ...


 
  
 MP3 (my bold)?
  
 Did you move the goal posts away from >16-bit/>44.1k _lossless_ audio? Move them to _lossy_ audio formats where you _might_ have some chance of demonstrating actually degraded  sound?
  
 Whether people can actually hear the difference between _high_-rate lossy and lossless is a whole other story. The goal posts in this thread are about 24/96-and-above _lossless_ audio.
  
 I've put the goal back in the right place: perhaps you'd like to take another kick?


----------



## arnyk

ffbookman said:


> Most of the stuff removed from an MP3 is the "polish" of the instruments, this provides the depth, the placement, the timbre of the instrument. When that is attacked through lossy compression, AFTER being degraded with downsampling and dithering there is a noticeable drop in overall data in the sound.  This 2-step degradation is what I am working to remove from existance.


 
 And the reliable evidence to support the above apparently unsupported claims is exactly where? Without reliable evidence the above is just heavy breathing, bragging, pontification, speculation, you pick the uncomplementary word. I'd bet money that if you were forced to save your life by ABXing a bunch of well-made MP3s from the high resolution recordings that for this test we will make them from, there would be a whole lot of random guessing followed by the unfortunate demise of someone we all know and love but know better to take his words at face value.


----------



## StanD

arnyk said:


> And the reliable evidence to support the above apparently unsupported claims is exactly where?Without reliable evidence the above is just heavy breathing, bragging, pontification, speculation, you pick the uncomplementary word.I'd bet money that if you were forced to save your life by ABXing a bunch of well-made MP3s from the high resolution recordings that for this test we will make them from, there would be a whole lot of random guessing followed by the unfortunate demise of someone we all know and love but know better to take his words at face value.


 
 Bookman doesn't require evidence, for his purposes, his imagination is sufficient.


----------



## arnyk

stand said:


> XXXXXXX doesn't require evidence, for his purposes, his imagination is sufficient.


 
  
 The phrase "Legend in his own mind" comes to mind somehow...


----------



## StanD

arnyk said:


> The phrase "Legend in his own mind" comes to mind somehow...


 
 Most of his statements defy logic and yet they keep on coming. At first they were somewhat humerous, now they give me a headache.


----------



## XenHeadFi

I have lurked for a bit on Head-Fi and have read my of the topics in the Sound Science subsection. I admit that I have not fully read this thread, and my points may have already been discussed in the previous pages.
  
  
 The problem I have with many of these "format wars" is the lack of a suitable control. I have a good understanding of the scientific process. Here I will make an appeal to authority: I have a PhD and have published in peer-reviewed journals researching molecular biology, bioinformatics, and molecular genetics. My first thought was that the master recording would be a suitable control/reference, however, this lead me to the "elephant in the room" that I have not seen discussed.
  
 I would think that one of the biggest constraints in recording is how the sounds are actually...err..recorded: the source of sound, the recording equipment and storage media. I assume that capturing the sound produced by people mechanically (playing instruments, singing, etc) is being done by a microphone. So, how good are studio microphones? Are they good enough to take advantage of a >16bit / >44.1khz waveform? I am admittedly naive in what happens during recording sessions, but to me, the weakest link is that first device, the microphone.
  
 To remove the microphone, we can take "sound recording" to the other extreme, all digital, like what Wendy Carlos did and still does. With modern computational clusters and their immense processing capacity, someone could craft a digitally pure waveform of a simulated string quartet in standard seating arrangement that, potentially, should be able to take advantage of the latest greatest audio format. Technically, this is not a "sound" yet, but I ask that you can grant me this latitude in the definition. Also, the simulated instruments would have to produce suitable harmonics from 0 Hz to 384 KHz (or higher?) that will be stored in the waveform at the highest resolution format (32bit/384KHz?). Remember, this is a test of audio resolution and, at this stage, should not be limited to what is generally accepted as audible. 
  
 Now transform this reference master "recording" to the different audio resolutions, all the way down to 16/44.1 and send it out as loss-less files that can support such content. I propose the venerable WAV format, which can handle up to 32 bits, 4.3 GigaHertz (~4,300,000 KHz) sample rate, and a maximum file size of 4GB. The WAV would be compressed by a suitable loss-less algorithm (LZMA/7Z/ZIP/etc). Now, we would have suitable files to test, and hopefully something that has been missing in many of these kinds of discussion, access to the unprocessed master waveform as a reference.
  
 These files could also be used for other tests, not just audio resolution.


----------



## icebear

Welcome to head-fi, nice first post 
	

	
	
		
		

		
			




 I am not familiar with Wendy C. I listen acoustic music mostly, jazz and classic. I just like the sound better and prefer live recordings in a natural room with everyone playing in one session and not something stitched together from multi track tapes or even completely generated with a computer. That being said, I agree that the microphone might be the weakest link, right next to choice of microphone and placement of the same
	

	
	
		
		

		
		
	


	




.
  
 The entire discussion about testing and finding evidence will never get us anywhere.
 The relevant question is not if there is a difference (of course if you look [listen] close enough you will find differences in everything) but if the quality currently offered and the format is satisfying and convenient to use for >90% of the consumers. That is not an actual question, I guess 
	

	
	
		
		

		
		
	


	




.
  
 What's the market share of music in higher rez. format >16/44.1 ?


----------



## StanD

@XenHeadFi Wendy Carlos works with Synthesizers which do not create the same quality of harmonic nuances that are found in natural instruments. Samplers are recordings so we're back to microphones and pickups. So don't sell modern microphones and instrument pickups short. Once one has a good recording creating playable product of different formats and resolutions is what we need to do. That's doable.


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## jcx

microphones for recording are often chosen for their "sound" - frequency response, sensitivity, directivity all are taken into account by a pro selecting and positioning mics in a recording session
  
 some vocal mics may start to roll off < 14 kHz, many more smaller condenser mics are pretty flat to around 20 kHz and one company does advertise a 50 kHz mic in its specialized "drum kit" package
  
 in large performance venues for Symphonic music and others air's increasing attenuation of high frequencies rolls off very high audio frequencies before they reach the audience
  
 modern close micing individual performers/instruments practice regularly captures high audio frequencies that wouldn't reach typical audience listening position
  
  
 there are few suggestive controlled human listening  studies indicating the need for > 20 kHz - and some of those few have been shown to be flawed - actual "conventional audio" frequency differences were caused by the "ultrasonic" content mixing down in speaker or amplifier nonlinearities
  
 so far no one is revising Psychoacousitc textbook human hearing sensitivity curves showing a steep drop by 20 kHz for the young college age populations most often tested - and age and hearing damage fairly quickly reduce most people's upper audio frequency sensitivity


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## XenHeadFi

icebear said:


> Welcome to head-fi, nice first post
> 
> 
> 
> ...


 
  
 Thanks! Wendy Carlos created the original TRON soundtrack from the 1980's. Whether you'll like it or not, you never know until you try. 
	

	
	
		
		

		
		
	


	




 (I have my guess, but I would not want to implant a bias!)
  
 No idea about market share and availability, however I thought this thread is about audio resolutions. Personally, I would love to see the science of audio updated a bit from down here at the consumer level. I admit I have no idea what is cutting edge acoustical research, so maybe they are already doing this kind of stuff.
  


stand said:


> @XenHeadFi Wendy Carlos works with Synthesizers which do not create the same quality of harmonic nuances that are found in natural instruments. Samplers are recordings so we're back to microphones and pickups. So don't sell modern microphones and instrument pickups short. Once one has a good recording creating playable product of different formats and resolutions is what we need to do. That's doable.


 
  
 Yes, synths as we know them are usually pretty artificial, but what I was inferring from "computational cluster" is a full on simulation of sound. Think of digital effects in movies. Many are pure digital constructs and when successfully done, you won't even know they were there. That visual simulation (rendering) has gotten to the point where they implement both photons AND waves of light. If you can simulate waves of light, you can start thinking about waves of sound, too. Also, the "quality of harmonic nuances" is why the simulation should be carried out to an extreme level of 0 Hz to 384 KHz if possible. I'm guessing that range should encompass just about any harmonic interactions from natural instruments.
  
 Also, my suggestion was a way to remove "what is a good recording?" from the test. Think of the debates that would cause! Theoretically, minor variations in attack/sustain/decay could also be simulated to make the sound more natural. Computational power is probably not the limiting factor for such a project.


----------



## XenHeadFi

jcx said:


> there are few suggestive controlled human listening  studies indicating the need for > 20 kHz - and some of those few have been shown to be flawed - actual "conventional audio" frequency differences were caused by the "ultrasonic" content mixing down in speaker or amplifier nonlinearities
> 
> so far no one is revising Psychoacousitc textbook human hearing sensitivity curves showing a steep drop by 20 kHz for the young college age populations most often tested - and age and hearing damage fairly quickly reduce most people's upper audio frequency sensitivity


 
  
 So recent studies were considered flawed...a good reason not to update textbooks. However, I do think the technology is now available to test again.


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## castleofargh

*/!\ IRONY WARNING /!\  *(any flawed reasoning resembling some from a real character would be coincidental, and I'll deny it in a court of headfi justice!) 
  
  
 if you use 2 microphones instead of 1 for the same sound, you get twice the data. do the math, it's better quality!
 dr chesky doesn't even understand the basics with his binaural records, real highres has at least 10 microphones per instrument, and you could never render all this inside 16/44-16/48!


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## castleofargh

xenheadfi said:


> jcx said:
> 
> 
> > there are few suggestive controlled human listening  studies indicating the need for > 20 kHz - and some of those few have been shown to be flawed - actual "conventional audio" frequency differences were caused by the "ultrasonic" content mixing down in speaker or amplifier nonlinearities
> ...


 
  
 you know I put myself in their shoes sometimes. if I could notice the added whatever from high sample rate, or truly could hear when a song lacks ultrasounds. oh boy! but I would go through all the existing test methods to try and get one that can prove it to the world. I'd be so exited from the idea, no way I wouldn't try everything and then some more. I would get the Guinness record for highest frequency heard, and each time I would go 2khz higher. I'd be the sergey bubka of ultrasound, breaking the world record more than 30times! I'd be famous like a kardashian without having to make a sex tape. total glory!
  
  some of the guys positive they can tell when it's not highres are pros. pros can be ignorant, pros can have a system with high IMD, and pros can have an agenda, but still they make noise on the matter so they're invested in the idea. several of those guys have the mean to test a great many things if they really wanted to. you think neil young couldn't make a campaign and get funding to show how highres is the bomb? it would take him 5 phone call and 2 tv shows.
  
 so why do you suppose we do not hear much about tests on ultrasounds or the audible benefits of higher sample rates outside of 2 papers by the same japanese guy? with the first one proved to be faulty and the second one being strangely specific on the ways to get a positive result(real loud ultrasounds, and not alone but with other music, and not asking people, but looking at their brain... ) why nothing more than this?
 I see a few possible reasons:
 -1/ the industry knows perfectly well how things really are, just like any audiologist.
  
 -2/ they did try a lot of things, failed all proper tests, and would rather not tell that they just spent a month and 50people to prove  they were wrong.
  
 -3/ those who can hear ultrasounds are space aliens, agencies all over the world work with the NSA and area 51. the NSA doesn't care about humans, they're tracking everybody to find the invaders! snowden is in mission for the NSA to create a cover so that aliens don't suspect a thing.  they set traps by opening a topic on ultrasounds, track online those who admit hearing something, find them and kill them! then they affect an agent that will keep pretending to be the dead alien and try to contact other aliens by means of talking about ultrasounds online and play world of warcraft.
  
 I believe that covers the question as to why we don't have more tests to look for ultrasound hearing and the importance of high sample rate for the listener.


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## StanD

xenheadfi said:


> Thanks! Wendy Carlos created the original TRON soundtrack from the 1980's. Whether you'll like it or not, you never know until you try.
> 
> 
> 
> ...


 
 I used to work with Electronic Music Synthesizers and knew Wendy as well as when she was Walter. She used Moog Modular Series which is a Subtractive Synthesizer. The control of harmonics and envelopes with such a device is predictable but limited. Predictable meaning easy for a musician to create sounds by understanding how the different modules work and the harmonic content of waveforms. These were analog computers.


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## XenHeadFi

stand said:


> I used to work with Electronic Music Synthesizers and *knew Wendy as well as when she was Walter*. She used Moog Modular Series which is a Subtractive Synthesizer. The control of harmonics and envelopes with such a device is predictable but limited. Predictable meaning easy for a musician to create sounds by understanding how the different modules work and the harmonic content of waveforms. These were analog computers.


 
  
 Wow. In my response to you I had a bit about my thoughts on Wendy Carlos' work, but I deleted it. I really enjoyed reading her blog as it was very informative and was a history of the beginnings of digital audio as written by someone who was right there. Fascinating reading to me. Too bad it hasn't been updated since 2009. I really enjoyed Switched-On Bach; like classical is to rock, Switched-On Bach is to electronica. I also have Boston Baroque performing Bach's Brandenburg Concertos on Telarc CDs (remember when DDD was actually something audiophiles looked for?). However, I thought her Switched-On Bach 2000 hit the "Uncanny Valley" for audio, which will probably happen if someone could simulate a string quartet. Still, I think it would make for very good reference material.


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## StanD

xenheadfi said:


> Wow. In my response to you I had a bit about my thoughts on Wendy Carlos' work, but I deleted it. I really enjoyed reading her blog as it was very informative and was a history of the beginnings of digital audio as written by someone who was right there. Fascinating reading to me. Too bad it hasn't been updated since 2009. I really enjoyed Switched-On Bach; like classical is to rock, Switched-On Bach is to electronica. I also have Boston Baroque performing Bach's Brandenburg Concertos on Telarc CDs (remember when DDD was actually something audiophiles looked for?). However, I thought her Switched-On Bach 2000 hit the "Uncanny Valley" for audio, which will probably happen if someone could simulate a string quartet. Still, I think it would make for very good reference material.


 
 A lot of the Electronic Music of the 60s and earlier were mostly bleeps, blops and tape splicing. Wendy took a risk of recording her renditions of classical music (OK, Baroque, Switched on Bach) and finally came out of it electronic music that normal people could enjoy as music. By the way, the oscillators on a Moog Modular Sythesizer of that day drifted like crazy so it was good to keep one's environment temperature stable.


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## interpolate

As featured in Clockwork Orange. William ørbit did something similar in the 90's with electronically recreating classical music.


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## MacacoDoSom

For all of you people that has doubts about bit depth and dither (and what's audible or not) I suggest you download the evaluation of the VST plugin from Stillwell "Psycho Dither", you can select from 6 bit to 32 bit with or without noise shaping (12 degrees of it), the red dithering is 'auto-black', (if there is absolute silence there is no dithering).
 It's a nice tool to check the audibility of dither.....


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## castleofargh

macacodosom said:


> For all of you people that has doubts about bit depth and dither (and what's audible or not) I suggest you download the evaluation of the VST plugin from Stillwell "Psycho Dither", you can select from 6 bit to 32 bit with or without noise shaping (12 degrees of it), the red dithering is 'auto-black', (if there is absolute silence there is no dithering).
> It's a nice tool to check the audibility of dither.....


 

 can we go below 8bit?(guess I'll try anyway, but am at a point in foobar where I need to remove stuff to add new components and the component folder is just a graveyard of all the crap I tried once and don't even remember what it is
	

	
	
		
		

		
			





). 
 I have http://www.toneboosters.com/tb-dither/ on foobar and it's as intuitive as it gets, but I can only "ruin" my sound down to 8bit and then use an idiotic noise shaping to bring up the noise where it's clearly audible.  still its the one plugin that gave me the most obvious "in practice" idea behind the concept of noise shaping.


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## MacacoDoSom

Yes


castleofargh said:


> macacodosom said:
> 
> 
> > For all of you people that has doubts about bit depth and dither (and what's audible or not) I suggest you download the evaluation of the VST plugin from Stillwell "Psycho Dither", you can select from 6 bit to 32 bit with or without noise shaping (12 degrees of it), the red dithering is 'auto-black', (if there is absolute silence there is no dithering).
> ...


 

 Yes it is very good, lots of noise shaping options, the stillwell goes down to 6 bit, thought... but tells you nothing about the noise shaping used...just a graph...


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## icebear

xenheadfi said:


> Thanks! Wendy Carlos created the original TRON soundtrack from the 1980's. Whether you'll like it or not, you never know until you try.
> 
> 
> 
> ...


 
  OK, gave it a listen (short only), sorry but I wouldn't listen to this type of music even if I got paid 
	

	
	
		
		

		
		
	


	




.
 Just the total opposite of my preference.


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## RRod

castleofargh said:


> can we go below 8bit?(guess I'll try anyway, but am at a point in foobar where I need to remove stuff to add new components and the component folder is just a graveyard of all the crap I tried once and don't even remember what it is
> 
> 
> 
> ...


 
  
 Another thing is to realize that you can "shape" truncation distortion as well; that is, if you truncate down to 8-bit, you can then move those distortion products around using various shaping algorithms/parameters. Of course you typically want to use dither to de-correlate errors from the signal, but it can be helpful to consider shaping as its own phenomenon, independent of dither.


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## KeithEmo

sonitus mirus said:


> That is easy.  For me, 10 songs at $10 is a bargain compared to 10 songs at $18.


 
  
 I agree - but that is always going to be a personal "value judgement". Sometimes I want "the best deal" and sometimes I want "the best - period" as long as I can afford it.
  
 A good $9 half-pound hamburger is also almost undoubtedly a much better buy than a $45 eight-ounce filet mignon (they are both eight ounces of USDA approved beef) and, while I sometimes spring for the filet, I usually settle for the hamburger. I would have to say that the same goes for music, although with me it tends to be related to the content itself. For many songs or albums, which I listen to "casually", I don't mind HD-FW, or iTunes quality. However, for the albums and groups that I really like, I want the best sounding version I can get... so I opt for the CD or the HD version. (Some people pay a fortune for signed items and autographs, which I won't; others will pay $120k for a Mercedes, when it won't get them to work any faster than my Nissan; so I guess we all have our personal quirks.)
  
 (To tell you the truth, if you want the best deal, it seems to me that a streaming service like Tidal, where you get zillions of songs, at CD quality, for about $20 a month is probably _THE_ best deal right now.... although I personally like to "own, have a copy of, and hold" any music I pay for. )


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## KeithEmo

thad-e-ginathom said:


> In other words, you have more or less the same expectations and biases that most of us have (err, more or less) which is why the marketing guys have such ranges, with different price points, available.
> 
> I'm not accusing you of being a marketing man, or anything really rude like that (  ), but you manufacture/sell stuff, right? You must be aware of at least some of the psychology of buying and selling stuff?
> 
> I try, as a buyer, to be aware, but that still, often, does not change my behaviour. I _want_ to buy a more expensive model when I go shopping. I can even get really pissed off if the salesman is an honest guy listing all the reasons why the cheaper one is just fine for me!


 
  
 Quite true. (And, in fact, I do work for the Marketing department). We all have our expectations and biases - and a large percentage of them are shared equally by all of us. And you're entirely right - we _CAN'T_ "turn them off"; at best we can recognize them and either compensate for them or simply acknowledge that they play a part in our decisions. And not all biases are bad. Is it really so awful that the next steak you buy will seem to taste better if you eat it in a nice restaurant, with soft music playing and dim lights, instead of in a cafeteria-style place with tiled floors and an annoying PA system? I'd be happy to pay the extra $20 to enjoy my meal more, but I like to know whether I'm paying for better meat, or better ambiance. Likewise, I've been known to pay a lot extra for a piece of audio gear because it has a knob that feels nice when you turn it, or a heavy metal face plate instead of a plastic box, and I really prefer equipment that accepts a power cord instead of adding another brick to the pile under my rack.
  
 Unfortunately, there's a "vicious cycle" today....  Many customers really want to believe that they _CAN_ avoid their biases. This, in turn, means that it would be "product suicide" for a vendor to say that their $500 DAC had the same parts inside it as their $200 one, but the metalwork and display were a lot nicer. That would offend some of their customers, many of whom would rather believe that they're paying for better performance, or better parts. So, in turn, the company has to add a few "audiophile parts", whether they really matter or not, or make up some imaginary reasons to justify the cost. (The problem is that we all pay the price - which is why you can never seem to find a lower-end product that has _ALL_ the features of the top model, but in a cheaper case, with a less fancy display. The manufacturer is almost obliged to remove at least a few important features from their cheaper product so customers will "see" a difference other than the externals.)
  
 Now, as for that sales guy, a pretty good sales guy will be honest; a _REALLY_ good sales guy will find out how you feel - and tailor his response to whether you actually want all the information or not. (He should be giving you what YOU want - or, at the very least, what he's calculated will make you most likely to buy the product.)


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## sonitus mirus

I don't know if I will ever purchase a lossless file at a premium price if I can still purchase the CD of the same music.   Other than the testing that I did, which required that I had some good quality HD files to ABX, I think the best option for me is to purchase a CD for $6 to $8 and then rip it to whatever format(s) I need.  Any of the streaming services work well for me, and I can't hear a difference between Tidal FLAC and the Google MP3 format with nearly every song.  I say nearly every song as I cannot possibly test every available song, but of those that were tested by me, I could not hear a difference.  
  
 Edit: Your cheeseburger to filet comparison does not make sense in my situation.  I cannot hear a difference between an iTunes AAC file and Red Book.  Any food would have to taste exactly the same and be an identical amount, but the cost would need to be significantly different.


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## KeithEmo

arnyk said:


> The problem here is that we're talking audio where sound quality is usually determined by sighted evaluations that are inherently debilitatingly flawed as I have explained here without credible rebuttal many, many times.
> 
> In the minds of many audiophiles the demonstrations of obvious superiority is any reviewer's claim, any blogger's claim, any audiophile's claim no matter how inherently flawed and therefore irrelevant.
> 
> ...


 
  
 I'm not convinced that it was fraud.
  
 Here's a hypothetical.... Let's assume that I buy a large quantity of some reasonably good "middle of the road wine". I put half of it in relatively normal bottles, which I sell for $12.99 each. I put the other half in really fancy bottles, designed by a high-end artist, with fancy corks and gold foil, and sell those bottles for $49.99 each. Now, further, let's assume that, when I survey my customers afterwards, many of the customers who bought at least one of each "type of wine" actually enjoyed the $49.99 bottle more. (This is almost certain to happen.)
  
 Was anybody defrauded? (The folks who paid more got more enjoyment for their money.)
  
 The legal standing for the wine would be that, as long as I didn't claim that the wine in the two bottles was actually different, I hadn't committed fraud.
  
 And I'm pretty sure that saying that "SACDs sound better than CD" constitutes a claim based on an opinion.
 In other words, there's an unspoken.... "We, and lots of other people, _BELIEVE_ that SACDs sound better than CDs".
 For that matter, they could probably have said, quite truthfully, that.....  "65% of the people we've surveyed told us that they think the SACD sounds better".
  
 It's only fraud if they _SPECIFICALLY_ said that those discs were created from high-resolution masters.
 (And neglected to mention, in the fine print, that those high-resolution masters were themselves made from low-resolution recordings.)
 I'm sure all of the claims were "reviewed by the legal department" and found to _NOT_ "cross the line into false advertising or fraud".
  
 I honestly don't see anything "more fraudulent" there than I see every evening on most TV commercials.


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## Ruben123

It's just very smart to ask $200 for an ordinary power cable and people actually buying them. Even better when they advertise that they hear enhanced bass, better soundstage and treble extension. That way you don't have to lie. I can only laugh at the cable threads any many times at the DAP section too. Not that I've heard any expensive DAP... But buying a $3500 DAP to go with a $200 earphone and talking about better performance is just weird. (Also I don't think differences between those devices are easily heard when DBT'ed)


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## Thad-E-Ginathom

Keith, you are quite right about the food, and I am even with you all the way on the audio gear. When I know that I'm entertaining my own biases, and that I can afford it, I'm really very happy to go for that finely-engineered knob or the beautiful case. I fully acknowledge, and subscribe to buyer/owner satisfaction coming in many different forms, shapes and sizes.
  
 People can even buy fake and fraud stuff, if that is what tickles them, but it doesn't make the sale right or ethical.
  
 PCM (or DSD) may very well not be, at any bit/sampling rate, the ultimate in audio storage and reproduction. One thing that makes me sad is that chasing the numbers in that department is not putting resources into whatever might come next. It is a dead end.


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## nick_charles

keithemo said:


> I honestly don't see anything "more fraudulent" there than I see every evening on most TV commercials.


 
  
 LOL - that's setting the bar pretty low for probity


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## RRod

keithemo said:


> I'm sure all of the claims were "reviewed by the legal department" and found to _NOT_ "cross the line into false advertising or fraud".


 
  
 There's quite a difference between having honest intentions and passing them by the legal department, and having nefarious intentions and passing them by the legal department.


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## interpolate

I have nothing further to add.


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## castleofargh

rrod said:


> castleofargh said:
> 
> 
> > can we go below 8bit?(guess I'll try anyway, but am at a point in foobar where I need to remove stuff to add new components and the component folder is just a graveyard of all the crap I tried once and don't even remember what it is
> ...


 
 yup that VST plug in I linked does just that, you shape the noise with what looks like a graphic EQ ^_^
 that's why I was saying it was intuitive.(and kind of fun)
  
  


sonitus mirus said:


> I don't know if I will ever purchase a lossless file at a premium price if I can still purchase the CD of the same music.   Other than the testing that I did, which required that I had some good quality HD files to ABX, I think the best option for me is to purchase a CD for $6 to $8 and then rip it to whatever format(s) I need.  Any of the streaming services work well for me, and I can't hear a difference between Tidal FLAC and the Google MP3 format with nearly every song.  I say nearly every song as I cannot possibly test every available song, but of those that were tested by me, I could not hear a difference.
> 
> Edit: Your cheeseburger to filet comparison does not make sense in my situation.  I cannot hear a difference between an iTunes AAC file and Red Book.  Any food would have to taste exactly the same and be an identical amount, but the cost would need to be significantly different.


 
 maybe that would work better with a bad pizza and a good pizza? ^_^ I'll have both anytime.


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## icebear

keithemo said:


> .....
> 
> It's only fraud if they _SPECIFICALLY_ said that those discs were created from high-resolution masters.
> (And neglected to mention, in the fine print, that those high-resolution masters were themselves made from low-resolution recordings.)
> ...


 
  Of course marketing always runs everything by legal to make sure that they can get away with it...
  


nick_charles said:


> LOL - that's setting the bar pretty low for probity


 
  
  


rrod said:


> There's quite a difference between having honest intentions and passing them by the legal department, and having nefarious intentions and passing them by the legal department.


 
  
  


interpolate said:


> I have nothing further to add.


 
  
* +1*


----------



## KeithEmo

nick_charles said:


> LOL - that's setting the bar pretty low for probity


 
  
 I don't disagree - but, then again, why should we expect audio companies to have any _HIGHER_ standards than any other major corporation selling a product. Even though we consider medicine to be sort of critical, and have a special agency to make sure we get to hear every claim, disclaimer, and side effect - you'll also notice that most of the late night "health remedies" make an end run around this by including some _VERY_ fine print saying that "they don't promise to treat any medical condition" - right after they spend the first three quarters of the commercial showing testimonials that sure sound like people claiming the exact opposite. Honestly, I would love to see some serious regulation of audio product claims, but we seem to live in one of the last few remaining corners of the wild west. While we're at it, maybe we should look into how truthful they are about how much better that $500 bottle of wine really tastes.
  
 Also, in the specific case of DVD-A and SACD, I wouldn't assume that there was any nefarious conniving going on. My guess would be that the engineers were convinced that both really are better (based on measurements which show that they are), and the marketing guys repeated this in their advertisements (like they were told to), and neither of those groups was directly involved in mastering the discs they were selling. In turn, the engineers who _WERE_ mastering the discs were probably told something like: "We want these masters converted to SACD/DVD-A at the best quality possible" - and they did their best to do just that. (And, of course, some philosophical types were so convinced that the format would "save the world from bad hi-fi" that they were willing to stretch the truth a bit to "convert the heathens".) I doubt anybody was actually told to lie, or that anybody actually sent a memo saying: "Let's sell them junk - they'll never know".
  
 Considering my personal experience, I would have to say I consider the guys who were saying that "SACD and DVD-A sound better than CDs" had more justification for making their claims than the guys who keep insisting that 256k streaming is "just as good as a CD".
  
 And, much as I hate to say it, I have very little sympathy for anyone who bought more than one SACD or DVD-A.... because they did so _AFTER_ having the opportunity to decide for themselves, so what they read in an ad really isn't much of an excuse. And, if their expectation bias caused the poor sods to actually get more enjoyment from that SACD, then just maybe they did get their money's worth.


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## icebear

keithemo said:


> ... *And, much as I hate to say it, I have very little sympathy for anyone who bought more than one SACD or DVD-A.... because they did so AFTER having the opportunity to decide for themselves, so what they read in an ad really isn't much of an excuse. *And, if their expectation bias caused the poor sods to actually get more enjoyment from that SACD, then just maybe they did get their money's worth.


 
  
 Somehow I wouldn't be surprised if like 5 or 10 years down the road, someone from Emotiva says the same about folks who bought high rez. DACs


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## arnyk

keithemo said:


> It's only fraud if they _SPECIFICALLY_ said that those discs were created from high-resolution masters.
> (And neglected to mention, in the fine print, that those high-resolution masters were themselves made from low-resolution recordings.)
> I'm sure all of the claims were "reviewed by the legal department" and found to _NOT_ "cross the line into false advertising or fraud".


 
  
 Strange coincidence - as the words danced around, you excluded the actual fraud that was actually committed.
  
 It was specifically said in many cases that the SACDs and DVD-as sounded better because they were on high resolution media.
  
 One example of many thousands:
  
http://store.sony.com/es-super-audio-cd-player-w-hdmi-output-zid27-SCDXA5400ES/cat-27-catid-EOL-Home-Theater-Speakers-Components
  
"Hear your music as you never have before. The SCD-XA5400ES Super Audio CD player delivers the highest standard in digital sound and clarity. It is the perfect complement to high quality, multi-channel receivers with uncompressed digital output for the Direct Stream Digital® signal."


----------



## old tech

Although skating close to the edge, that statement in itself does is not a misrepresentation. The Sony player does deliver higher quality digital sound and it does because the media format is (usually) mastered better. but it doesn't say the higher quality digital sound is because of the higher bit stream rate.


----------



## StanD

old tech said:


> Although skating close to the edge, that statement in itself does is not a misrepresentation. The Sony player does deliver higher quality digital sound and it does because the media format is (usually) mastered better. but it doesn't say the higher quality digital sound is because of the higher bit stream rate.


 
"Hear your music as you never have before." That implies it will play whatever I have better. So if I use one of my existing CDs, it will sound better?


----------



## old tech

Again, it is not specific enough to be misleading under law. It can imply a number of things, even the reality that to "hear your music as never before", you'll have to purchase it from a remastered source only available through the SACD media. Which of course can be played in the Sony player.

Just a disclaimer, I'm playing devil's advocate here. I think the marketing spiel is misleading in that they don't state specifically why the music may sound better. It is just not misleading in law. Put in another way, if someone or an Authority challenged Sony under consumer laws, Sony could easily demonstrate that SACDs (generally) played through their player will sound better than CDs. On the other hand, if Sony was making more specific claim that its SACD player sounds better than CDs because of higher sampling rates, DSD etc then it would be easier for someone to challenge them in a Court as there is direct technical/physiological and experimental evidence to refute it.

I think Sony's high end lawyers would have this area covered.


----------



## RRod

We've had some examples of shystery wording before on this very thread.
 http://www.head-fi.org/t/716822/why-24-bit-audio-and-anything-over-48k-is-not-only-worthless-but-bad-for-music/570#post_11750153


----------



## KeithEmo

stand said:


> "Hear your music as you never have before." That implies it will play whatever I have better. So if I use one of my existing CDs, it will sound better?


 
  
 Advertising is just - advertising. I don't see much of a qualitative difference between that claim and saying that "You'll enjoy your beer like never before in our new cut crystal beer mug". Of course, in the last 24 hours, I've seen TV commercials promising that "all natural cereal is better" (apparently because they squish their flakes so they look more like natural grain), and there's this makeup that "will make your skin look thirty years younger" (apparently because it uses an extract from some exotic melon that doesn't rot nearly as fast in the fields as regular ones), and let's not forget the one that is promising me "six pack abs in only 30 minutes three times a week". And, while we're at it, have you _EVER_ had a hamburger in a fast food restaurant that looked nearly as good as the one on the sign?
  
 My point is that, in order to be considered fraud, you have to make very specific factual claims that aren't true. So, does the XYZ-Super-Rez-disc-player make your CDs sound better? Well, perhaps in the opinion of the guy who designed it, it does. Was that claim misleading? Perhaps. Does it rise to the level of legal fraud? No way.


----------



## interpolate

I think the lab situations where an absolute null is achieved, the performance of the playback system will be unsurpassed or marred by involuntary inclusions of occlusions or interventions. In other words, you just can't predict the absolute outcome of every result. It's like Coffee, it can be the best in the region however if you are hopeless at making it then it'll likely won't be.


----------



## KeithEmo

icebear said:


> Somehow I wouldn't be surprised if like 5 or 10 years down the road, someone from Emotiva says the same about folks who bought high rez. DACs


 
  
 There's one _HUGE_ factor that everybody seems to be ignoring - specifically when we're discussing DACs......
  
 And that factor is simply that virtually every reasonably high quality DAC chip today that is "intended for audio use" supports 192k or higher. So, while it may be true that it costs considerably more to support sample rates above 44k with certain exotic designs (like R2R DACs, or those made with discrete components), and providing support for higher sample rates does require you to be more careful about following good design practices (which is always a good idea anyway), for the vast majority of companies selling DACs, it simply doesn't cost any more _TODAY_ to produce a DAC that supports 192k than to produce one that only supports up to 44k. (I'm ignoring the very few really low quality sub-$1 DAC chips still around that don't support the higher sample rates, but are also not considered to be good enough for audiophile products for other reasons as well.)
  
 So, if you don't believe that high-res files sound better, then simply don't buy them or listen to them (and that's the decision that might save you some money). And rest assured that, even though your DAC supports sample rates up to 24/192k, you didn't pay an awful lot for the capability you're not using.
  
 (Note that I'm not making any comment about whether, assuming there was no demand for high-res DACs a few years ago, perhaps the DAC chip makers would have spent their development money adding some other feature instead. I'm also not suggesting that "support for high-res files" isn't considered to be a "selling point"; we do have customers who specifically want that as a feature, and we're perfectly happy to provide it; and I'm sure some people have upgraded their "44k only" DACs so they could play high-res files. What I am saying is that, if everyone on the planet decided tomorrow that anything over 44k was a total waste of time, we would continue to use the same DAC chips we use now..... and it would be a purely marketing decision whether we painted over the lights for every sample rate above 44k or not. However, we would not be able to "offer you a DAC that was just as good, except that it didn't support anything above 44k, for less money". It simply doesn't work that way. )
  
 Personally, if I were shopping for a new DAC, I would only buy one that supported up to 192k. However, it wouldn't be a decision based on some specific belief that 192k is vastly superior - my reasoning is much simpler. I absolutely have a significant number of 192k files that sound clearly superior (to me) than their 44k counterparts. Without getting into that _OTHER_ discussion about whether they sound better _BECAUSE_ they're 192k or not, or whether I could convert them to 44k without hearing any difference, I consider my hardware to be "subservient to my software". In other words, I want a DAC that will play all the files I have without forcing me to convert them to suit its limitations. Since my current collection does include quite a few albums in 24/192k format, and only one of the player programs I use will convert files between sample rates automatically, that means that I "need" a DAC that supports up to and including 24/192k.
  
 (To put that a bit differently. When I bought my first car, air conditioning was a rather costly option, so I had to decide whether to spend an extra $1k to get air conditioning; and, when 44k DACs were the norm, and 192k DACs first came out, that high-resolution support was also an expensive upgrade. Nowadays, virtually every car comes with air conditioning standard, and most DACs support 192k, so neither of those is an important decision any more.)


----------



## KeithEmo

Quote:


old tech said:


> Again, it is not specific enough to be misleading under law. It can imply a number of things, even the reality that to "hear your music as never before", you'll have to purchase it from a remastered source only available through the SACD media. Which of course can be played in the Sony player.
> 
> Just a disclaimer, I'm playing devil's advocate here. I think the marketing spiel is misleading in that they don't state specifically why the music may sound better. It is just not misleading in law. Put in another way, if someone or an Authority challenged Sony under consumer laws, Sony could easily demonstrate that SACDs (generally) played through their player will sound better than CDs. On the other hand, if Sony was making more specific claim that its SACD player sounds better than CDs because of higher sampling rates, DSD etc then it would be easier for someone to challenge them in a Court as there is direct technical/physiological and experimental evidence to refute it.
> 
> I think Sony's high end lawyers would have this area covered.


 
  
 I agree.
  
_FRAUD_ is a very specific legal term - with a specific meaning (knowingly making a specific claim of fact, which you know to be untrue, for purposes of profit) and specific implications (you can get arrested or sued).
  
_BEING MISLEADING_ is something rather different..... I personally suspect that the right makeup will _NOT_ make the average fifty-year-old woman look thirty years younger, that I _WON'T_ look like the male fitness model in the ad if only I exercise "for thirty minutes three times a week", and that the hamburger I get at my favorite fast food place, while it certainly looks appetizing, won't ever look like the picture on the sign or in the TV commercial. I find all of those claims to be _misleading_, but they don't cross the line into fraud. (Apparently, even the ads attributing "psychic powers" to "psychic advisers" manage to avoid crossing that line.)
  
 In fact, I'm not at all sure that the wording of the claim you're quoting even crosses the line above "setting expectations for a good experience".
  
 I personally think it would be great if we had laws against "misleading advertising" (besides which, if you eliminated all the misleading commercials in an average TV evening, there wouldn't be many commercials left at all - which wouldn't hurt my feelings).


----------



## sonitus mirus

With regards to DACs and 24/192, most of the chips can handle 24/192, while the software is stuck in the past.  Even Windows 10 does not support USB Audio Class 2.0, so everyone that uses a Windows machine will still be required to install proprietary drivers to be able to play 24/192 in the native format.  No plug in play device will automatically be able to play 24/192 without special drivers.


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## StanD

keithemo said:


> Advertising is just - advertising. I don't see much of a qualitative difference between that claim and saying that "You'll enjoy your beer like never before in our new cut crystal beer mug". Of course, in the last 24 hours, I've seen TV commercials promising that "all natural cereal is better" (apparently because they squish their flakes so they look more like natural grain), and there's this makeup that "will make your skin look thirty years younger" (apparently because it uses an extract from some exotic melon that doesn't rot nearly as fast in the fields as regular ones), and let's not forget the one that is promising me "six pack abs in only 30 minutes three times a week". And, while we're at it, have you _EVER_ had a hamburger in a fast food restaurant that looked nearly as good as the one on the sign?
> 
> My point is that, in order to be considered fraud, you have to make very specific factual claims that aren't true. So, does the XYZ-Super-Rez-disc-player make your CDs sound better? Well, perhaps in the opinion of the guy who designed it, it does. Was that claim misleading? Perhaps. Does it rise to the level of legal fraud? No way.


 
 Perhaps at one point they'll have to add disclaimers at the end of high end audio adverts, just like pharmaceuticals. "Puchasing overpriced audio equipment can cause depression once you realize that you've been hoodwinked."


----------



## sonitus mirus

stand said:


> Perhaps at one point they'll have to add disclaimers at the end of high end audio adverts, just like pharmaceuticals. "Puchasing overpriced audio equipment can cause depression once you realize that you've been hoodwinked."


 
  
 It won't matter.  They can cherry pick the warnings that are emphasized and use small print and a fast-talking voice actor to gloss over the more worrisome issues.  "If listening to high quality music using our DAC causes an erection to last over 4 hours, be sure to seek medical attention, after you brag about it to your friends."  There could be a dozen or more health issues, and that is the one that they decide to spell out in all of their ads?  Good thing the audio industry does not have enough power and money to influence our government.  We see the effects on a smaller scale with this site, and the policies that are enforced in order to participate.


----------



## KeithEmo

sonitus mirus said:


> With regards to DACs and 24/192, most of the chips can handle 24/192, while the software is stuck in the past.  Even Windows 10 does not support USB Audio Class 2.0, so everyone that uses a Windows machine will still be required to install proprietary drivers to be able to play 24/192 in the native format.  No plug in play device will automatically be able to play 24/192 without special drivers.


 
  
 Sadly, that's specifically a Windows situation. Apple computers have had UAC2 support for quite some time, and some Linux distros now include it, but Microsoft still seems to insist that "there isn't enough interest". Note that UAC1 DACs, which work plug-n-play without a driver, can support up to 96k sample rates - just not 192k and above. And some current DACs allow the user the option of supporting up to 96k (UAC1) without a driver, or up to 192k - or above (UAC2) with a driver.
  
 From the vendors' point of view, drivers are a problem for two reasons....
  
 First, depending on how the design process is handled (contractually), the vendor may have to pay to have drivers written, or the USB drivers may be provided free-of-charge by the hardware vendor who sources their USB interface chips. (Obviously, in the first situation, supporting UAC2 sample rates will add some development costs.)
  
 Second, installing drivers is a nuisance for the end user, and adds support costs for the vendor. (While a DAC vendor is always going to get support questions about how to configure their DAC, requiring the end user to install drivers makes the process more complicated, and so makes for more support calls.)


----------



## KeithEmo

stand said:


> Perhaps at one point they'll have to add disclaimers at the end of high end audio adverts, just like pharmaceuticals. "Puchasing overpriced audio equipment can cause depression once you realize that you've been hoodwinked."


 
  
 I'd like to see disclaimers like that on a _LOT_ of ads.


----------



## Thad-E-Ginathom

keithemo said:


> There's one _HUGE_ factor that everybody seems to be ignoring - specifically when we're discussing DACs......


 
   
 There is one absolutely huge factor that people are now, ignoring in this thread. Go back to square one (or post one): these higher bit/sample rates are useless in every way other than audiophoolery.  Let's get back to basics!
  
 Whether or not I blame you and the rest of the equipment manufacturers for meeting demands for the fad is a moot point. Whilst I do think that all the fancy, ******** cable stuff really is nothing short of fraud, it is tough with equipment such as DACs. What would I do if I made them, and my living depending on supplying what people want? Looked at that way, it is a tough question.


> Personally, if I were shopping for a new DAC, I would only buy one that supported up to 192k. However, it wouldn't be a decision based on some specific belief that 192k is vastly superior - my reasoning is much simpler. I absolutely have a significant number of 192k files that sound clearly superior (to me) than their 44k counterparts. Without getting into that _OTHER_ discussion about whether they sound better _BECAUSE_ they're 192k or not, or whether I could convert them to 44k without hearing any difference, I consider my hardware to be "subservient to my software". In other words, I want a DAC that will play all the files I have without forcing me to convert them to suit its limitations.


 
  
 I also thin that, as a buyer, whether or not I accept that the high-sample-rate thing is ********, I am going to buy equipment that plays any music that is likely to come my way.  It is about playing music, and if I get a file that is high[-er] sample rate, I want to listen to it, right?
  
 I do blame the equipment manufacturers if and when they start leading the rush into lunacy, such as making DACs with DSD capabilites to play music that is not even available yet, but simply being able to play what is available is a different matter.
  
 Hands up manufacturers that refuse to play this game at all. Thank you Mr Lavry (if you still manufacturing digital equipment for up to 24/96 only?) and thanks for your honesty to your principles and your engineering principles --- but this is a niche market, and we are not about to see Lavry DACs in high-street stores.


> So, if you don't believe that high-res files sound better...


 
 What sad state of affairs that the hifi industry _is_ now belief-based!
  
 Back, as I said, to square one on this: there _are_ facts!
  
 EDIT: I seemed to have used a word which I regard as every-day English, but the bots don't like. It wasn't a _very_ bad word. The ****s probably make it look much worse that it was!


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## arnyk

keithemo said:


> There's one _HUGE_ factor that everybody seems to be ignoring - specifically when we're discussing DACs......
> 
> And that factor is simply that virtually every reasonably high quality DAC chip today that is "intended for audio use" supports 192k or higher.


 
 The above appears to just be more self-aggrandizing speculation on your part, Keith. How do you know what percentage of audiophiles have missed this rather obvious and old point? Market survey? Opinion survey? Personal bias?


----------



## arnyk

keithemo said:


> Sadly, that's specifically a Windows situation. Apple computers have had UAC2 support for quite some time, and some Linux distros now include it, but Microsoft still seems to insist that "there isn't enough interest". Note that UAC1 DACs, which work plug-n-play without a driver, can support up to 96k sample rates - just not 192k and above. And some current DACs allow the user the option of supporting up to 96k (UAC1) without a driver, or up to 192k - or above (UAC2) with a driver.
> 
> From the vendors' point of view, drivers are a problem for two reasons....
> 
> ...


 
  
 Looks like a lot of hand-wringing over nothing. I have a number of 24/192 DACs ranging in price over the range from $50 to about $1K, and they all came with windows 24/192 drivers at no extra change.


----------



## KeithEmo

arnyk said:


> Looks like a lot of hand-wringing over nothing. I have a number of 24/192 DACs ranging in price over the range from $50 to about $1K, and they all came with windows 24/192 drivers at no extra change.


 
  
 I don't see any "hand wringing" at all. (And you're agreeing with my main point - which is that nobody is "getting anybody to pay a lot extra for high-res support in a DAC when it may not be audibly superior" - because you aren't paying a lot extra for it at all. About the only "big deal" is having to include a driver - which isn't a big deal at all.)


----------



## RRod

keithemo said:


> Considering my personal experience, I would have to say I consider the guys who were saying that "SACD and DVD-A sound better than CDs" had more justification for making their claims than the guys who keep insisting that 256k streaming is "just as good as a CD".


 
  
 And considering my own blind tests on my own equipment and music, I think they all sound the same if they're made from the same original file. But the streaming people are asking ~$100–200 per year for access to entire catalogs, and the SACD/DVD-A people were asking $20–30 per disc, justifying the extra cost by "better sound" that may not even have come from a different master, but they weren't going to tell you that. I know who I'll put on the nice and naughty lists.
  
 People claiming that things like 256AAC are transparent for music can justify their claims by the boatloads of blind testing that go into developing and testing the perceptual models and filtering algorithms. That's not nothing.


----------



## KeithEmo

sonitus mirus said:


> It won't matter.  They can cherry pick the warnings that are emphasized and use small print and a fast-talking voice actor to gloss over the more worrisome issues.  "If listening to high quality music using our DAC causes an erection to last over 4 hours, be sure to seek medical attention, after you brag about it to your friends."  There could be a dozen or more health issues, and that is the one that they decide to spell out in all of their ads?  Good thing the audio industry does not have enough power and money to influence our government.  We see the effects on a smaller scale with this site, and the policies that are enforced in order to participate.


 
  
 Yup. (And isn't that a  great example of turning a "warning" into a "selling point".)
  
 However, interestingly, there was a point in history when the advertising practices of the audio industry were in the spotlight, and the government _DID_ step in to enforce "truth in advertising". Back "in the bad old days", there was no consistent way of rating amplifier power. Most reputable companies used what we now call "RMS power", but many "less reputable ones" did not. Instead, they used ratings like "instantaneous peak power". (What that meant was that, literally, they would take the output section of a power amp, connect it to a separate bench power supply and a power meter, turn the power up until it flamed out, and the highest number they could read through the smoke was "the IPP power rating".) It wasn't uncommon for a unit that delivered 25 watts/channel RMS to be rated "1000 watts per channel - IPP" - and they might or might not bother to put "IPP power" in very small print.
  
 As it turns out, the reputable companies found this practice offensive, and they didn't want to be in the position of having to lie to be competitive - so "complaints were files" and finally the Federal Trade Commission (FTC) stepped in and produced a standard for how amplifier power _MUST LEGALLY_ be measured. This standard was so specific that it even covered your concerns about how clearly the ratings were stated. The standard actually made it a legal requirement that, if a power specification was included, the RMS power rating must be the one printed in the largest font, and, any other alternative ratings, if they were also included, were legally required to be shown in smaller print. It also included requirements that the amplifier be "preconditioned" before the test (warmed up), that it must be able to deliver the rated power for a specified amount of time, and that you couldn't connect it to an external power supply to run the test.
  
 This standard is still in effect, although it ONLY applies to "home stereo equipment" (it applies somewhat differently to "home theater equipment"; and not at all to "portable equipment" and "car stereo equipment"). If you Google "FTC Power Ratings" and "1972" I'm sure you can find all the details (and the new "home theater revisions").


----------



## KeithEmo

arnyk said:


> The above appears to just be more self-aggrandizing speculation on your part, Keith. How do you know what percentage of audiophiles have missed this rather obvious and old point? Market survey? Opinion survey? Personal bias?


 
  
 You are quite right. I have no idea of the overall percentage. All I can state with certainty is that anyone who insists on complaining about how they were "scammed" by being charged a lot more for a DAC which includes "useless" support for high-resolution file formats doesn't know..... because, if they did, they would realize that they _DIDN'T_ pay a lot extra for it. (Let me rephrase that: It doesn't cost very much to include that feature; what the company they bought their DAC from charged for it, and they obviously agreed to pay, is between them and their accountant 
	

	
	
		
		

		
		
	


	




 )
  
 My point remains.... Adding "high-res file support" to a DAC is like adding an extra side rear view mirror to a car: some people won't use it - some will; some can't live without it - and some don't care at all; and, as long as some people want it, the car manufacturers will probably include it. (And, more to my point, I don't know too many people who insist on complaining about how they were "forced" to buy an extra mirror they don't use - it just isn't that big a deal.)
  
 The discussion _HERE_ certainly leads me to believe that many of the participants believe that "high-res file support" is an expensive feature that they have "paid extra for" - which is the mistaken impression I was correcting.


----------



## interpolate

Buying Vinyl or DSD quality recordings is also expensive, so you got to weigh it up whether buying a CD version as opposed to around the same when bought as AAC or MP3 format is good value for money. Compressed versus uncompressed PCM (or indeed dictionary compression on lossless versions) is a very moot point and will divide many people's opinions. 320K MP3/FLAC is where it is at for me right now. Although my preference would be CD or something physical and tangible. Simply because I started out with vinyl/cassette. 
  
 Whilst their sound quality wasn't always the best, modern mastering should have improved the process. I like both however much rather have a raw copy also.


----------



## sonitus mirus

keithemo said:


> Spoiler: Full Quote:
> 
> 
> 
> ...


 
  
 I do recall receiving tips when purchasing a car stereo to use RMS ratings when deciding on a amp.  I was taking advanced electronics training courses in the Navy at the time, and I understood the math behind it, but only enough to get me in trouble.


----------



## KeithEmo

rrod said:


> And considering my own blind tests on my own equipment and music, I think they all sound the same if they're made from the same original file. But the streaming people are asking ~$100–200 per year for access to entire catalogs, and the SACD/DVD-A people were asking $20–30 per disc, justifying the extra cost by "better sound" that may not even have come from a different master, but they weren't going to tell you that. I know who I'll put on the nice and naughty lists.
> 
> People claiming that things like 256AAC are transparent for music can justify their claims by the boatloads of blind testing that go into developing and testing the perceptual models and filtering algorithms. That's not nothing.


 
  
 I think you're exaggerating - just a bit. At least a few of the folks who sell high-res files actually _DO_ provide the details about what the file was mastered from. There was a two-page article describing in excellent detail exactly how the 24/192k re-release of the Grateful Dead studio albums was re-mastered. You can also usually find reviews from other people who've bought a specific release after it's been out for a while. (And, yes, some vendors don't provide any details, and a few have even been caught lying about provenance  occasionally - it happens in any industry - but it isn't typical.)
  
 Streaming services are like cable TV - if you don't like what they're offering then you shouldn't subscribe (or you should get that "month free trial"; some high-res download sites actually do give out a few free sample files so you listen for yourself). I can tell you for sure that the picture I get on cable isn't nearly as good as a well-recorded Blu-Ray disc... but the HD service I have does look a lot better than the SD service I used to have. And an awful lot of people insist that _THEY_ do hear an obvious difference between SACDs and CDs (I personally do not). In fact, you'll find whole threads - here and elsewhere - dedicated to that very topic. I really don't see this as any different than someone who buys a $200 bottle of wine based on some critic claiming it's worth it. As far as I'm concerned, if he doesn't taste a difference, then he shouldn't by it (or, at the very least, he shouldn't buy another bottle after he tastes the first one). There are people in this thread who act like buying a few high-res files is equivalent to buying a $500k house, and finding out after the paperwork is signed that it's on a flood plain, and is infested by giant radioactive scorpions.... and that selling high-res files is equivalent to selling that house to an unsuspecting old lady. 
  
 I personally don't think that a $500 bottle of wine tastes any better than a $25 bottle of wine... so I don't buy them. However, I _DO_ find that I hear a difference with many high-res remasters... so I _DO_ buy some of them. However, I don't mind if other people do buy $500 bottles of wine, and I also don't mind if some of my friends are perfectly satisfied with 256k AAC - and I don't think the folks who sell either one are "cheating anyone".
  
 If _YOU_ don't think that the expensive files sounds better, then _YOU_ shouldn't buy them.
  
 (You're also exaggerating the situation with 256k AAC/MP3 files. The actual claim is that "most people don't notice a significant difference with most music" - which was the goal in the first place. This is not at all the same as claiming that they are "audibly transparent". If you check out any technical description of how perceptual encoding works you will find it sprinkled with terms like "mostly" and "usually" and "typically inaudible"; I've never seen a technical description that claims the process to be "absolutely inaudible to any human under any circumstances". In fact, I have only met one or two people who were actually unable to hear any difference between the 256 MP3 and a 44k uncompressed versions of specifically chosen files. However, I know plenty of people who consider the fact that they can fit ten times as many MP3 files on their portable player to be a lot more important than the fact that they hear what they consider to be a slight difference on a few songs. )


----------



## RRod

keithemo said:


> I think you're exaggerating - just a bit. At least a few of the folks who sell high-res files actually _DO_ provide the details about what the file was mastered from. There was a two-page article describing in excellent detail exactly how the 24/192k re-release of the Grateful Dead studio albums was re-mastered. You can also usually find reviews from other people who've bought a specific release after it's been out for a while. (And, yes, some vendors don't provide any details, and a few have even been caught lying about provenance  occasionally - it happens in any industry - but it isn't typical.)
> 
> *You're just supporting my point. Buyers have often had to go to 3rd party reviews/analysis to determine exactly what they are getting with hi-res stuff, and it's not always a guaranteed improvement even in the mastering.*
> 
> ...


----------



## KeithEmo

RRod said:
			
		

>





> Quote:
> 
> 
> 
> ...


----------



## RRod

A few comments:
  
 .The current situation for me as an informed consumer is that I can't trust a CD release (in certain genres) to not have additional things done to it. The only way to be sure is to scour the internet for reviews and comparisons, so in the end it can be easier to just get the hi-res version (which is less likely to be mucked with) and do my own conversions. Unfortunately a lay consumer doesn't know how to do this, and is less likely to even know to look up comparisons of masters. At some point there is simply too much caveat put on the emptor for me to declare the system as fair and reasonable.
  
 .Comparing an un-vetted mp3 to the CD is the kind of uncontrolled listening that shouldn't count towards evidence towards the inferiority of the smaller format.
  
 .I agree that a cultural shift towards hi-res is fine and already feasible with the hardware. The thing is, where does it stop? Today it's 24/192. When will companies decide they want more money and re-release at 32/384 with similar promises of "better sound"? I know it's not necessarily a slippery slope, but companies have already unabashedly pushed out products with "improvements" that people just don't seem to hear in blind tests. They're basically running on the "bigger number has to be better" logic that some on here try to push.
  
 .The argument about keeping the "best" version is fine, but it's simply not true that a given hi-res master will actually be an improvement over any given Redbook release, especially if the first release was back before all this brickwalling nonsense. Again, can't trust nobody in this business!
  
 .The trickery I see is that companies *are* saying things will sound better in hi-res. No conditions, just straight up statement as fact, and thus people buy the stuff for extra coin. Then sweet, sweet expectation bias does the rest. Maybe that's a capitalistically fair system, but to me it stinks. If they're going to say things like "With this new hi-res version you will hear purity of tone that is impossible to get with a CD (which has a certain 'hardness' to the sound)," then it ought to actually be true.


----------



## arnyk

rrod said:


> The current situation for me as an informed consumer is that I can't trust a CD release (in certain genres) to not have additional things done to it.


 
  
 That's optimistic as heck!
  
 The current situation for us as informed consumers is that we can't trust _*any recording*_ to not have additional things done to it. 
  
 The only way out that I know of is to make your own recordings from scratch.


----------



## arnyk

keithemo said:


> I don't see any "hand wringing" at all. (And you're agreeing with my main point - which is that nobody is "getting anybody to pay a lot extra for high-res support in a DAC when it may not be audibly superior" - because you aren't paying a lot extra for it at all. About the only "big deal" is having to include a driver - which isn't a big deal at all.)


 
  
  


stand said:


> Perhaps at one point they'll have to add disclaimers at the end of high end audio adverts, just like pharmaceuticals. "Puchasing overpriced audio equipment can cause depression once you realize that you've been hoodwinked."


 
  
  
 More to the point:
  
 Purchasing overpriced audio equipment can lead to ludicrous denials of reality once you realize that you've been hoodwinked, but don't want to admit to having made a mistake.


----------



## arnyk

keithemo said:


> All I can state with certainty is that anyone who insists on complaining about how they were "scammed" by being charged a lot more for a DAC which includes "useless" support for high-resolution file formats doesn't know..... because, if they did, they would realize that they _DIDN'T_ pay a lot extra for it. (Let me rephrase that: It doesn't cost very much to include that feature; what the company they bought their DAC from charged for it, and they obviously agreed to pay, is between them and their accountant
> 
> 
> 
> ...


 
  
 The reason why we don't know what the _*high resolution tax*_ on the recordings and equipment that we buy is, is because it is a big secret.
  
 In the realm of consumer home audio, hi-rez mania is so pervasive that it is almost completely impossible to choose to buy anything less than a 24//192 DAC.  
  
 It is impossible to buy just about any modern recording that doesn't have a hi resolution master in its provenance.
  
 Virtually every home audio component has hi rez DACs in it by default.   This is often true whether the device can possibly play hi rez recordings or not.
  
 However, any well-informed consumer knows that there is no such thing as a free lunch, no matter how it is denied by hi rez advocates.


----------



## KeithEmo

arnyk said:


> The reason why we don't know what the _*high resolution tax*_ on the recordings and equipment that we buy is, is because it is a big secret.
> 
> In the realm of consumer home audio, hi-rez mania is so pervasive that it is almost completely impossible to choose to buy anything less than a 24//192 DAC.
> 
> ...


 
  
 What you're saying is indeed somewhat true - but I still think you're placing too much emphasis on it.... and here's why.
  
 If we limit the discussion to stereo parts, the bulk prices of "high performance audio DAC chips" range from about $2 to about $20, so that's the most you could save by using a cheaper part. (I'm talking about the current "high performance audio DAC" parts from AKG, Analog Device,s, TI, and ESS/Sabre used in the vast majority of consumer and audiophile equipment - and I'm excluding "exotic parts". Both the parts most manufacturers use, and the pricing for those parts, are readily available online.) And, as I mentioned in another post, while you do have to design more carefully to accommodate the higher bandwidth required for the higher sample rates, the additional care spent in design isn't that big a deal, and it has other benefits anyway. In most cases, if a company had a market for 1000 of 44k DACs and 1000 of 192k DACs, equal in everything except sample rate performance, the economies of scale would be such that it would usually literally be cheaper (per unit) to make 2000 of the higher performance units than to "split the SKU" (which is manufacturer-speak for making two different products, which use different parts, and have to be kept track of and warehoused separately) and make two separate products.
  
 And, in the mastering and recording process there are _OTHER_ benefits to using the highest resolution available for mastering and editing. Specifically, by using higher bit depths and sample rates for mastering, you avoid loss of quality from rounding errors and cumulative errors, as well as giving yourself a lot more headroom for adjusting levels during mixing and editing without having to worry about clipping and noise. So, even if they were to decide to only release music in 16/44k, most recording engineers would _STILL_ choose to master at 24/96, or even 24/192k anyway. (In other words, it still makes sense to master and edit at higher resolutions and bit depths - even if you plan to release in 44k.)
  
 I really think it's reasonable to equate a 192k DAC to a car with fuel injection. You probably can't find many cars today that still use a carburetor, but there's no real drawback to fuel injection, and lots of benefits, and the costs of inventing and developing fuel injection have already been spent, so, regardless of what might have been, and of whether you need the extra performance or not, fuel injection simply makes sense with today's technology.... and DACs that support 192k are pretty much at the same point in their life cycle.
  
 (Again, this is really a totally separate issue from whether it makes sense to pay extra for high-resolution downloads or discs.)


----------



## interpolate

I've only bought a few hi-res remasters and copies of music which are over 44.1K and I would say the difference is enough to justify the extra cost alas I wouldn't discard Redbook quality just yet.


----------



## MacacoDoSom

keithemo said:


> arnyk said:
> 
> 
> > The reason why we don't know what the _*high resolution tax*_ on the recordings and equipment that we buy is, is because it is a big secret.
> ...


 
 If most recording engineers still choose to master at 14/96 or 24/192 these recordings should be a lot cheaper than a CD no time spent on downsampling, no case, no logistics other than the server, no transportation, no manufacture etc, just take the files and make a PDF with a photo, as in most cases no credits, nothing written, just the artist name and the album name...
 This has nothing to do with cars... or carburetors, it's pure robbery... I don't think *arnyk *was talking about DACs... and I wonder why most professional DACs only go until 24/96?  and the ones that go up to 196 can't use SPDIF/TOSLINK at that rate?


----------



## arnyk

keithemo said:


> What you're saying is indeed somewhat true - but I still think you're placing too much emphasis on it.... and here's why.
> 
> If we limit the discussion to stereo parts, the bulk prices of "high performance audio DAC chips" range from about $2 to about $20, so that's the most you could save by using a cheaper part. (I'm talking about the current "high performance audio DAC" parts from AKG, Analog Device,s, TI, and ESS/Sabre used in the vast majority of consumer and audiophile equipment - and I'm excluding "exotic parts". Both the parts most manufacturers use, and the pricing for those parts, are readily available online.) And, as I mentioned in another post, while you do have to design more carefully to accommodate the higher bandwidth required for the higher sample rates, the additional care spent in design isn't that big a deal, and it has other benefits anyway. In most cases, if a company had a market for 1000 of 44k DACs and 1000 of 192k DACs, equal in everything except sample rate performance, the economies of scale would be such that it would usually literally be cheaper (per unit) to make 2000 of the higher performance units than to "split the SKU" (which is manufacturer-speak for making two different products, which use different parts, and have to be kept track of and warehoused separately) and make two separate products.


 
  
 Obviously, there's a lack of understanding of how audio gear is priced as compared to parts cost. Contrary to the naive statements above, using a $20 DAC chip in a piece of finished gear does not add a mere $20 to its retail price. Reality might be more like $ 60-120.


----------



## jcx

as a product developer I had some issues with the single parts cost multiplier pricing model - really is "too simple"
  
 for boutique small lot product the development time is everything - if you need a DSP function that requires a man year sw dev on low cost hardware or a $20 more expensive chip cuts it to a month or 2 then your product should be cheaper and faster to market with the higher horsepower, more costly chip rather than amortizing $100k salary over a few hundred pieces per year


----------



## Thad-E-Ginathom

interpolate said:


> I've only bought a few hi-res remasters and copies of music which are over 44.1K and I would say *the difference* is enough to justify the extra cost alas I wouldn't discard Redbook quality just yet.


 
  
 What difference? Did you make a comparison with the same material at 44.1K?
  
 Forgive me if I am unjustly including you among the many, many who post to the effect that their "High-Res" music sounds wonderful so _High-Res must be true,  _without having made _any, _not even sighted, comparison to a 44.1 copy.
  
 If you did compare, how do you know that the "difference" is attributable to the sample rate, rather than some other factor such as simply better mastering?
  
 Been there, done that: exclaimed how wonderful this 96K music is --- and realised that _so is my old CD collection,  _and no, there is no hidden toe-tapping musicality in higher bit rates that is not there in the 44.1 music. And more detailed testing of identical material (not even ABX) showed me that the oh-wow-extra-detail _was there in the 44.1 too.  _Every time.
  
 Mis-named "high-res" flies on gullibility. If the hifi community was not so open-mouthed, feed-me, obsessive, it would never float, let alone fly. 
  
 I prescribe a large dose of cynicism for us all, taken daily. It might even make those famous ears of ours a little more _trustworthy! _


----------



## Slaphead

interpolate said:


> I've only bought a few hi-res *remasters* and copies of music which are over 44.1K and I would say the difference is enough to justify the extra cost alas I wouldn't discard Redbook quality just yet.




And there is the key word - remaster.

I do some music production in my spare time, and as it's a hobby I naturally read up on it as well, so, while I would certainly not consider to myself to be an expert, I do know how to use the most common tools used in the mastering process, and use them relatively effectively.

The things that you can do with these tools can be astounding if done right and done subtly.

For instance using a plain old parametric EQ and creating a narrow 2 - 3db peak at around 14-15KHz will give a track a touch more air.

A stereo widener will give a bigger sound with more apparent soundstage, but overdoing it will introduce phasing problems which can be uncomfortable to listen to.

Using a multiband compressor, you can bring out micro details in various frequency ranges that would previously be masked by human hearing limitations (as a general rule you can't hear anything that's less than minus 30db from the current loudest sound)

A full range compressor with a modest compression ratio can help pull the track together. Again less is more here

Having a limiter at the end of the effects chain and set so it just tickles the point where the limiter kicks in can give a cohesiveness to the sound - we're not talking Death Magnetic brickwall limiting here, just a subtle implementation.

You can use a transient shaper to tame those overly sharp signals, give a track a touch more punch, or make it a little more laid back.

If you have access to the original separate tracks then you have a lot more latitude and precision on how you can apply these effects.

These things, when done subtly at a singular level will often be barely audible, but taken together they can bring a vast perceived improvement.

OK, so where does this fit into this thread. Well, if you were to take a master and from that master produce a Hi Res copy and a redbook copy they will sound identical. There will be no audiable difference between the two. 2 reasons. The first is that if you null those tracks (Normalise, invert one track and then sum) there will be differences, but those differences will be at or near the noisefloor of the analog section of your DAC let alone the noisefloor of the subsequent amp sections the signal will be pushed through. The second is that transducers even the top end ones can't even follow a 16/44.1 signal accurately - you only have to look at the response charts to realise this, although I would have to admit that the squarewave test is particularly punishing.

So, how do you sell a new and improved format where you can't hear the difference between that and the old format? The answer is that you make it sound different and that's where the mastering process comes in.

I played a particularly nasty trick on a friend of mine by "remastering" one of his favourite tracks. I then got him to listen to the normal version and then my hi-res version. He was so impressed with the hi-res version that I had to stop him dumping money on a new hi res DAP by coming clean with him. The original track was indeed played in WAV format at 16/44. However, my "hi-res" track was compressed to a 320Kbps MP3 just to make a point. He's still not forgiven me for that.

TL;DR

It's all in the mastering - nothing else.


----------



## RRod

You obviously don't seem to realize that dithering will do way more to the music than any of that


----------



## Slaphead

Oh, I do , not to mention shaping that noise that dither produces as well


----------



## KeithEmo

macacodosom said:


> If most recording engineers still choose to master at 14/96 or 24/192 these recordings should be a lot cheaper than a CD no time spent on downsampling, no case, no logistics other than the server, no transportation, no manufacture etc, just take the files and make a PDF with a photo, as in most cases no credits, nothing written, just the artist name and the album name...
> This has nothing to do with cars... or carburetors, it's pure robbery... I don't think *arnyk *was talking about DACs... and I wonder why most professional DACs only go until 24/96?  and the ones that go up to 196 can't use SPDIF/TOSLINK at that rate?


 
  
 No, actually it's capitalism, and "supply and demand" - which states, in simple terms, that you will be charged _WHATEVER YOU'RE WILLING TO PAY_. Now, with most products in a capitalist economy, there's a factor that holds down costs - competition. If your local high-end steak house decides to charge $250 for a filet mignon, there's probably one next door who will be happy to take a little less profit, and sell theirs for $150, and one next to them who'll do it for $100 - until, finally, you reach a point where someone offers one for $22.95 and makes little enough profit that nobody else thinks it's worth selling for less. With popular artists, each artist is usually signed with one label, so, at least for his or her latest album, there is a monopoly - if you want _THAT_ album, you'll pay whatever they're asking - or you'll do without. (It's not really much different than the reason that some people pay thousands of dollars for a Superbowl ticket when you can probably see plenty of excellent college and local games for free or close to it). In fact, there's even music you can legally download for free. You could also wait until that new CD is ten years old, and buy a copy used for $2. You're paying a premium for getting _specific_ music; at a _specific_ time; just like you pay cable extra to see your favorite shows when you want to see them. Classical music lovers also have the benefit of being able to choose between dozens (or sometimes hundreds) of different versions of a classical piece, many of which can also be had cheap or for free.
  
 The actual costs of manufacturing a physical CD disk are pretty low. Even in small quantities (500), the cost of having CDs pressed, with the jewel cases, and the inserts, all packaged for you, is under $2, and the price goes _way_ down from there in larger quantities... so you're not paying for the plastic. And, while it may require some expertise to choose the correct settings, and some folks prefer to do it using one program or another, the actual process of "downsampling" requires three or four mouse clicks, and takes a few seconds per track. (As in "export: sample rate:44k: depth:16: dither:t5: click: done".)
  
 There are several reasons why many professional DACs currently only go up to 96k. First, for anyone using Windows and USB, drivers are a nuisance. Second, most pros don't buy new equipment every year, and a lot of pro equipment remains in use for a very long time - and so most equipment currently in use was built and bought back when 96k _WAS _the highest sample rate available. Unlike consumers, pros don't always update their equipment every time a new feature comes out - they prefer to stick with equipment once they find stuff they like. This also ties back to drivers since drivers need to be updated when new versions of Windows come out, which makes updating drivers an ongoing project for equipment manufacturers, and increases the likelihood that a product will be "orphaned" if the manufacturer doesn't update the drivers to support the latest version of Windows - or whatever. This won't ever happen with a 96k UAC1 DAC.
  
 (My personal prediction is that you'll see a lot more pro DACs that support 192k very soon after Windows starts supporting it without separate drivers.)  
  
 As for supported sample rates, you've got that a bit turned around. Most recent DACs do in fact support up to 24/192 via electrical Coax (S/PDIF), and the Toslink (optical S/PDIF) inputs on lots of equipment do in fact support 24/192k. However, the Toslink outputs on most computers and computer-based devices top out at 96k. Also, to be honest, most industry folks consider Toslink to be "a low end consumer format" and simply aren't willing to spend the design effort, or the few cents more for a better part, to add 24/192k support to Toslink. (If your device has both electrical Coax and Toslink S/PDIF inputs or outputs, they share most of the same circuitry - with only a few different parts between them - a few pieces of actual output drive circuitry and the connector itself.) Since most high-resolution files are played on a computer, and most people who play those files on a computer use USB, the USB circuitry tends to "get all the attention" in terms of high-res support. (This is generally justified; there are technical reasons why asynchronous USB usually performs better - at least when your source device is based on computer technology.)
  
 And, I'm sorry, but "robbery" is when you _TAKE_ someone's money, or _FORCE_ them to give it to you. Since nobody's forcing anyone to buy music, in any format, the most it can be is "not a very good deal" - which, since we're all capitalists here, is up to the consumer to decide. (And there are a few groups - like Radiohead - who agree with that sentiment - and sell their music cheap - or even free - via download. If you want to change the system, then you can do so quite simply - by supporting the vendors you like and _NOT_ supporting those you don't.)


----------



## KeithEmo

arnyk said:


> Obviously, there's a lack of understanding of how audio gear is priced as compared to parts cost. Contrary to the naive statements above, using a $20 DAC chip in a piece of finished gear does not add a mere $20 to its retail price. Reality might be more like $ 60-120.


 
  
 When you look at the overall production cost and price of consumer products, that may in fact turn out to be true, but it's not nearly that simple - and the situation also isn't "symmetrical". 
  
 The reality is that there's what's called a "BOM cost"; that's the price of all the parts. Then there are other costs, like "pure" overhead, production, support, and shipping. Now, it may well be that a certain company may often price their products at "5x BOM cost" - but that's because, based on experience, they've found that _FOR THEM AND THEIR PRODUCTS_ it usually works out that way, so they use that number as a shortcut.
  
 In general, the cost of adding or changing parts is a combination of the cost of the part itself, the cost of modifying the design to use it, and the cost of changing production to use it. But, when you're designing a new product, choosing to use a more expensive, but otherwise equally simple or difficult to use part, probably won't affect the _COST_ by much more than the cost of the part itself. So, to "upgrade" a part with one that costs $10 more may well increase the price of the product by $100 (or it may not increase it at all). However, replacing a part with one that costs $10 less may also _INCREASE_ the cost, because the change still requires development and production changes, which probably cost more than the difference in the cost of the parts themselves. In the specific case of DAC chips, the cost of designing and building the circuitry associated with the DAC chip completely overshadows the cost of the DAC chip itself. 
  
 Now, of course, so far we're talking about production cost. Especially when it comes to "audiophile products", the cost to produce a product may have very little to do with its price, which is often based more on "perceived value" (which is simply another way of saying "whatever people are willing to pay for it" or "what the market will bear" - Economics 101 again).
  
 Therefore, by including a "premium" part that costs $10 more, a company may increase the "perceived value" of their product which, in turn, may allow them to charge $100 more for that product. However, make no mistake that the price increase is based mostly on the increase in the perceived value of the product and not on the cost of the part.


----------



## Thad-E-Ginathom

All this pontification about economics, society and the consumer has nothing at all to do with the technical superiority _or otherwise,_ of that product that the industry falsely labels "high-resolution" digital audio.


----------



## KeithEmo

thad-e-ginathom said:


> All this pontification about economics, society and the consumer has nothing at all to do with the technical superiority _or otherwise,_ of that product that the industry falsely labels "high-resolution" digital audio.


 
  
 Good point......
  
 The only place I can see where it is sort of tangent to the actual issue is when the discussion turns to whether it's "some sort of insidious plot to foist high-res audio on the innocent masses to sell more products" or simply a new and interesting technology that not everybody seems to agree is or isn't worthwhile.
  
 (The only relevance there being whether some - or all - of the people who think they hear a difference have really been duped into imagining it by the evil conspirators.)


----------



## jcx

no conspirators needed for human institutions, societies, voluntary associations to fail to pay attention to relevant Science on many issues
  
 of course suppressing discussion, restricting opportunities for education on Psychoacoustics, EE facts about consumer electronics and various human perception and reasoning fails is a decision


----------



## interpolate

DAC's still need programming to work within a circuit, it's not like installing memory or something self-automated with a SPD. As far as I understand it, manufacturers need to cover R&D, development outlay and the production costs in the first batch of units.
  
 RME Audio and Prism show how to do things like the big boys and you get what you pay for.
  
 I see Qualcomm Snapdragon 820 SoC has now been unveiled which might not mean much to us minions, However to electronic people it does mean better products.
  
 You can't really improve your hearing without other technology like a hearing aid.


----------



## Thad-E-Ginathom

keithemo said:


> Good point......
> 
> The only place I can see where it is sort of tangent to the actual issue is when the discussion turns to whether it's "some sort of insidious plot to foist high-res audio on the innocent masses to sell more products" or simply a new and interesting technology that not everybody seems to agree is or isn't worthwhile.
> 
> (The only relevance there being whether some - or all - of the people who think they hear a difference have really been duped into imagining it by the evil conspirators.)


 
  
 In other words, whether it is all a scam or not.
  
 It is.
  
 Which is _not_ to fling such accusations at guys like you on the equipment side: In this environment, you probably don't have a lot of choice but to supply what people want. But, when it comes to snake oil, it is not just the buyers that swept up in the rush.
  
 Mass hysteria is, by definition, _catching._ The madness grows.


----------



## icebear

In recent weeks I bought a couple of K2HD remastered CD.
 Some directly by JVC the proprietor of the process, some by Sony having the actual remastering job done by JVC.
 Examples are :
 "Passion, Grace&Fire" McLaughlin/DiMeola/Lucia (Sony)
 "Friday Night at SanFrancisco" McLaughlin/DiMeola/Lucia
 "Getz/Gilberto" (Lim)
 "Take Five" Brubeck (Sony)
 "Tchaikovsky #1 /Rachmaninoff #3" Argerich (Decca/Universal)
  
 These are clearly better than the regular CD issues BUT these are the same format i.e. the old fashioned 16bit/44.1kHz.
 Why especially Sony is showing off the capability of the red book format if the SACD / DSD is supposedly superior is beyond me.
  
 The explanations of the about 10 year old K2HD mastering process look at lot like the latest and greatest Meridian approach (MQA conveniently with built in DRM
	

	
	
		
		

		
		
	


	




) just that for MQA you will again need new equipment to decode the maximum quality and the K2HD CDs are spinning on any CD player. There are also xrcd editions by JVC, not sure how this again is different from K2 but still no special equipment necessary, all work within the redbook standard.
  
 For me these are proof that the redbook format can sound fantastic when the label wants it to sound like that.
 Whatever might be necessary or convenient at the production stage for recording, mixing and mastering, for playback anything beyond redbook is "nice to have" at best.


----------



## XenHeadFi

icebear said:


> In recent weeks I bought a couple of K2HD remastered CD.
> ...
> These are clearly better than the regular CD issues BUT these are the same format i.e. the old fashioned 16bit/44.1kHz.
> Why especially Sony is showing off the capability of the red book format if the SACD / DSD is supposedly superior is beyond me.


 
  
 Run it through the Dynamic Range program or look up the Dynamic Range of the mastering on http://dr.loudness-war.info/ . Researching and learning about DR was very enlightening on underlying reasons of certain hot-potato topics. Many of them can be answered simply by "Which was mastered in a sane way?"


----------



## Slaphead

Just as a follow up to my previous post on mastering. If any of you have an iPad (fairly recent recommended) and up to 20 bucks to spare you could have a go at doing some mastering yourselves.

Take a look at:

Final Touch from Postive Grid - $19.99

and

Audio Mastering by Igor Vasiliev - $12.99

Ok, these aren't going to be top grade mastering solutions, and if you push them too hard they'll let you know with crackles and breakup, but they can work surprisingly well.

You just load in a track from your music library on the iPad and off you go. Obviously it's best to feed them lossless tracks.

By the way I'd suggest you read this and then this before going near the dynamics section.

I figure if you guys are interested in music and the sound quality thereof then you'll probably be interested in a bit of behind the scenes stuff.

Disclaimer - the only connection I have with either of the two software developers is that I have their apps on my iPad.


----------



## castleofargh

slaphead said:


> Just as a follow up to my previous post on mastering. If any of you have an iPad (fairly recent recommended) and up to 20 bucks to spare you could have a go at doing some mastering yourselves.
> 
> Take a look at:
> 
> ...


 

 talking about master, I would certainly love to have one channel per instrument on the music I buy. I realize that it's the niche of a niche of a niche, but that's my utopia.
 my nemesis in audio is the drum guy/beat box(often cymbals too but they're mostly isolated and easy to EQ), to me it's almost always too loud. I end up listening to a musical genre not because I love it, but because it's low on drums and cymbals. if I could take down the drummer a few db on rock albums, I would listen almost exclusively to rock bands, that's a fact. sometimes I still EQ some kicks down on the stereo tracks(usually a small sharp EQ somewhere in the 3.5 to 6khz does it fairly well), but it's almost always at the cost of a voice or another instrument and that sucks 
	

	
	
		
		

		
		
	


	




.


----------



## icebear

xenheadfi said:


> Run it through the Dynamic Range program or look up the Dynamic Range of the mastering on http://dr.loudness-war.info/ . Researching and learning about DR was very enlightening on underlying reasons of certain hot-potato topics. Many of them can be answered simply by "Which was mastered in a sane way?"


 

 I am listening to music and decide which performance and sound I like. I don't waste any of my time trying to run analysis programmes and interpret any visual representation of a digital file. I listen to a music piece, a song. I can't imagine that ANY recording that has a dynamic range of greater than 96db. This stupid loudness war discussion is fruitless. Of course there is compression in certain kind of music. When I listen in the car to Jazz on WBGO and switch a station to some pop/rock station, these stations are double as loud and the pop music is mastered for outdoor use, mobile or radio, it is compressed - and it makes sense. How much noise do I have driving in the car? I don't really care about most of the pop music stuff and for listening on my home system I haven't encountered any heavy compression on my CDs. Possibly I am ignorant and just enjoy the music, I can live with that


----------



## KeithEmo

thad-e-ginathom said:


> In other words, whether it is all a scam or not.
> 
> It is.
> 
> ...


 
  
 I don't like the word scam - because it connotes something actually dishonest - rather than just "business a usual".... and "usual" simply means that product vendors need to come out with new products to sell every year or so - and that includes both hardware vendors and software (music) vendors. And, of course, the vendors are going to do their best to encourage people to buy the latest product. If there is a "problem" with audiophiles it's that (as consumers) they seem to "take it all so much more seriously" than many other folks, and seem much more willing to get swept up in the rush than "ordinary folks".
  
 Pretty much every old movie that was still in demand was sold on VHS tape. Then the ones that were still selling were re-mastered onto DVD (and the DVD version was usually - but not always - better). Then the ones that were still selling were re-mastered and offered again on Blu-Ray. And, since many of the masters really are about DVD quality to begin with, the Blu-Ray reissues often weren't much better than the DVD. And we all know that the movies that are still selling well on Blu-Ray will be out next year on 4k Blu-Ray. But, before you buy the "new Blu-Ray remaster" of your favorite movie, which you already have on DVD, you'll probably read a few reviews on your favorite store and see whether other people thought it was really better. (And you don't see too many heated arguments about whether "the Blu-Ray was really better, or it was really expectation bias, or the new release is really better - or they just brightened the colors up a bit". Sure, people will cheerfully discuss which version was better, and why, but they don't seem to get so _MAD_ about it. I know a lot of people who feel that 4k is unnecessary, and who don't plan to buy all their movies over again on 4k discs, and who may not even buy a 4k TV.... but they don't seem so annoyed about the people who _DO_ plan to do all those things.)
  
 And, like it or not, it is the job of advertising and marketing to _CREATE_ demand. The car I owned twenty years ago ran just fine, and it _STILL_ runs fine for its new owner, so I obviously haven't _NEEDED_ a new car in a very long time. It's up to the marketing guys to convince me that I want a new one. Sometimes the engineers come up with a great new feature; and everyone who sees it wants it. Other times the marketing guys figure out something people want and ask the engineers to create it. But, in reality, usually it's a bit of both..... some engineer says. "Ya know, they just came out with this new chip, so we could do this...". And then someone does a marketing study, formally or not, to decide if people would actually pay for that feature. Nobody in the car industry decides whether to design a new model year or not based on whether they've actually found any useful new features to add to last years model. We already _KNOW _that there's going to be a new model next year. And it _WILL_ have new features, which the marketing department will do their best to convince us we _NEED_. And, if they don't have any really useful new features to offer, they'll find something new to put into it, then spend a little more money on commercials than usual to convince us that we can't live without it. (You know your kids will hate you if they can't watch their favorite cartoons on their tablets in your new SUV, right?) I really like my backup camera; but I drove a car for thirty years just fine without one. So, the "new feature of the year" for DACs is support for 24/192k (next year it will be 32/384k), and the "new feature for this years remaster" is high-resolution (thirty years ago it was "virgin vinyl half-speed mastering"). 
  
 (The reason that it annoys me a bit when people talk like it's some huge conspiracy to "escalate things" is that the reality is that this year's 24/192k audio DAC probably costs a lot less than the 44k DAC chip that was current ten years ago. Notwithstanding a certain tendency towards nostalgia, technology generally does "get better and cheaper every year". NASA would have cheerfully spent $10 million twenty years ago for the technology that's in your $250 cell phone, mission control didn't own a monitor that was anywhere near as good as the cheap model in my living room, and you couldn't buy a 32/384k audio DAC back then either. So, every time I hear someone complain about how "44k is plenty good enough", I can't help thinking that in 1935 there were lots of "audiophiles" saying: "My one speaker sounds just like the real thing. What possible reason could I have for buying _TWO_ speakers?" and there were apparently a lot of folks who couldn't tell whether "Is it real or is it Memorex?". I'm not honestly sure whether 24/192k is "a huge step forward" from 44k or not, but I'm not prepared to declare that technology is "done" - and so there's no point in considering the possibility that there's room for improvement. )


----------



## Slaphead

Hi-Res audio isn't a scam as such, in fact it's needed at the production phase as working with 16/44 would soon introduce some audible quantisation errors depending on how many effects are being used. Working at 24/96 or 24/192 avoids this and also gives valuable extra headroom.

As to whether Hi Res audio is needed for the final product delivery then I'd say no, but at the end of the day there's a market for Hi Res audio as a good proportion of the members here and audiophiles in general have demanded it.

I personally wouldn't buy into Hi-Res audio as an end product as I see no benefit, but I wouldn't go as far to call it a scam. Certainly not from my perspective where I use it in the creation process.


----------



## RRod

keithemo said:


> I can't help thinking that in 1935 there were lots of "audiophiles" saying: "My one speaker sounds just like the real thing. What possible reason could I have for buying _TWO_ speakers?" and there were apparently a lot of folks who couldn't tell whether "Is it real or is it Memorex?". I'm not honestly sure whether 24/192k is "a huge step forward" from 44k or not, but I'm not prepared to declare that technology is "done" - and so there's no point in considering the possibility that there's room for improvement. )


 
  
 Note that there are certainly some of us who think that we would be better served with multi-channel releases at 16/44.1 rather than stereo releases at 24/192, so your two-speaker example isn't exactly the best one to make a point about the current hi-res market.
  
 On your second point, I think it's fair to point out that the equipment of the consumer needs to be considered, and not just the DAC. I would be my own real money that most of the people currently enjoying the latest Zep IV remaster have no idea what the frequency extension of their speakers/headphones are, how much IMD they might have, etc. I'm also pretty sure they don't have any equipment that's getting 24 actual bits of dynamic range. You can push the digital numbers as high as you like, it would be fine by me. 32/768 releases, whatever. But telling someone that will instantly derive benefit from such numbers *even on their current system* is getting into sketchy territory. This is one of the main things that irked me about the Pono release, for instance. They were basically telling people they would clearly hear the benefits of the hi-res even out of cheap ear-buds from the player's 5ohm output impedance. Gimme a break. (Note: you obviously don't have to answer for that).


----------



## RRod

slaphead said:


> Hi-Res audio isn't a scam as such, in fact it's needed at the production phase as working with 16/44 would soon introduce some audible quantisation errors depending on how many effects are being used. Working at 24/96 or 24/192 avoids this and also gives valuable extra headroom.
> 
> As to whether Hi Res audio is needed for the final product delivery then I'd say no, but at the end of the day there's a market for Hi Res audio as a good proportion of the members here and audiophiles in general have demanded it.
> 
> I personally wouldn't buy into Hi-Res audio as an end product as I see no benefit, but I wouldn't go as far to call it a scam. Certainly not from my perspective where I use it in the creation process.


 
  
 The scammy part is the insistence that the final conversion to 16/44.1 suddenly destroys all the hard work you just did with the effects. There's always justification for making the lives of the creators easier. It's still a fact, though, that there are Redbook releases from the earliest days of the format that smoke much of the stuff coming out in hi-res, and they certainly weren't recorded at 24/192. Perhaps they had less leeway for added effects, though.


----------



## Slaphead

rrod said:


> The scammy part is the insistence that the final conversion to 16/44.1 suddenly destroys all the hard work you just did with the effects. There's always justification for making the lives of the creators easier. It's still a fact, though, that there are Redbook releases from the earliest days of the format that smoke much of the stuff coming out in hi-res, and they certainly weren't recorded at 24/192. Perhaps they had less leeway for added effects, though.




To be honest virtually all the early releases on CD were AAD, meaning analog recorded, analoge mixed, and digitally transferred. On some of the first CDs I bought I remember a little warning that said to the effect of "The resolution of the CD is higher than that of the master tape. As a result there may be audible anomalies. This is no cause for concern". Also back then the tech was simply not available for many of the effects and processing we can do now on a bog standard computer.

While it's true some of the CDs that were released at the time were truly brillant, a lot of them absolutely sucked as a result of some studios just doing a digital transfer of their vinyl master tapes. This meant a lot of early CDs had screeching treble and no bass. It took a couple of years before they latched onto the idea that vinyl and CDs needed to be mastered differently.

But yeah, you're right. If the record companies are promoting hi-res to be audiably superior to the standard CD then it is somewhat misleading.


----------



## Labtek

silverears said:


> The controlled condition, I really want is the most resolving audio system known to man.  Then, if it still cannot distinguished, and only then, I will side with the rebel alliance here.


 
 I'm fed up listening to 2 dimensional recordings too.


----------



## Thad-E-Ginathom

keithemo said:


> I don't like the word scam - because it connotes something actually dishonest - rather than just "business a usual".... and "usual" simply means that product vendors need to come out with new products to sell every year or so - and that includes both hardware vendors and software (music) vendors. And, of course, the vendors are going to do their best to encourage people to buy the latest product. If there is a "problem" with audiophiles it's that (as consumers) they seem to "take it all so much more seriously" than many other folks, and seem much more willing to get swept up in the rush than "ordinary folks".


 
  Business as usual is to sell the latest... scam, then.
  
 Take another look at the heading, and the premise of this thread. It seems that you are a marketing man (sorry to be rude!) because it seems that youre "refutation" of that premise is that, hey, marketing men will be marketing men and they have a job to do. Points scored for serious argument: nil.
  
 Quote:
  


> Pretty much every old movie that was still in demand was sold on VHS tape. Then the ones that were still selling were re-mastered onto DVD (and the DVD version was usually - but not always - better). Then the ones that were still selling were re-mastered and offered again on Blu-Ray. And, since many of the masters really are about DVD quality to begin with, the Blu-Ray reissues often weren't much better than the DVD. And we all know that the movies that are still selling well on Blu-Ray will be out next year on 4k Blu-Ray. But, before you buy the "new Blu-Ray remaster" of your favorite movie, which you already have on DVD, you'll probably read a few reviews on your favorite store and see whether other people thought it was really better. (And you don't see too many heated arguments about whether "the Blu-Ray was really better, or it was really expectation bias, or the new release is really better - or they just brightened the colors up a bit". Sure, people will cheerfully discuss which version was better, and why, but they don't seem to get so _MAD_ about it. I know a lot of people who feel that 4k is unnecessary, and who don't plan to buy all their movies over again on 4k discs, and who may not even buy a 4k TV.... but they don't seem so annoyed about the people who _DO_ plan to do all those thing


 
  
 Do not recall the difference between VHS and DVD? Yes... it was very real. It not only brought us a better quality picture, it also brought us navigation facilities other than fast-forward and fast-back. This has no relevance whatsoever to high-sample-rate audio. None at all. You might as well justify it by saying that dynamos and starting handles gave way to reliable alternators on motor cars, so hey... hi-res audio.
  
 But you might have stopped there rather than heading straight into the _real_ reason for "high-res" audio... yet another copy of the same material gets sold by the music companies. _Now_ you are beginning to make sense! But possibly not the sense that you intended to make.


----------



## KeithEmo

thad-e-ginathom said:


> Do not recall the difference between VHS and DVD? Yes... it was very real. It not only brought us a better quality picture, it also brought us navigation facilities other than fast-forward and fast-back. This has no relevance whatsoever to high-sample-rate audio. None at all. You might as well justify it by saying that dynamos and starting handles gave way to reliable alternators on motor cars, so hey... hi-res audio.
> 
> But you might have stopped there rather than heading straight into the _real_ reason for "high-res" audio... yet another copy of the same material gets sold by the music companies. _Now_ you are beginning to make sense! But possibly not the sense that you intended to make.


 
  
 I'm not going to address your points one at a time. Suffice it to say that I don't believe that every product that you (or I) find to be useless is a scam. And, just because you seem totally convinced that there is no benefit whatsoever to high-resolution files, not everyone agrees with your opinion there - including me. We both agree that the improvement when going from VHS to DVD is obvious. However, I personally don't see any "real benefit" of re-mastering a movie that was originally mastered on DVD onto a Blu-Ray disc (after all, the picture is only DVD quality). However, someone else may feel that they'd cheerfully pay the $20 to have a copy that matches all the other little blue boxes on their shelf. And we can all read a few reviews and find out whether the picture is really any better, or there are wondrous extras on the new release, _BEFORE_ we buy it. Likewise, several companies who sell high-def files offer a sampler, or a few free file downloads, so anyone considering buying any can decide for themselves if they hear a difference or not.
  
 (I've even been known to buy a re-release of an album by my favorite group just to boost their sales figures. After all, for every extra copy they sell, they're that much more likely to get a good deal on their next album. And, for that matter, for every $25 high-def copy they sell, instead of a $10 AAC copy,  they're that little bit more likely to offer their next album at high-res, or even CD quality, rather than just go with a compressed version next time because they think nobody cares one way or the other. )
  
 It's pretty hard to make a credible claim that someone is "scamming you" if you actually get to sample the merchandise before you buy it. (And, if you're buying it _JUST_ because someone else recommended it, even if you can't tell the difference, then that's up to you.)


----------



## Thad-E-Ginathom

You keep coming back to "opinion." That is not what the thread is about.
  
 You seem to think that marketing issues are all that matter. Well, as you sell stuff, perhaps they are to you.
  
 There's no further to go here, except to remind myself why I am supposed to have given up these conversations.
  
 --- I call myself a "recovering audiophile."


----------



## KeithEmo

I disagree entirely.....
  
 The thread purports to be about "why anything over 48k is not only worthless but bad for music.... and that is clearly a statement of opinion. In fact, the thread seems to be a dispute about the claim that "you can't hear the difference - and so they are a scam".... and I see a whole lot of things going there.... most of which are opinion.
  
 1) Some people are quite certain that there is absolutely, positively, never an audible difference. (I, and many other people, do not consider this to be "proven" - and note that simply failing to prove that there is a difference does _NOT_ prove that there isn't one.)
  
 2) Even assuming that there is no audible difference, it's possible that there are other benefits to the file format itself. (For one thing, many modern DSP-based pre/pros process incoming digital audio at the same sample rate at which they receive it... So, even if a 44k file and a 96k file do sound exactly the same, the 96k file might sound better after being processed by the DSP because processing it at the higher sample rate introduces less degradation.)
  
 3) Even assuming that there is no audible benefit at all, one might suggest that any purchase of a "re-master" or "reissue" helps support the music industry, the record company, and the artist. (Note that I didn't say that the benefit would necessarily be to the customer.)
  
 4) Even ignoring those first three entirely, there is no credible reason to expect a negative impact from higher sample rates. (The one article which I've seen which makes that claim bases it on a bunch of rather shaky science. I've personally never encountered a high-res file that sounded worse on a specific piece of equipment because the high-frequency ultrasonic noise that it was able to contain because of the higher sample rate adversely affected the playback equipment; have you?)
  
 I can buy a $20 Casio quartz watch that keeps better time than any mechanical watch Rolex ever made..... but most people still wouldn't consider it to be a reasonable non-opinion-based claim if I were to say that "Rolex watches are a total waste - and, in fact, aren't as good as Casios". Apparently most people who like Rolex watches base their _opinion_ on other things besides keeping the most accurate time. 
  
 Quote:


thad-e-ginathom said:


> You keep coming back to "opinion." That is not what the thread is about.
> 
> You seem to think that marketing issues are all that matter. Well, as you sell stuff, perhaps they are to you.
> 
> ...


----------



## Thad-E-Ginathom

keithemo said:


> I disagree entirely.....
> 
> The thread purports to be about "why anything over 48k is not only worthless but bad for music.... and that is clearly a statement of opinion.
> ... ... ...


 
  
 Nope.
  
 This is in the sound science section. The first post featured material from a scientist. 
  
 If you want to argue the science, please do. It will probably go over my head, but I will try to follow and learn. However, marketing-man ifs, buts, maybes and if-onlies is _not_ what it is about except to the marketing men.
  
 You have the rest of the site for that.


----------



## FFBookman

rrod said:


> The scammy part is the insistence that the final conversion to 16/44.1 suddenly destroys all the hard work you just did with the effects. There's always justification for making the lives of the creators easier. It's still a fact, though, that there are Redbook releases from the earliest days of the format that smoke much of the stuff coming out in hi-res, and they certainly weren't recorded at 24/192. Perhaps they had less leeway for added effects, though.


 

 "Destroy" has gradients, it's not all or nothing. 
  
 Downsampling and dithering *reduces* your effects in your mix and your overall instrument timbre, it doesn't destroy it.  Lots of the mix lives out in that "unheard" space that downsample & dither seems to modify, and MP3 strongly damages.  MP3 almost destroys, but it's more like a reduction of 90% of the mix, which is why it sounds like it's emanating from a small paper bag most of the time.
  
 Mastering is all about getting the best sound out of format for the most devices. That and basic sequencing and filing, nothing else. They fall into the loudness trap to keep up with the Joneses.


----------



## FFBookman

rrod said:


> And considering my own blind tests on my own equipment and music, I think they all sound the same if they're made from the same original file. But the streaming people are asking ~$100–200 per year for access to entire catalogs, and the SACD/DVD-A people were asking $20–30 per disc, justifying the extra cost by "better sound" that may not even have come from a different master, but they weren't going to tell you that. I know who I'll put on the nice and naughty lists.
> 
> People claiming that things like 256AAC are transparent for music can justify their claims by the boatloads of blind testing that go into developing and testing the perceptual models and filtering algorithms. That's not nothing.


 

 Yeah but "****loads" is the right term because it's based on bad application of sensory data, bad testing, and corporate interests that have much invested in lossy DSP technologies. They all profit by convincing us less is more.
  
 People who make, love, create, mix, and master music -- the experts of recorded music -- hear resolution advantages easily.    Certain people on "head-fi" think all musicians are deaf and all audio producers and engineers are idiots.
  
  
 We would have had better than 16/44 for consumers long ago, but file sizes were too big to push it on the consumer in the 80's.  Also playback gear needs to be spec'd to deliver higher definition audio.  By the mp3 switchover you could put the crappiest DAC and analog you wanted in the device, it had nothing of any detail to playback anyway.


----------



## RRod

You've either tested this in some kind of controlled way or you haven't. If you reject the ways people typically do this, then finish developing your own method, convince people that its sound, then do the testing. Until then all you have is anecdote.


----------



## FFBookman

rrod said:


> You've either tested this in some kind of controlled way or you haven't. If you reject the ways people typically do this, then finish developing your own method, convince people that its sound, then do the testing. Until then all you have is anecdote.


 

 There has been some progress there but who knows what kind of timeline it has. My test procedure requires a bit more than a $20 switch box.
 In the meantime I have been reading up on other listening test formats, there's a couple of other formats that have been published I've been able to study so far.
 None of them claim to decipher musical enjoyment of the sample.


----------



## Ruben123

It's so lovely to use the cheapest DAC, cables and lowest quality sound (mp3 or mostly 16 bit CD though) but with greatest sounding speakers and earphones. 
Horrible sound though, fortunately I can't decide where it comes from


----------



## sonitus mirus

For me, it is not necessarily about being the cheapest, it is about maintaining audible transparency with regards to fidelity.  There are many options available to achieve this goal, and several of these would typically be considered affordable.   I know where my limits are through rigorous testing and from gaining reliable knowledge on the subject matter by perusing tons of material.


----------



## KeithEmo

thad-e-ginathom said:


> Nope.
> 
> This is in the sound science section. The first post featured material from a scientist.
> 
> ...


 
  
 If you really want to keep things in the realm of science, then you need to exclude human value judgements like "worthless" or "worthwhile".
  
 In terms of actual facts, we all know that high-res files have wider frequency response, and higher bit-depth files have greater dynamic range, so a 24/192k file has both wider frequency response and greater dynamic range than a 16/44k file. You seem absolutely certain that this difference is inaudible, while I (and many other people) are not.
  
 However, whether "high-res files are worthless or not" depends on a lot more than that. What you're trying to do is to say something equivalent to: nobody can legally drive over 70 mph on a freeway, therefore a Lamborghini is no better than a Toyota because the top speed for both on the freeway is the same.
  
 Personally, I haven't seen any conclusive proof that the difference between Red Book and high-res files is totally inaudible, to any human, with any source material, and under any conditions. And, to be quite honest, even assuming you could prove that, you couldn't possibly prove that someone won't invent a new speaker next week on which the difference is clearly audible. Therefore, to me, if I care about a particular recording, there is value in having "the insurance" of owning the best quality copy I can afford to buy.
  
 And, just for fun, I'm going to present a specific situation where a 24/192k recording would produce an obvious audible difference.....
  
 Let's assume that I have a collection of vinyl records that I want to archive. And, just for fun, let's assume that I make two copies of the archive, one at 16/44k and one at 24/192k. Let's even assume that, when I finish, I listen to both copies and don't notice an audible difference. Now, a year later, I decide to process my recordings to remove the occasional clicks and pops that were present on the records - and are still present on the recordings. Interestingly, several popular and effective click-and-pop-removal devices and programs will work quite well on the 24/192k recording, but won't work well, if at all, on the 16/44k recording (this happens because those devices rely on the ultrasonic spectrum of the individual ticks to tell the difference between a record scratch and, for example, a musician tapping on his music stand - and so they need to see the entire spectral content, up to about 40 kHz, to work properly). 
  
 Obviously, in this particular case, the extra bandwidth offered by the high-resolution format will give me a huge audible benefit after processing, because it will allow my tick-and-pop remover to function as intended, even though the difference before processing may not have been audible to human ears. As far as I'm concerned, that one example, and the possibility of others like it, gives ample justification for the "value" of having the best possible copy of any content - because you never know what errors may or may not be important later. (And, other than the Xaph article, I haven't seen a single experiment, or even a single anecdotal report, showing that anyone has ever experienced a loss in sound quality due to the excessive frequency response of high-res files, so I consider that article to be more like the opinion of a single engineer than "proven fact".)


----------



## Ruben123

sonitus mirus said:


> For me, it is not necessarily about being the cheapest, it is about maintaining audible transparency with regards to fidelity.  There are many options available to achieve this goal, and several of these would typically be considered affordable.   I know where my limits are through rigorous testing and from gaining reliable knowledge on the subject matter by perusing tons of material.




Well that's the other side of the story...  I don't "believe" in boutique audio magic things.


----------



## FFBookman

ruben123 said:


> It's so lovely to use the cheapest DAC, cables and lowest quality sound (mp3 or mostly 16 bit CD though) but with greatest sounding speakers and earphones.
> Horrible sound though, fortunately I can't decide where it comes from


 

 You've just described most modern people -- spending on speakers, spending on headphones, but they have a horrible DAC, horrible amp, and are playing horrible files. 
  
 The worse part about this is so-called audio experts recommending such things all over the internet.  
  
 "If you just upgraded your headphones or lowered your expectations due to noise..."
  
  
 all the while they just need better source and better rendering to make even budget speakers shine.


----------



## FFBookman

Just the facts, ma'am:
  
 Run time of a CD with 16 bit audio ~ 60 minutes depending on sample rate
 Run time of a CD with 18 bit audio ~ 35 minutes depending on sample rate
 Run time of a CD with 24 bit audio ~ 20 minutes depending on sample rate
  
 Cost of a 16bit DAC chip around 1980 ~ $100
 Cost of a 24bit DAC chip around 1980 ~ $500
  
 That is why they went with 16/44 as the consumer format in 1978.  _Sound science_ had nothing to do with it.
  
  
 Bandwidth of 24/192 PCM per second ~ 6.0Mbs
 Bandwidth of 24/96 PCM per second ~ 3.5Mbs
 Bandwidth of 24/44 PCM per second ~ 2.0Mbs
 Bandwidth of 16/44 PCM per second ~ 1.0Mbs
 Bandwidth of lossy 16/44 PCM per second ~ 0.3Mpbs
  
 This is why you have lossy - economics of bandwidth.
  
 If anything, the_ Sound Science_ has been used to justify degradations to audio quality standards over the years, not improve it.
  
 That's why it's not really a science when it comes to listening tests and perceptions- it can't separate and identify obvious business and economic influences.


----------



## Ruben123

Right now I listen to Mozart's piano concertos on a $40 Sansa Clip+ with $6 KZ Ed9 IEMs, can't remember I heard this concerto any much better.
Well a bit then, not much better. Even though my speakers are top notch.


----------



## interpolate

I always think of a MP3 as a budget brand of the same music. However with the prices based on convenience rather than overall sonic benefit.


----------



## cjl

ffbookman said:


> Just the facts, ma'am:
> 
> Run time of a CD with 16 bit audio ~ 60 minutes depending on sample rate
> Run time of a CD with 18 bit audio ~ 35 minutes depending on sample rate
> Run time of a CD with 24 bit audio ~ 20 minutes depending on sample rate


 
 You just pulled those numbers out of thin air. Given a fixed, 44.1 sample rate, a CD will hold:
  
 16 bit: 74 minutes
 18 bit: 66 minutes
 24 bit: 49 minutes


----------



## Thad-E-Ginathom

keithemo said:


> Personally, I haven't seen any conclusive proof that the difference between Red Book and high-res files is totally inaudible, to any human, with any source material, and under any conditions. And, to be quite honest, even assuming you could prove that, you couldn't possibly prove that someone won't invent a new speaker next week on which the difference is clearly audible. Therefore, to me, if I care about a particular recording, there is value in having "the insurance" of owning the best quality copy I can afford to buy.


 
  
 Well, no... the burden of proof is on you to prove that there _is_ an audible difference. That is especially true as you happen to be a member of the trade.


> And, just for fun, I'm going to present a specific situation where a 24/192k recording would produce an obvious audible difference.....
> 
> Let's assume that I have a collection of vinyl records that I want to archive. And, just for fun, let's assume that I make two copies of the archive, one at 16/44k and one at 24/192k. Let's even assume that, when I finish, I listen to both copies and don't notice an audible difference. Now, a year later, I decide to process my recordings to remove the occasional clicks and pops that were present on the records - and are still present on the recordings. Interestingly, several popular and effective click-and-pop-removal devices and programs will work quite well on the 24/192k recording, but won't work well, if at all, on the 16/44k recording (this happens because those devices rely on the ultrasonic spectrum of the individual ticks to tell the difference between a record scratch and, for example, a musician tapping on his music stand - and so they need to see the entire spectral content, up to about 40 kHz, to work properly).
> 
> Obviously, in this particular case, the extra bandwidth offered by the high-resolution format will give me a huge audible benefit after processing, because it will allow my tick-and-pop remover to function as intended, even though the difference before processing may not have been audible to human ears. As far as I'm concerned, that one example, and the possibility of others like it, gives ample justification for the "value" of having the best possible copy of any content - because you never know what errors may or may not be important later. (And, other than the Xaph article, I haven't seen a single experiment, or even a single anecdotal report, showing that anyone has ever experienced a loss in sound quality due to the excessive frequency response of high-res files, so I consider that article to be more like the opinion of a single engineer than "proven fact".)


 
  
 Oh dear, you really are clutching at straws!  Talk about ifs, buts and might be-s!  You might just as well say that you would have a huge audible difference if an operation were developed to _extend human hearing._ It's all flights of fancy. It's like, last time I did any digitising I _did_ actually use 24/96, just in case the bats and dogs preferred the results, but I couldn't hear the difference.  Mind you, I've researched a bit more since then.
  
 So you really have no case other than ifs, buts and maybes. I'm sure you don't want to be a representative of any other than your own corner of the trade, but I don't think I have heard any better (maybe it was inaudible) from any of your colleagues/competitors, whether they be purveyors of hardware or of music.
  
 I admit that I haven't heard of any reports of high-bit-rate music sounding _worse_ either, but hey, audio forums attract audiophiles, so it isn't likely, and, even if it is true (it is admitted that not all DACs are equal at all sampling rates, though), what kind of a sales line do you think this is: _Pay more for something that isn't actually worse!_
  
 That's some marketing, if you guys can manage it.
  
 Sadly, you seem to be.


----------



## old tech

keithemo said:


> I really think it's reasonable to equate a 192k DAC to a car with fuel injection. You probably can't find many cars today that still use a carburetor, but there's no real drawback to fuel injection, and lots of benefits, and the costs of inventing and developing fuel injection have already been spent, so, regardless of what might have been, and of whether you need the extra performance or not, fuel injection simply makes sense with today's technology.... and DACs that support 192k are pretty much at the same point in their life cycle.
> 
> (Again, this is really a totally separate issue from whether it makes sense to pay extra for high-resolution downloads or discs.)


 
 That is not a good analogy.  Comparing the advancement with fuel injection to carburetion is more like comparing a record player to a CD player (though like in audio, there are a minority in the automotive world that think carburetors are better, but I digress).
  
 "Hi res", compared to 16/44 is no doubt an improvement in storage and playback capability but 16/44 is alaready at the the limit of human hearing capacity.  Even assuming that there is musical content above 20khz and that it was captured in the recording and processing stages, it is beyond our physiological capacity to appreciate it.  24bit over 16bit makes even less sense given that the only advantage of it is an extra 48db of dynamic range over 98db.  How many recordings have you come across that have come close, let alone exceed a 98db dynamic range?
  
 A better analogy then is with a hypothetical new video format that promises to store and playback high light frequencies deep into ultra violet.  Like hi res music you can say that it is an improvement in video storage and playback capability but it is beyond the capability of human eyesight and can only be appreciated by a bee.  Likewise 24/96 over 16/44 playback might be useful for a dog and 24/192 over 24/96 is useless to a dog but might be appreciated by dolphins.


----------



## upstateguy

Maybe an analogy for high-resolution audio might be to film a movie using a full spectrum camera that captures both ultra violet and infra red wavelengths as well as the spectrum we can see.


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## dprimary

upstateguy said:


> Maybe an analogy for high-resolution audio might be to film a movie using a full spectrum camera that captures both ultra violet and infra red wavelengths as well as the spectrum we can see.


 
 I keep complaining to Christie Digital that their projectors are cutting out all that glorious extreme ultraviolet ionizing radiation, and they just ignore me. What little I get is absorbed by the air, come to think of it, same that air is sucking up all my 50KHz audio too. That is the problem, we must watch and listen to all high res media in a vacuum. It is Air that is preventing us from experiencing true high res.


----------



## upstateguy

dprimary said:


> upstateguy said:
> 
> 
> > Maybe an analogy for high-resolution audio might be to film a movie using a full spectrum camera that captures both ultra violet and infra red wavelengths as well as the spectrum we can see.
> ...


 

 the *Air* has always been the problem dp.


----------



## upstateguy

thad-e-ginathom said:


> keithemo said:
> 
> 
> > Personally, I haven't seen any conclusive proof that the difference between Red Book and high-res files is totally inaudible, to any human, with any source material, and under any conditions. And, to be quite honest, even assuming you could prove that, you couldn't possibly prove that someone won't invent a new speaker next week on which the difference is clearly audible. Therefore, to me, if I care about a particular recording, there is value in having "the insurance" of owning the best quality copy I can afford to buy.
> ...


 
  
 I am so tired of hearing about the burden of proof.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.


----------



## money4me247

upstateguy said:


> I am so tired of hearing about the burden of proof.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.


 
 mmm... I don't want to get caught up in this discussion too much as I find they usually don't end well, but the Nyquist-Shannon sampling theorem irrefutably proves mathematically that analog waveforms can be perfectly reproduced given that the sampling rate is at least twice the highest analog frequency.
  
 That being said, there are more factors involved with sonic reproduction and analog-to-digital conversion. Due to differences in design, some dac chips may not perform optimally or may hypothetically produce audible errors using lower sampling rates if they were optimized for usage with a higher sampling rate. There may be audible distortion during signal reconstruction/signal processing (such as aliasing) or temporal distortions such as jitter or quantization noise. Typically, digital forms of distortion is well-understood and kept to extremely low levels under the noise floor of the electronics with relatively no perceivable sonic impact.
  
 Realistically, this suggests that higher sampling rates above 44.1kHz will generally provide little to no real-world benefit compared to DAC changes (which also typically have extremely subtle real-world benefits compared to transducer upgrades). However, there may be instances where that is not true. From my personal experiences, I have found transducer changes to be much more noticeable over external component changes (dac/amp) and all three of those factors having much greater sonic impact over all other sonic tweaks (except for EQ) as long as your source file is already redbook quality.
  
 Hopefully, this perspective will not seem like an objectivist or subjectivist perspective but just my personal perspective on the issue based on the scientific information available and my own first-hand experiences.


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## Thad-E-Ginathom

<Crossposted with money4me247>
  
 ... an eminently sane, and economical, point of view 
  
 Quote:


upstateguy said:


> I am so tired of hearing about the burden of proof.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.


 
  
 I am so tired of hearing the burden of proof thing misrepresented and turned around simply by people who are representing something but don't have any substantive basis for it. I have a pill here that cures everything from the common cold to HIV: don't believe me? well, _you  _prove that it doesn't!
  
 I am not shutting down a subjective discussion: let me remind you, as I reminded Keith a few posts ago, that this thread is in the science section of the forum. You have the rest of the site for dreams, imagination, biases etc etc. (err... _We _have... I'm not immune to them either. Nobody is)
  
 Keith should really learn from the saying, "when in hole, stop digging." Did you notice that he has had to admittedly turn to realms of fantasy for his last post?
  
 Answering this sort of marketing stuff is so easy: it is like taking the proverbial sweets from the proverbial children. It is trivial. A more meaty conversation, with some sort of real, technical claims, would be much more interesting and lead to some potential of learning and taking the subject forward.
  
 Here's a gift for Keith: although 44.1 _can_ be said to be completely sufficient, there are real digital gurus who do support the idea that it was not an ideal choice. If I remember rightly, 'JJ' Johnston favoured something around 60 (I could be wrong on the actual number). What I have never seen favoured by any of the real scientists is the idea that the number should be doubled and doubled for ever-increasing satisfaction: 48, 92, 192, 384, and further into the skies of sampling-rate madness, with manufacturers making equipment that plays DSD rates than there isn't even any music for.
  
 Instead of this marketing madness, there could be proper, substantiated research into music recording and reproduction, as well as the other concern which is decent mastering, regardless of the medium/format. Our world has gone mad. The lunacy suits the sales people obviously: there is money in it for them.
  
 Monty says that anything over 48 is pointless; JJ says 60* would be ideal . Lets finance that research, instead of the "opinion" nonsense. Then, when and if the music industry actually comes up with _a good_ _reason_ to sell us the same music for the third or fourth time, it might actually be worth worth buying.
  
  
  
  
  
  
  
  
 =====
  
  
 *I'm really not sure of this exact number, but it would take an awful lot of digging to find it. Anyone who is at all interested in the actual science and technology, the theory and the practice, of digital audio and compression, lossy and lossless, would do well, if not already known, to read James D. Johnston's various writings and video presentations. For one thing, he is one of those rare people that can make the deepest of deep accessible to innumerates like me. I am hugely grateful to him for the fact that I have any understanding _at all_ of what digital sound is all about. And Monty, for his demonstration that digital audio is _not_ a series of steps, which is the misunderstanding that most of the nonsense is based on.


----------



## dazzerfong

upstateguy said:


> I am so tired of hearing about the burden of proof.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.


 
 Well, it's a good thing we're as far away in terms of subjectivity as is possible here.


----------



## Thad-E-Ginathom

> Did you notice that [Keith] has had to admittedly turn to realms of fantasy for his last post?


 
  
 Perhaps a little unfair of me. Keith makes the point that any digitisation, archiving, etc should be done in the best available format, and this is, of course, completely true. However good, for instance, 320 MP3 might be, and however indistinguishable from lossless compression it might be, it would, in my view, be a foolish choice for _archiving_ music.
  
 Yes, it is true that technology that is not available today, may be available tomorrow, for the improving of digitised archives, which is why they should always be a lossless format, and one that can be edited and processed without any change to the encoding.
  
 The mistake remains, though: the _assumption_ that 192 > 98 > 48.


----------



## OddE

money4me247 said:


> mmm... I don't want to get caught up in this discussion too much as I find they usually don't end well, but the Nyquist-Shannon sampling theorem irrefutably proves mathematically that analog waveforms can be perfectly reproduced given that the sampling rate is at least twice the highest analog frequency.


 
  
 -To be fair, Nyquist-Shannon has further requirements than just sampling rate > 2x analog bandwidth, most notably that the signal in question must be perfectly band-limited, which in turn implies infinite duration.
  
 I have not heard of anyone being able to demonstrate that easing on the band-limit criterion (within reasonable bounds) produces audible artifacts, though.


----------



## Slaphead

Rather then look at the technology, and personally for me there's no doubt that more information is beneficial from a working and/or future proofing point of view, let's look at the practicalities.

First off when you convert from the digital domain to the analogue domain you lose, and you lose big time. Your 24bits of dynamic range becomes around 20bits of dynamic range. You lose around 4bits at the DAC stage due to the noisefloor of the electronics. Boom gone. Then that signal has to be amped which results in even more loss. Traditional amplifier technology hasn't really changed that much since the transistor came about, it's just been incrementally improved and quite frankly not much since the 80's. Yes there are now class D amps but at the end of that chain the signal has to be passed through a smoothing unit before it's usable so you're still losing information.

Finally you have your transducers (headphones and speakers). This is where the vast majority of loss happens - as I've previously said, transducers can't even track a 16/44 signal correctly and without distortion, even the top ones (look at the response charts). So the bit that's delivering to your ears is causing a significant loss as well, such that you're listening to something that isn't even 16/44 resolution anymore.

All this happens before the sound actually hits your ears.

So is it worth having hi res audio? well the answer is yes and no. No because once the signal has reached your ears it's nowhere near the resolution it started as in the digital domain. However yes if you want to future proof your music collection, you want to archive, or you work with audio - there's no argument there.

Basically what we have at the moment is a wonderful digital system that's able to store audio in an amazingly high resolution that ultimately ends up getting fed through technology that was well understood and well implemented 40 or 50 years ago, and since then has not really been significantly improved (I still use HPs that are still on the market today, but were designed and introduced back in the late 80's). Personally I doubt we'll see a major shift in Hi Fi technology, at least in my lifetime, that will allow full hi res to be accurately rendered at the transducer stage.

Ok so armed with that information it gives you choice on where you get off the bus.

If you want hi res audio and you're convinced of it's benefits then go for it - I'm not going to argue with you, and you can ride the bus to the end station.

If you're happy with standard 16/44 and you're just listening as opposed to working with audio then great, and you can get off the bus here knowing full well it's the right choice for you.

If, like me, you honestly can't hear the difference between 256Kbps VBR AAC and lossless when consuming music then you just need a ticket for a short bus ride.

Now everybody can be happy


----------



## RRod

thad-e-ginathom said:


> Monty says that anything over 48 is pointless; JJ says 60* would be ideal . Lets finance that research, instead of the "opinion" nonsense. Then, when and if the music industry actually comes up with _a good_ _reason_ to sell us the same music for the third or fourth time, it might actually be worth worth buying.
> 
> 
> =====
> ...


 
  
 Lavry gives a nice overview of the topic. Whatever the theoretical limits are, every single person is perfectly capable of doing their own comparisons on their own equipment to see if they're getting what they think they're paying for. The trick is getting people to actually do this, and do it in a way that avoids bias.


----------



## KeithEmo

thad-e-ginathom said:


> Well, no... the burden of proof is on you to prove that there _is_ an audible difference. That is especially true as you happen to be a member of the trade.
> 
> Oh dear, you really are clutching at straws!  Talk about ifs, buts and might be-s!  You might just as well say that you would have a huge audible difference if an operation were developed to _extend human hearing._ It's all flights of fancy. It's like, last time I did any digitising I _did_ actually use 24/96, just in case the bats and dogs preferred the results, but I couldn't hear the difference.  Mind you, I've researched a bit more since then.
> 
> ...


 
  
 There's one thing you seem to have backwards. For the most part, people _don't_ buy high-res music files _because_ they have a 24/192k DAC; instead, they buy a 24/192k DAC _because they want to play high-res music files_. The company I work for (Emotiva) sells audio equipment; we don't sell music. Therefore, all we do is to sell people equipment that has the features they're asking for. Sure, there's always a "push/pull" of marketing involved.... just like, while some people probably really wanted WiFi in their SUV to begin with, the car companies have been doing their best to "inform people of that need" so they'll consider it to be a worthwhile feature when they go shopping. Personally I consider 24/192k support in my DAC to be a lot more useful than WiFi support in my car (and they don't seem to mention that you need a _subscription_ to use that WiFi; talk about a scam), but it doesn't annoy me all that much when I see that silly commercial where they guy's kids are sooooooo depressed when they can't watch the end of their movie in the competitors cheesy van. (However, it is great marketing, and I'll just bet that more than a few people buy that brand after imagining their kids whining like that... which they might do, after seeing the commercial, if daddy buys the van without WiFi. 
	

	
	
		
		

		
			





)
  
 The main part of the industry "driving the change to high-res music" is the people who produce and sell music - because it is their new "premium product". And, while "high-res" does seem to be the latest fad, and the excuse for the latest cycle of re-masters and reissues, I can't honestly say it seems like a huge huge factor there either. (The Dark Side of the Moon has been reissued and remastered dozens of times, and only a few of those were high-res; all the others sold just fine as "plain old remasters". It would be interesting to see exactly how big a "spike in sales" of that album occurred when each of the high-res versions was released - as compared to the regular ones - and as compared to yearly ongoing sales.)
  
 However, any discussions of how "worthwhile" high-res music is really need to be had with the music producers because, as I've pretty much beaten to death, as a "new technology", it hasn't really raised the price of hardware. (You might as well complain that power steering raises the cost of your car, and isn't really necessary.)
  
 As for the value of "future proofing" - I do think you also underestimate that (in general, although maybe not in specific). I'll never be feeding my digitized albums through a click and pop remover - because I managed to replace all the albums I actually care about with new digital editions, so I didn't digitize them. However, I have lots of old photos that I took as JPGs, and which look just fine, but now can't be practically edited in Photoshop (because JPGs look just fine to the human eye, but they have all sorts of artifacts that become apparent, and unsightly, when you try to perform certain editing operations on them). I also know quite a few people who were quite convinced that the AAC music they bought a few years ago "sounded fine", then ended up buying parts of their collection over again when they upgraded their equipment, and suddenly the shortcomings became apparent. Therefore, if I was creating digital content, I would master it at 24/192, for the same reason that I take photos that I think may turn out to be important as TIF or RAW instead of JPG - just as insurance against the possibility that it will matter in the future. (I'm guessing that you've never suffered any horrible consequences because you digitized your albums at 24/96k instead of 16/44k; even if you've so far not found any reason why it would matter, it didn't cost much extra effort and storage space is dirt cheap).
  
  
 To me, if you want to complain about "unnecessary technology", there are quite a few much more egregious examples in the "audiophile industry"....


----------



## Duncan

I have said this before, and will again, I would be (and am) happier with higher grade 'low resolution' equipment than I would be with lower grade 'high resolution' gear


----------



## KeithEmo

rrod said:


> Lavry gives a nice overview of the topic. Whatever the theoretical limits are, every single person is perfectly capable of doing their own comparisons on their own equipment to see if they're getting what they think they're paying for. The trick is getting people to actually do this, and do it in a way that avoids bias.


 
  
 Geeee, now there's a thought.
  
 Some people are quite convinced that they can, at least sometimes, hear a difference. I'm equally sure that there are more than a few people who think they can run faster, and sink more baskets, if they buy a certain brand of $350 sneakers. And, while the company who sells those sneakers probably has posters showing exactly how their "magic spring technology" works, most of the people who buy those sneakers will do so because their favorite athlete looked really cool in them on the commercial. (I very much doubt there will be much research done into sports kinematics and materials technology by potential customers.)
  
 As I've been pointing out - most DACs these days support at least 96k, so it'll cost ya $25 to buy a high-res album and find out for yourself (you can probably find a free demo to download, but I'd always rather use an album I'm familiar with and actually_ like_).
  
 And a final thought on bias..... While everyone goes on and on about how differences people think they hear are probably due to expectation bias, which is quite probably true a lot of the time, the reverse situation is also not only possible but likely. Even in the most perfect test protocol, where the subject can't possibly tell which sample is which, there's no way to avoid a possible expectation bias that the subject expects to hear _NO_ difference.


----------



## RRod

keithemo said:


> And a final thought on bias..... While everyone goes on and on about how differences people think they hear are probably due to expectation bias, which is quite probably true a lot of the time, the reverse situation is also not only possible but likely. Even in the most perfect test protocol, where the subject can't possibly tell which sample is which, there's no way to avoid a possible expectation bias that the subject expects to hear _NO_ difference.


 
  
 That would be a problem if you were pooling results across individuals, but a single person who is dead set in his belief that he CAN hear differences will probably be doing his best to do the test correctly. And "correctly" should involve some mode of blind testing.


----------



## sonitus mirus

I purchased a 24/96 file of Schostakovich's "The Coachman's Dance", that I enjoy immensely.  I converted this to 16/44 and did not hear any difference at all in an ABX.  Perhaps some hardware, software, or the format conversion process might create a situation where the 2 files sound different to some people, but I don't think this would qualify as a meaningful difference with regards to the topic of this thread.  24/192 adds nothing positive to the music with regards to sound quality.  Where is the evidence that it does when properly tested with the same material at 16/44?
  
 Higher resolution files merely allow the music industry to charge a premium price with almost no additional resources being spent on their side.  The music industry is run by a bunch of greedy a-holes with teams of political lobbyists and unscrupulous lawyers doing their bidding.  Their reputation precedes them.   
  
 My concern is that the best available options for sound quality will only be released on more expensive, high resolution formats.  Thank goodness the same greed that is driving the music industry, along with the ignorance of their consumers, has seen simultaneous releases of both high resolution and standard formats that sound exactly the same when properly tested against each other.  If things remains as they are, who am I to stand between a fool and their money?  I see a future where high resolution slowly takes root, while 16/44 becomes more difficult to obtain, eventually becoming obsolete.  Obviously, the price of music will go up since the quality will now be "much better', when no benefit will be observed, except for the wealthy music industry executives' financial portfolios.


----------



## RRod

keithemo said:


> And a final thought on bias..... While everyone goes on and on about how differences people think they hear are probably due to expectation bias, which is quite probably true a lot of the time, the reverse situation is also not only possible but likely. Even in the most perfect test protocol, where the subject can't possibly tell which sample is which, there's no way to avoid a possible expectation bias that the subject expects to hear _NO_ difference.


 
  
 To expand on this a bit:
  
 I just got a 24/96 version of the Steve Wilson Aqualung remaster. For fun, I downsampled the title track to 14/44.1 and then back up to 24/96, then did an ABX. As you can guess, I failed to hear a difference. Now, was this result due to my bias against hearing a difference? Perhaps, but here's the thing: if I downsample it to 8/44.1, or 16/30, I can easily ABX the difference. So indeed there comes a point where I can pass the test because I can actually hear a difference. Why should doing the test at 14/44.1 suddenly make me some bias-ridden Luddite who is out to destroy everyone's hi-res fun by faking results? It is certainly valid to say that I might not be listening as intently for differences as I might be if I were a "believer," but I can assure you I was listening more intently then I am during normal music listening.


----------



## sonitus mirus

rrod said:


> To expand on this a bit:
> 
> I just got a 24/96 version of the Steve Wilson Aqualung remaster. For fun, I downsampled the title track to 14/44.1 and then back up to 24/96, then did an ABX. As you can guess, I failed to hear a difference. Now, was this result due to my bias against hearing a difference? Perhaps, but here's the thing: if I downsample it to 8/44.1, or 16/30, I can easily ABX the difference. So indeed there comes a point where I can pass the test because I can actually hear a difference. Why should doing the test at 14/44.1 suddenly make me some bias-ridden Luddite who is out to destroy everyone's hi-res fun by faking results? It is certainly valid to say that I might not be listening as intently for differences as I might be if I were a "believer," but I can assure you I was listening more intently then I am during normal music listening.


 
  
 I do similar testing with lossy formats.  I convert to 96 kbps to identify artifacts and where these occur.  With most of the music that I have tested, Beginning at 160 kbps or 192 kbps, I have found that there is often not an obvious difference from the original FLAC created from ripping a CD.   At 192 kbps, I have successfully passed an ABX, but it was very difficult for me and I had to concentrate a great deal to the point where it was not enjoyable and not how I would typically listen to music.  To be safe, I use 320 kbps, which happens to be the same as the music service I use, Google.   
  
 For me, I wanted to know.  I did my best to ensure that I was conducting the tests correctly.  It took a long time before I could comfortably listen to lossy music without feeling like I might be compromising something with the quality.  Any blip, pop, or warble that I heard, I would immediately setup a test or listen to the lossless version to see if that same anomaly was in the original.  Every time, whatever I was hearing in the lossy was present in the original, and any ABX I conducted was null.   Sure, there may be something out there that I could hear a difference, or maybe just a certain section in a song might be noticeable, but I probably wouldn't be able to hear it through normal listening.  I probably impact the sound more significantly when I naturally swallow than any difference that could be heard between my lossy and the lossless version.


----------



## KeithEmo

rrod said:


> That would be a problem if you were pooling results across individuals, but a single person who is dead set in his belief that he CAN hear differences will probably be doing his best to do the test correctly. And "correctly" should involve some mode of blind testing.


 
  
 I agree entirely: if someone is convinced that they _CAN_ hear a difference, then proper blind testing will prove whether they're right or not.
  
 However, my point was that, if someone is dead set in _THEIR_ belief that there is no audible difference, there is no easy way I can think of to devise a test that will reliably "prove" that there is in fact an audible difference that they've - consciously or unconsciously - "chosen" to ignore (perhaps due to a bias to expect no difference).
  
 Therefore, it's relatively unlikely that either result is going to change the mind of a "true believer" either way (and, as with most other humans, audiophiles do tend to approach any question with biases and assumptions - in both directions - already somewhat firmly in place).
  
 It's very difficult to prove that a difference exists if the subjects themselves are unaware of it - either because they're deliberately lying, or because they are honestly not consciously aware of it. In the case of a drink or food, for example, instead of asking the subjects which one they prefer, you could simply place equal samples in front of a lot of subjects and see which ones subjects eat more of. Or, in the case of gauging "listener fatigue" with two pieces of equipment, you could simply allow subjects access to two identical rooms, one where each device is playing, and use the amount of time they spend in each room as a measure of how much they "enjoy" the device playing there - or how much it annoys them over time. In the case of different music files, you could perhaps hook the subjects up to a FMRI, and see if the actual patterns of neuron activity in their brains are different. (I picked that last example for a very specific reason.... because some recent studies have shown that, when a subject tastes the same identical wine, but from bottles with different price tags on them, not only do they say that the more expensive one tastes better, but MRI studies show that the pleasure centers of their brain actually do "light up brighter". In other words, the wine actually does taste different because of their bias - and they aren't "just imagining it". What if it turns out that high-res files really do sound better - but only if you know that they're "supposed to"? Does this negate the fact that the experience itself really is in fact different? Is a claim that a given product is better any less valid if it's only true _because_ you saw the commercial? )
  
 (Here's an interesting thought. Place a subject in a quiet room and monitor their vital signs - like blood pressure. After five minutes, play a single loud lion's roar, and measure how much their blood pressure jumps. Try the same experiment with a lot of subjects, and several different renditions of the lion's roar, but all carefully at the same loudness. You might find no difference in the response, or you might find that the more accurate recording of the lion provokes, on average, more of a rise in blood pressure. If so, then this data will allow you to accurately assess which version of the lion's roar is more accurate - even though it may turn out that people "like" the less accurate one better - possibly because it fails to provoke as much of a fear response. 
	

	
	
		
		

		
		
	


	




 )


----------



## Thad-E-Ginathom

No, Keith, I don't have it backwards: I'm very much aware of the fact that this is the _music industry's_ baby, and, except in exceptional circumstances, I don't really blame you guys for making gear that plays what the music industry is selling. What's more, even if I don't actively seek them, high-sample-rate music files come my way, and _even I_ want to actually play them. Principle and technical facts are for hammering out on forums, but there is a practical side to all this too. I don't see much 192 music, and when I do, I'm happy to down-sample it for playing on my up-to-96 DAC. On the other hand, I am not going to _reject_ a DAC just because it _will_ cope with >96k. 
  
 That's life, and listening to music in 2015. Stepping back, however, and asking what is going on here --- well, you know what I and many others think of the situation.


----------



## KeithEmo

rrod said:


> To expand on this a bit:
> 
> I just got a 24/96 version of the Steve Wilson Aqualung remaster. For fun, I downsampled the title track to 14/44.1 and then back up to 24/96, then did an ABX. As you can guess, I failed to hear a difference. Now, was this result due to my bias against hearing a difference? Perhaps, but here's the thing: if I downsample it to 8/44.1, or 16/30, I can easily ABX the difference. So indeed there comes a point where I can pass the test because I can actually hear a difference. Why should doing the test at 14/44.1 suddenly make me some bias-ridden Luddite who is out to destroy everyone's hi-res fun by faking results? It is certainly valid to say that I might not be listening as intently for differences as I might be if I were a "believer," but I can assure you I was listening more intently then I am during normal music listening.


 
  
 The thing that most people tend to forget is that biases tend to be most effective _in the absence of real or significant differences_. Biases tend to make us either imagine differences where none exist, or imagine no difference where a small difference exists, but they rarely override significant differences when they are actually there. (However, a bias can alter our evaluation criteria, so a bias can make a real and significant difference seem more or less important.)


----------



## Thad-E-Ginathom

Sorry, missed this...
  
 Quote:


rrod said:


> Lavry gives a nice overview of the topic. Whatever the theoretical limits are, every single person is perfectly capable of doing their own comparisons on their own equipment to see if they're getting what they think they're paying for. The trick is getting people to actually do this, and do it in a way that avoids bias.


 
  
 Yes, Lavry is pretty good at taking it step by step. I go so many steps  and am lost --- but each effort yields a little more than the last


----------



## KeithEmo

thad-e-ginathom said:


> No, Keith, I don't have it backwards: I'm very much aware of the fact that this is the _music industry's_ baby, and, except in exceptional circumstances, I don't really blame you guys for making gear that plays what the music industry is selling. What's more, even if I don't actively seek them, high-sample-rate music files come my way, and _even I_ want to actually play them. Principle and technical facts are for hammering out on forums, but there is a practical side to all this too. I don't see much 192 music, and when I do, I'm happy to down-sample it for playing on my up-to-96 DAC. On the other hand, I am not going to _reject_ a DAC just because it _will_ cope with >96k.
> 
> That's life, and listening to music in 2015. Stepping back, however, and asking what is going on here --- well, you know what I and many others think of the situation.


 
  
 On that note - let me say that I pretty much agree with you.
  
 More careful mastering and editing would clearly make a much more significant, and more positive, difference than something like playing with new and different sample rates.
 I do find, though, that re-masters do lately tend to be done at higher resolutions, and that a lot of them at least are rather an improvement (although quite probably not for that reason).


----------



## upstateguy

money4me247 said:


> upstateguy said:
> 
> 
> > *I am so tired of hearing about the burden of proof*.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.
> ...


 
  
  


thad-e-ginathom said:


> <Crossposted with money4me247>
> 
> ... an eminently sane, and economical, point of view
> 
> ...


 
  
  


dazzerfong said:


> upstateguy said:
> 
> 
> > I am so tired of hearing about the burden of proof.  It is nothing more than a way for an objectivist to try to shut down a subjective discussion.
> ...


 
  
 My post was only about evoking the "burden of proof" rubric. 
  
 Doing that accomplishes nothing because no one is going to go to the trouble to dig up the required proofs or post the required ABX-DiffMaker recordings usually asked for. 
  
 The other thing it does is open the door to what T-E-G said above about turning it around and asking for proof of the reverse.


----------



## money4me247

keithemo said:


> I agree entirely: if someone is convinced that they _CAN_ hear a difference, then proper blind testing will prove whether they're right or not.
> 
> However, my point was that, if someone is dead set in _THEIR_ belief that there is no audible difference, there is no easy way I can think of to devise a test that will reliably "prove" that there is in fact an audible difference that they've - consciously or unconsciously - "chosen" to ignore (perhaps due to a bias to expect no difference).


 
 Easy. Just offer a monetary reward high enough that they will want to hear a difference. Let's say a cash reward of 1 million for a 80% accuracy on an ABx test... pretty sure, the majority of testers will be trying as hard as they can to accurately hear the difference. Set-up a rigorously double-blinded test with a large sample size and a random sampling of participants. With properly done statistical studies, it is irrelevant whether the individual subjects believe they can or cannot hear a difference. A well-done sampling of the population should have enough variations in pre-existing beliefs that it doesn't matter. Same thing with well-done large-scale drug/treatment trials. Doesn't matter if the patient thinks the treatment will work or not work, a well-done sampling method should account for that variation.
  
 I would like to think that most people who approach ABx testing are generally aware that it is a tool to test a hypothesis and are curious to explore with an open mind... rather than simply using it as a tool to reinforce a pre-existing belief (...which would be the opposite of an objective scientific approach).
  
 of course, it is impossible to remove personal expectation bias and if a person is so stubborn in their belief either way... what is even the point of doing an A/B test?? The scientific method is just a tool to explore questions... not really supposed to be used to propagate one specific perspective or doctrine. I think the important thing to keep in mind is that the majority of individual findings are just a small insignificant data point. You really require a large-scale study to really begin to remove all the other confounding variables can possibly appear.
  
 just thought I would post this since this is the objectivist boardroom


----------



## RRod

keithemo said:


> The thing that most people tend to forget is that biases tend to be most effective _in the absence of real or significant differences_. Biases tend to make us either imagine differences where none exist, or imagine no difference where a small difference exists, but they rarely override significant differences when they are actually there. (However, a bias can alter our evaluation criteria, so a bias can make a real and significant difference seem more or less important.)


 
  
 I agree there, and I will gladly acknowledge that I might not be able to detect *small* differences via ABX due to bias affecting my approach to the test (lack of patience, sloppy controls, etc.) But small is small, and many on the hi-res side do not claim small difference, they claim night-and-day, which should be easy to detect.


----------



## RRod

> Originally Posted by *money4me247* /img/forum/go_quote.gif
> 
> of course, it is impossible to remove personal expectation bias and if a person is so stubborn in their belief either way... what is even the point of doing an A/B test?? The scientific method is just a tool to explore questions... not really supposed to be used to propagate one specific perspective or doctrine. I think the important thing to keep in mind is that the majority of individual findings are just a small insignificant data point. You really require a large-scale study to really begin to remove all the other confounding variables can possibly appear.
> 
> just thought I would post this since this is the objectivist boardroom


 
  
 This is exactly the kind of thing that should get hashed out in an actual journal, and the utility of such an undertaking would depend upon how flawed you think Meyer&Moran and all the other papers battling this out are. Still, it's pretty trivial for a given individual to plop a track into foobar and let it do its ABX thing and decide for themselves, at least as far as their own rig is concerned.


----------



## KeithEmo

rrod said:


> I agree there, and I will gladly acknowledge that I might not be able to detect *small* differences via ABX due to bias affecting my approach to the test (lack of patience, sloppy controls, etc.) But small is small, and many on the hi-res side do not claim small difference, they claim night-and-day, which should be easy to detect.


 
  
 I agree with you absolutely.
  
 I consider the differences to be relatively small, certainly less than the other differences frequently present when a recording is remastered. (For example, I'd much rather have a 44k version of the latest remasters of the Grateful Dead Studio Albums than a 24/192k copy of the previous version of the release.) 
 As with many things, both marketing folks and eager fans tend to exaggerate things, each for their own reasons. 
	

	
	
		
		

		
			




  
 As far as I'm concerned, the big problem is that the overwhelming majority of Red Book CDs deliver nowhere near the sound quality that the format is capable of. Perhaps, with luck, the current high-res fad will force music producers to look a little harder at that issue. (I'd be very happy if even a few more of the recordings I have in either format had production quality on par with the few really high quality Red Book CDs I have.)


----------



## Thad-E-Ginathom

The incentive is there. Anybody who can reliably and continuously double-blind identify higher-sample rate versions of the same material will make history. Their name will join Nyquist in the pantheon of digital sound. The _We-told-you-so_ will knock us cynics so flat on the ground that our heads will be spinning for weeks.
  
 I'd quite like to be that famous! Sadly, I know I won't be the one, not only because I don't believe it is possible, but because I have the hearing test results, and being able to hear 16K, let alone 20-plus-K would be a miracle for me.
  
 But, we have to ask why it has not happened. There is _no_ controlled, authenticated positive test result for the industry to shout from the roof tops about. None. And I am sure of that without digging the world's obscure research papers, because if it had happened, it would be the second post in every thread like this, if the first one ever got made.
  
 I am not saying that nobody has ever been able to identify a sample rate. There are guys out there (and in here, even) who work with digital music, lossy compression (just a few posts ago) and know artifacts like the rest of us know a bass drum beat, so maybe there is.


----------



## OddE

duncan said:


> I have said this before, and will again, I would be (and am) happier with higher grade 'low resolution' equipment than I would be with lower grade 'high resolution' gear


 
  
 -Besides, in most practical situations the source material is what is limiting your experience, not the hardware. (To rephrase: The CDs out there taking full advantage of the possibilities of Red Book are few and far between.).
  
 Obvious exception: headphones, where audible differences between various types verifiably exists.


----------



## Slaphead

The problem that we have with hi res audio when it comes to ABing is that, quite frankly, nobody has an audio system that can resolve the differences.

You can argue this day and night, but until theres an audio system that can track hi res (or even 16/44) accurately then there will be no definitive answers as to whether hi res actually makes a difference at the point of delivery.

I remember doing some some ABing with various compressed formats at relatively low bit rates, and I remember being shocked and totally disappointed with myself for not really hearing much difference beyond 160Kbps. at that point AAC had the edge, but move up to 192Kbps then there was really nothing to call.

I think a lot of you guys forget that to be able to hear a difference then you need a setup that can accurately resolve 24/96 or beyond, and IMO that audio setup does not exist... yet, even at the high end.


----------



## RRod

slaphead said:


> I think a lot of you guys forget that to be able to hear a difference then you need a setup that can accurately resolve 24/96 or beyond, and IMO that audio setup does not exist... yet, even at the high end.


 
  
 On the flip side, you have people who don't do A/B tests who claim to hear differences, even though their equipment can't possibly be resolving the high frequency content, and whose musical choices can't possibly be pushing more than 16bits.


----------



## KeithEmo

money4me247 said:


> Easy. Just offer a monetary reward high enough that they will want to hear a difference. Let's say a cash reward of 1 million for a 80% accuracy on an ABx test... pretty sure, the majority of testers will be trying as hard as they can to accurately hear the difference. Set-up a rigorously double-blinded test with a large sample size and a random sampling of participants. With properly done statistical studies, it is irrelevant whether the individual subjects believe they can or cannot hear a difference. A well-done sampling of the population should have enough
> 
> variations in pre-existing beliefs that it doesn't matter. Same thing with well-done large-scale drug/treatment trials. Doesn't matter if the patient thinks the treatment will work or not work, a well-done sampling method should account for that variation.
> 
> ...


 
  
 I agree entirely - that a cash reward is a great way to motivate everyone into trying their best to do something - like detect a difference.
  
 However, I don't agree with what you'd "like to think". I think it's basic human nature that people tend to approach most situations with an expectation, and, while they may be willing to keep an open mind, nobody likes to be proven wrong. Therefore, they're really hoping that their expectation will turn out to be right. However, that's good enough, as long as you've provided enough motivation for them to consider "the real prize" to be more important than "being right".
  
 The biggest problem with doing a large-scale data study is simple - someone has to pay for it.
  
 Same deal with this thread. If you read all the responses, the one thing that's perfectly obvious is that most participants _already think they know the result_. Some are quite convinced that the difference can;t possibly be audible; others are equally certain that, at least some of the time, the difference will be audible. I haven't heard a single person saying "I have no idea whether there's a difference or not; let's find out". Note that that sort of "pure science" isn't especially common anywhere else either. You won't find a pharmaceutical company saying "we have no idea whether chemical X will help with this disease or not - let's find out" either.
  
 In this case, though, the "money value proposition" is somewhat more obvious. High-res audio files have been promoted to the point where most people who have an opinion probably do believe that they are "somewhat better" - or at least that they are "this year's new and improved model". So, if a study were to confirm that most people heard a positive difference, most people would take that as confirmation of what they already assumed (and so it wouldn't sell any more albums). However, if the study were to show that nobody heard any difference, a lot of people might decide that there's no reason to buy this year's re-issue. And if, as I suspect, it showed that there was a slight audible difference at least sometimes, that still might be "inconclusive enough" to convince a lot of people not to spend the $25. In short, I think the risk of the results of such a study costing sales is far greater than the likelihood of its making additional sales - because the current state of public belief is probably more conducive to sales than any reality a study would be likely to discover. So, to put it bluntly, the risks of running such a study far outweighs the possible benefits - at least to anyone who might have the finances and the motivation to sponsor it.


----------



## KeithEmo

slaphead said:


> The problem that we have with hi res audio when it comes to ABing is that, quite frankly, nobody has an audio system that can resolve the differences.
> 
> You can argue this day and night, but until theres an audio system that can track hi res (or even 16/44) accurately then there will be no definitive answers as to whether hi res actually makes a difference at the point of delivery.
> 
> ...


 
  
 I agree with you - but you're missing part of the _BIG_ picture - at least the local version of it. Most of the arguments here are based on the idea that there wouldn't be any point in having an audio system that could resolve anything above 16/44k because 16/44k is plenty to resolve anything that we can hear - period; end.
  
 I'm more inclined to believe that, since most of the tests that claim to have "determined the limits of human hearing" were conducted on equipment of distinctly limited capabilities, many of them using what I would consider to be dubious methodology and insufficient numbers of samples and test subjects, they may in fact not be correct - or complete.


----------



## OddE

slaphead said:


> The problem that we have with hi res audio when it comes to ABing is that, quite frankly, nobody has an audio system that can resolve the differences.
> 
> You can argue this day and night, but until theres an audio system that can track hi res (or even 16/44) accurately then there will be no definitive answers as to whether hi res actually makes a difference at the point of delivery.
> 
> ...


 
  
 -Nice theory, but in reality, you don't.
  
 Adding more bit depth effectively increases the dynamic range. 24 bits? In theory, that would give you 144.48dB of dynamic range. Given that a VERY quiet room would have an ambient noise level around the 20dB(A) mark (This is incredibly generous; more likely, it will be closer to 30 for us mere mortals and the properties we can afford), to fully exploit the 24 bits capability, you'd need peaks at 165dB. That translates into instantaneous, severe hearing loss, blurred vision, problems breathing...
  
 Besides, the _instantaneous_ dynamic range of the human ear is only on the order of 30dB or so; if you are exposed to a SPL of, say, 100dB for even a very short time, your ear will effectively reduce its sensitivity to protect itself - and voila, your effective hearing threshold is increased, masking weaker sounds.
  
 As for higher sample rates, lots and lots of studies show that we simply aren't wired to take advantage of high frequency content beyond 20kHz (I'm feeling generous. Let's make that 25kHz to ensure all outliers with exceptional hearing are taken care of with margin to spare.
  
 So - as I don't feel like experiencing extreme pain before going deaf and neither do I feel like entertaining the local bats, there simply is no use for more than 16/44 or thereabouts. More bits would cause me pain if utilized; higher sampling rate would be lost on me. 
  
 So, short version - why bother?


----------



## sonitus mirus

keithemo said:


> I agree with you - but you're missing part of the _BIG_ picture - at least the local version of it. Most of the arguments here are based on the idea that there wouldn't be any point in having an audio system that could resolve anything above 16/44k because 16/44k is plenty to resolve anything that we can hear - period; end.
> 
> I'm more inclined to believe that, since most of the tests that claim to have "determined the limits of human hearing" were conducted on equipment of distinctly limited capabilities, many of them using what I would consider to be dubious methodology and insufficient numbers of samples and test subjects, they may in fact not be correct - or complete.


 
  
 And yet, there is nothing but anecdotal evidence to suggest that these results are not correct.  
  
 I concede the notion that it is possible for there to be differences, but to read about how some believe that CDs are irritating and fatiguing compared to the Pono offerings, or that well-encoded, higher bitrate lossy files make them nauseous only leaves me shaking my head and chuckling.


----------



## KeithEmo

sonitus mirus said:


> And yet, there is nothing but anecdotal evidence to suggest that these results are not correct.
> 
> I concede the notion that it is possible for there to be differences, but to read about how some believe that CDs are irritating and fatiguing compared to the Pono offerings, or that well-encoded, higher bitrate lossy files make them nauseous only leaves me shaking my head and chuckling.


 
  
 I agree.... and people love to exaggerate.
  
 A lot of people also don't seem to understand logic - and how specifics relate to generalizations.
  
 I've got about half a dozen Red Book CDs that sound - to me - ridiculously _GOOD_. Those CDs prove that the technology, and the Red Book format itself, is quite capable of sounding exceptionally good. (Which means that the CDs that sound bad are simply badly produced CDs.)
  
 Unfortunately, however, I have also heard a significant number of CDs - usually older ones - that sound especially bad. I've heard two possible explanations, both of which make sense to me. First, mastering at 16/44 requires that you apply a really sharp brick-wall low cut filter to your source at about 20 kHz, and many of the early attempts to do so didn't work very well. (Modern oversampling A/D converters eliminate the need for such sharp filters, and the technology improved over time anyway.) Second, when CDs first came out, many producers saw a need to "make sure the CD sounded obviously different than the vinyl version" to justify expecting people to buy things on CD that they already had on vinyl. And so, since vinyl is limited in terms of being able to record high frequencies at full volume, one obvious way to do this was to not only avoid the high-frequency limiting typically used with records, but to actually boost the treble even further, or to simply "take full advantage of being able to record 20 kHz at 0 dB", so the CD sounded "brighter and clearer". Together, it makes sense that early CDs might sound "shrill" or "harsh" - and I've certainly heard a few that live up to that threat.
  
 The other thing is that people tend to confuse the limitations of the sample rate and bit depth itself with the limitations of the technology - both then and now. For example, back when CDs were first introduced, the 22 kHz brick-wall low pass filter that was required as part of the A/D conversion process was a problem. The sharp-cutoff filters required were difficult to design, and even more difficult to produce, and had several serious flaws (generally poor phase response and severe ripple near the cutoff point) - which probably did contribute to the reasons why many early CDs sounded rather bad. (I recall having a few CD players, and one or two early DACs that didn't sound very good.) However, that's a limitation of the then-current technology, and not an inherent limitation of the sample rate used - and modern oversampling A/D converters avoid that limitation rather handily.
  
 I think I would go so far as to agree that _BACK WHEN CDS WERE FIRST DEVELOPED_ the choice of the 44k sample rate was a little too optimistic - because the requirement of passing audio up to 20 kHz, yet brick-walling everything above 22 kHz, was somewhat unreasonable considering the state of the art back then. It would have perhaps been sensible back then to have chosen a higher sample rate to avoid the stringent filtering requirements. However, with the advent of oversampling A/D converters, that problem was eliminated, and the format is able to deliver on its full potential quite well today.


----------



## icebear

keithemo said:


> I agree.... and people love to exaggerate.
> 
> A lot of people also don't seem to understand logic - and how specifics relate to generalizations.
> 
> ...


 
  
 I told you so


----------



## money4me247

keithemo said:


> *The biggest problem with doing a large-scale data study is simple - someone has to pay for it.*
> 
> Same deal with this thread. If you read all the responses, the one thing that's perfectly obvious is that most participants _already think they know the result_. Some are quite convinced that the difference can;t possibly be audible; others are equally certain that, at least some of the time, the difference will be audible. I haven't heard a single person saying "I have no idea whether there's a difference or not; let's find out". Note that that sort of "pure science" isn't especially common anywhere else either. You won't find a pharmaceutical company saying "we have no idea whether chemical X will help with this disease or not - let's find out" either.


 
  
 I actually started out in this hobby as a believer thinking that extremely high sampling rates and bit depth makes an enormous difference. however, when I tried to prove it to myself, I found that I couldn't.
  
 The "biggest problem" where someone has to pay for it, but no one has been willing to is actually quite revealing I think. With pharmaceutical companies, they are willing to pay for such studies because they are confident or at least optimistic that the results will be in their favor. I would think the incentive for companies selling "high-resolution" files to definitively prove that you can reliably identify the difference would be worth the costs of a large-scale study. There are actually quite a few sceptics who are offering large monetary rewards for people who can reliably pass an ABx test though I have not heard of anyone succeeding.
  
 Basically, a large-scale study will remove expectation bias as a factor as the participants will have a wide variety of different views. For individual ABx testing, yes expectation bias can play a factor in your results and stubborn people will believe whatever they want to believe. Really actually pretty pointless debating on this subject as most people participating in this discussion already have their minds made up and will not change their opinion. I would be happy to change my opinion if I could detect a difference, but honestly I personally find a lot of the subtle differences being discussed on the forums not to be relevant in real-world applications, so my perspective may be a bit different from others here.
  
 I do appreciate your perspective though.


----------



## Labtek

slaphead said:


> nobody has an audio system that can resolve the differences.





>


 
 You need 3D recordings and 3D playback. It's not real just cause it's clearer and clearer. If you want it to sound like your really there then you need to drop the whole 2D thing and go with three dimensional sound!


----------



## icebear

money4me247 said:


> *I actually started out in this hobby as a believer thinking that extremely high sampling rates and bit depth makes an enormous difference. however, when I tried to prove it to myself, I found that I couldn't.*
> 
> The "biggest problem" where someone has to pay for it, but no one has been willing to is actually quite revealing I think. ..


 
  
 There is a comment from Morten Lindberg, balance engineer and recording producer of "2L The Nordic Sound"
  
 QUOTE
 [I personally prefer extremely high resolution PCM over the DSD and I would claim that DSD is not transparent. But it all comes down to what the sound from your speakers can do to your body and mind. *I find that the placement of microphones has an infinite more important role in the final experience of music, than the difference between HiRes PCM and DSD.* Sometimes a lie can be more beautiful than the truth!]
  
 I am pretty sure that this company has their technical ducks in a very neat row when it comes to capturing the best possible sound. All this discussion about various formats clearly misses the bigger picture what is really important : That is the very first step to catch the sound: the microphone transfers a mechanical signal - sound i.e. air pressure into an electrical signal and into an A/D converter. All discussions about possible advantages afterwards about theoretical dynamic range of digital formats is pretty much blowin' in the wind.
  
 I will never listen at 96db above the ambient noise floor. I am not sure what would crash first my speakers, my headphones or my ears drums
	

	
	
		
		

		
			





. I am for sure NOT gonna find out.
 And unless it's some abstruse work using canons fired to push the max. SPL to insane levels, which kind of music has possibly a greater dynamic range than 96db?


----------



## esteboune

great thread....
  
 i do have a lot of HiRes music albums.
  
 I thought, *wrongly*, that my "expensive" audio set up will take the best of those HiRes files.
  
 it is ONLY marketing.
  
 i compared HiRes albums, to normal FLAC 16/44.1 and the results were appaling:
  
 on the best case: NO IMPROVEMENT at all
  
 on the worst CASE: LOSS of dynamic range !!!!!
  
 yes, you read correctly, *LOSS*
  
 i can tell you that i was deeply affected when i realised that the HiRes files i bought and downloaded from a well know HD website sounded worst compare to a trusty CD i ripped...
  
 The reason is simple:  if you work with poor products, you cannot expect a decent result. Those HiRes files, not all thanks God, have been created on poorly mastered album.
  
let me take and exemple:
  


 TOP: Money for nothing, Original CD, 16/44.1, 1985
 BOTTOM: same song, 24/192, from a remastered version of 2005
  
 Trust me, the original version sounds way better...
  
 i know, the problem is NOT the 24/192 files, the problem is the poorly remastered version victim of the loudness war. But all those HiRes sellers do not control the source, and do not communicate on that...
  
My conclusion is the following:
  
 IF source is the same: HiRes cannot harm, for sure. But Flac 16/44.1 is more convenient, take less memory on our devices.
  
 IF source are different: Original is usually the answer. 
  
*IF you want a good music experience, INVEST on the best possible recording/mastering, NOT on the bit rate and bit depth*


----------



## Slaphead

That Brothers in Arms album is one that almost certainly wouldn't benefit from a 24/192 version. The reason is, is that was one of the first CDs to carry the "DDD" rating, meaning that it was digitally recorderd, digitally mixed, and digitally transferred. However back in 85 there wasn't hi res tech availailable, so this all happened at 16/44 (or maybe 16/48). To create a hi res version they would have have to upsample the whole thing to 24/192. Given that upsampling on it's own would not bring about any improvement or difference in the sound it would then have been remastered to give the difference that hi res customers would expect.

If I remember rightly the original 1985 release was actually quite a quiet CD, even back then when virtually all CDs were coming out with around 6db headroom, so it's pretty inevitable that any remastering would have bumped the volume up.

As for the dynamic range thing, well less dynamic range is not always a bad thing as anything less than around 30db lower than the current loudest signal is typically not audible - it becomes masked to human hearing. If the dynamic range is reduced (compressed) then often it'll bring out details in the sound that were not previously audible - of course this can be, and often is, overcooked at the mastering stage in order to increase the perceived volume level of the track or album.


----------



## money4me247

icebear said:


> There is a comment from Morten Lindberg, balance engineer and recording producer of "2L The Nordic Sound"
> 
> QUOTE
> [I personally prefer extremely high resolution PCM over the DSD and I would claim that DSD is not transparent. But it all comes down to what the sound from your speakers can do to your body and mind. *I find that the placement of microphones has an infinite more important role in the final experience of music, than the difference between HiRes PCM and DSD.* Sometimes a lie can be more beautiful than the truth!]
> ...


 
  
 This is a really good point!!!!! +10 but ran outta kudos hahah


----------



## arnyk

money4me247 said:


> I actually started out in this hobby as a believer thinking that extremely high sampling rates and bit depth makes an enormous difference. however, when I tried to prove it to myself, I found that I couldn't.
> 
> The "biggest problem" where someone has to pay for it, but no one has been willing to is actually quite revealing I think. With pharmaceutical companies, they are willing to pay for such studies because they are confident or at least optimistic that the results will be in their favor. I would think the incentive for companies selling "high-resolution" files to definitively prove that you can reliably identify the difference would be worth the costs of a large-scale study. There are actually quite a few sceptics who are offering large monetary rewards for people who can reliably pass an ABx test though I have not heard of anyone succeeding.
> 
> ...


 
  
 There was a large scale blind test of high sample rate formats. For about the first 5-6 years it was believed that they were as sold - high resolution recordings. Millions of recordings were sold, thousands of titles were sold.  About then several people, armed with some pretty easy to make technical analyses, showed that about half of them were just low resolution recordings upsampled.
  
 Resolution can't be added back, once it is gone it is gone forever. Nobody seems to have noticed this by just listening.
  
 If there was a difference, after that goodly amount of time, large numbers of people would have noticed and critical listeners would have published lists of which recordings were truly high resolution, and which weren't., no?
  
 I think that a lot of people just want to jack up their standard for credibility higher than any available evidence so they can maintain their veil of denial.


----------



## arnyk

esteboune said:


> *IF you want a good music experience, INVEST on the best possible recording/mastering, NOT on the bit rate and bit depth*


 
  
 IF you want the best possible  music experience, INVEST on the best possible playback system, particularly transducers and room acoustics if relevant.
  
 Most recordings are available in one recording/mastering over the life of the recording.


----------



## dprimary

slaphead said:


> The problem that we have with hi res audio when it comes to ABing is that, quite frankly, nobody has an audio system that can resolve the differences.
> 
> You can argue this day and night, but until theres an audio system that can track hi res (or even 16/44) accurately then there will be no definitive answers as to whether hi res actually makes a difference at the point of delivery.
> 
> ...


 

 I would not say that. There is plenty of systems that can do 20/88 in studios and mastering suites. Not to say one wrong choice on equipment and you are back to 16/44. Yet there is still no proof that anyone can hear the difference. Twenty bit is about as far as the analog work can hope for.


----------



## arnyk

dprimary said:


> I would not say that. There is plenty of systems that can do 20/88 in studios and mastering suites. Not to say one wrong choice on equipment and you are back to 16/44. Yet there is still no proof that anyone can hear the difference. Twenty bit is about as far as the analog work can hope for.


 
  
 For extra credit, explain how the above is totally irrelevant to what I posted 
  
 Here it is again:
  
 "
 IF you want the best possible  music experience, INVEST on the best possible playback system, particularly transducers and room acoustics if relevant.
  
 Most recordings are available in one recording/mastering over the life of the recording.
 "
  
 Hint, there was no mention of bit depth.
  
 For extra credit, what is the actual active bit depth of a really good recording?


----------



## esteboune

arnyk said:


> IF you want the best possible  music experience, INVEST on the best possible playback system, particularly transducers and room acoustics if relevant.
> 
> Most recordings are available in one recording/mastering over the life of the recording.


 

 Obviously!
  
 That was an assomption for me.
  
 Let me rephrase!!!
  
 Should you have to choose between bit rate/depth and the recording quality, go for the recording


----------



## castleofargh

slaphead said:


> That Brothers in Arms album is one that almost certainly wouldn't benefit from a 24/192 version. The reason is, is that was one of the first CDs to carry the "DDD" rating, meaning that it was digitally recorderd, digitally mixed, and digitally transferred. However back in 85 there wasn't hi res tech availailable, so this all happened at 16/44 (or maybe 16/48). To create a hi res version they would have have to upsample the whole thing to 24/192. Given that upsampling on it's own would not bring about any improvement or difference in the sound it would then have been remastered to give the difference that hi res customers would expect.
> 
> If I remember rightly the original 1985 release was actually quite a quiet CD, even back then when virtually all CDs were coming out with around 6db headroom, so it's pretty inevitable that any remastering would have bumped the volume up.
> 
> As for the dynamic range thing, well less dynamic range is not always a bad thing as anything less than around 30db lower than the current loudest signal is typically not audible - it becomes masked to human hearing. If the dynamic range is reduced (compressed) then often it'll bring out details in the sound that were not previously audible - of course this can be, and often is, overcooked at the mastering stage in order to increase the perceived volume level of the track or album.


 

 it's the first CD I heard ^_^. it did quite a number on me TBH, a 11 or 12year old kid so proud of his k7 walkman turned into an instant hiss hater overnight. when the dobly stuff came out they did help for hiss though.


----------



## OddE

castleofargh said:


> it's the first CD I heard ^_^. it did quite a number on me TBH, a 11 or 12year old kid so proud of his k7 walkman turned into an instant hiss hater overnight. when the dobly stuff came out they did help for hiss though.


 
  
 -I still (20 years or so after last recording onto cassette) have a very ambivalent relationship to Dolby NR. As long as I played the tapes back on the same device they were recorded on, B worked pretty well, whereas I often heard a pumping sound if I used C. (Could be down to a poorly adjusted deck, though). Start using the tapes in another deck, though, and all bets were off. I don't know how portable Dolby intended NR-recorded tapes to be, but I found that results were hit-and-miss when swapping tapes between decks.
  
 I eventually landed on using only metal tapes (Maxwell MX-S, if memory serves) and no NR - then a couple of years later, a DAT deck solved all of those issues and then some.  (DAT is still (somehow) alive and well, by the way - I recorded onto a tape only the day before yesterday!)


----------



## KeithEmo

money4me247 said:


> I actually started out in this hobby as a believer thinking that extremely high sampling rates and bit depth makes an enormous difference. however, when I tried to prove it to myself, I found that I couldn't.
> 
> The "biggest problem" where someone has to pay for it, but no one has been willing to is actually quite revealing I think. With pharmaceutical companies, they are willing to pay for such studies because they are confident or at least optimistic that the results will be in their favor. I would think the incentive for companies selling "high-resolution" files to definitively prove that you can reliably identify the difference would be worth the costs of a large-scale study. There are actually quite a few sceptics who are offering large monetary rewards for people who can reliably pass an ABx test though I have not heard of anyone succeeding.
> 
> ...


 
  
 Concepts of scale - like "enormous", "significant", and even "large" are all relative. I know one or two old time NON-audiophiles who consider the change from old-school portable "transistor radios" to modern "hi-fi equipment" ones to be "a major step" because, on the new ones, the voices are actually clear and easily understood. Likewise, they rate a "music system" on whether you can tell what the announcers are saying most of the time or not, and they don't consider fidelity beyond that point to be important at all. (And a lot of modern listeners would probably say that the most important thing to them is whether the ear buds fall out in the gym or not.)
  
 As I mentioned in another post, I firmly believe that the reason most music vendors aren't that interested in sponsoring a major study is that, from their point of view, there is little to no chance that they will get a result that will significantly help sales. If a study were to prove that all - or even most - people couldn't hear the difference, then they would lose "high-res" as a purchasing incentive, and, even if such a study were to prove that a lot of people could hear a difference, but that most of them considered it to be a minor difference, it would still reduce the value of "high-res" as an incentive to buy. (And, ignoring whether any difference actually exists or not, I'm virtually certain that, in a world where the majority of people consider HD FM, MP3, and AAC to be "good enough", not that many of them would consider the difference between Red Book and HD to be "significant" or "important".) As long as only audiophiles are willing to pay a premium for high-res tracks, it will remain a small niche market and, as long as most people really think MP3 and AAC files are "good enough", it's likely to remain that way. And the only way a major study would help the music industry, and justify its cost, would be if it was actually likely to change minds - and so lead to an increase in sales. As it is, the "mystique" surrounding high-res audio files is probably worth more than the reality to the folks who are making the money.
  
 I would also be curious to know what the sales of high-res files really are, and how much of that is "marginal sales". (By which I mean how many _more_ copies of the latest re-master were sold because they were high-res, as compared to how many copies of every re-master are sold anyway. If it turns out that a large percentage of the sales of high-def files are simply to people who chose that format over the regular format, but would have bought the album anyway, then those sales benefit the companies who specialize in high-res file sales, but don't benefit the music industry overall nearly as much.) My personal suspicion is that, from a marketing perspective, simply having high-res re-masters as a "hot topic" reminds people to buy music, which is a net positive for the industry. (The 14th re-master of In the Heat of the Night wouldn't rate mention as a "news event", but the "high-res re-master" was worth a few articles, which fact, in and of itself, might have helped sales of the album.)    
  
 (To be totally honest here, the fact that a skeptic is willing to offer a reward that they are quite certain they will never have to pay out is not the same as someone paying money "up front" for a study that may cost them more than it profits them, and may actually cost them sales. )


----------



## FFBookman

The colossal problem is the ABX test itself.  Not the poor results. They are garbage and should be treated as such.
  
 The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners.  ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing. No one can hear anything.  There is no such thing as quality is it's primary conclusion.
  
 Since we all know that to be untrue, we must focus on the test that is proving this. It's because these studies don't deal with sound and music quality, they are either medical in nature, or deal with confusing consumerism and catch phrases. 
  
 ABX tests are very good for pharmaceutical companies and other product-based ventures.
  
 Music is so emotional and contextually personal to the listener that artists and purists with passion drive the industry of "what sounds good", not scientists with measurements.  
  
 In the last 5 years alone they are rewriting what they know about the inner cochlea, the nerves under hair follicles, and the speed and nature of the connection to the brain.


----------



## RRod

ffbookman said:


> The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners.  ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing. No one can hear anything.  There is no such thing as quality is it's primary conclusion.


 
  
 If you sample the music down where theory says you should hear differences (sample rate below twice you hearing threshold, bit depth insufficient for the dynamic range of the music), you'll be able to ABX the differences. Give it a try sometime.


----------



## castleofargh

ffbookman said:


> The colossal problem is the ABX test itself.  Not the poor results. They are garbage and should be treated as such. *as always where is the controlled reliable better alternative? what's the point of criticizing a system when you have nothing to offer that can actually hold a candle to ABX when it comes to reliability? how many times did we ask you for this?  *
> 
> The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners. *of course ABX is used to testing the audibility of sound differences , not for rating quality. maybe you also want to explain to us how spoons aren't good to cut down trees? start by learning what ABX is before pretending to explain it to us.  if you don't do it for yourself, do it for everybody else who has to read your nonsense over and over again.* ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing.* that's a blatant lie there are plenty of ABX tests giving positive results and I've used ABX to find out a threshold of my hearing plenty of times, with mp3 compression rates, with bit depth, with noise floor...  *No one can hear anything.  There is no such thing as quality is it's primary conclusion. *preparing straw man argument phase 1: "you all have emotions, therefore you must agree with me whatever my point was"*
> 
> ...


----------



## money4me247

ffbookman said:


> In the last 5 years alone they are rewriting what they know about the inner cochlea, the nerves under hair follicles, and the speed and nature of the connection to the brain.


 
 really? just curious what is your source for that statement? the majority of physiological and histological information on the ear has been pretty extensively covered and considered to be well understood for quite some time now. no real major rewrites/revolutions on our understanding of how the ear works to my knowledge. would love to learn more if you care to share.


----------



## old tech

money4me247 said:


> really? just curious what is your source for that statement? the majority of physiological and histological information on the ear has been pretty extensively covered and considered to be well understood for quite some time now. no real major rewrites/revolutions on our understanding of how the ear works to my knowledge. would love to learn more if you care to share.


 
 Me too!


----------



## arnyk

ffbookman said:


> The colossal problem is the ABX test itself.  Not the poor results. They are garbage and should be treated as such.
> 
> The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners.  ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing. No one can hear anything.  There is no such thing as quality is it's primary conclusion.
> 
> ...


 
  
  
 The above appears to me to be an poor and unconvincing  example of a collection of completely unsupported personal opinions belonging to a person with no credibility, or reputation. People who are unfamiliar with poorly-crafted rhetoric (hard to believe given this we are this close to national elections) should study it in order to find out how to systematically fail to convince reasonable people of anything.
  
 In fact thousands of amateurs and listeners have applied the ABX (1968) methodology to the problem of comparing musical selections that are thought to possibly be different and obtained useful and satisfying results both positive and negative in nature. The development of many audio products including loudspeakers and perceptual coders have greatly facilitated by the development of more reliable means for performing listening tests.
  
 BTW one of the common properties of anti-scientific writing is vague and untraceable  references to revolutionary scientific findings such as: 
 "In the last 5 years alone they are rewriting what they know about the inner cochlea, the nerves under hair follicles, and the speed and nature of the connection to the brain."
  
 Interesting assertion, but how would such detailed knowledge make global and earth shaking changes in what we know about relevant issues like hearing thresholds and masking?   Our current scientific understanding of hearing goes back to no later than Helmholtz   (August 31, 1821 – September 8, 1894) and has undergone VERY MANY  evolutionary changes and enhancements since then. That's the nature of science. This is almost 200 YEARS!
  
If the 5 years of actual work existed it would be citeable, but it does not appear to citeable by he author. Therefore it appears to be lacking in credibility.


----------



## dprimary

dprimary said:


> I would not say that. There is plenty of systems that can do 20/88 in studios and mastering suites. Not to say one wrong choice on equipment and you are back to 16/44. Yet there is still no proof that anyone can hear the difference. Twenty bit is about as far as the analog work can hope for.


 
 Quote: 





> Originally Posted by *arnyk* /img/forum/go_quote.gif
> 
> 
> For extra credit, explain how the above is totally irrelevant to what I posted
> ...


 
 It is completely irrelevant since it was not referring to any of your posts. I was referring to the straw man argument that there is no systems with flat extend frequency response. There is plenty of audio monitors that go out to 30-40k. Most studios have spent considerable amount of money on room acoustics and monitors. So it can't be claimed high bandwidth systems do not exist. 
 You would be lucky to achieve with great care 12 to 14 bits of depth even in a extremely low noise floor room most studios are in the NC25 range.


----------



## LajostheHun

slaphead said:


> That Brothers in Arms album is one that almost certainly wouldn't benefit from a 24/192 version. The reason is, is that was one of the first CDs to carry the "DDD" rating, meaning that it was digitally recorderd, digitally mixed, and digitally transferred. However back in 85 there wasn't hi res tech availailable, so this all happened at 16/44 (or maybe 16/48). To create a hi res version they would have have to upsample the whole thing to 24/192. Given that upsampling on it's own would not bring about any improvement or difference in the sound it would then have been remastered to give the difference that hi res customers would expect.


 
 BIA was recorded on a Sony 24 track digital  tape recorder, but mixed in analog then it was re-digitized using a Prism AD converters then recorded on a DAT machine. This is according to Neil Dorfsman the producer on the project. So it should be a DAD, not DDD. Having said that none of this alter  the point you're making here.


----------



## dprimary

ffbookman said:


> The colossal problem is the ABX test itself.  Not the poor results. They are garbage and should be treated as such.
> 
> The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners.  ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing. No one can hear anything.  There is no such thing as quality is it's primary conclusion.
> 
> ...


 

 What is the problem with ABX? Almost every mastering console made can level match and AB signal chains and processing. Are all mastering consoles garbage? That would make about every recording ever made garbage. I do not understand how anyone that records for a living has any problem with blind AB tests. The entire recording process is a constant AB test. Which mic sounds better, what pattern, which placement, which preamp, and on and on, hundreds to thousands of AB comparisons. After a few million AB comparisons I don't care who makes it, what it costs, analog, digital tube, solid state. I just use my ears and listen, having to see is just adding to the noise and distraction.


----------



## arnyk

dprimary said:


> It is completely irrelevant since it was not referring to any of your posts. I was referring to the straw man argument that there is no systems with flat extend frequency response. There is plenty of audio monitors that go out to 30-40k. Most studios have spent considerable amount of money on room acoustics and monitors. So it can't be claimed high bandwidth systems do not exist.
> You would be lucky to achieve with great care 12 to 14 bits of depth even in a extremely low noise floor room most studios are in the NC25 range.


 
  
 Who made the claim that there are no systems with flat extended frequency response?  Looks like another straw man to me!
  
 One problem with flat response beyond 20 KHz is that speakers that offer it (I have several) are usually very directional at those frequencies and that most things in a room that bounce sound are more absorbent of high frequencies, including the air in the room.
  

  
  
 This also affects recording studios and performance spaces.
  
 While there has been a slight increase in professional recording microphones with extended high frequency response, very many of them start rolling off agressively below 15 KHz.


----------



## dprimary

lajosthehun said:


> BIA was recorded on a Sony 24 track digital  tape recorder, but mixed in analog then it was re-digitized using a Prism AD converters then recorded on a DAT machine. This is according to Neil Dorfsman the producer on the project. So it should be a DAD, not DDD. Having said that none of this alter  the point you're making here.


 
 DAT machines did not come out till a few years later. DDD means digital multitrack, digital master, digital disk. There was not any production digital consoles in 85 it was the early 90's before we had a digital console that handle 24 track mix downs, and those did not sound that good.


----------



## LajostheHun

dprimary said:


> DAT machines did not come out till a few years later. DDD means digital multitrack, digital master, digital disk. There was not any production digital consoles in 85 it was the early 90's before we had a digital console that handle 24 track mix downs, and those did not sound that good.


 
 https://en.wikipedia.org/wiki/Brothers_in_Arms_(Dire_Straits_album)


----------



## dprimary

arnyk said:


> Who made the claim that there are no systems with flat extended frequency response?  Looks like another straw man to me!
> 
> One problem with flat response beyond 20 KHz is that speakers that offer it (I have several) are usually very directional at those frequencies and that most things in a room that bounce sound are more absorbent of high frequencies, including the air in the room.
> 
> ...


 
 I was responding to slaphead in original post. Other people have posted the same types of comments.
 There is no reason to claim we can't make a system that can go out to 30-40k is is not all that hard anymore. We can capture it record it and reproduce it yet so far nobody has been to prove they can hear it. 
 As you point out even I could hear 30-40 kHz going to symphony sitting in the sweet spot of the hall, I would not hear it because to air will have filtered it out. 
 You get to stadium sized venues and 8k is a miracle.


----------



## ExistentialEAR

with above 20khz the point is you may not hear it but you may sense it, it make allright to da brains.


----------



## arnyk

dprimary said:


> DAT machines did not come out till a few years later. DDD means digital multitrack, digital master, digital disk. There was not any production digital consoles in 85 it was the early 90's before we had a digital console that handle 24 track mix downs, and those did not sound that good.


 
  
  
 Quite a bit detail can be found about the equipiment they used particularly:
  
http://www.soundonsound.com/sos/may06/articles/classictracks_0506.htm
  
 and for more detail about the mixing console whose performance sets global limits:
  
http://thehistoryofrecording.com/Manuals/SSL/SSL_SL_4000G_Series_Manual.pdf
  
 Putting these documents together it seems easy to confirm that mixing was done in the analog domain as the primary SSL mixing console was purely analog.
  
 We also have information about the mics used on the drum kit, which were:
  
 "Utilising the limited space to best effect, the drum kit sat in the far left corner of the live area, facing the control room, and was miked with Sennheiser MD421s on the toms, an Electrovoice RE20 and AKG D12 on the kick drum, a Shure SM57 and AKG C451 with a 20dB pad on the snare, 451s for overheads and the hi-hat, and Neumann U87s set back a little in order to attain at least some kind of ambience."
  
 None of these have exceptionally extended response.   The 451 is among the more extended of the group and it starts rolling off at 15 KHz:


----------



## dprimary

lajosthehun said:


> https://en.wikipedia.org/wiki/Brothers_in_Arms_(Dire_Straits_album)


 

 DAT recorders did not come out till 87. He might have said DASH and the writer changed it to DAT. I have been recording digital and analog since 84. I never knew Omar Hakim played drums on that album.


----------



## arnyk

dprimary said:


> I was responding to slaphead in original post. Other people have posted the same types of comments.
> There is no reason to claim we can't make a system that can go out to 30-40k is is not all that hard anymore. We can capture it record it and reproduce it yet so far nobody has been to prove they can hear it.
> As you point out even I could hear 30-40 kHz going to symphony sitting in the sweet spot of the hall, I would not hear it because to air will have filtered it out.
> You get to stadium sized venues and 8k is a miracle.


 
  
  
 Obviously there's a problem with communication here. I have already said a few posts back that I own speakers with response up to 40 KHz, and while I did not brag about it, I did post the frequency and directional response of their tweeters as evidence.
  
 I have made studio recordings with observable response out to 40 KHz going back into the early 2000s. 
  
 Of course I am laughing inside when people  lecture me about the possibility of doing these things!
  
 However, doing good listening tests showing any audible benefits for any of is far easier said than done.


----------



## dprimary

arnyk said:


> Quite a bit detail can be found about the equipiment they used particularly:
> 
> http://www.soundonsound.com/sos/may06/articles/classictracks_0506.htm
> 
> ...


 
 It would have been a sony 3324 24 track and SSL 4000E in 85 with a 8" floppy disk (try to find one those today) More likely they mixed down to a 3402 sony or a Mitsubishi x-86. DDD only applied to the recording mediums, mixing was all analog.
 Prism didn't exist till 1988.


----------



## KeithEmo

ffbookman said:


> The colossal problem is the ABX test itself.  Not the poor results. They are garbage and should be treated as such.
> 
> The test format is completely inappropriate for determining the effect of sound quality of recorded music on listeners.  ABX applied to general hearing tests of music can only produce what is statistically insignificant results, aka nothing. No one can hear anything.  There is no such thing as quality is it's primary conclusion.
> 
> ...


 
  
 I disagree entirely - not with most of your supporting facts, but with the claim itself.
  
 Measurements are _NOT_ something different from experience. Measurements are simply a way of quantifying experience... often for purposes of enabling communication.
  
 It's not especially meaningful for me to tell you that "System X is loud"; because "loud" isn't well defined. Therefore, it's more useful to say "System X is playing at 102 dB SPL"; because we both agree on what that means. Saying that "something sounds good" is also rather meaningless for purposes of communication; because it means different things to different people. Therefore, if you and I wish to conduct a _MEANINGFUL_ dialog about how various equipment sounds, we need to have some agreed-upon framework to use. The problem I see with most audiophiles is that they simply don't understand the science, and so don't do this very well... and this is usually due to a combination of insufficient understanding _AND_ insufficient information.
  
 A perfect example is "THD" (distortion). I see endless discussions about "whether things with low distortion sound good" and "whether low-feedback designs sound better even though they have higher distortion". The problem with any discussion at this level is that the terms of the discussion are being overgeneralized. You might as well discuss whether green fruits taste better than red ones. The reality is that there are a huge variety of different types of distortion, and most circuits have several of them in varying amounts, and, in order to have a useful discussion, a single number value simply isn't good enough.
  
 To say that the difference between a typical low-feedback single ended triode tube amplifier and a high feedback solid state design is that the tube amp has much higher distortion is about as specific - or as useful - as saying that most Ferraris are orange or red. A more useful level of detail would be to say that the triode amp has much higher levels of distortion, but it's distortion spectrum is such that it has mostly second order harmonics, while the solid state amp has much lower total distortion, but much higher percentages of higher harmonics. Even more useful would be to provide a spectrum analysis showing the actual amounts of each harmonic... Even more useful that that would be to provide several such graphs, taken at different frequencies and power levels. (And we also need to include the fact that those statements are generalizations: I could deliberately design a high-feedback solid state design that mimicked the distortion spectrum of a low feedback tube design, and would sound exactly like one.)
  
 Your statement about "statistically insignificant" is also almost the exact opposite of the truth - as is your implication that audio is not "a product-based venture".
  
 If I were planning to sell music downloads, the results of an ABX test showing that the majority of my potential customers were unable to tell the difference between a 24/96k download and a 128k MP3 would be _VERY_ statistically significant. To me, it would show that I needn't waste a lot of effort and space on higher quality downloads; and, to my customers, it would show that they probably won't hear anything wrong if they buy my smaller, lower quality, and cheaper downloads. Audio and music companies are in fact "product-based ventures"... and do in fact base the particulars of their products on what the majority of their customers want. (Note, however, that a company who sells :high-resolution" music to "audiophiles" is going to be matching their products to a different target audience than "a popular on-the-go streaming service".)  The trick is to avoid misinterpreting "results show that most people don't hear a difference" as meaning that "there is no difference" or even that "_NOBODY_ can hear a difference". My point here is that a particular ABX test may be very useful, and very accurate, for a big music company like Sony, but still be meaningless for an audiophile; but that doesn't mean that the test is "bad", simply that it may not be "the right test" for what you're hoping to find out. (The important thing is to understand the science well enough to figure out how well that test pertains to your particular needs.)
  
 Once you learn to avoid overgeneralizations, and to understand what the results of a given test actually mean, that information is still a lot more useful than "Keith thinks product A sounds great" or "Joe likes product B better". The simple fact is that information about someone else's "emotional response" to something is relatively useless (unless you know that person very well, and know for a fact that _YOUR_ emotional response will be similar to theirs. (So standardized results are usually more useful in the end.)
  
 I have friends who like the way tube equipment sounds; personally I do not. When we listen, we both hear the same thing; but they find it pleasant while I find it annoying and unpleasant. However, since we both understand the measurements behind the differences in sound, they can still predict reasonably well whether I will like a given piece of equipment, and I can predict what they will like. Without those measurements as common ground, we really couldn't have a meaningful conversation about audio equipment; but with them we can.
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
 There is nothing whatsoever "wrong" with a properly conducted ABX test. A properly run ABX test will in fact tell you if a specific group of people were able to distinguish between two test samples or devices under specific conditions, which is precisely what it was designed to do. The "flaw" is in how many people choose to interpret that data.


----------



## castleofargh

+1 but you're talking about interpreting the data when he doesn't even understand what ABX is used for(doesn't stop him from judging though...). we're not at the level of "how to objectively interpret data", or "why we can't prove a negative with science". we're more at a RTFM stage with the guy.


----------



## old tech

You're right with most things that you say re measurements, common standard etc. However while it is true that a tube amp can be measured and sound different to different people, I don't see its relevance to your example of hi res files. A high resolution file may sound different to me and you, but if neither of us can tell difference between 24/96 and 16/44 under a controlled ABX, then we those sound differences to us will be exactly the same if it was a 16/44 file. Or put it another way, if you ABX the sound from your friend's valve amp with a good SS amp, you will pick the difference in an ABX test and so to will your mate. But if you ABX'd the same valve amp on two identical amps you should get the same result as ABX'ing 16/44 vs 24/96, ie no difference.


----------



## Speedskater

_But if you ABX'd the same valve amp on two identical amps you should get the same result as ABX'ing 16/44 vs 24/96, ie no difference._
  
 Is there a typo here?


----------



## old tech

Yep, got my tongue twisted as well.


----------



## arnyk

speedskater said:


> _But if you ABX'd the same valve amp on two identical amps you should get the same result as ABX'ing 16/44 vs 24/96, ie no difference._
> 
> Is there a typo here?


 
  
  
 Whatever, there seems like a strong possibility of one or more false claims.
  
 Without knowing a lot more details it is impossible to know the outcome of a listening test before you do it.
  
 For example, all high resolution recordings don't sound the same and they don't compare the same the same to all low resolution recordings.
  
 For another example, all tubed amps don't sound the same and they don't compare the same the same to all SS amps.
  
 Things like listener training and qualifications, recordings used for the comparison and the actual quality of the monitoring systems used for the comparison as well as the actual products being compared can strongly influence the outcome of the listening test.


----------



## upstateguy

castleofargh said:


> +1 but you're talking about interpreting the data when he doesn't even understand what ABX is used for(doesn't stop him from judging though...). we're not at the level of "how to objectively interpret data", or* "why we can't prove a negative with science"*. we're more at a RTFM stage with the guy.


 
  
 This got me thinking on a basic level.
  
 If someone says that they can hear the difference between two amplifiers, that claim is either provable or not.
  
 If someone says that they don't hear a difference between the same two amplifiers, that claim is a negative and can't be proven.
  
 But what about the claim that the two amplifiers can be made to sound the same?
  
 Then, after achieving that goal, can the claim be made that both amplifiers sound the same? 
  
 Claiming that both amplifiers sound the same is a positive claim and should be able to be proven, but isn't it really  just another way of restating the "don't hear a difference" claim that can't be proven.
  
 Confusing?


----------



## Ruben123

Problem is even bigger when you think that "not hearing a difference" might be problematic for other parties too - who DO hear a difference. While scientifically the party that does hear a difference has to prove it, someone who hears a difference but says to do not so gives even more problems. By placebo. By example: comparing 64kbps mp3 vs 320kbps and saying "its mp3 it sounds the same". How to prove he is wrong.


----------



## jcx

positive controls are allowed - have tests that from the literature are audible in DBT included in the trials - ignore all results from any subject not discriminating the known audible differences
  
 training is allowed, recommended too - humans seem to be able to succeed at quite remarkable things that almost everyone fails at on 1st try, things that are unremarked parts of a practiced pro or serious amateur's ordinary activity in their area of expertise
  
 cash rewards for positive results should help too - which is why we point out that there are no golden ears out there boasting that they paid for their cables by taking Richard Clark's $10k from him


----------



## icebear

There will never be a conclusive verdict on this topic*.
  
 It always matters under which circumstances the music is just "consumed" or devotedly listened too. If it's just background noise to get someone through the day, the commute, whatever or if the music listening experience is the sole point of attention.
 Beyond a certain level of quality there is no objectively better, it's just a tiny bit different and a pure matter of personal preference what someone likes better.
 If the music speaks to you, stirs emotion, stops you in your tracks and calls your full attention ... this is good enough, at least in my book.
 Be bold and make a decision what is good enough for yourself and stop endless comparisons and running after the latest and greatest marketing claim.
 Enjoy the music.
  
  
 * Even if there was, would it change anything one way or the other?
 If there was proof of no audible difference, would this end sales of high rez files? No.
 If there was proof that high rez is noticeably better than 16/44.1, would it be the end of CD and comparable loss less downloads? No.
 As always the consumer votes with their disposable income, ... or at least their credit card.


----------



## interpolate

CD is definitely better than regular MP3 and high-res is definitely remarkably defined. Can we just say it's way better than a tin-can and string.


----------



## Thad-E-Ginathom

icebear said:


> There will never be a conclusive verdict on this topic*.
> 
> It always matters under which circumstances the music is just "consumed" or devotedly listened too. If it's just background noise to get someone through the day, the commute, whatever or if the music listening experience is the sole point of attention.
> Beyond a certain level of quality there is no objectively better, it's just a tiny bit different and a pure matter of personal preference what someone likes better.


 
   
 But what if it isn't, even a _tiny_ bit different? If it then comes down to preferring the  label that says 192 to the label that says 44.1,  we might as well talk about which colour we like best.
  
 Quote:


> If the music speaks to you, stirs emotion, stops you in your tracks and calls your full attention ... this is good enough, at least in my book.
> Be bold and make a decision what is good enough for yourself and stop endless comparisons and running after the latest and greatest marketing claim.
> Enjoy the music.


 
  
 In practice, and when I am listening to music, I agree entirely. I've seen people file their music collections under formats, sample rates, etc --- and I wonder why! Once it is in my collection, and especially if it was the only copy I could get, I don't care if it is allegedly high-res, a scratchy vinyl rip, or even grabbed from youtube: it's music.


----------



## sonitus mirus

thad-e-ginathom said:


> ...we might as well talk about which colour we like best.


 
  
 If you can even see all of the colors.
  
 https://www.linkedin.com/pulse/25-people-have-4th-cone-see-colors-p-prof-diana-derval


----------



## Gr8Desire

upstateguy said:


> This got me thinking on a basic level.
> 
> If someone says that they can hear the difference between two amplifiers, that claim is either provable or not.
> 
> ...


 
  
 If you are trying to rationalize a hobby then these questions are worthwhile. 
  
 AB testing will easily show that all amps don't sound the same. I don't think anyone disputes that. Unfortunately ABX testing also shows that many listeners think the same amp sounds different. The false positive rate increases when you lengthen the time between samples, as does the rate of false positives._ What conclusion should you draw?_

 Perhaps a better question to ask: *Which amp sounds better?

 Conclusion: *If you get a consensus, you can look for measurable attributes like wattage and %THD to help define thresholds for _better_.  

*Problem*: After you pass the lowest thresholds you can't get a consensus on _better_. Go a bit higher and you can't get a consensus on _different_. So you are back to all amps sounding the same _once they are good enough_.

 Simple?


----------



## Thad-E-Ginathom

sonitus mirus said:


> If you can even see all of the colors.
> 
> https://www.linkedin.com/pulse/25-people-have-4th-cone-see-colors-p-prof-diana-derval


 
  
 Looks like I have four kinds of cone. Does that mean I can hear stuff that others can't?


gr8desire said:


> If you are trying to rationalize a hobby then these questions are worthwhile.
> 
> AB testing will easily show that all amps don't sound the same. I don't think anyone disputes that. Unfortunately ABX testing also shows that many listeners think the same amp sounds different. The false positive rate increases when you lengthen the time between samples, as does the rate of false positives._ What conclusion should you draw?_
> 
> ...


 
_SImple_ is being rational from the outset. There should be no need to rationalise this hobby: how and why did it get _irrational?  _My diagnosis: mass hysteria aggravated by mass-ive egos.
  
 What's so tough about accepting that we might not have heard something we think we heard? It is hardly life-threatening, but audio forums and meetings give the impression that it strikes at the very basis of reality. What humbug!


----------



## dazzerfong

thad-e-ginathom said:


> What's so tough about accepting that we might not have heard something we think we heard? It is hardly life-threatening, but audio forums and meetings give the impression that it strikes at the very basis of reality. What humbug!


 

 Simple: when you sunk $100k into something, and found out it's all for nought, that's generally the first thing you fall into: denial.


----------



## Thad-E-Ginathom

Well, yes... Been there; suffered the various forms of post-purchase psychology. Never had the chance to have it cost me _that_ much though. 
  
 And, of course, being but a _recovering_ audiophile, part of me still wonders...


----------



## StanD

Once the mob's loudmouth spews a story, all bets are off.


----------



## Slaphead

sonitus mirus said:


> If you can even see all of the colors.
> 
> https://www.linkedin.com/pulse/25-people-have-4th-cone-see-colors-p-prof-diana-derval




And

http://www.snopes.com/politics/medical/tetrachromacy.asp

You can't test for tetrachromacy on a computer screen as a computer screen is not capable of generating the range of colours needed to test for the condition.


----------



## KeithEmo

thad-e-ginathom said:


> In practice, and when I am listening to music, I agree entirely. I've seen people file their music collections under formats, sample rates, etc --- and I wonder why! Once it is in my collection, and especially if it was the only copy I could get, I don't care if it is allegedly high-res, a scratchy vinyl rip, or even grabbed from youtube: it's music.


 
  
 I have to both agree and disagree on your statement there - and quite strongly (but not for aesthetic reasons). I always file my music by file type, but the reason is purely practical. If I always have to go over to my MP3 folder to find the only copy I have of some of my old favorite songs, then that reminds me that I should, at least occasionally, be checking to see if a better quality copy has become available. (Between re-masters and uploads, new versions of songs or albums frequently become available, so that's not an unlikely possibility.) And that's doubly true for something like Youtube, where there is a huge amount of variation between different copies of "the same" music or video. (So, yes, if I notice that I've listened to that same poor quality MP3 five times this week, then that prompts me to go on an "expedition" on eBay for a better copy, or even a better quality video on Youtube.)


----------



## Thad-E-Ginathom

I'll do that if the music quality is irritating, and I get the sudden inspiration to look for better. I would never do it _because_, for instance, hey, only 64kbps. If it is listenable, it is listenable, and I don't even consider the file format.
  
 Irritating could be a bad vinyl rip (eg from one of my own scratched LPs!) as often as it is a symptom of low bit rates.
  
_disclosure: _My relative-to-many-others fairly-small digital music library is mostly of an _arbitrary  _standard that says, lossless FLAC wherever possible, and 320- or 180-MP3 where not. The reason I say arbitrary, is that I have never taken the trouble to _find out_, by listening tests, at what point I, personally, really can hear the difference.
  
 I do keep lossy copies for portable use, in another location. Probably nothing _over_ 180-MP3.


----------



## KeithEmo

thad-e-ginathom said:


> I'll do that if the music quality is irritating, and I get the sudden inspiration to look for better. I would never do it _because_, for instance, hey, only 64kbps. If it is listenable, it is listenable, and I don't even consider the file format.
> 
> Irritating could be a bad vinyl rip (eg from one of my own scratched LPs!) as often as it is a symptom of low bit rates.
> 
> ...


 
  
 That's pretty well the way I look at it.... although I do extend that to include "high-res versions whenever they happen to be available". Ignoring the heated discussion about whether there is an audible difference in high-res files _because_ they're high-res, I have found that many of the recent high-res re-masters do sound significantly better to me than previous releases. For example, I find the recent Grateful Dead Studio Remasters to be far superior sounding - to me - than any previous release (and there have been a lot). Now, whether that's because they're high-res, or simply because they're the latest re-master, or because a little more effort was spent to make them sound good because they're "an audiophile release", doesn't specifically influence my decision to buy them; I'm buying them because they sound better _for whatever reason_. I would also note that many of the recent high-res re-masters _DON'T_ sound any better to me than any other versions that are available - for whatever reason - but I'm basically taking that same gamble whenever I buy any re-master, whether it's high-res or not (which is why it's always a good idea to follow the reviews).


----------



## Thad-E-Ginathom

Grateful Dead


----------



## Ruben123

Woa some science guys (+ me one time) have had discussions in the pono topic about the benefits of 24 bits (and other things) and we got flamed because it was off topic and "discuss it elsewhere". Now it's almost 2 pages talk about Win 7,8,10... How is that not off topic ?!


----------



## RRod

ruben123 said:


> Woa some science guys (+ me one time) have had discussions in the pono topic about the benefits of 24 bits (and other things) and we got flamed because it was off topic and "discuss it elsewhere". Now it's almost 2 pages talk about Win 7,8,10... How is that not off topic ?!


 
  
 Because that topic doesn't go against the underlying philosophy of the Pono. We Redbook people are infidel defilers, and any mention of bits or samples is thread-crapping.


----------



## jcx

> I find the recent Grateful Dead Studio Remasters to be far superior sounding - to me - than any previous release (and there have been a lot). Now, whether that's because they're high-res, or simply because they're the latest re-master, or...


 
 Plangent Processing? - I know GD has used them in the past - if you can get original tapes you can strip out the "FM" modulation, "scrape flutter" mechanical tape motion errors from bearing noise, rubbing, stick-slip by digitizing at high enough sample rate to capture the recorder's AC Bias tone and use that as your timebase to back out the tape speed errors


----------



## interpolate

Celemony offer an effective yet very expensive solution to tape transfers with Capstan.


----------



## sonitus mirus

keithemo said:


> That's pretty well the way I look at it.... although I do extend that to include "high-res versions whenever they happen to be available". Ignoring the heated discussion about whether there is an audible difference in high-res files _because_ they're high-res, I have found that many of the recent high-res re-masters do sound significantly better to me than previous releases. For example, I find the recent Grateful Dead Studio Remasters to be far superior sounding - to me - than any previous release (and there have been a lot). Now, whether that's because they're high-res, or simply because they're the latest re-master, or because a little more effort was spent to make them sound good because they're "an audiophile release", doesn't specifically influence my decision to buy them; I'm buying them because they sound better _for whatever reason_. I would also note that many of the recent high-res re-masters _DON'T_ sound any better to me than any other versions that are available - for whatever reason - but I'm basically taking that same gamble whenever I buy any re-master, whether it's high-res or not (which is why it's always a good idea to follow the reviews).


 
  
 I find the Grateful Dead remasters to sound outstanding as well, even with the Google Play 320 kbps streaming mp3 files.  I am very familiar with this music, and I prefer the latest remasters to the CDs that I own.  I had all of the studio releases on vinyl back in the day, as well, and the latest remasters sound better than anything I can recall hearing before.  Maybe the hi-rez versions are even better, but I doubt it.  If I wasn't already extremely happy with the streaming version, this would be another opportunity for me to test a hi-rez file converted to Red Book or lossy.  I think I will save my money this time, as I am fairly certain how the ABX will result.\
  
 Edit: This is the HDTracks version of the music in Google Play that I have been listening to with the wonderful sound quality.  http://www.hdtracks.com/complete-studio-albums-collection


----------



## OddE

sonitus mirus said:


> I find the Grateful Dead remasters to sound outstanding as well, (...)


 
  
 -Excellent news, I guess I'd better pick a few of them up - while I've got just about every album they ever put out, plus most of Jerry's pursuits outside the Dead, the earlier remaster offerings have left me underwhelmed, to say the least. If they finally got it right, this'll be a classic case of 'Shut up and take my money!!!'


----------



## sonitus mirus

odde said:


> -Excellent news, I guess I'd better pick a few of them up - while I've got just about every album they ever put out, plus most of Jerry's pursuits outside the Dead, the earlier remaster offerings have left me underwhelmed, to say the least. If they finally got it right, this'll be a classic case of 'Shut up and take my money!!!'


 
  
 With regards to the studio releases, it is the Complete Studio Album Collection which I find to sound the best on Google.  I have not listened to any others lately, but I would have confidence with any of the latest Rhino releases.


----------



## FFBookman

Keep running people through those stupid tests and you'll continue with this horrible sounding "perceptual coding" world of lowest common denominator audio. Netflix streams 6Mbs to most of the residences in america but 1.4Mbs of audio should be reduced to 0.25Mbs?
  
 That's why effective bitrate is the only true measurement of the format, assuming it's the same encoding.
  
 If you have stereo PCM data - at what native resolution was it recorded at? What was it mixed and mastered at? That is the resolution it should be heard at. Simple. 
  
 Downsampling is only done for convenience, aka storage size and bandwidth used.  Always has been this way.
  
 Quality vs convenience is as old as time itself. 
  
  
 No studio I have ever been in, including the cheapest bedroom jobby, records at less than 1.4Mbs/sec. Most pro studios record at well over 2Mbs/sec per track.
  
 Reducing that is done for economic reasons only.
  
  
 Keep changing the subject and applying all this BS to me and my motivations. I'm not rich, I don't own high end gear, I don't listen to loudness wars stuff, I don't think vinyl is the best format ever, I don't think the earth is flat evolution doesn't exist. Your name calling is petty.
  
 I simply think reducing things "for the consumer" because no one cares is ancient thinking, and those of you on this board that continue to push this out of context argument about what people can and can't hear are ultimately hurting all of us.
  
 It's digital - it's all bandwidth. Storage, real-time transmission, and ultimately cost is all determined by bandwidth used.


----------



## jcx

you should talk with Monty about his Ogg Vorbis development - what he could hear with training, how he qualified the listening abilities of the core testers - how many standard deviations beyond "average hearing", how many didn't make the cut
 likewise the Harmon "Golden Ears" free hearing test is only the preliminary screen for their internal listening testers
  
 I would say your "because no one cares" claim is a strawman


----------



## castleofargh

ffbookman said:


> Keep running people through those stupid tests and you'll continue with this horrible sounding "perceptual coding" world of lowest common denominator audio. Netflix streams 6Mbs to most of the residences in america but 1.4Mbs of audio should be reduced to 0.25Mbs?
> 
> That's why effective bitrate is the only true measurement of the format, assuming it's the same encoding.
> 
> ...


 

 it's you who's changing the subject. nobody here has hatred toward highres, and nobody is pushing for low resolution if it doesn't serve any benefit. the simple basic truth is that highres costs more than cds and we don't hear a difference. a 32giga µsd costs nothing, a 128giga is still pretty expensive(unless you buy a fake one on ebay). and I don't really notice the difference between the 16/44 flac and the max vbr mp3 , so why would I pay more for the same number of songs in higher format but no audible benefit? it's all about not being an idiot throwing money out the window for something I do not hear.  the day I can hear a difference I'll buy highres, a bigger hard disc and get bigger µSD cards. for my pictures I need more space and see the difference when I shoot in raw, so I have several big capacity compact flash. because it serves an actual purpose outside of me masturbating on my bigger stuff.
 just like the day I find a pizzeria that makes consistently better pizza, I'll go there and get those. but I won't go buy a pizza that cost 10more bucks again and again when it tastes like the cheap one I'm used to eat.
 it's not a conspiracy or trying to drag audio down, it's common sense. but you buy what you buy, I buy what I buy, and it's really not a problem at all.
  


> That's why effective bitrate is the only true measurement of the format, assuming it's the same encoding.


 
 and every time it's not? ^_^
 you're making all that argument to conclude with something useless. if it's the same encoding then anything can let you measure the format. bit rate, sample rate, how whatever number is bigger in one format... what point are you making here? if it's 16/44pcm vs 16/48pcm, I also know everything I need to know, how is bitrate a better measurement?
  
  


> and those of you on this board that continue to push this out of context argument about what people can and can't hear are ultimately hurting all of us.


 
 ????? What are you talking about? what we hear is out of context?
	

	
	
		
		

		
			




 that's the kind of total nonsense that's been turning you into a very annoying troll. you're doing this to yourself don't blame us for getting fed up by your nonsense and answering in kind.
 when you pretend to understand something you don't, and try to explain to the very guy who developed ABX in audio(Arny) how abx is wrong. when you claim you hear the difference in highres while all tests I know of, or have done myself, seem to show the opposite. when we offer you ways to demonstrate you can hear the difference, and you respond by systematically rejecting all controlled testing methods offered to you with the most ludicrous reasons. like saying that abx is testing memory. or bringing emotion into the mix when the subject is testing for audible differences. who's the one running away and changing the subject?
 when you mistake bitrate and sample rate, several times, even though we all explained it to you. if you didn't understand what we were saying or simply didn't trust us, any wiki page would have made it clear that your reasoning was using 2 different units and so was irremediably wrong. but you persisted with your total denial, never admitted to being wrong even once when the occasions really were not lacking. it's not even being wrong that's a problem, we all are wrong several times a day. it's how you pretend to educate us on subjects you don't understand, and how you never admit to being wrong yourself even with the nose pushed against your own crap. that's what's making you pretty insufferable in those threads.
  
 try posting something really factual and with a point that is really related to your argument, like maybe something you can actually back up with evidence for once. and offer those evidences without us having to ask. or simply try to present your opinions in ways that don't look so damn much like claims. do that and I swear I'll give you all the respect your posts deserve. just like the day you show me a way to test highres that's not a joke, and I do hear a difference, I'll start replacing all my audio files with higher resolution. but empty claims and fallacies, that you can stop, I would really appreciate.


----------



## Don Hills

castleofargh said:


> ...


 
  
 (Applause)


----------



## dprimary

slaphead said:


> And
> 
> http://www.snopes.com/politics/medical/tetrachromacy.asp
> 
> You can't test for tetrachromacy on a computer screen as a computer screen is not capable of generating the range of colours needed to test for the condition.


 
  
 I have not been able to find any information on the extended color space they are looking for. There is displays with greatly extended color spaces, none of them are cheap.
  
 Here is another article with more detail http://unreasonablydangerousonionrings.com/2015/03/03/you-are-not-a-tetrachromat-and-this-graphic-is-********/
  
 so far it seem researchers have found only one women with the proven ability.
  
 Which seems to be one more person in the world then is that can prove they can hear over 24k. In reality 20k is an outlier in ability, not an average that audiophiles seem to think it is. 
  
http://wiki.hydrogenaud.io/index.php?title=Absolute_Threshold_of_Hearing
  
 Find one person that prove they can hear the difference between 48k and 96k and there is going to be plenty of people that will want to understand what is going on.


----------



## FFBookman

castle of argh -- that's a lot of words to admit it's all about convenience and about what you think you can hear, and the value judgement you make on whether you want your music files as a full format or otherwise reduced somehow.  you did not mention quality or quality measurements, only convenience.
  
 scream kick and yell all you want, you know i'm right. *quality ≠ convenience.*
  
 if the music is tracked at 16/48 you should hear it and buy it at 16/48.
 if the music is tracked at 24/44 you should hear it and buy it at 24/44.
 if the music is tracked at 24/192 you should hear it and buy it at 24/192.
 if the music is tracked as 256k mp3.... oh wait, no one does that. not even horrible modern popdubrap is created at such a low resolution. 
  
 if you think you can save drive space because you can't tell the difference, then reduce them yourself.  take them as far down as you want to, no one cares! LAME is right.
  
 this debate is about a consumer format, a standard (16/44) whose time has come and gone, and you types that hang around here convincing yourself the rest of the world is stupid because no one can detect more than 1.0Mbs of stereo vibration coming at their body. 
  
 when experts that work in sound, that mix sound, that make sound, that have worked their whole lives in recording studios tell you that you are misguided and just not hearing correctly you run to ABX test results to make yourself feel better.   
  
 i think the whole lot of you have a real listening problem and can't find the obvious tells 
  
 If Arny helped develop the ABX listening test no wonder he's arrogant and nasty towards me -- i have said that that test is horribly misaligned with that use case, giving such useless results that normally calm people get into huge flame wars about those results.  Many of us won't touch one of those ABX hi-res challenges with a 10 foot pole because it walks you directly into the trap of the ABX test - nothing sounds better than anything else because it can't be proved repeatedly through ABX tests of the masses.
  
 And I've set about designing a better test format instead of trying to fight through those that parrot the useless results.
  
 Until then I'll just remind you all that 320k PCM < 1400k PCM < 3000k PCM < 5600k PCM.  That's available bandwidth at the various resolutions. What the artist and producer choose to do with that bandwidth is up to them.
  
 I am against artificial bandwidth restrictions on my music.


----------



## RRod

ffbookman said:


> That's available bandwidth at the various resolutions. What the artist and producer choose to do with that bandwidth is up to them.


 
  
 Just because they choose a format doesn't mean the music itself needs what the format can provide, especially given that it's destined for the filter that is the human ear/body. Once again you simply assume that every part of the chain can make use of potentially infinite resolution, which isn't even close to being true.


----------



## money4me247

lol, only on headfi wld failure to pass a controlled doubleblinded test mean that the test is flawed.

everywhere else (esp in scientific communities) it means that differences arent very noticeable or significant.


----------



## OddE

ffbookman said:


> castle of argh -- that's a lot of words to admit it's all about convenience and about what you think you can hear, and the value judgement you make on whether you want your music files as a full format or otherwise reduced somehow.  you did not mention quality or quality measurements, only convenience.


 
  
 -I beg to differ. As has been explained to you numerous times by numerous, exceedingly patient people the point is not convenience - and neither is it what Castleofargh can or cannot hear.
  
 The point is that study after study after study finds that human beings are not bats. We simply cannot hear the extra information (potentially) provided by upping the sample rate. We cannot. Or, at the very least, no-one has as of yet proven or even substantiated on the claim that some people have extraordinary hearing abilities. Not one. Neither will we be able to appreciate the increased dynamic range potentially provided by increasing the bit depth; what we have is already plenty more than our ears can handle without suffering permanent damage.
  
 While people may buy their music fifteen times over up to 128 bits/14MHz for all I care, in a forum dedicated to science, you had better be able to back your opinion with more than subjective opinion and hot air. You aren't. THAT is what irks people more than anything - well, at least, that is what irks me more than anything.
  
 That being said, I am eagerly awaiting your new and improved test methodology. Should you be able at some point to demonstrate that hi-res actually makes an audible difference, I'll be all ears - pun intended.
  
 I'm just not holding my breath waiting for it to happen.


----------



## KeithEmo

jcx said:


> you should talk with Monty about his Ogg Vorbis development - what he could hear with training, how he qualified the listening abilities of the core testers - how many standard deviations beyond "average hearing", how many didn't make the cut
> likewise the Harmon "Golden Ears" free hearing test is only the preliminary screen for their internal listening testers
> 
> I would say your "because no one cares" claim is a strawman


 
  
 I would add to that, in this day and age, that "the extra cost of streaming or storing high-resolution" is also a straw man. The last USB hard drive that I bought cost me $116 for 3 tB (that's 3,000 gB - which can hold somewhere between 500 and 1,000 complete albums at 24/96k). If you're paying somewhere between $10 and $25 for an album on CD or as a high-res download, the extra ten cents it costs you to store the high-res version is insignificant, and the same is true for the extra server space the music company has to use to store it, and the extra bandwidth you use to download it. (The major "cost" for the seller is the extra effort of creating and keeping track of multiple formats - and, if you look at most online stores, rather than try to minimize this, most do their best to make everybody happy by offering several different formats at each resolution.) The "benefit" of compressed formats like MP3, and even the benefit of Red Book files over high-res, is simply _convenience_; you can fit more of them onto a device with limited storage, which was probably limited to keep the size of the device down, and smaller files copy faster. The main incentive to develop formats like MP3 and AAC was to be able to claim that "you could fit your entire music collection on an iPod"; all of that other stuff about using up less bandwidth, and taking up less server space, is mostly rationalization. Being able to carry your entire album collection in your pocket, instead of having to decide which 100 albums you want to pack this morning, is a _convenience feature_. If anything, offering that album on AAC or MP3 costs a few cents extra, because someone had to make that copy from the master, and someone had to add it to the shopping cart.
  
 I think we can probably all agree that nobody _needs_ high-res audio, and, in fact, _most people_ are perfectly satisfied with AAC or MP3.


----------



## icebear

ffbookman said:


> castle of argh -- that's a lot of words to admit it's all about convenience and about *what you think you can hear*, ..
> 
> scream kick and yell all you want, *you know i'm right.*  quality ≠ convenience...
> 
> ...


 
  
 I'm still on the fence about someone who wants to tell others
 -what they can hear,
 -what they should listen to
 -what they should buy
 -that they have listening problems
 -what calm people are
 -and who is about to revolutionized the science of comparative consumer research
 And on top of it everybody knows he is right!
  
 Entertaining though but -
 Eh, honestly no ... I just decided on which side of the fence I am on and it's the other one, just getting to hit that personal peace button


----------



## KeithEmo

money4me247 said:


> lol, only on headfi wld failure to pass a controlled doubleblinded test mean that the test is flawed.
> 
> everywhere else (esp in scientific communities) it means that differences arent very noticeable or significant.


 
  
 I agree with your first statement; but the truth of that statement doesn't prove the second one.
  
 A well designed test proves (or fails to prove) _what it was designed to test_. If I was starting a music streaming service, then my goal would be to optimize the sound quality and the rest of the overall experience for _most_ of my customers, while minimizing my costs and maximizing my profits; this means that I would quite possibly make choices based on multiple criteria. For example, if it turns out that 95% of my customers think a 128k Ogg file sounds just fine, and 20% of them find the extra twenty seconds it would take to download the album at 24/96k to be annoying, then it would clearly make sense for me to use 128k Ogg and not 16/44k PCM to sell music to my customers. And, if I decide to start an "audiophile streaming service", my marketing department may do a survey and find out that 29% of my potential customers consider "24/96k" to be "an important consideration" when they're deciding whether to switch to my service instead of someone else's. They may also figure out that an ad campaign to convince people that "it's worth waiting for better quality" will serve to "neutralize" the annoyance most of my customers feel about the extra wait.  However, note that _NONE_ of those is a technical consideration, and they can only be "tested" or "proven" by asking people, or by seeing how my customers react.
  
 My point, as regards your statement, is that a test can be designed to show whether "most people notice a significant difference" or whether "anybody can ever detect a difference"... and those two tests serve very different purposes. What many of us here seem to lose track of is that both the design of a test, and the interpretation of the results, depend on our goal when the test was devised. If I were to test 1000 people, with 25 different test files each, and were to find out that one of those people could reliably recognize the high-resolution version of one of those files, how would we interpret that data? If I was trying to determine "if there was a significant difference", then my 1/25000 positive result wouldn't even statistically be "a blip on the line". However, if I was trying to determine "if any human can tell the difference", then my result is 100% positive, because I've found a human who can in fact tell the difference. Neither interpretation is _wrong_; they just serve _different purposes_.
  
 I'm guessing that you could conduct a lot of tests that would show that peanut butter is safe for humans - statistically. However, I once worked with a fellow who was so allergic that one teaspoonful of the stuff would send him into shock - and probably kill him. Therefore, you can't say that peanut butter is 100% safe for humans to eat. Likewise, until you test an awful lot of people, under a lot of different conditions, you aren't going to be able to definitively declare that "there is no audible difference between high-res and non-high-res files. It's a lost cause. (Unless your real goal is to determine if _most_ people can or cannot hear a difference - _most _of the time. (And, if that's really your goal, then an ABX test is a great way to go about it... as long as you state your goals, conclusions, and justifications clearly.)


----------



## Ruben123

320k PCM < 1400k PCM < 3000k PCM < 5600k PCM

Only on head fi this is true and doesn't get censored!


----------



## RRod

keithemo said:


> I'm guessing that you could conduct a lot of tests that would show that peanut butter is safe for humans - statistically. However, I once worked with a fellow who was so allergic that one teaspoonful of the stuff would send him into shock - and probably kill him. Therefore, you can't say that peanut butter is 100% safe for humans to eat. Likewise, until you test an awful lot of people, under a lot of different conditions, you aren't going to be able to definitively declare that "there is no audible difference between high-res and non-high-res files. It's a lost cause. (Unless your real goal is to determine if _most_ people can or cannot hear a difference - _most _of the time. (And, if that's really your goal, then an ABX test is a great way to go about it... as long as you state your goals, conclusions, and justifications clearly.)


 
  
 To be fair, deaths due to anaphylaxis really don't have false positives. The point of a statistical test is to have enough sample to meet the false positive/negative rates that will, in your own mind, convince you of a course of action. It is true that we'll probably never be able to tell if 0.00000001% of humans can hear hi-res, but we can be pretty darn sure that less than 1% can. And we can certainly tell *for ourselves on our own equipment* if we can hear difference, to within our tolerances for error. Just saying "oh we can never be sure so f' it" is basically poo-pooing the whole field of statistics, after all.
  
 Some of us would be just fine to have, say, 32/192 as a standard for audio. Whatever. But the point is to have a standard, and one set upon scientific principles and statistical studies. That was supposed to be what 16/44.1 got us, and what it seems to get us in blind testing. So hopefully one can understand the skepticism to the mantra of hi-res, which seems to be to push up numbers purely for the sake of pushing up numbers.


----------



## dprimary

keithemo said:


> I would add to that, in this day and age, that "the extra cost of streaming or storing high-resolution" is also a straw man. The last USB hard drive that I bought cost me $116 for 3 tB (that's 3,000 gB - which can hold somewhere between 500 and 1,000 complete albums at 24/96k). If you're paying somewhere between $10 and $25 for an album on CD or as a high-res download, the extra ten cents it costs you to store the high-res version is insignificant, and the same is true for the extra server space the music company has to use to store it, and the extra bandwidth you use to download it. (The major "cost" for the seller is the extra effort of creating and keeping track of multiple formats - and, if you look at most online stores, rather than try to minimize this, most do their best to make everybody happy by offering several different formats at each resolution.) The "benefit" of compressed formats like MP3, and even the benefit of Red Book files over high-res, is simply _convenience_; you can fit more of them onto a device with limited storage, which was probably limited to keep the size of the device down, and smaller files copy faster. The main incentive to develop formats like MP3 and AAC was to be able to claim that "you could fit your entire music collection on an iPod"; all of that other stuff about using up less bandwidth, and taking up less server space, is mostly rationalization. Being able to carry your entire album collection in your pocket, instead of having to decide which 100 albums you want to pack this morning, is a _convenience feature_. If anything, offering that album on AAC or MP3 costs a few cents extra, because someone had to make that copy from the master, and someone had to add it to the shopping cart.
> 
> I think we can probably all agree that nobody _needs_ high-res audio, and, in fact, _most people_ are perfectly satisfied with AAC or MP3.


 
  
 I would disagree, while it is a minor cost impact to stream one HD stream when you are looking at millions a streams from the streaming services. 3-4 times the bandwidth for HD is costly. 
  
 Tracks for sell though should all be lossless, there is no reason to still be selling lossy audio.
  
 I do agree that raw storage is cheap. Limited storage on laptops and portable devices is a larger limiting factor. Play back software is another. I have yet to find software that manages large external collections, is easy to use, sounds good, and doesn't use half the processing power just to play an audio file. Some of this software is so poorly written it uses more processing power for two tracks then multitrack software with 20-30 tracks during mix down. 
  
 I could never fit my entire music collection on an iPod, it was one of the first players I could fit 8-10 albums on, vs the 2 songs on the other portable players at the time that likely did not support uncompressed formats like the iPod did. What apple did was give you choice of sound quality vs quantity, and anyone could pick the balance that was best for them.


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## arnyk

keithemo said:


> I find the recent Grateful Dead Studio Remasters to be far superior sounding - to me - than any previous release (and there have been a lot). Now, whether that's because they're high-res, or simply because they're the latest re-master, or because a little more effort was spent to make them sound good because they're "an audiophile release", doesn't specifically influence my decision to buy them; I'm buying them because they sound better _for whatever reason_. I would also note that many of the recent high-res re-masters _DON'T_ sound any better to me than any other versions that are available - for whatever reason - but I'm basically taking that same gamble whenever I buy any re-master, whether it's high-res or not (which is why it's always a good idea to follow the reviews).


 
  
 Problem is Keith for all of your past and present brave talk about testing, you appear to cherry pick your testing methodology (IOW, sighted and and all of the other common audiophile testing errors thrown in for good measure) so that the only evidence that you gather agrees with your well-known agenda.


----------



## castleofargh

castleofargh said:


> when you mistake bitrate and sample rate, several times, even though we all explained it to you.


 
  


ffbookman said:


> Until then I'll just remind you all that 320k PCM < 1400k PCM < 3000k PCM < 5600k PCM.  That's available bandwidth at the various resolutions.


 
 and once again I don't know if you really lost it or if you're the troll of the year doing it on purpose to get me banned.
 obviously I can't say what you wrote here is wrong, because yes a bigger number is bigger and even if your numbers are a very poor choice, they have the same unit for once.
  
 but last post you went with streaming speed and bitrate as the true value that tells us something. the very next post you're now talking bandwidth. so which is it? I won't make up your mind for you.
 but maybe this will help, the first one was a silly claim because any lossless compression would prove you wrong by making a superior file smaller when sent. this time all I need to do is give you a 3bit/192khz track and you'll say it's better than a 16/44 because it has the bigger bandwidth.
 once more you try to convince us of something with examples you didn't think through. who on earth would take you seriously when you do that?
  
 as a personal request, if you could keep around all the annoying little letters you usually find after the K, you know bps, hz, B... they hold a sentimental value and I'm sad when they're forgotten.
  
  
 oh and obviously the all idea that we should listen in the format chosen by the guy doing the master is just another half backed justification.
 1/ when you're a pro you pick a working format that is the best you can use because you know you will degrade the signal while working on it. you know you might just end up with a guy who didn't play as in testing and only some extra dynamic range will salvage the track. that's IMO the kind of real life reasons that make a pro pick a high res format to work with. just like I wouldn't shoot my pictures in 800*600 jpg even if I knew the guy only wanted them for facebook.
  
 2/ now the very obvious flaw in your idea, all the old stuff were mastered on tapes. so I guess everybody's an idiot for buying the vinyls all those decades ago. it was made on tapes! and we're bigger idiots now for buying the digital versions!
 my grandma had some vinyls of classical performances at the time recorded on wax cylinders, what an idiot she was. "you should hear it and buy it "as wax!
 and the guy worked on 24bit, but after a few years they had to transfer it onto other tapes, and then digitalize it, now the original track has lost some bits of resolution in the process, I guess we should just throw it away as the guy doing the mastering intended it to have at least 6 more bits... there is no way your caricature of a reason can resist the ruthless attacks of real life.
  as a pro you get the best stuff available at the time. your argument makes no sense outside of saying not to upconvert a 16/44 to 24/96. but that wasn't the message you were trying to get through. right?
  
  
  


ffbookman said:


> when experts that work in sound, that mix sound, that make sound, that have worked their whole lives in recording studios tell you that you are misguided and just not hearing correctly you run to ABX test results to make yourself feel better.


 
 I see plenty of those on hydrogen and a few here (when not banned)saying that you're the one who's wrong. guess we don't look up to the same guys.


----------



## KeithEmo

rrod said:


> To be fair, deaths due to anaphylaxis really don't have false positives. The point of a statistical test is to have enough sample to meet the false positive/negative rates that will, in your own mind, convince you of a course of action. It is true that we'll probably never be able to tell if 0.00000001% of humans can hear hi-res, but we can be pretty darn sure that less than 1% can. And we can certainly tell *for ourselves on our own equipment* if we can hear difference, to within our tolerances for error. Just saying "oh we can never be sure so f' it" is basically poo-pooing the whole field of statistics, after all.
> 
> Some of us would be just fine to have, say, 32/192 as a standard for audio. Whatever. But the point is to have a standard, and one set upon scientific principles and statistical studies. That was supposed to be what 16/44.1 got us, and what it seems to get us in blind testing. So hopefully one can understand the skepticism to the mantra of hi-res, which seems to be to push up numbers purely for the sake of pushing up numbers.


 
  
 I agree entirely.
  
 My point with the peanut butter was to set a sort of criterion for reasonability. While I certainly don't recall being nervous the first time I ate peanut butter, and I might be surprised if someone were to tell me they had such a severe allergy to it, I still wouldn't automatically fail to believe someone who said they were that allergic "because nobody is". And, likewise, just because the majority of humans don't seem to be able to hear a difference between a high-res file and a 16/44k file, you're going to have to present me with some truly overwhelming statistical evidence before I accept that "nobody can". (Many of the arguments I'm hearing here sound a lot like: "Peanut butter doesn't bother me, and none of the fifty people we tested were allergic, so I'm pretty sure those allergies are a myth - probably started by companies trying desperately to trick people into buying higher priced non-pb spreads".)
  
 As for Red Book as a "standard" - I think that it is seriously flawed. For one thing, even modern technology, with DSP-based digital filters, and oversampling, has difficulty delivering good performance near the Nyquist frequency. Therefore, I have trouble believing that any CD hardware that was practical and available back when the standard was established could really deliver frequency response flat to 20 kHz, without significant phase errors and distortion. This also might explain why many early CDs seemed to not sound very good. And, even if I were to agree that "most humans probably can't hear anything above 20 kHz", I would still expect a viable standard to include a safety margin. (Can you imagine trying to establish a top speed for automobiles at 60 mph "because nobody is allowed to go over 55 anyway"?)
  
 The stories I've heard (oft repeated) seem to suggest that the standard was established based on the fact that it would allow a 70 minute concert to be recorded onto the amount of space that was currently available on a disc - and that, as long as those two requirements were met, nobody was interested in establishing how practical it was, or in adding a bit of safety margin. In other words, the so-called standard was set based on political and business rather than scientific reasoning. (We can get our concert on a disc, which we can manufacture today, and it squeaks by inside the bare minimum of what people should be able to hear.) This isn't exactly what I would call "a standard based on careful research"... and, to me, it simply doesn't seem worthy of defense. (If we really believe that the limits set by 16/44k are "_just barely_ good enough", then let's posit a standard at 24/96k, so we have some safety margin and some room for error.)


----------



## Thad-E-Ginathom

In my school life, I failed most of the exams I took. I conclude that exams must be really, really stupid.


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## KeithEmo

arnyk said:


> Problem is Keith for all of your past and present brave talk about testing, you appear to cherry pick your testing methodology (IOW, sighted and and all of the other common audiophile testing errors thrown in for good measure) so that the only evidence that you gather agrees with your well-known agenda.


 
  
 Actually you're sort of right - however, in point of fact, I'm not making a specific claim regarding high-res files at all. I do specifically claim that certain specific high-res files sound better than their supposedly equivalent standard-res versions, but I'm _not_ saying that I'm convinced that they sound different because they're high-res, and I'm certainly _not_ presenting any evidence to that effect. I'm perfectly open to the possibility that they may simply sound better because the company selling them expended a little extra effort on the mastering, or even that they deliberately made sure to degrade the quality of the 44k version so that an audible difference would exist. My claim is simply that nobody has done a test that is thorough enough to prove the fact one way or the other... and that none of the tests being quoted ad nauseum really is sufficient.
  
 In fact, personally, I consider the question to be somewhat moot. I am quite certain that specific high-res remasters I own sound obviously better than previous non-high-res versions, and a lot of recent re-masters are being done in various high-res formats. And the fact that many of the good quality music files I encounter lately happen to be high-res is sufficient reason for me to make sure that any DAC I purchase can play them. (Because it's inconvenient to have to convert a file that I acquire so it will play on my equipment; I'd much rather have equipment that will play any file I might encounter.) I'm just opposed to seeing the spread of "information" based on inadequate testing - because I consider it to be nearly as likely to cause errors in judgment as its opposite (snake oil). 
  
 (And, yes, I think it would be interesting to do some proper testing to settle the question once and for all. And, if I was still in college, and was looking for a subject for a term paper, I would probably pick that one. And, yes, I would probably be willing to take part in such testing. However, I would only do so if I was convinced that the test methodology was sufficiently well thought out to give meaningful results, and if there was a requirement that the results would be published - regardless of the outcome.)


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## RRod

keithemo said:


> And, likewise, just because the majority of humans don't seem to be able to hear a difference between a high-res file and a 16/44k file, you're going to have to present me with some truly overwhelming statistical evidence before I accept that "nobody can".


  
 It's fine to want overwhelming evidence, but what counts for that is up to you, and I don't it's unreasonable for someone to ask what your standards are. For instance, a one-off 9/10 that someone reports online hardly convinces me. Someone getting 23/25 in an actual controlled study gets my attention.
  


keithemo said:


> As for Red Book as a "standard" - I think that it is seriously flawed. For one thing, even modern technology, with DSP-based digital filters, and oversampling, has difficulty delivering good performance near the Nyquist frequency. Therefore, I have trouble believing that any CD hardware that was practical and available back when the standard was established could really deliver frequency response flat to 20 kHz, without significant phase errors and distortion. This also might explain why many early CDs seemed to not sound very good. And, even if I were to agree that "most humans probably can't hear anything above 20 kHz", I would still expect a viable standard to include a safety margin. (Can you imagine trying to establish a top speed for automobiles at 60 mph "because nobody is allowed to go over 55 anyway"?)


  
 I don't know enough about old ADCs/DACs to know how well they performed. Anyone care to chime in? As far as modern stuff, I've seen plenty of things that were purty-near flat from 20-20k (I would post a link but it's from a "he-who-shall-not-be-named"). That *some* early CDs sounded bad doesn't mean they all did. I just picked up a copy of the famous 1979 Telarc 1812 overture and the sonics are great. So are they from various other early ventures I have lying around.
  


keithemo said:


> The stories I've heard (oft repeated) seem to suggest that the standard was established based on the fact that it would allow a 70 minute concert to be recorded onto the amount of space that was currently available on a disc - and that, as long as those two requirements were met, nobody was interested in establishing how practical it was, or in adding a bit of safety margin. In other words, the so-called standard was set based on political and business rather than scientific reasoning.


  
 I've never found a reliable source on the LvB 9th anecdote. Besides, the 44.1 standard was around just a bit before the CD. The Wiki on 44100 gives some alternative theories, but that's not a reliable source either I guess.


----------



## FFBookman

castleofargh - very simple - make FLAC files, look at the total K size divided by the run-time of the song.  that's approx how many K per sec are used to render that song.
  
  if 5:00 takes 30mb it's approx 1000k per second.  this is redbook range.
  
 if 5:00 takes 85mb it's approx 2800k per second.  this is the 24/88 & 24/96 range
  
 if 5:00 takes 170mb it's approx 5600k per second.  this is the 24/192 range.
  
 there's a little bit of overhead in the FLAC file container so i'm using approximations.
  
  
 the point is that 256k mp3 is only pushing 256k at you, that's bandwidth, that's fixed at a number that 21st century laymen know.
  
 i'm giving you the HD numbers on their bitrate scale and you can't accept the truth.
  
 256 might be _good enough for you_, but it's not the same as the master. 1000 might be _good enough for arnyk_, but it's also probably not the same as the master.
  
 i just want to purchase the best copy available, and that means no needless degradations for convenience or bandwidth restrictions.


----------



## KeithEmo

(I'm replying to RRod's replies to me - I can't seem to get the quote mechanism to handle it correctly. RRod's replies will be in regular text, and my replies to them after - and flagged with >>'s
  
 Originally Posted by *RRod*
  
 It's fine to want overwhelming evidence, but what counts for that is up to you, and I don't it's unreasonable for someone to ask what your standards are. For instance, a one-off 9/10 that someone reports online hardly convinces me. Someone getting 23/25 in an actual controlled study gets my attention.
  
 >> I agree, and what I consider sufficient depends on the circumstances. If I'm buying a bottle of wine, I'm perfectly willing to go with a majority of reviews on a wine blog; after all, it'll be gone tomorrow anyway. However, when someone is telling me that "there is absolutely no difference between a high-res file and a CD", here's what that means to me..... If they're right, I get to save $5 by buying the CD instead of the high-res download. If they're wrong, I save $5, but end up enjoying that song just a little less than I might have. Even worse, I'll have to buy it all over again when I find out my mistake or, even worse than that, the good copy might no longer be available by then. Paying $5 more today to buy the high-res version that just might possibly be better seems like the safest bet there to me.
  
 I don't know enough about old ADCs/DACs to know how well they performed. Anyone care to chime in? As far as modern stuff, I've seen plenty of things that were purty-near flat from 20-20k (I would post a link but it's from a "he-who-shall-not-be-named"). That *some* early CDs sounded bad doesn't mean they all did. I just picked up a copy of the famous 1979 Telarc 1812 overture and the sonics are great. So are they from various other early ventures I have lying around.
  
 >> A lot of early CDS sounded rather bad - probably for a variety of reasons. However, the requirements of "properly" converting analog to digital include absolutely filtering out ALL content above the Nyquist frequency before doing the conversion. That means that, in order to make a CD without really bad distortion, you must use a filter that is flat to 20 kHz, yet is down at least 80 dB at 22 Khz. In the days of analog filters this was virtually impossible to achieve, and so some compromises were always involved, which prevented the reality from performing anywhere near the theoretical performance. (Oversampling avoids this requirement, but oversampling wasn't available as a technology when the red Book standard was written.)
   I've never found a reliable source on the LvB 9th anecdote. Besides, the 44.1 standard was around just a bit before the CD. The Wiki on 44100 gives some alternative theories, but that's not a reliable source either I guess.
  
 >> Yeah, I've heard a lot of variations on the story. However, I'm pretty sure that sample rates significantly above 48 kHz weren't readily available at the time. This means that when they tested "whether Red Book standard was audibly identical to the original" what they were really testing was whether it was audibly identical to an analog master tape. However, today we have "originals" that are far better than analog master tapes, and most people I know don't believe that analog master tape is "audibly identical to the original". (And the fact that a CD could be made to sound indistinguishable from 1970's vintage analog master tapes, when played through 1970's vintage amplifiers and speakers, really doesn't convince me that they're "audibly indistinguishable _FROM THE ORIGINAL_". In short, I'm not convinced that "the best equipment in 1976 was audibly perfect - and any and all improvements claimed since then are either snake oil or wishful thinking".


----------



## KeithEmo

ffbookman said:


> castleofargh - very simple - make FLAC files, look at the total K size divided by the run-time of the song.  that's approx how many K per sec are used to render that song.
> 
> if 5:00 takes 30mb it's approx 1000k per second.  this is redbook range.
> 
> ...


 
  
 Thank you for stating that very clearly....
  
 Today's technology makes it relatively simple for me to own a perfect digital copy of the original master version... and, since it is identical to the original, I know, beyond any doubt, that it will sound exactly the same. We all agree that reducing the sample rate, or performing any sort of conversion, is going to result in a copy that is less accurate to the original, but we don't seem to agree at what point the difference will become audible. Therefore, to me, it makes the most sense to simply stick with the perfect copy of the original. We can all agree that a perfect copy will absolutely definitely be audibly identical to the original. (I don't find the extra storage space or bandwidth to be an issue, and I don't mind paying an extra dollar or two for the added insurance of not having to worry about it, so I don't see much reason to try and find out how much accuracy I can discard before it becomes audible.)
  
 (In the old days, back when we had analog master tapes, and masters, and mothers, and pressing masters, we had an endless chain of loss - where each generation was a little less accurate than the previous one, and getting one generation closer to the original cost a lot of money. I know that I personally regretted the fact that I couldn't afford to buy a copy that sounded as good as the one the mixing engineer handed to the pressing house. Well, we've finally reached a point where, thanks to modern technology, I _CAN_ buy a copy that has the very same bits as the one he handed to the mastering house. I personally think that's a good thing, and I have little desire to figure out, in absolute detail, exactly how much of that extra quality I can sacrifice before it becomes audibly noticeable. Is it really so awful to pay an extra dollar or two for "quality that you might not need"?)


----------



## old tech

keithemo said:


>


 
 True, many old CDs were not that great but I don't think it has anything to do with 44.1 or the earlier ADC/DAC technology.  It that was the case it cannot explain all the great sounding CDs released around the same time period.  I have quite a few early CDs which sound great, and sometimes the defitinitve version of that album.  Pink Floyd, David Bowie, Joan Armatrading, Dire Straits and so on.  I suspect the real reason why there were many bad ones had more to do with the lack of mastering to the format (often using an EQd version of the vinyl mix-down) in a rush to the market than any percieved deficiencies in redbook or the technology supporting it.  I suspect there would be more poor CDs over the past 15 years than the first 15 years due to overcompression and loudness.


----------



## RRod

keithemo said:


>


 
  
 Yeah a good inline replying method eludes me :/
  
 Hi-res price is tangential to the point of having tests and statistical outcomes from those tests that we deem as good enough evidence for audibility. And audibility is apropos to any conversation on the benefits and pricing of hi-res, because if we can't actually hear a difference with blinders on, then such non-benefits shouldn't be touted as rational for charging us even $1 more.
  
 Re old equipment. Whatever attenuation old players actually managed, it's still true that many people simply don't need bone-flat response up to 20kHz, because our hearing drops off before then. I mean really, how bad could a filter that's aiming for 19k and allows aliasing up to 24k going to sound to those of us who are struggling to hear 18k at normal listening levels? But I was -1 for the first batch of CD releases, so people with personal experiences with the earliest equipment should chime in.
  
 It's true, I expect a modern 24/192 master to outperform tape in every way. This still doesn't change the fact that quantizing many tracks to 14bit does squat to the signal, and that people who actually hear 20kHz seem a rare breed.


----------



## Don Hills

rrod said:


> Yeah a good inline replying method eludes me :/
> ....


 
  
 I cut and paste the original quote as many times as I wish to divide it, then delete the irrelevant parts from each.
  


rrod said:


> ... (The rest of your post.)


 
  
 Like that...
 It can get cumbersome if you're quoting a long post. We need to be able to put the cursor in the quoted text and click a button labeled "split quote here."


----------



## RRod

don hills said:


> I cut and paste the original quote as many times as I wish to divide it, then delete the irrelevant parts from each.
> 
> 
> Like that...
> It can get cumbersome if you're quoting a long post. We need to be able to put the cursor in the quoted text and click a button labeled "split quote here."


 
  
 Thanks; good to know that at least works. I tried doing my own cuts by hacking the divs in the source, but it didn't like that.


----------



## castleofargh

keithemo said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
 @bookman you're still mixing mp3 bitrate with pcm bitrate, so I rest my case. you don't understand anything of what is said to you however how many times it's done. you're obviously a bot and an admin should remove your account to prevent more spamming of the exact same mistaken message over and over again.
  
 @keith and of course you buy whatever you want, and there is nothing wrong with getting the perfect reproduction.
 but I really don't see the rational behind paying for something I can't hear. if using only my human senses I fail to hear a difference and have no way to know unless I look at the numbers. how is it different from suggestion and placebo? at least when I buy a screen with a better gamut, it's something within the threshold of my senses that will be upgraded. I may find the upgrade meaningless and not worth the money, but it will be noticeable. with highres the changes are clearly outside of my hearing threshold. just like radio frequencies are outside of my hearing threshold and I wouldn't care to pay for an album that includes them just like I wouldn't pay more to have infra red on my screen. if it's outside of what my senses can get, it's outside of my subjective experience and audio is only that to me. I don't care about the life of the singer, and I don't care if there are plenty of sound at 24khz. that's not part of my experience of the music.
 so while I certainly understand your reasoning, I don't share it.
  
  
  
  
 my second problem with your post is the sound system. how many amps can resolve highres files at normal listening level into a real load? and of course the obvious, how many transducers can hope to get close to even cd quality? so what are we talking about here when we say perfect copy? if in the end our sound system is way below the CD resolution, and we fail to hear a difference, it's really just paying more for an idea.


----------



## dprimary

Bit rate is pretty meaningless for anything other then calculating network bandwidth and storage. 5000kbps could also be 8bit mono 655k sample rate. It could be 10 tracks of 14 bit /36k sample rate.


----------



## Thad-E-Ginathom

dprimary said:


> Bit rate is pretty meaningless for anything other then calculating network bandwidth and storage. 5000kbps could also be 8bit mono 655k sample rate. It could be 10 tracks of 14 bit /36k sample rate.


 
  
 Leave the book man to his books. Don't expect him to understand stuff like this.
  
 (yeah... I know, attacking the person is not a good argument either, but when, as has been pointed out, they continuously fail to understand, it's hard to know where to go next. Except perhaps _end of conversation._ except that there never really was one)


----------



## FFBookman

dprimary said:


> Bit rate is pretty meaningless for anything other then calculating network bandwidth and storage. 5000kbps could also be 8bit mono 655k sample rate. It could be 10 tracks of 14 bit /36k sample rate.


 

 It could be but it's not because I STATED exactly what I was talking about.
  
 Network bandwidth and storage is EVERYTHING in digital audio.  It's the main count of everything, assuming you are using the same encoding and file containers, and in this case I clearly said STEREO PCM at what resolution and if lossy or not.  I was very clear and guess what - out came the personal attacks again.
  
 I didn't say 8-bit mono 655k sample rate. No one cares about that for music production.
 I didn't say 10 tracks of 14bit/36k sample rate.  No one care about that for music production.
 Stay on topic.  Commercial music production, stereo, PCM. Thats all I'm talking about.
  
  
 KeithEmo is right, I'm right, the rest of you are choosing to degrade your audio because you _don't think you can hear_ the degradation.  That's your choice but you should degrade your own files, not continue to claim that I'm imagining things, and not add to the confusion about what people can and should hear.
  
  
 I have to ask, is there anything else your body can clearly and repeatedly do that you don't have math or testing to confirm?
  
 You "16/44 is the highest" people are the most ridiculous lot of brainiacs I've come across. Continue to deny the sensations of the physical world, it could take another few hundred years to devise the math behind what we all do all the time.  I've done many things in 1 hour this morning that you have no mathematical model for.
  
 Sadly you won't be alive to buy the upgraded robot version of yourself.  Until then enjoy listening to your degraded music files. There's other pleasure in the world, hopefully you don't degrade every piece of art given to you.


----------



## FFBookman

thad-e-ginathom said:


> Leave the book man to his books. Don't expect him to understand stuff like this.
> 
> (yeah... I know, attacking the person is not a good argument either, but when, as has been pointed out, they continuously fail to understand, it's hard to know where to go next. Except perhaps _end of conversation._ except that there never really was one)


 

 I understand this stuff on a level you only seem to dream about.  I don't need a formula or a white paper to prove to me what is true and obvious and easily recreated whenever I choose. 
  
 Or you think I'm dreaming. Either way don't accuse me of ignorance, I make and mix music and I can prove the differences between resolutions easy. I've been working in production with digital audio since 1994, with analog audio since 1989.
  
 This is relatively easy stuff, stuff I figured out 20+ year ago. Yet here I am in 2015 arguing with you folks on Head-Fi.  What does the "Fi" stand for in your world?  
  
 Keith Emo seems to be making a stronger case than me, that's cool. I'm fed up because this anti-resolution movement is killing the art of recorded music.
  
 This thinking brought about MP3's and their "perceptual coding" and it brought about this plug-in saturated world of fake audio being blasted ever louder and louder though smaller and smaller bandwidth restrictions.


----------



## arnyk

ffbookman said:


> I understand this stuff on a level you only seem to dream about.  I don't need a formula or a white paper to prove to me what is true and obvious and easily recreated whenever I choose.
> 
> Or you think I'm dreaming. Either way don't accuse me of ignorance, I make and mix music and I can prove the differences between resolutions easy. I've been working in production with digital audio since 1994, with analog audio since 1989.
> 
> ...


 
  
 Two words: Sighted evaluations.
  
 Until you understand why they are grossly invalid and act on it...


----------



## FFBookman

You wanna do a test?  Set up 3-4 mics on a drum set. Botnick style or however you choose.  Or hell, 1 mic hanging over the snare periscope-style will do the trick.
  
 Record at the highest resolution your AD converters can do.  Mine do 24/96 reliably (24/192 is still wonky on my rig).
  
 Just hit the snare hard 4 times. Then do some rolls on the snare. Then 30 seconds of hi hat work. Open and close the high hat as you roll on it. Now for your big finishing move, smash each cymbal and let it decay all the way out. Get to silence. Then do a few more  with some stick work on the top of the cymbal.  
  
 Overall, record about 4 minutes of smashing drums.  No compression, no effects, just raw microphone data with basic stereo pan if using more than 1 mic. If you can play beats with fills go for it, but it's not needed.
  
 OK back to the computer -- mix it to 2-track stereo and output it at native resolution. Then dither and downsample it down to 16/44. Make different copies for as many dithering algorithms as you have on your rig (I have 3 installed).
  
 Now go back to your full resolution version and study it.  Listen to how long the decays are. Listen to the high hat detail (there's tons of it). Listen to the attack and decay of the snare. Pay attention to how the snare hits you, how much snap it has. Listen for stick noises, breathes, room hum, ambient sound off mic, etc. Determine the virtual placement of the drums in the mix, make notes if you want. How wide is the soundstage? Where is that drum/cymbal located at in the mix? 
  
 Once you've made yourself familiar with the drum sound at full resolution, picking out the degradations in the 16/44 files should be easy.
  
 You will hear everything get smaller. All decays will get shorter and harsher, with a cut off at the end that wasn't there.  Room noise and breathes will also be less than before. The soundstage will get slightly narrower and the instrument placement will be slightly fuzzy and appear less focused on one particular spot. The detail of the high-hat will start to be compromised.
  
 If you continue this test and take a 16/44 file into LAME and make MP3s out of it, you will hear your beautiful acoustic drums turn into samples of cardboard. You will hear your full splashy cymbals turn into 808-sounding digital recreations of a cymbal.
  
 This happens for almost every instrument as you degrade the signal. Given that multitrack recording is the art of making between 4 and 200 tracks all work together in a layered and musical way this degradation of every instrument really takes it's toll on the final mix.
  
 You are hearing the effects of ultrasonics on music, and what happens when you remove them.  This is why musicians that spend big money learning their craft and recording it are all for high resolution digital. They want what they worked so hard making heard by the world. Why else bother making it?


----------



## dazzerfong

ffbookman said:


> You wanna do a test?  Set up 3-4 mics on a drum set. Botnick style or however you choose.  Or hell, 1 mic hanging over the snare periscope-style will do the trick.
> 
> Record at the highest resolution your AD converters can do.  Mine do 24/96 reliably (24/192 is still wonky on my rig).
> 
> ...


 
 One hell of a convenient way to shut out the test to most people. Why don't you record it for us, if you don't mind?


----------



## RRod

And then you do it blind…


----------



## FFBookman

rrod said:


> And then you do it blind…


 

 and it's even easier to hear


----------



## RRod

ffbookman said:


> and it's even easier to hear


 
  
 Yeah I'm sure. Here's another even easier test to try. Take any track you think actually makes good use of the capabilities of 24/192, resample it to Redbook, and then do whatever comparison test you want that doesn't involve knowing which file is which. Bet you won't try it.


----------



## FFBookman

dazzerfong said:


> One hell of a convenient way to shut out the test to most people. Why don't you record it for us, if you don't mind?


 

 If you can't and have never taken part in something like this you aren't qualified to tell people what's the highest resolution needed.
  
 I've said from day one -- if you don't believe resolution matters, go into a professional recording studio and tell them that.
  
 Stop telling the creators to create the lowest common denominator.  There's room for quality.  16/44 is a nice baseline - no one should be fed less quality than that. But it's just the beginning of what recorded sound is capable of.


----------



## castleofargh

Quote:


ffbookman said:


> rrod said:
> 
> 
> > And then you do it blind…
> ...


 
  
 ahahahahahahahahahahahah. this from the very guy failling to abx a 24/96 vs a 16/44. this is just pure gold in a post.


----------



## RRod

castleofargh said:


> Quote:
> 
> ahahahahahahahahahahahah. this from the very guy failling to abx a 24/96 vs a 16/44. this is just pure gold in a post.


 
  
 Yeah I had to put my eyes back in their sockets.


----------



## FFBookman

rrod said:


> Yeah I'm sure. Here's another even easier test to try. Take any track you think actually makes good use of the capabilities of 24/192, resample it to Redbook, and then do whatever comparison test you want that doesn't involve knowing which file is which. Bet you won't try it.


 

 I can do that all the time, very easily.  Just close my eyes and hit this:
  

  
 That's even tougher than a regular ABX because the ponoplayer's signal chain plays all those formats about as good as it can. Some people complain that the Revealer feature is counterproductive because the PP plays MP3's so well. 
  
 Fast switching between songs is always a problem. I prefer to let it roll and listen for overall room sound and decays.


----------



## FFBookman

I mix with my eyes close.
  
 You guys listen with your eyes.
  
 Laugh all you want. I'm going around the world, you are falling off the edge.


----------



## RRod

ffbookman said:


> I can do that all the time, very easily.  Just close my eyes and hit this:
> 
> 
> That's even tougher than a regular ABX because the ponoplayer's signal chain plays all those formats about as good as it can. Some people complain that the Revealer feature is counterproductive because the PP plays MP3's so well.
> ...


 
  
 Yeah I'm sure Pono did its best to make a neutral app for blind testing. Never mind that Amazon mp3 wouldn't even be using the same master. Never mind that you can see which one you are selecting (at least from the interface you are showing). What's this supposed to show us exactly?


----------



## OddE

dazzerfong said:


> One hell of a convenient way to shut out the test to most people. Why don't you record it for us, if you don't mind?


 
  
 -Oh, yes, please. I'd be more than happy to sox the heck out of the original, convert, downsample, drop bits &c to my heart's content, then turn everything back to 24/192 and let people try to listen for differences.
  
 Downside to doing this over the Internet, of course, is that anybody can have a look at the files in, say, a suitably plugined Foobar and tell which sample is which.


----------



## KeithEmo

castleofargh said:


> @bookman you're still mixing mp3 bitrate with pcm bitrate, so I rest my case. you don't understand anything of what is said to you however how many times it's done. you're obviously a bot and an admin should remove your account to prevent more spamming of the exact same mistaken message over and over again.
> 
> @keith and of course you buy whatever you want, and there is nothing wrong with getting the perfect reproduction.
> but I really don't see the rational behind paying for something I can't hear. if using only my human senses I fail to hear a difference and have no way to know unless I look at the numbers. how is it different from suggestion and placebo? at least when I buy a screen with a better gamut, it's something within the threshold of my senses that will be upgraded. I may find the upgrade meaningless and not worth the money, but it will be noticeable. with highres the changes are clearly outside of my hearing threshold. just like radio frequencies are outside of my hearing threshold and I wouldn't care to pay for an album that includes them just like I wouldn't pay more to have infra red on my screen. if it's outside of what my senses can get, it's outside of my subjective experience and audio is only that to me. I don't care about the life of the singer, and I don't care if there are plenty of sound at 24khz. that's not part of my experience of the music.
> ...


 
  
 Actually, I can think of several "rationales" for buying something that doesn't sound "provably" different to me - some practical, some potentially practical, and some just a matter of "self satisfaction"....
  
 1) Even assuming that I can't hear any difference today, I might actually be able to hear one later. While I don't subscribe to the whole "golden ear idea" in general, it is true that our abilities do change over time, and we do sometimes "learn how to listen better". I may attend a live concert and suddenly discover that the speakers I thought sounded "just like live" really don't, or I may simply start paying more attention to certain aspects of the sound that I hadn't noticed before, or I may decide I like binaural recordings of chamber music with lots of natural ambiance as well as multi-tracked pop-rock. (I have one recording that has an odd little sound at one point, which I had always assumed was simply the recording microphone clipping; it turns out that it's a vibration coming from part of the drum; once I realized that, and heard a real drum make that sound, I started noticing whether it sounded natural or not on my speakers and headphones.)
  
 2) Even assuming my abilities don't improve, I might buy better recordings someday, or change other equipment which renders the difference audible. (Certain speakers or amplifiers tend to make certain errors more audible - whether because they're more accurate, or simply because they emphasize them. For example, bright speakers make recordings with poorly recorded high-end sound more obvious. And many people here will surely attest to the fact that they notice things when listening on headphones that they don't when using speakers.) 
  
 3) I may have a current or future technical justification. When I take pictures with my camera, a high-quality JPG (lossy) version of most pictures will often look just as good as a true lossless RAW frame - out of the camera. However, when I try to adjust it later in Photoshop, the artifacts on the compressed picture will often become obvious due to the processing. Most of us don't "re-master" our music, but many of us do use "processing" like surround-sound decoders, or spatial processors, or even noise removers, some of which may be affected by differences we can't hear, and they may be affected in ways that we can hear. (To use an example from the days of vinyl and SQ surround sound. The decoders used to play SQ encoded material use phase relationships to decide which parts of the audio belongs in which speaker. In many cases, if you have a record that has been mechanically damaged, the distortion from the damage is "pushed into the rear channels" at a boosted level by the decoder, which can make an album that sounds only slightly damaged in stereo virtually unlistenable through the decoder.) In the current context, next year's surround decoder may use high frequency phase cues present in the music to locate various instruments, and so may work with 96k recordings and not 44k ones. (While I agree that we can never know for sure where that would end, making recordings with at least a little bit of safety margin seems prudent.) 
  
 4) Not all "limits" are as black-and-white as many people think. To take your example; most people would agree that buying a monitor that would display a gamut up to 850 nm would be silly (that's the "color" used by many IR remote controls). However, to say that "you can't ever see it" is wrong. In fact, light of that color is visible to most people, but only if it's bright enough. (If you look at the dot from "an invisible 850 nm LASER", you will indeed see it as a faint pink visible dot, because it is in fact _slightly_ visible to most people.) In that case, I would agree that being able to see that color on your monitor probably serves no useful purpose, but I'm not so sure that everything that isn't directly audible is "useless".
  
 5) Sometimes extra "safety margin" serves other benefits. For example, even though most of us probably don't hear much above 20 kHz, someone with an engineering background would still avoid an amplifier that was only able to amplify "20 Hz to 20 kHz" or that had a distortion plot that rose sharply right above 20 kHz - because good performance up to 50 kHz or so almost always signifies excellent performance inside the audio band, and performance that fails to extend past 20 kHz tends to suggest that problems exist inside the "audio band", even though they may not be visible on standard measurements. And, for another possibility, there was a recent AES paper that suggested, although I wouldn't say that it rose to the level of proof, that some people notice shifts in the sound stage on recordings that are band-limited to 20 kHz. (Their test showed that, even though their test subjects reported that the recordings "sounded the same", the location of instruments in the sound field was sometimes shifted when a recording was band-limited to 20 kHz. They suggested that, even though a 44k sample rate can record all audible sound, it may not be able to record the phase cues that our brains and ears use to determine location accurately enough.
  
 6) As for your final point..... I simply disagree with the premise. Many of us get better equipment "as we progress in the hobby", and the technology itself improves. It would be foolish to buy something that is audibly inferior simply because my current system isn't able to let me hear its flaws, when the system I own next year may make them obvious. (I hear lots of details on my electrostatic headphones that I never noticed before on my speakers.) It would even be foolish to buy something that doesn't sound any better on _ANY_ system available today, if that situation is likely to change. Twenty years ago you couldn't buy a TV that would let you see how much better the picture is on a Blu-Ray disc than on a DVD; yet the difference was really there, and now can be seen on most TVs.
  
 In that last situation, if I bought my entire collection of movies as DVDs when they were "the current technology", I might end up buying them all over again as Blu-Ray discs. However, if I'd had the opportunity to buy them as a direct copy of the digital theatrical master, which is better than both DVD and Blu-Ray discs, then I would still have "the best copy available". (That option isn't available for video, but it is equivalent to buying a 24/96k or 24/192k copy of the audio master.)
  
 I know lots of people who bought a significant amount of music on 128k AAC files then, after upgrading their music system, and realizing that the difference was audible to them after all, ended up having to buy it all over again (or pay the upgrade fee). Buying a version that's "a lot better than our ears" rather than one that's "just barely better than we believe is audible" seems like a good form of insurance against that (and, in most other situations, most people I know would consider a "safety margin" to be a good thing.)
  
 I'm going to offer two very different examples - in other contexts - to support my point.
  
 1) There are several devices designed to automatically remove ticks and pops during vinyl playback. Many of them use the ultrasonic content of the audio signal to decide the difference between a "legitimate tick" and a record scratch; because scratches have significant ultrasonic content while music recorded on records does not. You could use one of these devices (or equivalent software) to remove ticks and pops from the archive recordings you had made of your favorite albums if you'd made those recordings at 96k. However, it wouldn't work if you'd recorded them at 44k because the ultrasonic content the device relies upon would be missing.
  
 2) Since you mentioned monitors and visible gamut.... There is a system quite similar to the click and pop remover that is commonly used to automatically detect and repair scratches on slides. It works by recognizing that certain colors of light that are invisible to the human eye are blocked by the surface coating on slide film. (Basically, by scanning the slide at these frequencies, you can produce an "image" of scratches in that surface, and use that information to control the correction process.) Of course, you could only use this system on a scanned and stored image if it included a gamut much wider than the range of the human eye. (So, if someone was archiving important photographs before this system existed, it would benefit them today if they'd scanned them using IR and UV light as well as visible light, even though, at the time, there was no apparent reason to do so.)
  
 To me, all of this simply suggests that "getting the best quality copy you can afford" rather than "one that's just good enough" really does make sense. Now, in that second example, it might not pay to buy a special scanner to record a whole lot of information you might never use. However, in the case of music, where there is already a "starting point" that is limited by the ability of the microphones and mixing equipment we're using, it seems to make sense to hold onto a little extra information - since we already have it anyway - just in case we may want it later.


----------



## Ruben123

ffbookman said:


> You wanna do a test?  Set up 3-4 mics on a drum set. Botnick style or however you choose.  Or hell, 1 mic hanging over the snare periscope-style will do the trick.
> 
> Record at the highest resolution your AD converters can do.  Mine do 24/96 reliably (24/192 is still wonky on my rig).
> 
> ...




With what headphones ? And could you do the same blindfolded?


----------



## KeithEmo

rrod said:


> Yeah a good inline replying method eludes me :/
> 
> Hi-res price is tangential to the point of having tests and statistical outcomes from those tests that we deem as good enough evidence for audibility. And audibility is apropos to any conversation on the benefits and pricing of hi-res, because if we can't actually hear a difference with blinders on, then such non-benefits shouldn't be touted as rational for charging us even $1 more.
> 
> ...


 
  
 I sort of agree that touting the differences as "audible" if they aren't is somewhat misleading (even if they are otherwise "desirable" for other reasons). However, I think it falls well within the bounds of "ordinary advertising claims".....
  
 As for the filters, the problem is that creating an analog filter with those characteristics is difficult, and manufacturing one that performs to specification is expensive. Even a "perfectly" designed filter is going to be a compromise between a sharp roll off and a smooth frequency and phase response (any practical filter that is sharp enough will almost certainly have unacceptable frequency response and phase ripple inside the audio band, and any filter with good frequency and phase response will not be sharp enough). And, once you have a decent filter designed, you're going to be stuck with another tradeoff between performance and cost - because the components accurate enough to manufacture an analog filter that precisely are extremely expensive. It's sort of like being tasked with designing a racing car that can go 500 MPH _AND_ get 100 MPG - and then sell it for $35k. I'll bet that the older recordings that sound really good were done with a filter that accepted a slight loss of high frequencies in return for a smoother overall frequency and phase response.
  
 Modern oversampling technology totally avoids those tradeoffs (at the cost of a few other minor ones), but oversampling was only developed later, as a way to avoid those compromises, _after_ the standard was set.


----------



## KeithEmo

ruben123 said:


> With what headphones ? And could you do the same blindfolded?


 
  
 As for which headphones..... If you can hear the difference with _ANY_ headphones, then that proves that the difference is real, at which point you're just testing headphones. Personally, whether you happen to like the way they sound or not, I've found electrostatic phones to usually do the best job of making differences audible and exposing tiny details.
  
 Sadly, most of the commercially available test discs are in fact Red Book CDs, although a few are starting to appear at higher resolutions.
 (A lot of places who sell high-res downloads also offer various sample discs and even free download samples.)


----------



## RRod

keithemo said:


> I sort of agree that touting the differences as "audible" if they aren't is somewhat misleading (even if they are otherwise "desirable" for other reasons). However, I think it falls well within the bounds of "ordinary advertising claims".....
> 
> I'll bet that the older recordings that sound really good were done with a filter that accepted a slight loss of high frequencies in return for a smoother overall frequency and phase response.
> 
> Modern oversampling technology totally avoids those tradeoffs (at the cost of a few other minor ones), but oversampling was only developed later, as a way to avoid those compromises, _after_ the standard was set.


 
  
 Agreed that it's hard to point to anything and say it's at the height of advertising shystering. Still, oversold is oversold.
  
 As far as older recordings: I've read that pre-emphasis was used to help with older DACs, but the language isn't so clear on how exactly it helped either the digital or analog components. As far as testing out older tracks, one could do a comparison of relative energy in the last 1/2 or 1/3 octave for recordings of the same material, but eh. It would be great if someone had specs for older ADCs or DACs lying around. 
  
 Certainly oversampling / Σ-Δ helped bring cheap flatness to consumer level products, but that doesn't mean that pro gear back in the earliest days didn't perform just as well, and I would think that companies like Telarc were using the best they could get their hands on.


----------



## KeithEmo

rrod said:


> Agreed that it's hard to point to anything and say it's at the height of advertising shystering. Still, oversold is oversold.
> 
> As far as older recordings: I've read that pre-emphasis was used to help with older DACs, but the language isn't so clear on how exactly it helped either the digital or analog components. As far as testing out older tracks, one could do a comparison of relative energy in the last 1/2 or 1/3 octave for recordings of the same material, but eh. It would be great if someone had specs for older ADCs or DACs lying around.
> 
> Certainly oversampling / Σ-Δ helped bring cheap flatness to consumer level products, but that doesn't mean that pro gear back in the earliest days didn't perform just as well, and I would think that companies like Telarc were using the best they could get their hands on.


 
  
 You're sort of mixing apples and oranges now. Pre-emphasis is a process of boosting certain frequencies when you record them, then reducing them by an equal amount at playback, and serves the purpose of "noise shaping". If I know my tape is going to introduce high frequency hiss, I boost the highs when I make the recording. Then, when I play it back, I reduce the highs by an equivalent amount. This lowers the high frequencies in the music to the level they were at originally, but reduces the high frequency noise added by the tape. As long as the process is handled properly, there should be no net alteration in the sound. Pre-emphasis is usually used on tapes, and both Dolby noise reduction and RIAA equalization are really just very specialized types of it). There were some issues with early CDs because not all devices handle the pre-emphasis and de-emphasis properly (so some CDs end up sounding very shrill if they were recorded with pre-emphasis but end up being played without the matching de-emphasis - usually because the player doesn't support it or the flags inside the file aren't set correctly).
  
 A simple high frequency roll off is actually easy to compensate for - by simply boosting the highs by the appropriate amount. (Which means that an ADC with a slightly rolled off high end is easy to correct for - by boosting the highs slightly in the mix. However, an irregular high end would be much more problematic.)
  
 The usual issue with early filters was what we would call "an irregular response". I can have two filters that both "spec out" at "-1 dB at 20 kHz, but one may be flat to 15 kHz, then slope smoothly down to -1 dB at 20 kHz, while the other is flat to 18 kHz, -1 dB at 18.5 kHz, 0 dB at 19 kHz, -1 dB at 19.5 kHz, 0 dB at 20 kHz, and -3 dB at 21 kHz. Simple filters tend to have smooth response curves (like the first example), while the complex ones you need to get flat in-band response and a sharp drop off afterwards tend to have a lot of ripple (which is what that second example has).
  
 Some early equipment did in fact perform very well - but not all of it. (And, yes, some early digital equipment had a tendency to be designed for "good specs" rather than sound quality.) And, in the beginning, that equipment was largely unfamiliar to most of the pros who were using it. So you ended up with equipment that was far from perfect, and which required someone to be knowledgeable of the various tradeoffs involved if you wanted to get the best performance from it, but which was often being operated by people who weren't intimately familiar with those quirks and tradeoffs. Add to that the fact that there was also a tendency to specifically adjust the sound to fit what "customers expected the digital version to sound like" so people would have incentive to replace their albums with CD versions, and you had a sort of formula for inconsistency.


----------



## castleofargh

@Keith I don't hold my breath for a headphone/speaker that out resolve CD. I most certainly hope I'm wrong and it will happen in my lifetime, but I really don't hold my breath.
  
 nothing wrong about thinking about the future, and even though bookman wants to make me the champion of promoting mp3, I have never bought music in mp3 format and would always keep 2 archives of my music in flac. it's most likely unnecessary, but it's the kind of habit I got from my past mistakes in photography ^_^.
  
 I agree with several things you said on principle as motives to buy higher resolution, it's simply that I don't see my own hearing with whatever listening experience I will gain, changes the fact that I won't be able to pass an abx. not that I'm pessimistic, just that I've seen enough people fail around the world not to think I'm some uberman who will do it. and as I said I don't think my sound system will in my lifetime outperform CD resolution. so just like I don't see the point of getting better than my odac given how much worse is the rest of my gears, I don't believe that highres can benefit me so I don't pay for it.
 but of course it's my own judgment call and I'm perfectly fine with people thinking otherwise. if anything the idea of having the best available does appeal to a great many people. I'm fine with that.
 I just wish the publishers would stop their game of restricting some very good masters to highres format.
  
  
  
  
  
  
 about booky's pono format test, here is the post that got deleted and got me locked out of the pono topic.. because "hearing test should be kept to sound science" and we were warned about it. fear all the evil of my post!!!!!!!!!  I was answering to a guy asking about the very pono listening test and asking if people heard a difference. so obviously it was easy for me to explain how it wasn't a proper test without explaining what is... 
	

	
	
		
		

		
			




 I'll end up banned soon because you only get so many strikes before being out. but the idea of being vetoed for posting something both on topic and factual, while half of the posts before me were pure fallacies, false claims and anti science hatred, that really tells a lot about what currawong thinks and cares about. and good for us, that's who we have to do most of the moderation in the sound science section \o/ yay!!!!!! go science!!!!
 anyway here it is, don't let your kids read it, it's hardcore stuff, or really not. maybe ask your favorite moderator:


> except that the "revealer" fails to provide unbiased observation. the prime point of making a correct comparison is not to know what kind of file you're listening to. just like when tasting 2 cakes, you don't tell people that the first one cost 15$ and the second one cost 55$, because from there, even if the 2 cakes are in fact the same, more people will pick the expensive one as being better. demonstration:
> 
> 
> 
> ...


----------



## RRod

keithemo said:


> You're sort of mixing apples and oranges now. Pre-emphasis is a process of boosting certain frequencies when you record them, then reducing them by an equal amount at playback, and serves the purpose of "noise shaping". If I know my tape is going to introduce high frequency hiss, I boost the highs when I make the recording. Then, when I play it back, I reduce the highs by an equivalent amount. This lowers the high frequencies in the music to the level they were at originally, but reduces the high frequency noise added by the tape. As long as the process is handled properly, there should be no net alteration in the sound. Pre-emphasis is usually used on tapes, and both Dolby noise reduction and RIAA equalization are really just very specialized types of it). There were some issues with early CDs because not all devices handle the pre-emphasis and de-emphasis properly (so some CDs end up sounding very shrill if they were recorded with pre-emphasis but end up being played without the matching de-emphasis - usually because the player doesn't support it or the flags inside the file aren't set correctly).
> 
> A simple high frequency roll off is actually easy to compensate for - by simply boosting the highs by the appropriate amount. (Which means that an ADC with a slightly rolled off high end is easy to correct for - by boosting the highs slightly in the mix. However, an irregular high end would be much more problematic.)
> 
> ...


 
  
 The wiki page just didn't make clear the connection to filtering; from what you say it sounds like it was only used for abating noise.
  
 A bit of variation between 0 and -1dB up above 18kHz may be "bad" in the universal sense, but I doubt that is what people are hearing as sounding bad. If I play a mix of a 12kHz and 18kHz tones, I can't tell you when the 18k pops in. That's not justification for having non-flat response, of course, but it does illustrate a point about audibility versus measurement.


----------



## Thad-E-Ginathom

Mr Bookman, if you really would like to make your test files as prescribed ---and perhaps Arny will tell you the technical stuff that you have to comply with--- there are plenty of folk who will take the test.
  
 I'll even try myself, but with audiometry showing failing hearing from 8kHz, the normally-audible spectrum is more more than I can cope with.
  
 Really. People would accept the ABX challenge willingly.  But... you go first.


----------



## RRod

thad-e-ginathom said:


> Mr Bookman, if you really would like to make your test files as prescribed ---and perhaps Arny will tell you the technical stuff that you have to comply with--- there are plenty of folk who will take the test.
> 
> I'll even try myself, but with audiometry showing failing hearing from 8kHz, the normally-audible spectrum is more more than I can cope with.
> 
> Really. People would accept the ABX challenge willingly.  But... you go first.


 
  
 He won't accept ABX as the means of testing it.


----------



## Thad-E-Ginathom

Well, that's that then.
  
 My "end of conversation" was only a few posts premature.


----------



## Gr8Desire

ffbookman said:


> ...
> You wanna do a test?  Set up 3-4 mics on a drum set. Botnick style or however you choose.  Or hell, 1 mic hanging over the snare periscope-style will do the trick.
> 
> Record at the highest resolution your AD converters can do.  Mine do 24/96 reliably (24/192 is still wonky on my rig).
> ...


 

 Umm... Err...  People have been recording drums since the beginning of the digital era. 

 16 bit PCM** easily covers the dynamic range of every drum set I have ever encountered - including the insane range from the PCM-out port on my Roland V-Drums.  I have done thousands of such recordings.

 The problem is NOT the number of bits. * It is the playback devices.* If I use the full compliment of 16 PCM bits, the music is either too quiet or it distorts too much on loud passages. There is no solution in the digital realm (i.e., more bits or samples will not help). The analog devices that reproduce the sound cannot adequately cover the dynamic range of 16 bit PCM. Oddly enough, 24 bits helps marginally. 24 bits has better resolution by offering more samples within the 16 bit PCM range. Range beyond 16 bits is not really useful - again because analog devices can't reproduce all of them.

*BECAUSE OF ANALOG PLAYBACK DEVICE LIMITATIONS* VIRTUALLY EVERY DRUM RECORDING IS COMPRESSED to FIT INTO ABOUT 12 PCM BITS.  

 Someday we might create better playback devices (BTW: I'm thinking mainly speakers and HPs).  Until then, recordings and remastering efforts will continue to use little of the dynamic range available.
   
 ** I refer to 16 BIT PCM (LPCM really) because it defines linear spacing between adjacent quantized values (same distance between quiet and loud values) and therefore uses its 16 bits VERY INEFFICIENTLY. Other encoding standards use variable spacing between quantized values focusing more bits where there is actually sound.  As such, they need even less than 16 bits to cover all recording requirements.


----------



## KeithEmo

rrod said:


> The wiki page just didn't make clear the connection to filtering; from what you say it sounds like it was only used for abating noise.
> 
> A bit of variation between 0 and -1dB up above 18kHz may be "bad" in the universal sense, but I doubt that is what people are hearing as sounding bad. If I play a mix of a 12kHz and 18kHz tones, I can't tell you when the 18k pops in. That's not justification for having non-flat response, of course, but it does illustrate a point about audibility versus measurement.


 
  
 The purpose of filtering in A/D and D/A conversion isn't at all intuitive.... so perhaps a proper explanation would be in order.
  
 I'm sure you're familiar with the basic fact that a digital audio file can _ONLY_ be used to store information up to just under the Nyquist frequency - which is 1/2 of the sample rate. For a 44.1 kHz file, the Nyquist frequency is 22.5 kHz, which is why a CD can't contain any information above that frequency (and using 20 kHz as the cutoff frequency does give a tiny safety margin there).
  
 However, in reality, any source is going to contain some information above 22 kHz; which will include high order harmonics produced by some instruments like cymbals, as well as actual noise from equipment like preamps and other electronics, hiss from analog master tapes, and even noise present in the room where the recording was made. The problem is that this extra "unrecordable" content doesn't simply disappear when you feed the content into your ADC. In fact, the opposite is true; if there is any content present above the maximum frequency that can be encoded "properly", it is converted into a very audible and unpleasant distortion during the encoding process. The actual process involved, and the exact results, are somewhat complicated, but the net result is that a significant portion of it is "folded back around the Nyquist frequency" into the audio _INSIDE THE AUDIO RANGE OF THE ENCODED AUDIO_.
  
 Let's take our CD as an example: the sample rate is 44 kHz, and the Nyquist frequency is 22 kHz. Now let's assume that, mixed in with my "audible source material", there is a 28 kHz tone (it could simply be some ultrasonic harmonic of some instrument, or just some 26 kHz hiss in the microphone preamp). If I were to feed that source into an ADC which lacked the proper filtering, that 28 kHz ultrasonic would be "folded down around the Nyquist frequency". This means that most of the energy contained in that 28 kHz tone, which starts out being 6 kHz _ABOVE_ the Nyquist frequency (22+6 kHz), and which cannot be encoded into our output file, will be converted by the encoding process into an equivalent amount of energy 6 kHz _BELOW_ the Nyquist frequency (22-6 kHz). So the energy from our ultrasonic 28 kHz noise source, which was totally inaudible, will appear in our encoded file as noise at 16 kHz. This process will occur with all signal energy which is present in the source that is above the Nyquist frequency - and will essentially end up as noise/distortion in the final file. The actual process is somewhat "messier" than my simplified explanation, so that 28 kHz tone will actually cause noise spikes at other points inside the audible range. And, of course, I used a single tone as an example, while the reality is somewhat more complicated. (Think of it sort of like a distorted and inverted reflection in a camera lens interfering with the desired image.) The purpose of the filter in the ADC is to remove any content outside the frequency range which the encoder can handle properly so this doesn't happen (which means anything above the Nyquist frequency).
  
 Unfortunately, most real-world recordings contain all sorts of energy above 20 kHz, ranging from harmonics of actual instruments, to room noise and such, and even to analog distortion products. _ANY_ of this which the filter fails to remove will end up as distortion and noise inside the audio band in the resulting encoded file, which is why it is critical that the filter remove it - or at least reduce it to a very low level.
  
 (Oversampling avoids this issue by shifting the Nyquist frequency upwards, thus making it easier to design a filter that is flat to all audible frequencies, yet still has sufficient attenuation above the Nyquist frequency. In order to make a "clean" conversion at 44 kHz, to the level required for a CD, you would need a filter that is flat to 20 kHz, yet down about 80 dB at 24 kHz, which is very difficult to achieve in practice. To get a similarly noise and distortion free output at a 96k sample rate, the filter would have to be flat to 20 kHz, and down 80 dB at 50 kHz, which is a far gentler slope, and so much easier to design and build.)


----------



## arnyk

ffbookman said:


> You wanna do a test?


 
 Yes, and it seems from the above obvious attempt to distract, that it is very likely that you don't want to do a valid test.
  
 All you appear want to do is yet another sighted evaluation.
  
 You're not dealing with a novice. I am fully aware with all of the well-documented fatal issues with that your ancient and highly flawed listening comparison technology involves, and have already documented it here many times.
  
 I'd be glad to review the matter again for your benefit.
  
 All this gratuitous discussion about microphone technique may impress novices, but my professional experience with live recording of music probably vastly exceeds yours, so of course I'm not the least bit distracted by it.


----------



## arnyk

keithemo said:


> The purpose of filtering in A/D and D/A conversion isn't at all intuitive.... so perhaps a proper explanation would be in order.


 
 The proper explanation is that recording at sample rates up to 4 times the standard CD sample rate is a simple matter of lining up the readily available 24/192 recording gear (I've had the highly regarded LynxTWO gear for over a decade), and make the recording. Been there done that many time. If the 16/44 has any inherent sound quality loss, then downsampling the 24/192 or 24/96 recording to 16/44 should be instantly discernible. If you eliminate the usual audiophile crutches of sighted evaluations, there is nothing that is discerned. In this day and age 24/192 recordings are a simple matter of a little google searching and downloading. Downsampling can easily be done with readily available high quality freeware such as Sox. The proper listening test can be coordinated and run for the interested individual with readily availble high quality freeware such as Foobar2000. Why waste time with pedantry when it is so simple to obtain the evidence of your own ears?


----------



## RRod

keithemo said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
  
 I'm well aware of both aliasing and imaging and how things like higher sampling rates and upsampling can help make the job of avoiding them easier for the analog components (by shifting some of the duty to the digital realm). That doesn't change the question: how bad exactly were the pro-level ADCs used for making earlier PCM recordings for CD release. If you were saying that ±1dB up above 18kHz was the problem, then I'm saying that for many of us such issues are already negligible in effect due to the limits of our hearing.


----------



## arnyk

rrod said:


> He won't accept ABX as the means of testing it.


 
  
  
 Probably an example of cherry-picking testing methodologies in order to obtain the desired results.
  
 The problem is not the rejection of ABX, because ABX is just one of many good listening test techniques.  Some are more ideal for some tests than others, but they all can be used to obtain valid results.
  
 The probable desire is to continue with some variation on the now-totally-discredited sighted listening evaluation technique or some variation on it. Of course, that can't be accepablable because of the extreme propensity for both false positives and negatives.


----------



## arnyk

rrod said:


> I'm well aware of both aliasing and imaging and how things like higher sampling rates and upsampling can help make the job of avoiding them easier for the analog components (by shifting some of the duty to the digital realm). That doesn't change the question: how bad exactly were the pro-level ADCs used for making earlier PCM recordings for CD release. If you were saying that ±1dB up above 18kHz was the problem, then I'm saying that for many of us such issues are already negligible in effect due to the limits of our hearing.


 
  
 While frequency response limitations can profoundly affect listening tests, they aren't the core of the problem. The core of the problem is masking,  a profound effect that affects hearing sample rate related effects even among listeners with ideal or near-ideal hearing.


----------



## Gr8Desire

ffbookman said:


> I can do that all the time, very easily.  Just close my eyes and hit this:
> 
> 
> 
> ...


 
  
 Fascinating feature!  

 Downsampling 192/24 to 96/24 will create a near perfect result (care to verify?).  But downsampling to a non-integral clock rate of 44.1 and for MP3 (also 44.1kHz) with induce significant dithering (introduction of random sample values to make up for missing time-specific data) during the conversion process.  

 Pono is PURPOSELY choosing inappropriate sample rates to make themselves look good. They should be using 48/16 for their so called 'low-res' playback.  That would produce a GREAT result. Not what Pono wants.  

_Made my day,_ this one.


----------



## RRod

arnyk said:


> While frequency response limitations can profoundly affect listening tests, they aren't the core of the problem. The core of the problem is masking,  a profound effect that affects hearing sample rate related effects even among listeners with ideal or near-ideal hearing.


 
  
 True, but for masking to even matter you have to be able to hear the particular frequency to begin with. What I'm trying to get at with my questions to Keith are exactly what kind of frequency response issues did early ADCs have (perhaps you have some data on this?).


----------



## jcx

its possible some "worse" pieces of the chain could have distortion that makes otherwise ultrasonic content audible as difference IMD products
  
 this was a demonstrated error of some ultrasonic hearing listening tests - both tweeters and some amps were found to be at fault - "audiophile" reputation doesn't guarantee great linearity, particularly of "no feedback" amps driven to high levels
  
 so its entirely possible for someone who hasn't verified low distortion with a mic and headphone coupler to really hear a difference even in a properly controlled, blind test - because there could be conventional audio frequency sound differences when the amp or headphone has high levels of ultrasonic signal to distort


----------



## Ruben123

Is there audible dither when comparing 44 vs. 48khz? I know of problems when the sound card can't handle one or the other though


----------



## arnyk

ruben123 said:


> Is there audible dither when comparing 44 vs. 48khz? I know of problems when the sound card can't handle one or the other though


 
  
  
 Sample rate conversion with no other changes does not require  changing the dither. The dither gets resampled along with everything else.  Decreasing the number of bits per sample point does require changing the dithering ..


----------



## Ruben123

arnyk said:


> Sample rate conversion with no other changes does not require  changing the dither. The dither gets resampled along with everything else.  Decreasing the number of bits per sample point does require changing the dithering ..




OK thanks !


----------



## arnyk

rrod said:


> True, but for masking to even matter you have to be able to hear the particular frequency to begin with.


 
  
  
 Not a  problem because it is generally the low frequency content that masks the high frequency content.
  
  
 Originally Posted by *RRod* /img/forum/go_quote.gif
  
  


> What I'm trying to get at with my questions to Keith are exactly what kind of frequency response issues did early ADCs have (perhaps you have some data on this?).


 
  
 I've previously posted such data as I was able to acquire myself here.  
  
 Digital Audio started out with short data words (8-9 bits) and low sample rates (8-12 Khz) if memory serves, but the purpose was telephone service.
  
 Prior to the introduction of the CD  data words were generally 12 bits or more, and sample rates were 32 Khz or higher with some of the better examples up around 50 KHz.
  
 The first two CD players on the market (Sony CDP 101 and Philips CD 100)  had pretty close to 16 bit resolution (measured) and reasoanbly flat frequency response.
  
 Here's the worse of the two - the analog filtered CDP 101:
  

  
 About a half dB down at 20 KHz.
  
  
 The Philips CD100 had a digital filter and far better response.


----------



## RRod

arnyk said:


> Not a  problem because it is generally the low frequency content that masks the high frequency content.
> 
> 
> Originally Posted by *RRod* /img/forum/go_quote.gif
> ...


 
  
 Interesting, thanks!


----------



## KeithEmo

rrod said:


> True, but for masking to even matter you have to be able to hear the particular frequency to begin with. What I'm trying to get at with my questions to Keith are exactly what kind of frequency response issues did early ADCs have (perhaps you have some data on this?).


 
  
 Unfortunately I'm repeating this from memory - and it's mostly based on engineering design articles (as quoted in articles with titles like "here are the problems with old style ADCs and here's how to avoid them"). Since I don't produce recordings, and so have no control over what equipment was used to convert a particular CD, I haven't bothered to keep track of individual examples. Basically, the situation where the gain of a filter rises and falls several times over a relatively narrow range of frequencies near the cutoff frequency is known as "in band ripple", and "engineering best practices" suggest that it is something to be minimized. I can tell you that this type of ripple is easily audible at lower frequencies, where it actually produces an audible "warble" in instruments or voices that move up and down across the frequency range over which it occurs, but I haven't heard of any specific testing to determine the limits of audibility.
  
 Since modern oversampling technology has rendered most of this subject moot anyway, it doesn't receive much discussion lately. I did have an opportunity recently to compare specifications on several "professional sample rate converters" sold during the early days of CD production. (I was wondering if any "old pro units" available on the used market would be suitable for use with a DAC.) One unit I recall was very proud of the fact that their unit could reduce jitter "to as low as a mere 2 nanoseconds". 2 nanoseconds is 2000 picoseconds, which would be considered barely passable for a piece of low end consumer gear today. I suspect that many of the really early ADC units probably didn't even include specs for things like jitter, because their importance simply wasn't recognized at the time, which would make comparisons difficult unless you actually were to secure samples of early units and measure them. In all fairness, I should also mention that some particular early models of ADCs had a reputation for sounding very good, and some, like one model from Pacific Microsonics, continue to be used today.
  
 (Many of the articles published on this subject predate the popular use of the Internet, so they're difficult to search for and locate.)


----------



## dprimary

ffbookman said:


> It could be but it's not because I STATED exactly what I was talking about.
> 
> Network bandwidth and storage is EVERYTHING in digital audio.  It's the main count of everything, assuming you are using the same encoding and file containers, and in this case I clearly said STEREO PCM at what resolution and if lossy or not.  I was very clear and guess what - out came the personal attacks again.
> 
> ...


 
  
 Originally Posted by *FFBookman* 


  
  
 Until then I'll just remind you all that 320k PCM < 1400k PCM < 3000k PCM < 5600k PCM.  That's available bandwidth at the various resolutions.
  
  
 Where do you see anything about bandwidth or resolution? Bit rate is not bandwidth. I know the sample rate and bit depth I'm perfectly capable of calculating storage and network bandwidth requirements. 
  
 I have needed to use both sample rates and bandwidth in my examples while the 8 bit one was for a biologist. Needing 10 tracks at 14 bit at 36k is not that unusual. In fact the early digital music recordings were 14 bit and roughly 36 k sample rates.  
  
 Here is little history
  
http://www.aes.org/aeshc/pdf/fine_dawn-of-digital.pdf
  


ffbookman said:


> KeithEmo is right, I'm right, the rest of you are choosing to degrade your audio because you _don't think you can hear_ the degradation.  That's your choice but you should degrade your own files, not continue to claim that I'm imagining things, and not add to the confusion about what people can and should hear.
> 
> 
> I have to ask, is there anything else your body can clearly and repeatedly do that you don't have math or testing to confirm?
> ...


 
  
 The topic is not about lossy or lossless compression. It is about is anything beyond 16bit 48k even needed for a release format. Bit rates are just a distraction, it is 16/48 we know the bit rate.
  
 Confirm it! all we need is one person in billions on the planet to be able to constantly pick out 16/44.1 from a 16/88 or 24/44 recording in a blind repeatable single variable test. Yet Arthur never shows up to pull the sword from the stone.
  
 I have enough reason to believe people might be able to detect above 20k that I'm open to the possibilities. Till I can repeat the test.
  
 Anyone jumping up and down claiming it is night and day my BS meter is so pegged the needle broke. Yes I can reliably  pick out a well encoded 320kbs AAC from a well recorded raw PCM file but it not trivial. Can I pick between and PCM and AAC of every pop/rock song, the answer is no,  please refer back to the "well recorded" requirement. If it is trivial maybe the encoder is garbage.
  
 Don't worry mathematicians are generally about 200 years ahead of the rest of us. The math is figured out centuries before we can possible build it.


----------



## icebear

dprimary said:


> ...
> *Anyone jumping up and down claiming it is night and day my BS meter is so pegged the needle broke.*


 
 ROFL 
	

	
	
		
		

		
			





 made my day!


----------



## arnyk

ffbookman said:


> I was very clear and guess what - out came the personal attacks again.
> 
> KeithEmo is right, I'm right, the rest of you are choosing to degrade your audio because you _don't think you can hear_ the degradation.  That's your choice but you should degrade your own files, not continue to claim that I'm imagining things, and not add to the confusion about what people can and should hear.
> 
> ...


 
  
 Pretty ironic all of the personal attacks that are part of this post that starts out by complaining about personal attacks.
  
 For example:
  


> You "16/44 is the highest" people are the most ridiculous lot of brainiacs I've come across


 
 .


> Sadly you won't be alive to buy the upgraded robot version of yourself.


 
  
  
 I can only conclude that the poster is not very self-aware.


----------



## FFBookman

"i can only conclude"   
  
 see what i mean?
  
 yeah i said i was part of it, mr reading comprehension.  you really are special.


----------



## FFBookman

Lemme say, for the record, I don't think I've ever heard 16/48.  Perhaps that's part of the disagreement here?  
  
 You think 16/48 is all you need, I say 16/44 isn't enough.  It's not the same argument on both ends.
  
 I hear an improvement with 24/44 over 16/44 on the same material, but I've never tested 24/44 against 16/48. I will have to try that sometime.
  
 I used to think "well that should be enough" and "i'm sure the high end is a bunch of BS".   See I'm poor, independent, and would rather spend on analog instead of digital gear.  So I wanted to believe 16/44 was enough.
  
 [and 16/44 is good, decent, not a major degradation from the master. When used right it's nearly perfect. It was a good choice in 1981]
  
 Until you move up to 24bit played from a real chain, then you hear more depth and air. More movement. More accurate timbre. Far more movement across the soundstage. More of the mix, and more of the humans behind the mix. 
  
 I mix music and it is a passion of mine -- where things sit in the mix, how they blend, how they bounce off the walls, how the mic was placed -- this is critical for me. The history of recorded music is important to me.  
  
 MP3 was supposed to be temporary and CD was temporary as well. CD had it's 20 year run and when it came time for the upgrade the internet drove us off a cliff. It's been about 20 years of this lossy crap and I just hope we upgrade to better than what we had 40 years ago.


----------



## castleofargh




----------



## interpolate

castleofargh said:


>




 This.


----------



## arnyk

ffbookman said:


> Lemme say, for the record, I don't think I've ever heard 16/48.


 
  
 Since 16/48 is an audio format that is widely used for the audio tracks associated with video, a person would have to live in a cave to have not heard it by accident in these days.
  


ffbookman said:


> You think 16/48 is all you need, I say 16/44 isn't enough.  It's not the same argument on both ends.


 
 It's true. I base my belief on generally accepted science and reliable evidence and you seem to base your opinon on some theories that you seem to have made up and maybe some inherently flawed listeni ng evaluations. Not the same thing.
  


ffbookman said:


> I hear an improvement with 24/44 over 16/44 on the same material,


 
 So how is it that you can compare musical selections like this, when you've already said that:
  


> By the authority of where our brain processes sound verses where our brain processes the recollection of sound. They aren't even in the same regions of the brain.


 
  
 It would appear that were the above true, no comparison of 24/44 over 16/44  would be possible.  Yet, you claim that you have done it. What magic was employed?


----------



## Thad-E-Ginathom

> Originally Posted by *arnyk* /img/forum/go_quote.gif
> ... ... ...
> 
> 
> ...


 
  
 The only way around that one would seem to be _simultaneously_ listening!


----------



## arnyk

thad-e-ginathom said:


> The only way around that one would seem to be _simultaneously_ listening!


 
  
 Been there, done that, had a horrible time.
  
 It appears that Mr. Bookman's problem is to demonstrate or specify an effective means for comparing musical selections processed by different technical processes that doesn't have all of the inherent flaws of sighted evaluations but can't be done blind.


----------



## Ruben123

thad-e-ginathom said:


> The only way around that one would seem to be _simultaneously_ listening!




Or just looking at the file extension


----------



## Thad-E-Ginathom

My media player _tells_ me when I am listening to good music. In fact, I don't even have to put the headphones on!


----------



## RRod

thad-e-ginathom said:


> The only way around that one would seem to be _simultaneously_ listening!


 
  
 Actually, if you invert one of the files then simultaneous listening can actually be instructive:


----------



## Thad-E-Ginathom

(But, in practice, isn't it quite difficult to get a really clean null? Not that I ever tried...)


----------



## RRod

thad-e-ginathom said:


> (But, in practice, isn't it quite difficult to get a really clean null? Not that I ever tried...)


 
  
 I would forgive someone for mixing the tracks together digitally.


----------



## money4me247

rrod said:


> Actually, if you invert one of the files then simultaneous listening can actually be instructive:


 
  
 You can also set-up to play one channel per side (each channel with different format) to see if you can hear any differences. if you are using IEMs that look the same on both sides, you can do a semi-'blinded' test having someone else cover the L/R markings with a different sticker and swapping to see if you can call out which format is which.


----------



## RRod

money4me247 said:


> You can also set-up to play one channel per side (each channel with different format) to see if you can hear any differences. if you are using IEMs that look the same on both sides, you can do a semi-'blinded' test having someone else cover the L/R markings with a different sticker and swapping to see if you can call out which format is which.


 
  
 If your right ear has more frequencies gone than your left then that could mess things up, unless you just use the same ear each time. Still seems like way more work than a straight up switcher.
  
 The main point I was trying to make is that it can be helpful to see and hear exactly what gets lost when you drop bits or samples. This example was from dithering a 24/96 track down to 16/96… oh I'm sorry, I mean from truncating a 24/96 track down to 14/96 ^_^ Noise below -100dB, that's what you're missing, yet I'm sure people will tell you all kinds of amazing audible things the extra 10bits are doing for this remaster (anyone care to guess what it is 
	

	
	
		
		

		
		
	


	




)


----------



## Thad-E-Ginathom

rrod said:


> If your right ear has more frequencies gone than your left then that could mess things up, .. ... ...


 
  
 Yep. My ears are more than a bit lacking when using them both, and are certainly not matched.
  
 Oh for the perfect hearing of youth (and the lucky few).


----------



## arnyk

ruben123 said:


> Or just looking at the file extension


 
  
  
 Yes - the _*advantages*_ of sighted listening!   Your identification of the UUTs is always perfect and your pet theories, no matter how far off the wall,  are always verified.
  
 This contrasts with messing around with Lady Science, who will bite your hand for the heck of it, falsify your theories on the spot, and generally lay waste to your illusions in a heart beat.


----------



## interpolate

Why bother with high-fidelity if your only aim is to point out the negatives of superlative testing?
  
 To be fair, people should just accept what they like and forget what the scientists are saying.


----------



## Mambosenior

interpolate said:


> ...people should just accept what they like and forget what the scientists are saying.




Absolutely! The trouble, many times, is that some people attempt to equate "what they like" WITH scientific principles WITHOUT understanding scientific principles.

I never understood why there are subjective posts in the (clearly marked) "Sound Science" forum. A war of words always ensues.


----------



## Ruben123

Just found it at pono's thread and dropping it here... http://www.audiostream.com/content/abx-tests-prove-hi-res-audio-legit#iIDZ557EtGQgqfPt.97


----------



## RRod

ruben123 said:


> Just found it at pono's thread and dropping it here... http://www.audiostream.com/content/abx-tests-prove-hi-res-audio-legit#iIDZ557EtGQgqfPt.97


 
  
 If memory serves there were issues found with that particular test. If Arny pops on I'm sure he can give all the sordid details. It just goes to show that there is some import in making sure that the ultimate answers to such questions come from some kind of rigorous testing environment, not, you know, the interwebz.


----------



## goodyfresh

So recently I used dBpoweramp to convert/downsample some 24/96 and 24/192 files down to 24/48, and proceeded to do some A/B testing.  Wasn't quite sure if I really was hearing a difference or not, so I had my roommate help me with doing some blind-testing, and sure enough, nope, no real audible difference, at least for me.  Not using any of my equipment. . .whether straight from the computer, or using my Fiio X3ii as a USB DAC (which sounds better than the computer on its own, obviously), and whether using the V-Moda M-80's, the Shure SE215, or the Sony MDR-1A, I was completely incapable of identifying any difference between the higher-and-lower res lossless files in blind testing.  I then proceeded to convert some tracks all the way down to 320kbs Mp3 lossy, and again. . .I couldn't hear a difference, seriously!  Is somethign wrong with my ears that I couldn't hear a difference between the 320kbs lossy and the lossless, or is it really just not an audible difference, objectively speaking, for anyone?


----------



## RRod

goodyfresh said:


> So recently I used dBpoweramp to convert/downsample some 24/96 and 24/192 files down to 24/48, and proceeded to do some A/B testing.  Wasn't quite sure if I really was hearing a difference or not, so I had my roommate help me with doing some blind-testing, and sure enough, nope, no real audible difference, at least for me.  Not using any of my equipment. . .whether straight from the computer, or using my Fiio X3ii as a USB DAC (which sounds better than the computer on its own, obviously), and whether using the V-Moda M-80's, the Shure SE215, or the Sony MDR-1A, I was completely incapable of identifying any difference between the higher-and-lower res lossless files in blind testing.  I then proceeded to convert some tracks all the way down to 320kbs Mp3 lossy, and again. . .I couldn't hear a difference, seriously!  Is somethign wrong with my ears that I couldn't hear a difference between the 320kbs lossy and the lossless, or is it really just not an audible difference, objectively speaking, for anyone?


 
  
 In honest testing the difference should be subtle if it is detectable, and probably only then with certain kinds of "killer" sound bits (castanets seem to get a lot of mention). One thing to try is to gradually reduce the MP3 bitrate down until you can hear differences, note where those differences are, then see if you can pick them up at the next higher rate.
  
 The issue with many people's claims about something like 320mp3 isn't so much that they claim they hear a difference, but that this difference is *night and day*, which if you play around with testing these things you might end up having a hard time believing.


----------



## goodyfresh

rrod said:


> In honest testing the difference should be subtle if it is detectable, and probably only then with certain kinds of "killer" sound bits (castanets seem to get a lot of mention). One thing to try is to gradually reduce the MP3 bitrate down until you can hear differences, note where those differences are, then see if you can pick them up at the next higher rate.
> 
> The issue with many people's claims about something like 320mp3 isn't so much that they claim they hear a difference, but that this difference is *night and day*, which if you play around with testing these things you might end up having a hard time believing.


 

 I see.  Have YOU ever been able to hear differences, however subtle, in objectively-done A/B (preferably blind) testing between 320kbs lossy and lossless FLAC files?


----------



## RRod

goodyfresh said:


> I see.  Have YOU ever been able to hear differences, however subtle, in objectively-done A/B (preferably blind) testing between 320kbs lossy and lossless FLAC files?


 
  
 No, but I also am not god's gift to hearing. 256mp3 seems fine enough for me, but I run with AAC anyway because why not use the better codec for the same bitrate? At 192 I can start picking out things if I really, really tune in. Some people would claim to hear differences with 320mp3 whilst listening to earbuds on a train…


----------



## money4me247

goodyfresh said:


> I see.  Have YOU ever been able to hear differences, however subtle, in objectively-done A/B (preferably blind) testing between 320kbs lossy and lossless FLAC files?


 
 It is possible. I've tried a ton of blind testing. Depends more on the source track. Some masters/source tracks have elements that you can pick up differences, but it does take quite a while to find spots that you can reliably consistently identify. Always an extremely extremely subtle difference from my experience and hard to believe that certain people view it to be a 'night-and-day' difference after blind testing. Basically takes a while even finding a spot on a track where you think you can hear that type of subtle difference before you can even run your back-and-forth ABx guessing.
  
 I would agree that generally for pleasure listening or musical enjoyment, there is no real need to go above 320kbps. The difficulty to consistently reliably identify differences cannot be overstated at this level. Hard enough here that I am extremely skeptical of subjective claims of being able to detect above CD-quality or lossless FLAC files. That along with the scientific explanation behind higher bit depth and sampling rates (which basically strongly implies if not already conclusively states) that it makes to difference chasing higher bits depths & sampling rates provided that the bit depth & sampling rate used covers the audible spectrum of frequency ranges that human can hear. The standard CD bit depth and sampling rate already covers that, so hard to really say what the true benefits of beyond CD quality audio does.
  
 However, the importance of having a nice master cannot be understated and that can & will make a extremely large difference in sound quality. Funny how that sort of thing ever gets discussed though.
  
 edit: also would like to note that some dacs have filters or whatever that can make differences between different formats more exaggerated. but that is really a problem with the dac... not the source file.


----------



## goodyfresh

money4me247 said:


> It is possible. I've tried a ton of blind testing. Depends more on the source track. Some masters/source tracks have elements that you can pick up differences, but it does take quite a while to find spots that you can reliably consistently identify.
> 
> I would agree that generally for pleasure listening or musical enjoyment, there is no real need to go above 320kbps. The difficulty to consistently reliably identify differences cannot be overstated at this level. Hard enough here that I am extremely skeptical of subjective claims of being able to detect above CD-quality or lossless FLAC files. That along with the scientific explanation behind higher bit depth and sampling rates (which basically strongly implies if not already conclusively states) that it makes to difference chasing higher bits depths & sampling rates provided that the bit depth & sampling rate used covers the audible spectrum of frequency ranges that human can hear. The standard CD bit depth and sampling rate already covers that, so hard to really say what the true benefits of beyond CD quality audio does.
> 
> However, the importance of having a nice master cannot be understated and that can & will make a extremely large difference in sound quality. Funny how that sort of thing ever gets discussed though.


 

 This seems like as good a place as any to once again post this meme:
  


 LMAO!  That guy can totally hear the difference between high and low sample-rates, guys!  And just look at how smart and sophisticated he looks!  Wow!


----------



## money4me247

goodyfresh said:


> This seems like as good a place as any to once again post this meme:
> 
> 
> 
> LMAO!  That guy can totally hear the difference between high and low sample-rates, guys!  And just look at how smart and sophisticated he looks!  Wow!


 
  
 hahaha! actually dogs may be able to hear that high. humans though... nope. our ears just can't do it.


----------



## XenHeadFi

goodyfresh said:


> I see.  Have YOU ever been able to hear differences, however subtle, in objectively-done A/B (preferably blind) testing between 320kbs lossy and lossless FLAC files?


 
 A long time ago, I test a LAME MP3 (256 VBR Q9) made from a CD that I ripped using EAC that was encoded in FLAC. I used Foobar's ABX plugin and could not tell the difference. I then read that differences might be found in high frequency transients, so I listened extremely intently to just high frequency transients. I finally heard a "flutter"/"wobble" of the top ring from a bell/triangle-type thing in the MP3 that was not in the FLAC. From this, I decided that my MP3 settings were more than sufficient for portable use (high sound quality, small-ish file sizes). I only heard the difference when I wasn't listening to the music, but trying to find a sound signature.
  
 EDIT: Oops, forgot my LAME settings. Here is part of the command from my PYTHON script: lame --quiet --vbr-new --add-id3v2 -V 0 -h -b 256. So "-V 0" that means highest quality VBR and "-q 2", recommended PsychoAcoustical modeling.


----------



## goodyfresh

money4me247 said:


> hahaha! actually dogs may be able to hear that high. humans though... nope. our ears just can't do it.


 

 That's the point, man.  Dogs CAN hear that high.  I've heard many stories (and in one case, actually seen it happen in real life, a while back!) about certain high-res (96Khz or above) masters of certain songs that, when played back on certain systems with frequency-responses going all the way up to 50Khz or more, will make dogs start flipping-out and going berserk, because they can hear the higher ultrasonic content in the music that humans can't   My buddy back in West Virginia had a nice system with JBL monitor speakers, and when he'd play certain Pink Floyd and other songs in hi-res, his Shiba Inu would start to totally lose its ****, it was hilarious 
	

	
	
		
		

		
			




  
  


xenheadfi said:


> A long time ago, I test a LAME MP3 (256 VBR Q9) made from a CD that I ripped using EAC that was encoded in FLAC. I used Foobar's ABX plugin and could not tell the difference. I then read that differences might be found in high frequency transients, so I listened extremely intently to just high frequency transients. I finally heard a "flutter"/"wobble" of the top ring from a bell/triangle-type thing in the MP3 that was not in the FLAC. From this, I decided that my MP3 settings were more than sufficient for portable use (high sound quality, small-ish file sizes). I only heard the difference when I wasn't listening to the music, but trying to find a sound signature.


 

 That's a pretty enlightening story! Thanks for sharing


----------



## Thad-E-Ginathom

money4me247 said:


> ... ... ...
> However, the importance of having a nice master cannot be understated ... ... ...


 
  
  


goodyfresh said:


> LMAO!  That guy can totally hear the difference between high and low sample-rates, guys!  And just look at how smart and sophisticated he looks!  Wow!


 
  
  
 I expect he has a nice master too!


----------



## flognarde

No one say you can't or you don't hear the difference but that's audiofoolery. In my tests I only noticed differences when the album had been remastered, which is a totally different subject.
 On some DSD re-encoded albums it even degraded the sound.


----------



## goodyfresh

thad-e-ginathom said:


> I expect he has a nice master too!


 
 He's a dog, though, so he can hear the difference between hi-res and low-res even without different masters, since he can hear ultrasonic frequencies up to and above 50Khz


----------



## RRod

What a bad master might look like:


----------



## goodyfresh

rrod said:


> What a bad master might look like:


 
 LMAO that's in such poor taste. . .I love it


----------



## sonitus mirus

goodyfresh said:


> LMAO that's in such poor taste. . .I love it


 
  
 It reminded me of tonight's Ohio State vs Virginia Tech game.


----------



## castleofargh

metallica: master of puppies.


----------



## frodeni

I do not have the time to read through 1162 posts, but from the hundreds I have read, I sort of wonder if people know why we ended up with 16/44.1? It never had anything to do with human thresholds, or any of the arguments just thrown out there now, supporting this as the human threshold of ultimate hearing. It sort of have something to do with being able to play Beethoven in the car, and the capacity given the laser in 1984-85.
  
 Is there any real science in this thread upon the ability of the ear? What did I miss?
  
 I have seen plenty just claiming this or that, but what are the bases this is founded on? Apart from pure speculation that is.
  
 From a scientific point of view, there is only need for singe proper ABX test to foul the heading of this thread. Given the subjective data out there, the claim of this thread, is probably false.


----------



## RRod

frodeni said:


> I do not have the time to read through 1162 posts, but from the hundreds I have read, I sort of wonder if people know why we ended up with 16/44.1? It never had anything to do with human thresholds, or any of the arguments just thrown out there now, supporting this as the human threshold of ultimate hearing. It sort of have something to do with being able to play Beethoven in the car, and the capacity given the laser in 1984-85.
> 
> Is there any real science in this thread upon the ability of the ear? What did I miss?
> 
> ...


 
  
 So auditory testing doesn't show an upper frequency hearing limit of near 20kHz? Certainly the specific choice of 44.1 might be due to other factors, but it's still not 35kHz.
  
 Subjective data are certainly handing for coming up with hypothesis and getting a sense for what direction research should go, but as you said ultimately objective standards must be set for deciding when paradigms have been broken. How well do you fair in ABX tests of hi-res vs. Redbook on your own system?


----------



## goodyfresh

rrod said:


> So auditory testing doesn't show an upper frequency hearing limit of near 20kHz? Certainly the specific choice of 44.1 might be due to other factors, but it's still not 35kHz.
> 
> Subjective data are certainly handing for coming up with hypothesis and getting a sense for what direction research should go, but as you said ultimately objective standards must be set for deciding when paradigms have been broken. How well do you fair in ABX tests of hi-res vs. Redbook on your own system?


 

 Well, the majority of people CAN'T hear as high as 20Khz.  I'm in my late 20's, and already the upper-limit of my hearing has dropped as low as 16.5Khz.  HOWEVER, there are a select few humans out there who can apparently hear stuff as high as 21 or 22Khz.


----------



## Thad-E-Ginathom

goodyfresh said:


> He's a dog, though, so he can hear the difference between hi-res and low-res even without different masters, since he can hear ultrasonic frequencies up to and above 50Khz


 
  
 I wonder if he needs any of those frequencies to identify _His Master's Voice?_
  


frodeni said:


> I do not have the time to read through 1162 posts, but from the hundreds I have read, I sort of wonder if people know why we ended up with 16/44.1? It never had anything to do with human thresholds, or any of the arguments just thrown out there now, supporting this as the human threshold of ultimate hearing. It sort of have something to do with being able to play Beethoven in the car, and the capacity given the laser in 1984-85.


 
  
 I'm not sure, even though I pre-date the CD by a couple of decades, that Beethoven (or anything else) from a CD player _in a car_ wasn't beyond the wildest dreams of the early makers of CDs and CD players.
  
 I do vaguely remember some of the stuff that went into the possibly-not-optimal choice of 16/44.1, but hey, "Covers the human audible range," was, I think, part of it. And, if by any remote chance it wasn't, then, hey, _What an amazing coincidence! _
  
 Whatever next? Some guy will be along with some mathematical formulae to explain how it works, but not perfectly, of course; his numbers won't explain everything we hear. 
  
  


> Is there any real science in this thread upon the ability of the ear? What did I miss?


 
  
 The first post?
  


> From a scientific point of view, there is only need for singe proper ABX test to foul the heading of this thread.


 
  
 You are absolutely right: there is only the need of one ABX blind test that shows that, all other considerations aside (see the first post), high sample rates can be _reliably and repeatable_ (and _readily,  _seeing as how this is a marketable product we talking about) identified.
  
 When it happens, it will make audio history. But don't hold your breath, because it has had a while so far.
  


> Given the subjective data out there, the claim of this thread, is probably false.


 
  
 Did you come to give us answers, or just to restate the problem?


----------



## frodeni

Well guys, Sony demanded that a specific symphony by Beethoven could be played in a car stereo. That defined the length of a CD, and the width. Then if 20-20k was to be covered, and the amount of storage was a given by the laser, the end result simply had to be 16/44.1. Because of the requirement of one major Sony exec.
  
 And no, I do not contest the frequency range of the human ear, but what the ear might hear within that range. It is not a given that the 16/44.1 has the resolving power within that range, as to outdo the ear.
  
 And no, I have not stepped into high res music yet. No rush. When I do, I will down sample a great recording, by the book, and listen carefully. I am in no rush, as I simply know that I will hear great differences. I always do, if others consistently have so before me. As to detect differences, I sort of always end up on the extreme end of the scale. Yet to ever been beaten, but I am not young anymore. So no rush.
  
 I know what I hear, and do not care what others think. Especially those who try to corrupt a healthy touch with the real world, and my senses. And where is that bloody proof of yours. Stills eludes me like nuts. Where is that proof that I cannot sense what I in-fact do? Or is it going to still be this insane head game?
  
 I simply enjoy the music on Tidal, and hardly lift a finger to acquire music. Not like it used to be. I rather stick with 16/44.1 than spend all that time again.


----------



## castleofargh

goodyfresh said:


> rrod said:
> 
> 
> > So auditory testing doesn't show an upper frequency hearing limit of near 20kHz? Certainly the specific choice of 44.1 might be due to other factors, but it's still not 35kHz.
> ...


 

 and a few guys are born with 3 arms. but it doesn't warrant to go making all the shirts in the world with one more hole.


----------



## castleofargh

frodeni said:


> Well guys, Sony demanded that a specific symphony by Beethoven could be played in a car stereo. That defined the length of a CD, and the width. Then if 20-20k was to be covered, and the amount of storage was a given by the laser, the end result simply had to be 16/44.1. Because of the requirement of one major Sony exec.
> 
> And no, I do not contest the frequency range of the human ear, but what the ear might hear within that range. It is not a given that the 16/44.1 has the resolving power within that range, as to outdo the ear.
> 
> ...


 

 the best answer you can hope for is by doing the test yourself. I can't claim that nobody can hear the difference, but I sure can claim that I can't. so it's case closed as far as my audio library is concerned.


----------



## RRod

frodeni said:


> Well guys, Sony demanded that a specific symphony by Beethoven could be played in a car stereo. That defined the length of a CD, and the width. Then if 20-20k was to be covered, and the amount of storage was a given by the laser, the end result simply had to be 16/44.1. Because of the requirement of one major Sony exec.
> 
> And no, I do not contest the frequency range of the human ear, but what the ear might hear within that range. It is not a given that the 16/44.1 has the resolving power within that range, as to outdo the ear.
> 
> ...


 
  
 I've never seen any actual agreement on the Beethoven story. Here is an alternative theory.
  
 You'll have to define what you mean by "resolving power" before we can discuss a claim about Redbook not having enough.
  
 Where did Tidal come into this?


----------



## Thad-E-Ginathom

rrod said:


> Here is an alternative theory.


 
 Yes, I thought that I remembered something that was full of numbers and much more boring than getting a symphony on one "side," as it would probably have been thought of at the time.
  
 Being able to do that, of course, is one of the great advantages of CD over LPs. Not that there are not symphonies that take up more than one CD (Mahler 2 comes to mind, simply because it is a favourite, and one of the classical CDs that I own).
  
 So, I don't mind imagining some Sony executive that picked his favourite piece of music and used it as the yard stick. Sony executives were a different breed, anyway, back in the days when it was a vibrant, innovative engineering company. Wasn't it the requirement of a "Sony executive" that gave us the Walkman, which was really the birth of _pocket-portable_ personal music listening.


----------



## goodyfresh

castleofargh said:


> and a few guys are born with 3 arms. but it doesn't warrant to go making all the shirts in the world with one more hole.


 

 So we should just drop sample-rates down to 35Khz in order to save room on people's hard-drives?  Seems silly to me


----------



## RRod

goodyfresh said:


> So we should just drop sample-rates down to 35Khz in order to save room on people's hard-drives?  Seems silly to me


 
  
 It's more that we're saying that Redbook is already the "3-legged pants" that should cover everyone. Things like AAC and MP3 are seeing if we can sew up one of the legs without anyone noticing


----------



## frodeni

rrod said:


> I've never seen any actual agreement on the Beethoven story. Here is an alternative theory.
> 
> You'll have to define what you mean by "resolving power" before we can discuss a claim about Redbook not having enough.
> 
> Where did Tidal come into this?


 
  
 As for Beethoven, we would never see any agreement, given the people in question. It has been a while, so I cannot recollect my sources, but they were pretty solid. In the end, it was a nice trade-of anyway.
  
 As for resolving power, that could be as in bits, thus higher resolving power among the y-axis. That would resolve finer details, and offer more dynamic range. Higher sampling frequency, would also offer more accuracy along the x-axis. I have lived long enough to see theories come and go in this field. What once seemed rock solid, might suddenly be disrupted by a discovery disrupting it. Like when jitter first came to be. Or this bloody digital noise over USB, that comes on top of the signal. What once was jibberish, is now common knowledge.
  
 My guide is what I sense, and that is a bumpy road. Like when that USB cable worked. I am still working on adjusting to that.
  
 It has been a while since I looked at Nyquist and the red book. But thinks have changed over the years. On both my and the industry understanding. The sonic experiences are more or less the same. SPDIF beats optical, to name one. Also, my guess, well probably not even a guess, is that for vector sound, 16/44,1 is not close to cutting it.
  
 Tidal comes into this, as I play my music by Tidal, and it is expected that Tidal and some others, will offer high def in a few years time. Like is the case in central Europe by other services. That is my real reason for not going there yet.


----------



## goodyfresh

frodeni said:


> As for Beethoven, we would never see any agreement, given the people in question. It has been a while, so I cannot recollect my sources, but they were pretty solid. In the end, it was a nice trade-of anyway.
> 
> As for resolving power, that could be as in bits, thus higher resolving power among the y-axis. That would resolve finer details, and offer more dynamic range. Higher sampling frequency, would also offer more accuracy along the x-axis. I have lived long enough to see theories come and go in this field. What once seemed rock solid, might suddenly be disrupted by a discovery disrupting it. Like when jitter first came to be. Or this bloody digital noise over USB, that comes on top of the signal. What once was jibberish, is now common knowledge.
> 
> ...


 

 Not really more than tangentially on-topic but, in the interest of all of us agreeing on something. . .One thing we can all agree on and not need to debate, I suppose, is that the Nyquist Sampling Theorem is one of the downright COOLEST results in all of Applied Math


----------



## castleofargh

about the CD format to fit one symphony, it's sweet and makes for a good tell. but even if it was true, they could have just made the CD slightly bigger... after all there wasn't yet a standard for size and from vinyl to CD they had quite a margin to play with if they really wished to have longer play time.
 and if they didn't care to reduce the resolution, then why make something as big as a CD?  they could have saved money making something smaller.
 to me the symphony story doesn't make more sense than saying some guy randomly picked a number and got away with it.
  
  
  
 Quote:


goodyfresh said:


> castleofargh said:
> 
> 
> > and a few guys are born with 3 arms. but it doesn't warrant to go making all the shirts in the world with one more hole.
> ...


 
 my point was that we don't have a legitimate reason to go up. but there are already plenty of available ways to go down for those who want to. so the format doesn't matter to me.
 as long as I'm not asked to pay more for music I don't hear, I will be fine even with 64/356khz.


----------



## goodyfresh

castleofargh said:


> my point was that we don't have a legitimate reason to go up. but there are already plenty of available ways to go down for those who want to. so the format doesn't matter to me.
> as long as I'm not asked to pay more for music I don't hear, I will be fine even with 64/356khz.


 

 LMAO 64/356?  The file-sizes would be ASTRONOMICAL!


----------



## bfreedma

The link below is the history of the CDs development and rational for technical decisions as recalled by the lead Phillips engineer from the Phillips/Sony consortium responsible for the development of the format and Red Book standard. He very specifically debunks the myth about the length of Beethoven's 9th.

https://web.archive.org/web/20141104160226/http://www.exp-math.uni-essen.de/~immink/pdf/cdstory.htm

Not absolute proof, but certainly a credible source.


----------



## RRod

bfreedma said:


> The link below is the history of the CDs development and rational for technical decisions as recalled by the lead Phillips engineer from the Phillips/Sony consortium responsible for the development of the format and Red Book standard. He very specifically debunks the myth about the length of Beethoven's 9th.
> 
> https://web.archive.org/web/20141104160226/http://www.exp-math.uni-essen.de/~immink/pdf/cdstory.htm
> 
> Not absolute proof, but certainly a credible source.


 
  
 Interesting; thanks for the read. Already thinking of oversampling even when this stuff first came out; to be a fly on the wall when these guys were planning the digital audio revolution.


----------



## RRod

goodyfresh said:


> LMAO 64/356?  The file-sizes would be ASTRONOMICAL!


 
  
 People are already 3/8 of the way there.


----------



## sonitus mirus

goodyfresh said:


> LMAO 64/356?  The file-sizes would be ASTRONOMICAL!


 
  
 Let me grab my calculator.
  
 Pink Floyd's album, _Dark Side of the Moon_, was about 43 minutes in length.  For 2 channels, that would be an uncompressed bit rate of around 45,000 kbps with a file size just over 14GB.  
  
 I probably still could never hear a difference between that and a 320 kbps MP3 version, with a total file size just under 100MB for the same album.
  
 My math could be wrong.  If it is, the calculator must be broken.


----------



## goodyfresh

sonitus mirus said:


> Let me grab my calculator.
> 
> Pink Floyd's album, _Dark Side of the Moon_, was about 43 minutes in length.  For 2 channels, that would be an uncompressed bit rate of around 45,000 kbps with a file size just over 14GB.
> 
> ...


 

 AHAHAHAAH 14gb for Dark Side of the Moon, seriously?  That's insane.  But you just know that once that bit-depth becomes available, people are going to be all like "OMG I LOVE MY MUSIC IN 64/356, it sounds SOOOOO much better [because I'm a total fool who falls for the placebo-effect every time like a dumbass]!"  LMAO


----------



## RRod

goodyfresh said:


> AHAHAHAAH 14gb for Dark Side of the Moon, seriously?  That's insane.  But you just know that once that bit-depth becomes available, people are going to be all like "OMG I LOVE MY MUSIC IN 64/356, it sounds SOOOOO much better [because I'm a total fool who falls for the placebo-effect every time like a dumbass]!"  LMAO


 
  
 Yes, that's one of the issues that's at the heart of this kind of debate. It would be all fine with me if we decided today that 24/96 would be the new standard, but I would have 100% confidence that 2 years down the road the record companies will be pitching a new DSotM at gajillion bits/bazillion Hz to make new money on what will probably be a mastering that is already available.


----------



## XenHeadFi

rrod said:


> ... I would have 100% confidence that 2 years down the road the record companies will be pitching a new DSotM at gajillion bits/bazillion Hz to make new money on what will probably be a mastering that is already available.


 
 More like a WORSE mastering than what is currently available. Bigger and Louder!


----------



## goodyfresh

rrod said:


> Yes, that's one of the issues that's at the heart of this kind of debate. It would be all fine with me if we decided today that 24/96 would be the new standard, but I would have 100% confidence that 2 years down the road the record companies will be pitching a new DSotM at gajillion bits/bazillion Hz to make new money on what will probably be a mastering that is already available.


 

 Ugh *vomits violently*


xenheadfi said:


> More like a WORSE mastering than what is currently available. Bigger and Louder!


 

 *Even More vomit*


----------



## inthere

I approached several major labels about adopting a new standard developed at Princeton by a Dr. Edgar Chouri which is basically an algorithm that's now able to reproduce Binaural data through normal speakers. Before, some of you know binaural recordings could only be heard properly on headphones. 
  
 it would introduce a new revenue stream because binaural recordings are very distinguishable from ordinary recordings. 
  
 The problem is it's not backwards compatible with old recordings; if it wasn't originally recorded with a dummy head binaural mic you won't be able to simulate it well, so it would probably be limited to newer artists.


----------



## goodyfresh

inthere said:


> I approached several major labels about adopting a new standard developed at Princeton by a Dr. Edgar Chouri which is basically an algorithm that's now able to reproduce Binaural data through normal speakers. Before, some of you know binaural recordings could only be heard properly on headphones.
> 
> it would introduce a new revenue stream because binaural recordings are very distinguishable from ordinary recordings.
> 
> The problem is it's not backwards compatible with old recordings; if it wasn't originally recorded with a dummy head binaural mic you won't be able to simulate it well, so it would probably be limited to newer artists.


 

 I've always wished that ALL music in studios would be recorded as binaural recordings, or perhaps with two microphone setups at once in the studio, one for normal recordings for 2.1 speakers and one with a dummy-head for binaural.  There's such a shortage of good binaural music out there, but I much prefer it because of the soundstage and realism of the imaging.


----------



## KeithEmo

I just bit the proverbial bullet and BOUGHT the AES paper (AES Convention Paper 9174). (Apparently AES papers remain copyrighted, and cannot be reprinted for free, which is why everyone is talking about the paper but not many people seem to have read it.) Note that the title of the paper refers to "filters", but what they're testing is whether the necessary application of band-limiting, as applied to CD content, is audible. Basically, for the study, they took some 24/192k content and band limited it using filters equivalent to those you would use to record CD and DVD audio content (filtered to cut it off at the Nyquist frequencies of 22 kHz and 24 kHz respectively). The results were that the subjects _WERE_ in fact able to tell the "CD version" from the "high-def version" - with a reliability that exceeded random chance (in general, it was between 56% and 66%, which may not be overwhelming, but is statistically better than random). The study also found that the difference was more audible with certain passages than with others, and included some interview data (taken afterwards) where the subjects explained subjectively the differences they claimed to hear (which do coincide with the differences many audiophiles also claim to hear).
  
 After reading the paper itself, and some of the "critiques" of it, I came to the following conclusions:
  
 1) The test definitely concluded that some subjects, with some music samples, can in fact hear a difference with a statistical certainty beyond random chance.
  
 2) The test itself was well designed, and specifically avoided some of the limitations and weakness that were present in previous tests that came to the opposite conclusion. (There was discussion of the inadequacies of some previous tests, and of how this test was designed to overcome those, including specifications showing that the equipment used by the current test was in fact able to resolve the differences that were being tested for audibility.)
  
 3) All of the critiques I saw referred to claimed errors in the choice of appropriate test methodology. However, none of them specifically suggested any way in which such errors, if they actually were errors, could possibly have caused any sort of "false positive". In other words, I didn't see any claims that there was anything wrong with the test that could reasonably have caused it to "find" audible differences that weren't in fact real. They seem to have done an excellent job of ensuring that, if differences were found, those differences would be due to actual audible differences in the samples - as heard by the test subjects.
  
 4) The test did use very few subjects, and a wider test, with a lot more subjects, would certainly be more conclusive...
  
 (Since the paper is copyright, I can't post it - but anyone who wishes to dispute the conclusion really owes it to themselves - and the authors - to read the paper before criticizing it or challenging the results. To do so will cost you $20 - assuming you're not an AES member - which is probably "money spent in a good cause".)
  
  
 Quote:


rrod said:


> If memory serves there were issues found with that particular test. If Arny pops on I'm sure he can give all the sordid details. It just goes to show that there is some import in making sure that the ultimate answers to such questions come from some kind of rigorous testing environment, not, you know, the interwebz.


----------



## RRod

I was thinking of the online "keys" test that also ended up having problems. I see the paper you bought is the one by Meridian; it's been a while since I read the discussion on hydrogen on that one. Guess I should go back.**
  
 *The take home, given by the admin when he closed the thread:
 "I think it was well established in this thread that the authors of this paper (intentionally?) used inappropriate practices to push Meridian's agenda to sell expensive gear and their new format MQA."


----------



## castleofargh

keithemo said:


> I just bit the proverbial bullet and BOUGHT the AES paper (AES Convention Paper 9174). (Apparently AES papers remain copyrighted, and cannot be reprinted for free, which is why everyone is talking about the paper but not many people seem to have read it.) Note that the title of the paper refers to "filters", but what they're testing is whether the necessary application of band-limiting, as applied to CD content, is audible. Basically, for the study, they took some 24/192k content and band limited it using filters equivalent to those you would use to record CD and DVD audio content (filtered to cut it off at the Nyquist frequencies of 22 kHz and 24 kHz respectively). The results were that the subjects _WERE_ in fact able to tell the "CD version" from the "high-def version" - with a reliability that exceeded random chance (in general, it was between 56% and 66%, which may not be overwhelming, but is statistically better than random). The study also found that the difference was more audible with certain passages than with others, and included some interview data (taken afterwards) where the subjects explained subjectively the differences they claimed to hear (which do coincide with the differences many audiophiles also claim to hear).
> 
> After reading the paper itself, and some of the "critiques" of it, I came to the following conclusions:
> 
> ...


 

 was the filter used a hard one (at 20khz or something), or what we usually see starting gently around 16khz? because to me the roll off alone at high frequency could sometimes be enough to notice a difference (if we have the ears and transducers that can keep up after 16khz). and that difference would become clearly small with the use of 48khz and starting to filter a little higher.
 I'm not sure there is something else to hear?


----------



## RRod

castleofargh said:


> was the filter used a hard one (at 20khz or something), or what we usually see starting gently around 16khz? because to me the roll off alone at high frequency could sometimes be enough to notice a difference (if we have the ears and transducers that can keep up after 16khz). and that difference would become clearly small with the use of 48khz and starting to filter a little higher.
> I'm not sure there is something else to hear?


 
  
 See the hydrogen thread I posted in the edit to my last post. "Rectangular dither" gets a lot of mention…


----------



## castleofargh

yup I saw it after I posted. in fact I remember reading that topic at the time, just didn't know it was the same thing ^_^.


----------



## RRod

castleofargh said:


> yup I saw it after I posted. in fact I remember reading that topic at the time, just didn't know it was the same thing ^_^.


 
  
 The webpage linked a few pages ago had references both to the (in)famous hydrogen key test and this convention paper. Sometimes I wonder what life is like in an atmosphere of hydrogen rather than uranium hexafluoride.


----------



## KeithEmo

castleofargh said:


> was the filter used a hard one (at 20khz or something), or what we usually see starting gently around 16khz? because to me the roll off alone at high frequency could sometimes be enough to notice a difference (if we have the ears and transducers that can keep up after 16khz). and that difference would become clearly small with the use of 48khz and starting to filter a little higher.
> I'm not sure there is something else to hear?


 
  
 "Two kinds of linear-phase FIR filters...... For both FIR filters, the ripple depth over the passband was a maximum of 0.025 dB, and the stopband attenuation was 90 dB. The frequencies of the transition bands were 23500-24000 Hz and 21591-22050 Hz, corresponding to the standard sample rates of 48 kHz and 44.1 kHz respectively. These parameters were chosen to offer a reasonable match to the downsampling filters used in good-quality A/D converters or in the mastering process..."
  
 These are quite sharp filters....
  
 Note that they presented ALL the samples themselves at 24/192k - which should effectively eliminate any "difference" contribution by the DAC being used. (The samples were filtered to limit them to the same bandwidth as if they had been filtered for recording onto a CD, but, by them leaving the samples at 192k, both the original 24/192k sample and the "filtered CD and DVD samples" could then be played on the same DAC, under the exact same conditions, and any contributions to the sound by the DAC would be the same for both - so the only difference would be the filtering applied.)
  
 They named the speakers used, which were "digital speakers" with internal DACs, and were rated to be "flat" to about 40 kHz - and included graphs and impulse response measurements for the system they used. (Thus proving that, if there were audible differences, they wouldn't be obscured by the electronics or speakers used for the test being unable to reproduce them. This is something that I've noted as being missing from every other test I've seen published - proof that the test system was able to reproduce, and deliver to the air in the listening room, the differences that we're supposedly testing...  and, to me, in order to make a valid test as to whether something is audible or not, you first have to demonstrate that your test equipment is in fact presenting it to your subjects to be - potentially - heard. If your test equipment isn't up to the task, and so obscures the difference whose audibility you're trying to test, then of course nobody is going to hear it - but it doesn't prove anything. Likewise, if the test samples you're using have no content above 20 kHz, then you can't expect your test subjects to even potentially notice whether any of the content that wasn't there to begin with is audibly missing after being filtered out.)
  
 (Another thing I thought was interesting.... In the interviews, the sample clips that the subjects found to be the most noticeably different were those containing echoes and reverberance - with subjects claiming that the sound character of the reverberant tails in those segments was altered by the filters.... which happens to agree with a lot of the "anecdotal" claims I've heard put forth by various audiophiles about differences in digital audio content.)
 


----------



## RRod

Except that you can play a 20kHz tone on a system that CAN reproduce it, and most people still won't be able to hear it, and those that can won't be able to at anything other than a roar, so these things happening due to frequencies higher than that can't be due to single frequencies, and thus questions about things like IMD come up. For filters that flat and sharp (♯♭?), the ringing should also be up in the 20kHz area, so now we're saying that people are hearing differences at these frequencies in reverb tails, where the amplitude should be relatively low. Just none of it adds up, and the thread over at hydrogen doesn't make me feel less uneasy.


----------



## Thad-E-Ginathom

rrod said:


> I was thinking of the online "keys" test that also ended up having problems. I see the paper you bought is the one by Meridian; it's been a while since I read the discussion on hydrogen on that one. Guess I should go back.**


 
  
 Oh god, no... once was enough. More than enough, even.
  


> *The take home, given by the admin when he closed the thread:
> "I think it was well established in this thread that the authors of this paper (intentionally?) used inappropriate practices to push Meridian's agenda to sell expensive gear and their new format MQA."


 
  
 Of course, there are audio forums where such promotion is not only acceptable, but normal (I'm not pointing the finger, or at least, not _one_ finger: I can think of several).


----------



## sonitus mirus

Are convention papers even peer reviewed?  It should create suspicion that the authors of this paper all work for the same entity that stands to benefit financially from their supposed results.


----------



## KeithEmo

rrod said:


> Except that you can play a 20kHz tone on a system that CAN reproduce it, and most people still won't be able to hear it, and those that can won't be able to at anything other than a roar, so these things happening due to frequencies higher than that can't be due to single frequencies, and thus questions about things like IMD come up. For filters that flat and sharp (♯♭?), the ringing should also be up in the 20kHz area, so now we're saying that people are hearing differences at these frequencies in reverb tails, where the amplitude should be relatively low. Just none of it adds up, and the thread over at hydrogen doesn't make me feel less uneasy.


 
  
 The problem there is that you're expecting a very specific correlation - either you hear the frequency or not. However, the way our brains perceive sound, and how we react to it, isn't always that simple. To pick a silly example (that might turn out not to be entirely silly in the end), what if, when I played the sound of a violin, with a bandwidth of 20 kHz and a bandwidth of 40 kHz, it sounded exactly the same to you, but the one with the 20 kHz filter sounded like it was six inches further to the left? (That could happen if the part of your brain that "listens to music" couldn't "hear" the 30 kHz overtones, but the part that processes cues for determining source location was affected by them. Since we do know that different information is processed by different parts of our brains, this may seem unlikely, but it isn't impossible.)
  
 And here's another interesting thought. Several people have suggested that extra inaudible ultrasonic content might become audible if it interacted in the amplifier or speaker to produce IM products which are audible. Now, what if ultrasonic content that is encoded properly on the recording, and played faithfully by the DAC and other playback equipment, results in IM distortion in your ears, and that IM product is used somehow by our brains - and so is audible by its absence when it's missing? (If we are used to hearing the sound of a cymbal being modulated by the reaction of our ears to high-level ultrasonic transients that are present, then cymbals heard without that modulation will sound wrong - even though the CAUSE of the modulation may in fact be totally inaudible.) I'm not specifically suggesting that this happens; merely offering it as one of many possibilities that cannot reasonably be ruled out without further testing.


----------



## RRod

Frequency hearing limits have been tested. Your violin example would be easy enough to test, but I don't see why some kind of test signal wouldn't be better for the task. But until it's done it's just possibilities. I'll add that the literature I've read on binaural technology doesn't worry a whit about anything over about 20kHz for positioning.
  
 Ear-induced IM is of course a possibility, but having tested tones from 20 to 30kHz (which both my E-MU and HD800 should be good for) with 1kHz differences, I've never heard any magic 1kHz or otherwise IMD product pop out. That proves nothing, of course, but again it's something that would seem easy enough to eek out with test signals, instead of positing things we hear with cymbals.
 You mentioned the import of having a system flat enough to make all these frequencies manifest, but there are tons, TONS, of people claiming to hear these exact same phenomena (reverb differences, ringing, schmear) on ear-buds and any given hi-res DAC. So I have a hard time accepting "agreement with audiophile anecdotal claims" (my scare quotes) as necessarily a good thing


----------



## OddE

keithemo said:


> (...) what if, when I played the sound of a violin, with a bandwidth of 20 kHz and a bandwidth of 40 kHz, it sounded exactly the same to you, but the one with the 20 kHz filter sounded like it was six inches further to the left? (That could happen if the part of your brain that "listens to music" couldn't "hear" the 30 kHz overtones, but the part that processes cues for determining source location was affected by them. Since we do know that different information is processed by different parts of our brains, this may seem unlikely, but it isn't impossible.)


 
   
 -Wouldn't this thought experiment be torpedoed by the cilia which translates the vibrations captured by our ear into signals for the brain to interpret? That is, at 30kHz they simply do not pick up anything to forward to the brain?
  
 Quote:


keithemo said:


> And here's another interesting thought. Several people have suggested that extra inaudible ultrasonic content might become audible if it interacted in the amplifier or speaker to produce IM products which are audible. Now, what if ultrasonic content that is encoded properly on the recording, and played faithfully by the DAC and other playback equipment, results in IM distortion in your ears, and that IM product is used somehow by our brains - and so is audible by its absence when it's missing? (


 
  
 -IM is a non-linear phenomenon. It is not generated in our ears. (To the best of my knowledge; I'd be happy for someone who knows more about the human ear than I do to chime in...)


----------



## KeithEmo

rrod said:


> I was thinking of the online "keys" test that also ended up having problems. I see the paper you bought is the one by Meridian; it's been a while since I read the discussion on hydrogen on that one. Guess I should go back.**
> 
> *The take home, given by the admin when he closed the thread:
> "I think it was well established in this thread that the authors of this paper (intentionally?) used inappropriate practices to push Meridian's agenda to sell expensive gear and their new format MQA."


 
  
 I'd have to say that, while I certainly can't speak to the honestly of the authors of the paper, the practices they followed seemed to me to be reasonable. (While they did in fact do a few things differently than I would have, and I would have liked to see a much bigger sample population, they also corrected several things that I considered to be obvious issues with the previous papers on the subject.)
  
 If you're going to test whether differences above 20 kHz are audible, then you must first validate your test procedure by demonstrating that your test equipment is capable of faithfully reproducing the frequencies whose audibility you're testing. This was done in the recent AES paper - and it was NOT done in the previous papers claiming to show the opposite (at least not in the ones I've read). Saying that a test proves that something is inaudible, while not first showing that your test setup was even capable of presenting it to be potentially heard, is a textbook formula for an invalid test.... At least these guys got that part right.
  
 And, as for peer review, or improvements on the test.....  I guess we'll have to wait until someone else does it better.
 I would also suggest that anyone wishing to "debunk" their results can do so quite easily..... by performing an equally valid, or preferably better, test that shows them to be wrong.


----------



## Thad-E-Ginathom

A person can be 100% reasonable, sincere and even honest --- and still be wrong.  Surely, most of us, in one way or another have been there.


----------



## RRod

keithemo said:


> I'd have to say that, while I certainly can't speak to the honestly of the authors of the paper, the practices they followed seemed to me to be reasonable. (While they did in fact do a few things differently than I would have, and I would have liked to see a much bigger sample population, they also corrected several things that I considered to be obvious issues with the previous papers on the subject.)
> 
> If you're going to test whether differences above 20 kHz are audible, then you must first validate your test procedure by demonstrating that your test equipment is capable of faithfully reproducing the frequencies whose audibility you're testing. This was done in the recent AES paper - and it was NOT done in the previous papers claiming to show the opposite (at least not in the ones I've read). Saying that a test proves that something is inaudible, while not first showing that your test setup was even capable of presenting it to be potentially heard, is a textbook formula for an invalid test.... At least these guys got that part right.
> 
> ...


 
  
 Yes, for the best test you would want flatness into the hi-res frequencies throughout the chain, though 24-bits of actual dynamic range might be a tall task. I'm fine with people not being convinced by, say, Meyer and Moran for the audibility of hi-res. That doesn't mean that useful things didn't come out of it, namely testing the assertion that people were hearing benefits from SACD/DVD-A on equipment that people actually use. As I said before, people all across this board and others are claiming audibility on the gamut of kit, much of which I'm sure isn't delivering hi-res content faithfully.
  
 In the end, though, a non-reviewed paper from people with a definite stake in the game probably isn't where a hat should be hung. Of course, it would be nice if people on the other side actually made their own attempt at a new test. Perhaps they feel the case is already closed.


----------



## El Zilcho

keithemo said:


>


 
  
  
 You say that the test did use very few test subjects, but that the results above 50% is statistically better than random.  But random will have groups and clusters and will not always be a perfect representation of true statistical probability in small samples.  I could flip a coin ten times and get heads nine times.  Doesn't mean I'm more likely to get heads, that's just "random" in action.
  
 In fact, randomness itself will make tests with few samples go one way or another.  You need large sample sizes to filter out the random, and show true statistics.  Maybe I flip that coin 20 times I would get heads 15 times.  Maybe I flip it 50 times and get heads 30 times.  Maybe I flip it 2000 times and get heads 1043 times.  That tells me the random is getting filtered out and a better picture of a 50/50 chance is showing through.  If I flip it 2000 times and get heads 1900 times, that would tell me something is affecting the odds, and there must not be a true 50/50 chance.  I guess what I'm getting at is, if there was no audible difference and people were "randomly" choosing, there is a 50/50 chance of hitting one song or the other.  In that case, it could take a lot of samples to show the true statistics, and randomness could account for a percentage above 50 if the sample size was small.
  
 I realize you did say that more samples would be better.  I also concede that the results could show a slight bias towards the ability to hear a difference.  Just some thoughts I couldn't help thinking as I read through the posts. 
  
 Edit:  I don't know why quoting isn't working.  I couldn't even get copy/paste to work.  Stupid Edge browser.


----------



## KeithEmo

el zilcho said:


> You say that the test did use very few test subjects, but that the results above 50% is statistically better than random.  But random will have groups and clusters and will not always be a perfect representation of true statistical probability in small samples.  I could flip a coin ten times and get heads nine times.  Doesn't mean I'm more likely to get heads, that's just "random" in action.
> 
> In fact, randomness itself will make tests with few samples go one way or another.  You need large sample sizes to filter out the random, and show true statistics.  Maybe I flip that coin 20 times I would get heads 15 times.  Maybe I flip it 50 times and get heads 30 times.  Maybe I flip it 2000 times and get heads 1043 times.  That tells me the random is getting filtered out and a better picture of a 50/50 chance is showing through.  If I flip it 2000 times and get heads 1900 times, that would tell me something is affecting the odds, and there must not be a true 50/50 chance.  I guess what I'm getting at is, if there was no audible difference and people were "randomly" choosing, there is a 50/50 chance of hitting one song or the other.  In that case, it could take a lot of samples to show the true statistics, and randomness could account for a percentage above 50 if the sample size was small.
> 
> ...


 
  
 As for the number of samples - and stating whether a given result "is random or not" - a lot of that falls into the "magical language of statistics". Statistical analysis of results falls into "probabilities" and "probabilities _OF_ probabilities"... which basically means that they have math that they apply _TO_ the math. The simple reality is that most of us mere humans don't understand statistics. The actual statistical "prediction" of what happens when you flip a "fair coin" isn't that it will come up 50/50 heads and tails. Rather, they say that, if you flip a fair coin ten times, then repeat that entire test ten times, there is a 3 in 10 chance it will come up 5/5, and a 6 in 10 chance that it will come up somewhere between 4/6 and 6/4, and a 9 in 10 chance that it will come up somewhere between 3/7 and 7/3. So it tends to come up 5/5, and, as you consider results different than that, the further away from 5/5 you look, the less likely it will occur.
  
 This is those statistical certainty numbers that we usually refer to as "statistical significance".
  
 So, if you flip a coin ten times, and it comes up 3/7, you probably shouldn't be surprised. But, if you repeat that whole test ten times, and it comes up 3/7 on half of them, then there's reason to suspect that something's wrong (because the odds against _THAT_ happening may be 50 to 1 against). And, if you repeat it 1000 times, and your coin comes up heads 7 out of 10 times half of the time, then it's probably a trick coin.
 So, for example, if one of our ten subjects guesses correctly 29 times out of 47 trials, they will say that the expected result is 23/24 or 24/23, that the result they got is xx% away from that, and that the likelihood of that difference being due to random chance is some other percentage. 
  
 In the case of the test described in that AES paper, I believe they used 8 or ten subjects, and each listened to a few hundred samples of music; the results showed that, on average, they were correct between 55% and 65% of the time, which was judged to be relatively unlikely to occur by random chance. (The statistical numbers describing how likely this was to occur by random were quoted.)
  
 In general, in the real world, when "testing on a tight budget", you usually see very few subjects, and a small but not unreasonable number of test runs (perhaps 10 or 20 subjects listened to 100 or 200 music samples each). This is usually determined simply by how many volunteers you can get into one room at the same time, and how long you can convince them to participate before they get bored or tired and wander off. Any more than that and you need a budget to pay people (or, as in a drug test, people who have their own motivation to participate).  
   
And, a final comment to those among us who seem certain that "the fix is in" and the few tests that seem to suggest that the difference is audible "were conducted by folks with an ulterior motive - or at least a bias", I would remind them that most of the original tests which showed that "CDs were faithful to the original within the limits of audibility" were also conducted by people with a bias. (Most of the folks who ran the earliest tests had the "agenda" of proving that CDs were "audibly perfect" - so they could convince audiophiles to replace all their vinyl with CDs without worrying about their CDs becoming obsolete in a few years. Therefore, they had very little incentive to exhaustively verify that there were no differences that were _ever_ audible - under _any_ conditions, using _any_ test samples, and using _any_ test equipment. Rather, their agenda, paid for by CD backers like Sony, was to prove that CDs were "good enough that nobody would ever need anything better".)

  
 And, since today there are several companies who are making good money based on the _assumption_ that high-res files are better, and nobody I can think of who stands to make money if that were to be proven wrong, we can all guess who's likely to be sponsoring paid testing any time soon. (And, for anyone who suggests that we can do some sort of "crowd sourced test" - where thousands of people can download some test files and report their results - and one or two honest folks can tabulate the results - I ask..... How are we going to run the test to make sure that nobody cheats? How will we know that half of the responses aren't really sent in by "bots" written by someone with an agenda, and that a few participants won't cheat just to convince everyone else to believe what they "know" to be true? Ensuring that the test really is fair, and preventing this sort of thing, is one of those factors that makes reliable legitimate testing expensive and labor-intensive.)
  
 And I'll leave this post with a final thought......
  
 Perhaps someone should think about starting up a "campaign" on something like KickStarter - to collect enough donations to sponsor a real test, with enough participants to be meaningful, and test equipment proven to be up to the task, under fair and properly controlled conditions. (Perhaps they could set up a room at the next audio show, where volunteers could try fairly to hear the difference that may or not be there, and offer a prize of an expensive piece of audio equipment as incentive for each participant to try to prove that they do hear a difference. Of course, the test conditions, and the equipment used, would have to be decided upon and agreed to be a _FAIR_ application of the test....  which tends to be easier to ensure when the funding is provided by an impartial group whose only presumed goal is to find out the truth. I might suggest having a panel of real audiophiles set out the test requirements, so there's no question about what we're trying to prove, but having the actual tests run by a separate panel of non-audiophile scientists, paid simply for accurate results regardless of the outcome.)
  
 I know that I personally would be interested in _BOTH_ whether there is even the slightest difference which is audible under very specific conditions, _AND_ whether, in practical terms, if such a difference is proven to exist, it would actually matter to most people or not. To that end, with sufficient funding, the test could be extended to both "any equipment" and "typical home equipment".


----------



## KeithEmo

rrod said:


> Frequency hearing limits have been tested. Your violin example would be easy enough to test, but I don't see why some kind of test signal wouldn't be better for the task. But until it's done it's just possibilities. I'll add that the literature I've read on binaural technology doesn't worry a whit about anything over about 20kHz for positioning.
> 
> Ear-induced IM is of course a possibility, but having tested tones from 20 to 30kHz (which both my E-MU and HD800 should be good for) with 1kHz differences, I've never heard any magic 1kHz or otherwise IMD product pop out. That proves nothing, of course, but again it's something that would seem easy enough to eek out with test signals, instead of positing things we hear with cymbals.
> You mentioned the import of having a system flat enough to make all these frequencies manifest, but there are tons, TONS, of people claiming to hear these exact same phenomena (reverb differences, ringing, schmear) on ear-buds and any given hi-res DAC. So I have a hard time accepting "agreement with audiophile anecdotal claims" (my scare quotes) as necessarily a good thing


 
  
 I agree with you entirely.
  
 My comment about "anecdotal claims" was simply that the specific _types_ of differences which subjects in the test claimed to have heard agreed with the _types_ of differences that many audiophiles claim to hear. As you say, this may mean nothing, but I consider it to be at least interesting - and perhaps "slightly suggestive".
  
 Also note that I'm not at all discounting the possibility that a lot of the people who claim to hear differences really are imagining them. Considering how suggestible we humans are, I would be surprised if a lot of people didn't imagine they hear what they expect to hear. However, that doesn't really matter in terms of the overall result - except that you have to set up your test in such a way that you "control out" that factor. (In a controlled test, some people will experience headache relief with a placebo sugar pill, but significantly more will experience relief with aspirin. Those results show both that the placebo effect is real, and that aspirin is more effective than a placebo, and those results are in no way contradictory. However, if you're testing aspirin, if you want a valid result, you'd better compare the results with aspirin with those with a placebo, to determine the _difference_ between them.)
  
 My personal prediction would in fact be that, under some certain conditions, with certain test samples, and certain playback equipment, some people can hear a difference beyond random chance - but that, for most test samples, and under most conditions, most people only imagine they do. To me, this result would be plenty of justification for offering high-res versions of albums for sale to those who hear - or think they hear - a difference.... but would probably result in many individuals deciding that they don't feel the necessity to pay extra for them, and a good percentage agonizing about whether they "really" hear the difference or not (which result would annoy just about everyone).


----------



## RRod

keithemo said:


> I agree with you entirely.
> 
> My comment about "anecdotal claims" was simply that the specific _types_ of differences which subjects in the test claimed to have heard agreed with the _types_ of differences that many audiophiles claim to hear. As you say, this may mean nothing, but I consider it to be at least interesting - and perhaps "slightly suggestive".
> 
> ...


 
  
 Well I think the point of something like ABX is to weed out the imagination. But that's just to make things official; the hi-res industry is free to run on a imagination scheme if they want. I mean, there's the whole nutritional supplement industry that exists.
  
 I can't think of a corresponding concept to placebo in a sample-spec ABX*, since there really isn't an effect being measured. Can you elaborate on what you mean by controlling-out imagination in the ABX context?
  
 *Hydrogenaud.io seems to run with this definition.


----------



## KeithEmo

odde said:


> -IM is a non-linear phenomenon. It is not generated in our ears. (To the best of my knowledge; I'd be happy for someone who knows more about the human ear than I do to chime in...)


 
  
 Our ears are not linear. At the least, they include a very powerful "automatic gain control" effect - which is what gives our hearing its huge dynamic range. This is why you can easily hear a coin drop in a quiet room at midnight, but you can't hear that same coin drop in the lobby after exiting a loud rock concert - because the gain of your ears has adjusted itself based on the current audio input level - which is another name for "modulation". (Even air is subject to some - small - modulation effects.)


----------



## El Zilcho

I appreciate the additional info in your reply Keith.
  
 Honestly, if they had each subject listened to hundreds of samples, that's a lot more than I imagined when you said there were few samples.
  
 I realize that my example of statistics using the coin flip was very simplistic, but I guess I'm just a natural skeptic, and when I pictured eight or ten people listening to ten different samples, I couldn't help but think that a result of 55-65 percent wasn't exactly definitive.
  
 I also realize that conclusions are often drawn from small samples for practicality reasons (which limit larger sample sizes), and they simply base it on what is by far the most likely answer for the results.  
  
 I would also would be interested to know what the individuals scored on the tests.  If the average was 55-65 percent, I would be interested to know if anyone scored 75 percent or better.  It's possible that due to variations in hearing between individuals, some are more able to hear differences.  Also, if they were simply asked to pick which one they thought was the "high res" track, someone getting a very low result (25 percent or less) could suggest that they were in fact hearing a difference, but simply preferring the lower rate file for one reason or another, and therefore assuming it was the higher one.
  
 Interesting stuff either way.


----------



## KeithEmo

rrod said:


> Well I think the point of something like ABX is to weed out the imagination. But that's just to make things official; the hi-res industry is free to run on a imagination scheme if they want. I mean, there's the whole nutritional supplement industry that exists.
> 
> I can't think of a corresponding concept to placebo in a sample-spec ABX*, since there really isn't an effect being measured. Can you elaborate on what you mean by controlling-out imagination in the ABX context?
> 
> *Hydrogenaud.io seems to run with this definition.


 
  
 Actually, the whole point of ABX testing _IS_ to "control out" expectation and any related "placebo effects"... and it should work pretty well at doing so.
  
 The only real area where ABX testing falls short is that it may actually suppress reporting of legitimate differences... which seems to be the claim that most of those who oppose it make. The problem with ABX testing is that it specifically asks the subject to identify similarities - rather than differences - and it does so in a specific way which is based on at least some expectation of "accurate acoustic memory".
  
 1)
  
 We humans are usually better at recognizing differences than similarities. In a different context, let's assume I choose to do an ABX test to determine how well you can recognize color differences, and I do it by the simple expedient of holding up 12" square tiles in various colors. I hold up Tile A, then I hold up Tile B, then I hold up Tile X, and ask you to tell me which of the first two tiles Tile X most closely matches - and, from this, I can get a number for exactly how small a color difference you can detect. However, if, instead of doing it that way, I hold up Tile A and Tile X next to each other - at the same time - and ask you if they are the same, then I repeat this with Tile B and Tile X, we will find that you are able to determine differences that are _MUCH_ smaller (we're talking several full orders or magnitude). The simple fact is that, when it comes to colors, or shades of grey, humans are _much_ more sensitive to differences in samples presented side by side than to differences in samples that are presented sequentially. (Presumably this is because our ability to do "simultaneous comparisons of visual stimuli" is much better than our ability to do so sequentially.) In other words, if you want to test the ability of humans to detect minor differences in colors, ABX testing is _NOT_ the most sensitive way to do so, so it does not give you the most accurate results.
  
 ABX tests test our ability to match similarities in sounds - but that's not exactly the same thing as "distinguishing differences". In fact it's a somewhat more complex task. If you actually want to test whether someone can hear a difference between two signals or not, then the most direct way to do so, with the least extra complications, is to actually present two signals which may or may not be different and ask the test subject whether they're different or not. Since our ears are different, which precludes simply playing the two signals simultaneously in different ears, I can't think of a perfect analogue of simply displaying two color tiles side by side, but there is a close equivalent. Simply play PAIRS of audio samples, and ask the test subject whether they are the same or not. Present the subject with simple random pairs of test samples, each of which can be AA, AB, BA, or BB, and ask the subject whether the two samples in each pair are the same or different (allow the person to replay each pair as many times as they like before answering). Because this would both minimize any requirement of "acoustic memory", and also minimize the "additional thought processing involved", I suspect it would be much more sensitive than the ABX methodology. (In each sample pair there should be a "tick" added between the samples, to avoid any possibility of a discontinuity between the samples providing a "cue" that they're different.)
  
 2)
  
 (this really relates to the first point).... Humans tend to be uncomfortable when asked to report about anything they're unsure of. There have been reports that, with various ABX tests, the subject feels pressure to experience "well defined" results. In other words, if they really don't hear a difference, they experience stress when asked to "pick one" and, if they're told to "just make your best guess", they tend to either overthink the response or refuse to answer. Both of these can tend to obscure "low level responses". For one thing, if they don't consciously notice a clear and distinct difference, they may "mentally flip a coin", which is _NOT_ the same as guessing - because guessing may be influenced by a slight unconscious preference. (In other words, they may have consciously decided that there is no difference, then may consciously decide what to "guess"... which is not the same as a true random response.) And, obviously, having a subject who is uncomfortable doing the test presents all sorts of problems with their responses (they may consciously "make up answers" rather than give their best answer).
  
  
 Another, and more basic issue, which is a problem with both ABX testing, and my proposed alternative, is that both are testing for "consciously experienced short term responses". To state a whole range of possibilities informally.....
  
 What if Sample A gives me a headache after listening to it for a half hour but Sample B does not?
 What if Sample A raises my Serotonin level, or lowers my blood pressure, more than Sample B?
 What if Sample A makes me feel more alert, or Sample B makes me sleepy faster?
  
 And several of these relate to the second point above. Most people (at least most audiophiles), if presented with two seemingly identical test samples, will tend to ignore minor and seemingly unrelated differences. Subjects of drug tests tend to both report unrelated symptoms and side effects, and under-report side effects that they don't perceive as being related. Just as many audiophiles tend to report things like "listener fatigue" which may well be imaginary, many non-audiophiles would probably fail to report it if it were to occur, because they wouldn't consider it to be related to what they were listening to. It would be quite difficult to design a test that would eliminate both of those possibilities (and you can't simply ignore legitimate possibilities because they're difficult to test).


----------



## RRod

A couple of comments (not gonna try to quote that one 
  
  
 .I'm trying to think what simply playing a difference file for someone would miss out on. Possibly IMD between the hi-res and Redbook components, but seems like not much else. Another thing one could try is to play a fixed sample in one ear, and switch between the two samples in the other ear (maybe inverting those files). Maybe I'll play around with this. Still, the color example is nice but comparisons between senses don't always work. Looking at the two cups on my desk is not the same as hearing two sounds coming into my right ear.
  
 .There is no time limit to the ABX protocol that I know of, so I don't see why it wouldn't be up to the task. And control almost always gets harder as time increases (consider actually testing your sleepiness example).


----------



## sonitus mirus

keithemo said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
  
 While perhaps not practical, this is not an issue with an ABX test, even using the popular ABX tools provided as a plugin for the free Foobar2000 music player.  There is nothing preventing you from making large enough files to accomplish exactly what you are describing.  I think you would have a difficult time establishing that a lower quality version is actually resulting in giving you headache.


----------



## KeithEmo

sonitus mirus said:


> While perhaps not practical, this is not an issue with an ABX test, even using the popular ABX tools provided as a plugin for the free Foobar2000 music player.  There is nothing preventing you from making large enough files to accomplish exactly what you are describing.  I think you would have a difficult time establishing that a lower quality version is actually resulting in giving you headache.


 
  
 I agree - but that is one argument in favor of "non-self scored testing".
  
 In other words, rather than to judge the test based on what people reported they experienced, it might be better to split them into two groups, let one group listen to the high-res sample in a room for as long as they like, let the other group listen to the low-res sample in another room, also for as long as they like, then see if either group - on average - abandoned the room much sooner than the other. Another way to do it would be to take one group of people, and offer them a choice of two waiting room, with a different sample playing in each, but otherwise identical - and observe whether, over time, the vast majority seemed to prefer one room over the other - and an easy metric there would be to simply count how many "person hours" were spent in each room. If it turned out that the total number of person-hours spent in one room was very different than the other, and the rooms were otherwise identical, then one could conclude that the music had somehow influenced the participants - consciously or not. This is the type of test used to test other "ambient factors" - like how the color of walls affects people... you present the stimulus and observe directly how it affects behavior - and avoid the complicating factor of asking the participants to score the results themselves.
  
 This is pretty much the same, and for the same reasons, why some people will test which soda flavor is preferred by asking a bunch of people which one they liked better, while others insist that the results are more accurate if, instead of asking the people their choice, they simply leave five bottles of each in a room with a crowd of people, let each drink whatever they want, and see which five bottles get emptied first. The presumption being that, because subconscious factors may be involved, the subjects may in fact drink more of the flavor that they consciously don't prefer..... (and, to a soda company, they're more interested in which one you drink than in which one you SAY you like).
  
 In our case, it might turn out that, even though the participants can't hear a difference consciously, they find one or the other "more enjoyable" or "less annoying". There is also the potential that the results just might vary with things like sound level and the type of sample used. (For example... people might find high resolution classical music more soothing, but find high-resolution rap music more annoying... and that balance might be different at different listening levels.)
  
 Again, though, my overall point is that I don't think even showing that a certain sample of people say they hear no difference proves that no audible difference exists.....
  
 To go back to my original example - most prisons and similar institutions have their walls painted certain colors - because tests have supposedly established that those colors are more conducive to "positive moods and tranquility" while other colors tend to result in "more conflict". I'm pretty sure those results were obtained by observing populations in rooms painted various colors over time - and NOT by asking people whether they felt calmer in the red room or the green room after fifteen minutes. (Because, in that situation, the actual behavior of the participants is a much more accurate indicator than their conscious perceptions.)
  
 Now, a REALLY interesting test would be to monitor the actual pleasure centers in the brains of the participants (using an FMRI).
 I would then compare the neurological responses of participants who were listening to high-res and low-res files, but include some of each who had been informed, both correctly and incorrectly, which they were listening to. Wouldn't it be interesting if it turned out that the people who THOUGHT they were listening to high-res files enjoyed them more, but it didn't correlate with what the files really were at all.


----------



## castleofargh

we all know that abx isn't perfect, how can anything depending on human senses while self administrated, hope to be perfect? even masturbation can always be improved.
 but while you're explaining how some things may elude an abx test, we have a vast majority of people who are misinformed about sighted evaluation and read, and believe
	

	
	
		
		

		
		
	


	




, reviews from guys who claim to be beyond bias. "oh I'm experienced enough to blablablah" kind of nonsense even from professionals.
 to me this is the real problem in audio communities. not that a few guys out of the minority who uses ABX, sometimes think too highly of the conclusion of a test.
  each time someone write about ABX not being a panacea, you have 10 dudes who misunderstand it for "I should keep doing no control at all and trust myself because it's better".
  
  
 let's first take care of the ship sinking, after that we'll argue about the best way navigate .


----------



## old tech

All this discussion on ABX, Moran and Myer, later experiments etc is all good and well, but I can't get my head around the first order givens and hence why the burden of proof should not fall squarely on those making claims that high res audio sounds better.
  
 If 24bit compared to 16bit only offers an increase in dynamic range which is not of any practical use, and lowers the noise floor from an already very low and masked by any music content, why should there be any theoretical advantage in reproducing sound?
  
 Secondly, a 100years of testing have proved that the hearing range of humans is within a 20hz to 20khz band.  So why should there be any theoretical advantage in extending the range for music playback?  I understand the arguments, none of them convincing or proved since digital oversampling, that a steep cut-off may result in errors below the threshold.  Surely if that ever was an issue, that would have been resolved in the early 80s?  I know of no peer review standard papers proving otherwise?
  
 Lastly, if we look at progress in hi res video technology, eg OLED displays etc, the high res bit depth here makes a difference due to smaller pixellation, hence sharper pictures.  This is not the same as audio where higher bit depth does not increase the accuracy of the sound wave in the band limited frequency.
  
 As an analogy with 44.1 vs 96 or 124 debate, what would everyone think if TV manufacturers started marketing sets claiming higher quality pictures becuase they increased the frequency bandwidth deep into infra red or ultra violet?  It appears only in audiophile land that people think such claims are credible.


----------



## frodeni

Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
  
 Given that I can pinpoint the sound trailing a plane high in the sky, a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that. That again implies that the hearing of humans is more accurate than 16/44.1.


----------



## cjl

frodeni said:


> Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
> 
> Given that I can pinpoint the sound trailing a plane high in the sky, a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that. That again implies that the hearing of humans is more accurate than 16/44.1.


 

 Care to share that math? I too have done the math, and 16/44.1 seems more than sufficient based on my calculations. As for timing, 16/44.1 has timing accuracy far, far better than just one sample. I've seen very credible evidence showing that its timing accuracy is significantly into the nanoseconds, despite only having one sample every ~23 microseconds or so.


----------



## castleofargh

old tech said:


> All this discussion on ABX, Moran and Myer, later experiments etc is all good and well, but I can't get my head around the first order givens and hence why the burden of proof should not fall squarely on those making claims that high res audio sounds better.
> 
> If 24bit compared to 16bit only offers an increase in dynamic range which is not of any practical use, and lowers the noise floor from an already very low and masked by any music content, why should there be any theoretical advantage in reproducing sound?
> 
> ...


 
 also screens are on a few parameters still inferior to human eye's thresholds. getting better luminosity range, or bigger gamut, those are still clearly visible compared to screens with lower contrast or lower gamut(well as long as the picture uses a rich color profile and dynamic of course).
 on the other hand, for most songs we could just cut to maybe 10bit and 25hz-17khz and the average guy would already have a real hard time noticing a change.
  


frodeni said:


> Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
> 
> Given that I can pinpoint the sound trailing a plane high in the sky, a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that. That again implies that the hearing of humans is more accurate than 16/44.1.


 
 I love my mum, rain is wet and it's proven by science so 16/44 is clearly enough.
 there I made a demonstration just as good and rational as yours.
  
  
 why do I feel more and more like you're not at all new here but just an old "friend" using a smurf account?


----------



## goodyfresh

castleofargh said:


> also screens are on a few parameters still inferior to human eye's thresholds. getting better luminosity range, or bigger gamut, those are still clearly visible compared to screens with lower contrast or lower gamut(well as long as the picture uses a rich color profile and dynamic of course).
> on the other hand, for most songs we could just cut to *maybe 10bit* and 25hz-17khz and the average guy would already have a real hard time noticing a change.
> 
> I love my mum, rain is wet and it's proven by science so 16/44 is clearly enough.
> ...


 
 Well yeah, since the vast majority of music out there has no more than 20dB dynamic range, and a pretty big majority has as little as 10 to 15.  Meanwhile, I myself cant' hear anything above 16.5khz, so. . .there's that.

 I've often made that argument before, the one about how silly it would be for monitor manufacturers to advertise that their monitors can do Infrared and Ultraviolet.  Lmao.  It's pretty funny if you think about it.  If we were making televisions and computers for Mantis Shrimps (they have the most awesome vision in the world, look them up if you want to have your mind blown), but we aren't, we're making them for us humans, so such things would just be superfluous and add to R&D costs.  Not to mention give us cancer or some crap when people start making a big deal about how stuff looks "so much better" when it can produce gamma-rays, which is the essential equivalent of people talking about moving up to 384khz sample-rates in audio and everyone raving about their current 192khz rates is like if fokls wanted TV's that produce X-rays


----------



## old tech

frodeni said:


> Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
> 
> Given that I can pinpoint the sound trailing a plane high in the sky, a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that. That again implies that the hearing of humans is more accurate than 16/44.1.


 
 I too wouldn't mind seeing the maths or credible evidence for that.  If it was the case, we would have had half a century of similar complaints with vinyl and other analogue media, as it is technically inferior to 16 bits (even the best 24 track analouge masters were only about 13 bits).


----------



## frodeni

cjl said:


> Care to share that math? I too have done the math, and 16/44.1 seems more than sufficient based on my calculations. As for timing, 16/44.1 has timing accuracy far, far better than just one sample. I've seen very credible evidence showing that its timing accuracy is significantly into the nanoseconds, despite only having one sample every ~23 microseconds or so.


 
  
 That is actually fair to ask, but in this environment in here, I will not be the first one to share. It feels plain wrong. I actually had written a full post about it, but then old tech entered his post, and all my willingness to share in this thread, sort of disappeared.
  


old tech said:


> I too wouldn't mind seeing the maths or credible evidence for that.  If it was the case, we would have had half a century of similar complaints with vinyl and other analogue media, as it is technically inferior to 16 bits (even the best 24 track analouge masters were only about 13 bits).


 
  
 Yeah. This one. This prohibits a culture of sharing, and most people posting, behave the same way. I sort of had enough by now.
  
 cjl, I do not have a lot of math, as that is lost years ago. My understanding of things has grown by the years as well, and the simple math I just did, seems rather to open doors than to slam them. But this is not my field of interest any longer, and the math is now lost on me. If you got something to share at hand, please do. In here or by PM. I ask as if you do have some more proper math than the Pytagoras I just did, with no differential or vector math at all, it will be easier for me to explain the point to the people doing this at the university. They might be working it as it is, but if not, I simply want to give them a heads up.
  
 I got full access to a few scientific databases, so if you have a reference to a paper on those timing accuracy numbers of yours, these university people would love that.
  
 My point is that the time is now ripe. In particular the WR side of things.
  
 For those of you who do not get the reference I made up to describe this: The listeners head is scanned physically. The distance the sound travel is then known, and it is pure vector physics to calculate the difference form the source to each of the listeners ears. Since the distance is known in space vectors, both time delay and sound level can be calculated pretty accurately. This can then be fed into a closed set of headphones. If you still do not get it, sound can be reproduced virtually around you, in all directions. A motion sensor, can even keep the illusion that they are placed physically in the room, at a fixed position, even if you turn your head. The monster breathing down your neck, you will hear his roar in fixed postion as you turn around to deal with him. And yes, the monsters coming from under the floor, you will hear them down there, before the break trough the floor.
  
 This is the way the industry will go, and there will be no singing into any plastic scull. A transition for the music industry will be smooth as well, given that the tech is software based, and hardly require much new gear in any studio. Work practices remain as they are. Producing a classic stereo record, is pretty straight forward.
  
 So, cjl, please share if you have something that will help me explain this to applied math at UIO. If applied math is not working on this, they should.


----------



## Don Hills

frodeni said:


> ... I do not have a lot of math, as that is lost years ago.


 
  
 I'll keep it simple then: Achievable digitally sampled time resolution = 1 / ( 2 pi bandwidth number_of_levels). You should be able to find this formula and its derivation in the scientific databases you have access to. (I don't have access to your databases, so I can't do your homework for you.)
  
 There have been various figures suggested for the lowest perceivable time resolution, from 40 microseconds or so down to about 5 microseconds.
  
 For 44.1/16 at 0dBFS = 1 / (6.282 * 20000 * 65536) = 1.2^-10 seconds = 0.1 nanoseconds.
  
 Even if you go all the way to -80 dB with 16/44.1, you still get: 1 / (6.28 * 20000 * 8) = 1 microsecond. 
  
 Hopefully you can see that humble 16/44.1 is still more than good enough to provide time resolutions exceeding that of the human auditory system.
  
 If you have difficulty visualising this, I suggest you watch this video:
 https://www.youtube.com/watch?v=cIQ9IXSUzuM
 You ought to watch the whole thing, but if you skip ahead to 20:52 you'll see a signal delay being varied smoothly by less than one sample period.


----------



## frodeni

Thanks Don.
  
 I look into it when I find the time. Thanks for taking the time.
  
 I still hear nothing about using headphones for 3D, like for vector generated sound. Does anybody know of projects looking into that?


----------



## RRod

frodeni said:


> Thanks Don.
> 
> I look into it when I find the time. Thanks for taking the time.
> 
> I still hear nothing about using headphones for 3D, like for vector generated sound. Does anybody know of projects looking into that?


 
  
 https://www.openal.org/
  
 The library comes with a utility for formatting and applying HRTFs. Just the other day I started up a game in Linux and knew it had to be using OpenAL, since one of the first things I heard was someone behind me.


----------



## RRod

old tech said:


> All this discussion on ABX, Moran and Myer, later experiments etc is all good and well, but I can't get my head around the first order givens and hence why the burden of proof should not fall squarely on those making claims that high res audio sounds better.
> 
> If 24bit compared to 16bit only offers an increase in dynamic range which is not of any practical use, and lowers the noise floor from an already very low and masked by any music content, why should there be any theoretical advantage in reproducing sound?
> 
> ...


 
  
 The problem is that those who are convinced they hear differences in hi-res will often claim that there is some heretofore unmeasured or poorly understood human sense that is the cause. And of course this regularly comes with the notion that ABX can't possibly be used to judge differences based on these senses. This is how critiques of ABX tend to go anyway: "I sense a difference, I can't sense it in ABX, therefore ABX must be wrong."
  
 Myself, I find it not at all surprising that my ability to ABX a lower-res version of a track depends upon exactly the limits you mention above. If I decimate the track to below 2x my threshold of hearing (18k with the volume cranked, but more like 15k at normal listening volumes), or if I quantize the track down to a number of bits that cannot support its dynamic range, then I can actually pass the test. Now of course people will say that I am merely deliberately failing the test once the specs get higher, which is maybe true, maybe not. But when outcomes support theory, well that should mean something.


----------



## goodyfresh

rrod said:


> The problem is that those who are convinced they hear differences in hi-res will often claim that there is some heretofore unmeasured or poorly understood human sense that is the cause. And of course this regularly comes with the notion that ABX can't possibly be used to judge differences based on these senses. This is how critiques of ABX tend to go anyway: "I sense a difference, I can't sense it in ABX, therefore ABX must be wrong."
> 
> Myself, I find it not at all surprising that my ability to ABX a lower-res version of a track depends upon exactly the limits you mention above. If I decimate the track to below 2x my threshold of hearing (18k with the volume cranked, but more like 15k at normal listening volumes), or if I quantize the track down to a number of bits that cannot support its dynamic range, then I can actually pass the test. Now of course people will say that I am merely deliberately failing the test once the specs get higher, which is maybe true, maybe not. But when outcomes support theory, well that should mean something.


 

 People aren't going to listen to/believe you, man, no matter how logical you are being.  Haha.


----------



## RRod

goodyfresh said:


> People aren't going to listen to/believe you, man, no matter how logical you are being.  Haha.


 
  
 Well yeah. But we're not even allowed to vent science on any other sub-boards, let alone have serious discussions 
	

	
	
		
		

		
		
	


	




 So might as well let it out here.


----------



## goodyfresh

rrod said:


> Well yeah. But we're not even allowed to vent science on any other sub-boards, let alone have serious discussions
> 
> 
> 
> ...


 

 Lmao, yeah


----------



## cjl

frodeni said:


> That is actually fair to ask, but in this environment in here, I will not be the first one to share. It feels plain wrong. I actually had written a full post about it, but then old tech entered his post, and all my willingness to share in this thread, sort of disappeared.
> 
> 
> Yeah. This one. This prohibits a culture of sharing, and most people posting, behave the same way. I sort of had enough by now.
> ...


 
 So you have evidence, but you won't share because... it feels wrong? I really don't see anything wrong with that post you quoted either, as it is simply making the point that if the timing resolution and dynamic range of digital are insufficient, so too will be the timing resolution and dynamic range of analog, as digital is far superior in both respects.
  
 As for my math, Don has already demonstrated that 1 microsecond is an extremely pessimistic estimate for the time resolution of 16/44.1 digital audio. Let's think about what that means in the real world though. The speed of sound in normal, room temperature air is just a bit faster than 1 foot per millisecond. Therefore, in 1 microsecond, a sound wave will travel a bit over 1/1000 of a foot, or a bit less than 1/64 of an inch. This means that greater than 1 microsecond of error in timing can be generated if one headphone driver is 1/64 of an inch closer to your eardrum than the other is. Worrying about 1 microsecond of timing is pointless unless you always position your speakers/headphone drivers in exactly the same place relative to your eardrums to this level of precision, and I don't know of anyone who has ever done this.
  
 When it comes to amplitude/dynamic range, a broadband signal like the noise from an aircraft can easily be recorded at -80dBFS or lower. If you assume a maximum playback level of 115dBA, this means that 16 bit digital audio can record a jet aircraft at ~35dBA easily, and if you lower the required maximum level (because I really don't understand why you'd want the simultaneous capability to record an aircraft at cruising altitude and a noise as loud as many rock concerts), you can improve this arbitrarily. No, 16 bit audio isn't (quite) capable of recording everything from the quietest audible sound up to the threshold of pain without changing levels at some point, but that capability really isn't necessary, as you would never want to have that kind of dynamic range in a recording.
  
 Finally, what does Pythagoras have to do with anything, and where did you use any math at all? You ask if I have anything more proper than the Pythagoras you did, but I don't see any math from you at all, nor do I think anything here will be interesting or surprising to anyone studying at a university. None of this requires study at a university level, and it is already well understood.


----------



## castleofargh

frodeni said:


> Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
> 
> Given that I can pinpoint the sound trailing a plane high in the sky, *a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that*. That again implies that the hearing of humans is more accurate than 16/44.1.


 
  
  


frodeni said:


> cjl said:
> 
> 
> > Care to share that math? I too have done the math, and 16/44.1 seems more than sufficient based on my calculations. As for timing, 16/44.1 has timing accuracy far, far better than just one sample. I've seen very credible evidence showing that its timing accuracy is significantly into the nanoseconds, despite only having one sample every ~23 microseconds or so.
> ...


 
 QED. 
 I started asking for you to substantiate your claims when posting in sound science, because it's annoying to see people make empty claims all day long in a section that is supposed to deal with reality and facts. now it's even gotten to the point where you make points using what you then admit you don't know. so what's up with that?  those empty nonsense claims have nothing to do in this section! stop it!


----------



## KeithEmo

old tech said:


> All this discussion on ABX, Moran and Myer, later experiments etc is all good and well, but I can't get my head around the first order givens and hence why the burden of proof should not fall squarely on those making claims that high res audio sounds better.
> 
> If 24bit compared to 16bit only offers an increase in dynamic range which is not of any practical use, and lowers the noise floor from an already very low and masked by any music content, why should there be any theoretical advantage in reproducing sound?
> 
> ...


 
  
 But your analogies aren't actually correct.
  
 When HD TVs (1920 x 1080) first appeared, it was NOT universally agreed that the extra resolution actually made a significant difference for most customers. Many people in fact argued that there was very little HD content available, and that using "full 1080p HD resolution" on a screen smaller than 30" was a total waste anyway - because nobody could see the difference between 720p and 1080p on a screen that small. However, today, almost every TV of any size is full 1080p HD, and we're having the same argument about 4k.
  
 The problem with your argument is that the basic premise is limited. Yes, if there was absolute reliable proof that frequency response above 20 kHz absolutely, positively produces no audible difference, then it would be unnecessary (although I'm still not convinced that having a "safety margin" above the bare minimum isn't still a good idea). However, the proof you're offering isn't at all "absolute" or "conclusive". In fact, most of those tests were conducted with inadequately sized sample groups, using obsolete equipment, and frequently conducted using dubious test methodology. The fact that twenty or thirty people, using 1980's vintage technology, and 1980's vintage recordings is NOT compelling proof that the difference doesn't exist - at least not to me. And, if we were in fact to prove, with properly sized and run tests, that the difference wasn't audible with the best equipment available today, that wouldn't constitute evidence about whether there might be a difference that is audible with the equipment available in twenty years. I simply don't believe that we actually understand 100.0% of how human hearing works; especially since human hearing takes place partly in the brain - and we certainly don't understand anywhere near 100% of how THAT works.)
  
 (The reality is that there have been several tests run in recent times which tend to suggest that frequency response above 20 kHz can in fact produce audible effects - in different ways and with different implications. The recent AES paper seems to show that a small sample of individuals was able to "beat the odds" in terms of telling whether a given sample was high resolution or not. Another test I recall reading about produced a result that demonstrated that, while the participants didn't hear what they considered to be an audible difference with band-limited content, the location of instruments in the sound stage was perceived as being shifted with the band-limited version, which is in fact "an audible effect". Note that I don't consider either of those results to be "compelling" either but, when balanced against tests run decades ago, with the audio equipment then current, I think they raise enough questions to make it unreasonable to "fall back" on those outdated results as being "absolute facts" without confirmation.)


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## castleofargh

yup, tv resolution alone means nothing, in the end what matters is the angular resolution of the eye, so unless it's in conjunction with screen size and distance from the viewer, it's not working as an analogy and we can't even determine a threshold of visibility.


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## KeithEmo

castleofargh said:


> we all know that abx isn't perfect, how can anything depending on human senses while self administrated, hope to be perfect? even masturbation can always be improved.
> but while you're explaining how some things may elude an abx test, we have a vast majority of people who are misinformed about sighted evaluation and read, and believe
> 
> 
> ...


 
  
 I agree with you wholeheartedly.
  
 So, in case there's any confusion about my opinion here.....
  
 We are _ALL_ subject to expectation bias (sometimes called the placebo effect). What we see, hear, taste, and feel is influenced to a significant degree by what we expect. This is true for all humans (at least all humans tested so far). None of us is immune and, while we can acknowledge it, and even do our best to avoid it or compensate for it, it is inescapable. I would also agree that a significant percentage of the "subjective" opinions of most audiophiles - including myself - is quite probably based on bias or mis-perception.
  
 There is also a tendency for we humans, especially audiophiles, to confuse science with practicality... and I'm considering this to be a discussion of science rather than practicality. To put it bluntly, I've owned dozens of pairs of speakers in my life, and about a dozen different makes and models of headphones, and I currently own several thousand CDs - and probably about a hundred "high-res remasters" of various albums. Of those, if I exclude situations where the difference is likely to be due simply to obvious differences in the mastering itself, there are about a dozen instances where I'm quite certain that I can hear an actual differences, and only then if I listen using one particular pair of speakers, or two specific pairs of headphones, and only then if I listen very carefully and concentrate on listening _for_ a difference. And, even then, I only notice it by direct A/B comparison - and I almost certainly wouldn't notice it if I were to walk out of the room and come back in.
  
 Therefore, if you want to discuss whether there is a "significant difference" or "a difference most people would hear" I would probably agree that it would be unlikely. And, when and if actually asked whether high-res remasters are in fact better, my answer is virtually always that: "I can tell you that I have lots of high-res remasters that sound obviously better than the original; but I'm not sure whether it's because they're high-res or simply because they're mastered better." However, as a point of science, I certainly haven't seen proof that convinces me that there is absolutely positively no audible difference - and, even if someone were to prove to me that I personally really can't detect an audible difference, that still wouldn't prove that nobody else on Earth can do so.
  
 I also agree that unbiased testing is the best way to determine for sure, and that ABX testing works pretty well to eliminate a significant amount of the bias that normally leads people to reach unreasonable and inaccurate conclusions. (It's also about the only type of testing you can do for yourself; without a huge budget and a very large group of cooperative friends.) And, in fact, if anyone wishes to determine for themselves whether _THEY_ can hear a difference, using their equipment, and their favorite source material, then performing an ABX test is almost certainly the best way to go about it (and the ABX test plugin for Foobar2000 is an excellent way to do it.) I would also consider it to be a very fair argument that any difference that cannot be detected using a plain old ABX test isn't "significant" or "important" to most people (and, even if limiting the bandwidth to 20 kHz really does shift the violin three inches to the left, I don't personally care about that either).


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## KeithEmo

sonitus mirus said:


> While perhaps not practical, this is not an issue with an ABX test, even using the popular ABX tools provided as a plugin for the free Foobar2000 music player.  There is nothing preventing you from making large enough files to accomplish exactly what you are describing.  I think you would have a difficult time establishing that a lower quality version is actually resulting in giving you headache.


 
  
 Agreed. The problem with testing some of these things is the number of test subjects and the time required. For example, in order to test whether "version x" is more fatiguing than "version y" with any degree of certainty and accuracy we need to get four or five hundred people in a room, get them all to listen to each for three or four hours, then repeat that every day for a few weeks. Unless you have a huge budget, it would be impractical to do this in the format of a proper ABX test.
  
 However, a less formal version, which might consist of playing "version x" through the loudspeakers in your local library on Mondays and Wednesdays for a month, and "version y" on Tuesdays and Thursdays, keeping track of how many people complain that the music is annoying, and the average time each customer stays in the library before they leave, and correlating the results, might actually be something that could be arranged (perhaps by the library in cooperation with a local university).
  
 And, if you didn't notice the flaw there, we should alternate different days of the week for each version sample to rule out the possibility that there's some other reason why patrons stay for less time on Tuesdays or get more headaches on Wednesdays. This is the sort of detail that people who do these sorts of tests for a living have to account for.


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## sonitus mirus

keithemo said:


> Agreed. The problem with testing some of these things is the number of test subjects and the time required. For example, in order to test whether "version x" is more fatiguing than "version y" with any degree of certainty and accuracy we need to get four or five hundred people in a room, get them all to listen to each for three or four hours, then repeat that every day for a few weeks. Unless you have a huge budget, it would be impractical to do this in the format of a proper ABX test.
> 
> However, a less formal version, which might consist of playing "version x" through the loudspeakers in your local library on Mondays and Wednesdays for a month, and "version y" on Tuesdays and Thursdays, keeping track of how many people complain that the music is annoying, and the average time each customer stays in the library before they leave, and correlating the results, might actually be something that could be arranged (perhaps by the library in cooperation with a local university).
> 
> And, if you didn't notice the flaw there, we should alternate different days of the week for each version sample to rule out the possibility that there's some other reason why patrons stay for less time on Tuesdays or get more headaches on Wednesdays. This is the sort of detail that people who do these sorts of tests for a living have to account for.


 
  
 I wasn't necessarily thinking about large-scale, authoritative testing.  If I was getting headaches and suspected it was due to the format of the file, I'd want to to at least try to test this on myself.  Maybe simply making copies of my favorite few albums at different quality levels and then having the play list shuffled would be enough.


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## RRod

keithemo said:


> Therefore, if you want to discuss whether there is a "significant difference" or "a difference most people would hear" I would probably agree that it would be unlikely. And, when and if actually asked whether high-res remasters are in fact better, my answer is virtually always that: "I can tell you that I have lots of high-res remasters that sound obviously better than the original; but I'm not sure whether it's because they're high-res or simply because they're mastered better." However, as a point of science, I certainly haven't seen proof that convinces me that there is absolutely positively no audible difference - and, even if someone were to prove to me that I personally really can't detect an audible difference, that still wouldn't prove that nobody else on Earth can do so.


 
  
 The way to test hi-res isn't to compare it to a CD that might be a different mastering; you reduce the hi-res version to Redbook and then have a go. I mean, any remastering that required that I AB quick-switch to hear a difference would make me question just how much work the mastering engineer put into the thing!


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## KeithEmo

rrod said:


> The way to test hi-res isn't to compare it to a CD that might be a different mastering; you reduce the hi-res version to Redbook and then have a go. I mean, any remastering that required that I AB quick-switch to hear a difference would make me question just how much work the mastering engineer put into the thing!


 
  
 Agreed - although "remastering" is a very flexible term, and seems to mean different things to different people. To me, the 24/192k remasters of the Grateful Dead Studio Albums sound a lot different - and a lot better - than all of the previous versions of the same albums I've heard (and I've read descriptions of the significant processing and signal "repair" that was done along the way). However, I've definitely got a few other 24/192k "remasters" that sound so identical to previous versions that I'm pretty sure the _ONLY_ difference is that they ran the same exact master tape through the converter at 24/192k instead of 16/44k. (And that's giving them the benefit of the doubt that they didn't simply upsample the 16/44k version.)
  
 The problem with simply down-converting from the 24/192k version to produce an "equivalent 16/44k version" is that ANY sample rate conversion involves some filtering, and so the conversion process itself will in fact alter the signal slightly. Even taking a 16/44k signal, converting it to 24/192k, then converting it back to 16/44k, using the same program, which should produce no difference at all, is usually audible... so the conversion process itself is NOT audibly transparent. (If you can't hear a difference, then it will prove that the conversion and the difference in sample rate taken together aren't audible with your sample content, which would be sufficient for the folks looking to prove that the difference doesn't exist; but, if you do hear a difference, it won't be possible to tell whether the difference is due to the difference in sample rate, or to the conversion process itself, or both.)
  
 Therefore, as I've said before, anyone considering whether to purchase a "remastered" version of an album is probably better served by reading a few reviews about the particular version they're considering buying, and deciding in general whether it is likely to be an improvement, than to worry about the sample rate it happens to be offered at.... (and I personally tend to be willing to spend a few dollars more for the higher-resolution version - but more as a matter of "insurance" than out of any specific expectation that it will be better). The simple reality is that, technical realities aside, the whole "high-res file craze" has provided an excellent excuse for the latest wave of "remasters" and "reissues" and, for whatever reason, many of them are in fact very good. Everyone should also remember that, in the end, even if it turns out that 24/192k is capable of sounding audibly better than 16/44k, that's still only going to be true in a specific situation if the master is good enough for the difference to matter, and if the engineering and conversion are good enough to preserve that difference.


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## RRod

> The problem with simply down-converting from the 24/192k version to produce an "equivalent 16/44k version" is that ANY sample rate conversion involves some filtering, and so the conversion process itself will in fact alter the signal slightly. Even taking a 16/44k signal, converting it to 24/192k, then converting it back to 16/44k, using the same program, which should produce no difference at all, is usually audible... so the conversion process itself is NOT audibly transparent. (If you can't hear a difference, then it will prove that the conversion and the difference in sample rate taken together aren't audible with your sample content, which would be sufficient for the folks looking to prove that the difference doesn't exist; but, if you do hear a difference, it won't be possible to tell whether the difference is due to the difference in sample rate, or to the conversion process itself, or both.)


 
  
 Of course it involves filtering, but at some point you have to say: this is how we test the same content at different rates. Even feeding the recorded signal into two different paths for hi-res and Redbook could lead to differences not due only to the bit depth or sample rate. But I strongly disagree that interpolating to 192 and back down to 44.1 will be "usually audible." Once again, you can look at and listen to the difference between the original 44.1 and the up/down version, and here the comparison is even more apt because there wasn't any hi-res material to begin with.


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## KeithEmo

rrod said:


> Of course it involves filtering, but at some point you have to say: this is how we test the same content at different rates. Even feeding the recorded signal into two different paths for hi-res and Redbook could lead to differences not due only to the bit depth or sample rate. But I strongly disagree that interpolating to 192 and back down to 44.1 will be "usually audible." Once again, you can look at and listen to the difference between the original 44.1 and the up/down version, and here the comparison is even more apt because there wasn't any hi-res material to begin with.


 
  
 I guess "usually" is a vague term.
  
 I've tried that test (converting a 16/44k original up to 24/192k and then back to 16/44k) with a few programs - and the difference was sometimes rather audible. However, most of the "higher end" programs offer multiple options for dithering and filtering whenever you do a sample rate conversion, as well as various tradeoffs between cutoff frequency, cutoff sharpness, impulse response, and processing time, and at least some of those options are in fact audibly different. I wouldn't rule out the possibility that at least some of those combinations and options may turn out to be inaudible, but that would itself need to be tested. (And I'm not specifically aware of a certain combination of program and settings that I would assume to produce an inaudible conversion.)
  
 Of course, as I mentioned, if the test were to conclude that there was no audible difference, then that conclusion would support both claims: that the differences between the different sample rates were inaudible, and that any anomalies caused by the conversion process itself were also inaudible (ignoring the slight possibility that differences caused by each individually might cancel out).


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## RRod

keithemo said:


> I guess "usually" is a vague term.
> 
> I've tried that test (converting a 16/44k original up to 24/192k and then back to 16/44k) with a few programs - and the difference was sometimes rather audible. However, most of the "higher end" programs offer multiple options for dithering and filtering whenever you do a sample rate conversion, as well as various tradeoffs between cutoff frequency, cutoff sharpness, impulse response, and processing time, and at least some of those options are in fact audibly different. I wouldn't rule out the possibility that at least some of those combinations and options may turn out to be inaudible, but that would itself need to be tested. (And I'm not specifically aware of a certain combination of program and settings that I would assume to produce an inaudible conversion.)
> 
> Of course, as I mentioned, if the test were to conclude that there was no audible difference, then that conclusion would support both claims: that the differences between the different sample rates were inaudible, and that any anomalies caused by the conversion process itself were also inaudible (ignoring the slight possibility that differences caused by each individually might cancel out).


 
  
 You'll have to describe what rather audible means, because what I tend to get from sox (for a difference) using the typical methods is some content at -70dBFS between 20-22kHz and then dither noise at about -110dB for the rest of the frequency range.


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## Roly1650

rrod said:


> Of course it involves filtering, but at some point you have to say: this is how we test the same content at different rates. Even feeding the recorded signal into two different paths for hi-res and Redbook could lead to differences not due only to the bit depth or sample rate. But I strongly disagree that interpolating to 192 and back down to 44.1 will be "usually audible." Once again, you can look at and listen to the difference between the original 44.1 and the up/down version, and here the comparison is even more apt because there wasn't any hi-res material to begin with.



Yes, I'd have to agree. I've taken "puported" 24/96 or 24/192 files, down/up converted with Audacity's default settings and then nulled in ADM, which reports a null down around -85dB. Sure, when you listen to the difference track there is something there, but, with the best will in the world, *nobody is going to hear it in the presence of the original signal at that sort of level.

There's a guy on Youtube who's made video's showing the same thing when he nulls down/up converted files, so I'd have to say any software that produces audible differences due to down/upconverting is pretty lousy and needs to be avoided.*


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## goodyfresh

roly1650 said:


> Yes, I'd have to agree. I've taken "puported" 24/96 or 24/192 files, down/up converted with Audacity's default settings and then nulled in ADM, which reports a null down around -85dB. Sure, when you listen to the difference track there is something there, but, with the best will in the world, nobody is going to hear it in the presence of the original signal at that sort of level.
> 
> There's a guy on Youtube who's made video's showing the same thing when he nulls down/up converted files, *so I'd have to say any software that produces audible differences due to down/upconverting is pretty lousy and needs to be avoided.*


 
 Yeah, that's why I use dBpoweramp, it's great software.  If your conversion software actually produces an audible difference when downsampling from 192 to 48, the fact is it's a problem with the software, not with the act of downsampling itself.


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## KeithEmo

rrod said:


> You'll have to describe what rather audible means, because what I tend to get from sox (for a difference) using the typical methods is some content at -70dBFS between 20-22kHz and then dither noise at about -110dB for the rest of the frequency range.


 
  
 To me "rather audible" means that, when I switch back and forth, I hear an obvious difference.
  
 Unfortunately, unless you take actual measurements, the various settings in the various conversion programs aren't comparable. For example, in Izotope RX3's Resample module, I get to pick a new Sample Rate, a Filter Steepness (from 0 to 2000 with a default of 834), a Cutoff Shift (from 0.7 to 1.4) that adjusts the cutoff frequency up or down, and a pre-Ringing setting (that goes from 0 to 1.0), and in Adobe Audition I get to choose between "high quality" and "fast processing" - which I'm pretty sure means that the high quality option uses a filter with more taps. And I'm sure I could look up whether each of those uses FIR or IIR filters, how many taps each uses, and at least some of the filter parameters, although most programs tend to omit many of those important details. And, of course, other programs offer other options. My point there is that the settings you choose when doing resampling, and how audible (or not) any of them are, is a whole subject in and of itself.... so the first part of determining whether the difference in sample rates was audible would be to choose a method for performing the conversion, and then testing whether _THAT_ was in fact inaudible or not. (And, if you go to some of the pro-audio forums, you'll find lively discussions about which converters sound better, and which ones are more transparent, and which settings are best on each for particular kinds of music.)
  
 I also notice that your chosen resampler - SoX - offers quite a few options; in fact, their example graph of a 96k to 44k conversion shows the transient response plots for twelve different combinations of settings, all of which look - and potentially sound - different. (There are several other threads where the audibility of those types of differences are discussed and argued.) Which of those should we consider to be "right"? And that's just one parameter that can be adjusted. (And we should note that all of those graphs show ways in which SoX _ALTERS_ the original signal during the conversion process.) There's also a link on the SoX page to a website that shows the performance of a whole slew of sample rate converter programs (SoX ranks very well with certain settings; but many other popular programs do not.)
  
 (I'm not trying to derail this discussion onto a siding.... I'm merely pointing out that converting a 96k file to 44k without introducing any audible artifacts isn't at all a trivial proposition, and even such a seemingly simple step as "making a good 44k version of a 96k file" is a lot less simple than it seems at first.)
  
 Another factor that most of the participants here seem to ignore is that different DACs process different sample rates differently (most DACs are programmed to use different oversampling multipliers depending on the sample rate of the input - because hardware limitations prevent them from applying high oversampling multipliers to inputs that are already at high sample rates). This means that, even assuming that you did have two samples, at different sample rates, but otherwise audibly identical, they could sound audibly different when played on a specific DAC because that particular DAC responds differently to the different sample rates. (The much maligned recent AES paper addressed that issue by having all of their samples played at 192k - with the "44k" samples filtered AS IF they were prepared for being recorded at 44k. Thus their samples could be expected to exhibit any audible differences due to the bandwidth limitation of the filtering, but, at the same time, avoid differences due to how the individual DAC used handles different sample rates. In other words they used "44k audio played at 192k" for their "44k samples".)
  
 Again, however, finding that no audible differences existed in spite of these other variables would prove that none of them was audible.


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## old tech

This looks bad, but how do you quote a part of a post in reply?


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## old tech

Re - Frodeni

With respect, I think it is a bit unfair to accuse me of shutting down debate. This is after all a "sound science" forum. The questions I raised are quite fundamental to the science. If you are putting forward a proposition that violates what we know about human hearing, and logic eg why an apparent issue with 16bit would not manifest itself more with analogue 13bit then it is perfectly valid to the debate to ask for evidence, and the maths that you say support this proposition. If these questions regarding logic and evidence behind propositions which challenge what we know technically and physiologically about music playback and music perception cannot be asked on a sound science forum then of what use is it?


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## RRod

keithemo said:


> To me "rather audible" means that, when I switch back and forth, I hear an obvious difference.


 
  
 So then the question is, how are you listening to hear these obvious differences. As both myself and Roly have pointed out, the artifacts from *common* decimation/quantization methods in widely-available software are way down below the actual level of the music, and the "loudest" parts are way up in the spectrum, past the edge of the commonly held audible range. Thus the only possibilities I can fathom are a) you are cutting out an extremely quiet section of music and jacking up the volume, and also have an upper hearing limit actually near 20k or b) you are switching between resampling algorithms that have passband features that would not be considered flat within the audible range. And there seems to be an underlying assumption that because you can see pre-ringing on a linear low-pass filter then that immediately implies audibility *at actual music listening levels*, which just isn't the case.
  
 No arguments on your last point; that's exactly how I do things for my own ABX: convert to 16/44.1 then go back up to the original hi-res spec.


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## old tech

keithemo said:


> But your analogies aren't actually correct.
> 
> When HD TVs (1920 x 1080) first appeared, it was NOT universally agreed that the extra resolution actually made a significant difference for most customers. Many people in fact argued that there was very little HD content available, and that using "full 1080p HD resolution" on a screen smaller than 30" was a total waste anyway - because nobody could see the difference between 720p and 1080p on a screen that small. However, today, almost every TV of any size is full 1080p HD, and we're having the same argument about 4k.
> 
> ...



 


I don't believe that the analogies are incorrect around the basic point. Increased bit depth does increase the quality of video due to pixilation, but does not increase sound quality in audio beyond 16bit because the sound wave is already perfect and all you are doing is increasing the dynamic range and lowering the noise floor. That extra video resolution of 1080p was not agreed to be an improvement back in the day is neither here or there. The limitations were largely as you say the screen size. With audio, the limitation, beyond 16bit, is not the hardware but our ears. Whereas the eye can determine high res video due to smaller pixilation, there is no pixilation in audio. If you can actually find a home consumer DAC that can actually resolve 24bits, and have the equipment to go with it, it will not make the sound wave more perfect than 16bit digital or 8bit digital for that matter.

The analogy of going beyond 20khz in audio with TVs being able to reproduce frequencies into infra red is valid. In both cases there is no point as it is outside the human range. I don't disagree that higher frequencies than 20khz may produce audible effects, but this is more likely to be distortions in the stereo system reacting to ultrasonic content. There is documented evidence that 24/192 can cause distortion is some stereos because of it.

The bottom line is that you appear to be challenging the established known facts that humans can't hear well above 20khz and can't resolve a dynamic range greater than 16bits. This is a bit like challenging the notion that humans are unable to pick up sonar signals like dolphins and bats. Surely the burden of proof is on those making the claim. And using special pleadings such as "we do not know 100% how human hearing works" to somehow justify then that means we can hear ultrasonic sound waves or have machine like resolution is a well know ploy used in most psuedosciences.


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## goodyfresh

old tech said:


> I don't believe that the analogies are incorrect around the basic point. Increased bit depth does increase the quality of video due to pixilation, but does not increase sound quality in audio beyond 16bit because the sound wave is already perfect and all you are doing is increasing the dynamic range and lowering the noise floor. That extra video resolution of 1080p was not agreed to be an improvement back in the day is neither here or there. The limitations were largely as you say the screen size. With audio, the limitation, beyond 16bit, is not the hardware but our ears. Whereas the eye can determine high res video due to smaller pixilation, there is no pixilation in audio. *If you can actually find a home consumer DAC that can actually resolve 24bits*,


 
 This is the best point of all.  Namely, there aren't any DAC's which ACTUALLY resolve past 20 or at most 21 bits, anyway, folks.  They may be able to receive and process signals at 24 or 32 bits, but they can't actually fully process and output anything beyond the first 21.
  
 The other thing is the difference between resolution in video and sampling in audio.  They are fundamentally different.  Thing is that video with pixels is always going to be a DISCRETE approximation of what is, when ignoring the quantum-mechanical scale at least, a CONTINUOUS phenomenon in the real world.  However, audio sampling is different.  Namely, while it uses a discrete approximation of a continuous phenomenon, the waveform can be reproduced EXACTLY within the audible frequency-range from that approximation.  This is due to the Nyquist Sampling Theorem, which says that any waveform with a frequency which never at any point goes higher than the value *f* can be perfectly reconstructed without any error at all by using samples with a rate of *2f*.  Thus why 44.1khz sampling-rate is able to perfectly reproduce audio up to 22Khz in frequency, for example.
  
 https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

 Anyone proposing that there is an actual audible difference from sample-rates higher than 44.1Khz is fundamentally misunderstanding the concept of the Nyquist Sampling Theorem and how the very idea of Sampling in Fourier Analysis even works, mathematically speaking.  Such misunderstanding makes sense given that the math is fairly high-level and far above the high-school precalculus or calculus that most people top-out at in their lives.  Higher sampling rates are only better if you want extra "room" for DSP's to play around with the waveform during mastering due to the availability of accurate reproduction of much smaller wavelengths, i.e. higher frequencies than 22Khz.  In terms of the actual PLAYBACK of the music, they simply cannot, mathematically speaking, produce an audible difference within the range of human hearing.
  
 Bit-depth is a bit more complicated of an issue, but comes down in the end to something similar:  Our ears don't hear as precisely as our eyes see, so unlike the difference between 16-bit and 24 or 32-bit color in video (where our eyes can clearly see the difference in the range of different colors and their relative brightness levels), the actual maximum dynamic range of human hearing when listening to a significantly audible signal (which should be noted to be different than the dynamic range when suddenly hearing noises after being in a completely quiet environment for an extended time) is able to be completely reproduced by 16-bit or PERHAPS, in the most extreme cases, 20-bit audio.  But the latter, the extreme cases, dont' even apply in the case of music, as it is simply impossible to find music with a dynamic range beyond about 25 to 30 decibels at most, and the vast majority of music has less than 15 decibels dynamic range.
  
 It's too bad this won't convince people, though.  They'll still be convinced that they are hearing a difference with 32/192 audio compared to 16/44.1.  The placebo-effect and expectation-bias are very strong things when it comes to human perceptioon.


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## dprimary

keithemo said:


> I just bit the proverbial bullet and BOUGHT the AES paper (AES Convention Paper 9174). (Apparently AES papers remain copyrighted, and cannot be reprinted for free, which is why everyone is talking about the paper but not many people seem to have read it.) Note that the title of the paper refers to "filters", but what they're testing is whether the necessary application of band-limiting, as applied to CD content, is audible. Basically, for the study, they took some 24/192k content and band limited it using filters equivalent to those you would use to record CD and DVD audio content (filtered to cut it off at the Nyquist frequencies of 22 kHz and 24 kHz respectively). The results were that the subjects _WERE_ in fact able to tell the "CD version" from the "high-def version" - with a reliability that exceeded random chance (in general, it was between 56% and 66%, which may not be overwhelming, but is statistically better than random). The study also found that the difference was more audible with certain passages than with others, and included some interview data (taken afterwards) where the subjects explained subjectively the differences they claimed to hear (which do coincide with the differences many audiophiles also claim to hear).


 
 If you are member of AES you can download 25 papers this month for free. Having read the paper I think the test was pretty well done, They give enough detail that you can  try to copy the test. They do not provide the much detail on the matlab filters. There is a few issues I see, first of all the reduced the bandwidth and resolution at the same time. I would have like to see them change one variable at a time. They did not also filter the audio at 96k to see the the filter they created were audible. If a 96k filter was audible then it is not the bandwidth limit that is audible but the filter. In this test though the purpose was to test the audibility of typical downsampling and bit reduction.
  
 I have purchased the tracks they used and since I am testing some large studio monitors that go out to 40k, I plan to test the same tracks. I will using isotope to process the tracks to see if can repeat their results


----------



## Thad-E-Ginathom

frodeni said:


> Most people discuss the difference in the audible range of frequency. Personally, I think 16/44.1 is far too inaccurate for 3D sound. Both for amplitude and for timing.
> 
> Given that I can pinpoint the sound trailing a plane high in the sky, a quick look at the math needed to do so, strongly supports that 16/44.1 is insufficient for that. That again implies that the hearing of humans is more accurate than 16/44.1.


 
  
  
 Damn good thing that stereo is _not  _3D then. It has never claimed to be.
  
_Imagination_plays a huge _valid_ part in listening to music, just as it does in reading a book. It is supposed to happen. The engineers may be able to suggest depth by varying levels, and even timing, but there is no up/down pan on control board.
  
 Congratulations on your imagination: you are using it in _exactly_ the right way!
  
 When I was a very young child, I preferred radio to TV. According to me, the pictures were better.


----------



## frodeni

goodyfresh said:


> This is the best point of all.  Namely, there aren't any DAC's which ACTUALLY resolve past 20 or at most 21 bits, anyway, folks.  They may be able to receive and process signals at 24 or 32 bits, but they can't actually fully process and output anything beyond the first 21.
> ...


 
  
 And they should not. The threshold of pain is at about 120dB.
  
https://en.wikipedia.org/wiki/Audio_bit_depth
  
 Still does not change much, but sure raises a lot of new questions.
  
 Great point.
  
 It seems to me that the human ability to hear, is not where the problem is, rather more the complexity of digital sound reproduction, including jitter and noise effects. This is going nowhere, as the difference going HD is clearly heard by many.


----------



## RRod

frodeni said:


> It seems to me that the human ability to hear, is not where the problem is, rather more the complexity of digital sound reproduction, including jitter and noise effects. This is going nowhere, as the difference going HD is clearly heard by many.


 
  
 Clearly heard by many when they know the track is HD. Curiously the effect seems to disappear when blinders are put on and they can only use their ears.


----------



## goodyfresh

rrod said:


> Clearly heard by many when they know the track is HD. Curiously the effect seems to disappear when blinders are put on and they can only use their ears.


 

*exactly *


----------



## KeithEmo

old tech said:


> keithemo said:
> 
> 
> > But your analogies aren't actually correct.
> ...


 
  
 Just to be clear - I'm not actually challenging anything one way or the other - because I haven't run a properly controlled test (and, again, even if I personally couldn't hear a difference, that wouldn't prove that nobody can). My main point is that "the established science" may simply not be right. Five hundred years ago, the established science they taught in school was that the Earth was flat, and tomatoes were poisonous; now we know better. When I went to high school, they taught in science class that all matter was made up of protons, neutrons, and electrons - which were the smallest indivisible "pieces" of matter; and that model was good enough to bring us nuclear power plants and the fusion bomb; but now we find that notion quaint, and there's an active debate about whether matter is "really" vibrating 11-dimensional energy strings, or a collection of smaller particles called quarks, or something not quite either one. And, the last time I looked, we _still _don't know exactly how the human brain works (and "hearing" occurs in both the ears and the brain).
  
 Incidentally, for an interesting experiment, go buy yourself one of those new souped up half watt LASER pointers that operates at 720 nm or 840 nm; that's the "invisible infrared" color used by a lot of remote controls; and a LASER puts out a very clean single frequency. Shine the dot somewhere and you will probably find that the "invisible" dot is in fact clearly visible; I can see it quite clearly as a pale pink - and so can most people. So I guess the "science" about IR light being "invisible" is wrong too. (Actually, in order to be visible to most of us, it has to be so bright that it is somewhat dangerous to look at for more than a few seconds, but my point stands - the "commonly accepted fact" is in fact _wrong_. And, in fact, a TV that was actually able to display long-wave IR, and so make the bright sun in the picture of the desert actually feel warm on your face, would - at least to me - have much better fidelity than the one I have now.)
  
 I don't know for sure whether the difference between 16/44k and 24/192k is audible - everything else being exactly equal, but I'm absolutely positive that I don't necessarily trust the "truth" as "discovered" by scientists back when most audiophiles were certain that a Dynaco Stereo 70 and Koss pro4AA's "sounded audibly perfect" because both "covered the entire audible spectrum". And, with many modern DACs with selectable filters, there are differences that many people find audible which _seem_ to coincide with different sample rates and different filter responses producing audible differences. Perhaps there's something there; or perhaps what we're hearing is simply that a given DAC handles 16/44k differently than it handles 24/192k - because it uses a different oversampling multiple; and perhaps the endless discussions in one or two pro sound forum about how certain sample rate converters sound better or worse with certain types of music is all superstition as well (audiophiles have nothing on pros for superstitious beliefs). However, I'm not quite prepared to say that "audio science is at its end because there's lots of equipment available today that's audibly perfect, so there's nothing to improve."
  
 Personally, since the science shows clearly that high-res files are in fact superior in quality (frequency response and dynamic range) - whether that superiority is audible or not - then to me that's enough justification for continuing to improve things.... and for studying whether those technical improvements lead to some sort of audible improvements. I can also say that, personally, I'm willing to pay a bit extra for a technical improvement even if that improvement doesn't yield anything that's currently important - or even noticeable. (If it turns out that nobody can hear the difference, that still won't prove that the extra information that's there won't be useful to some new "3D decoder" someone comes out with next year, or some other gadget neither of us can guess at, and so won't prove it "totally useless".) I also simply see the latest "fad" for high-res remasters as being generally a good thing - because at the very least it encourages people to listen to music carefully enough that they are actually hearing it. (I'd rather see people spending money on high-res players that don't sound different than on cheesy 128k MP3 players which they imagine "don't sound _much_ different" - because the latter is a slippery slope I'd rather avoid approaching.) 
  
 Now, if you want to start a new thread entitled "What is the best and most practical sample rate and bit depth to use for distributing consumer music?" then I might well be inclined to agree with you on a lot more things.


----------



## KeithEmo

dprimary said:


> If you are member of AES you can download 25 papers this month for free. Having read the paper I think the test was pretty well done, They give enough detail that you can  try to copy the test. They do not provide the much detail on the matlab filters. There is a few issues I see, first of all the reduced the bandwidth and resolution at the same time. I would have like to see them change one variable at a time. They did not also filter the audio at 96k to see the the filter they created were audible. If a 96k filter was audible then it is not the bandwidth limit that is audible but the filter. In this test though the purpose was to test the audibility of typical downsampling and bit reduction.
> 
> I have purchased the tracks they used and since I am testing some large studio monitors that go out to 40k, I plan to test the same tracks. I will using isotope to process the tracks to see if can repeat their results


 
  
 I agree there.... there are many different filter types and options - and they only tested one of them (and proving that a single filter is audible doesn't prove that others are). I would have also liked it if they had included a few more types of test equipment (since they basically used one DAC and one pair of speakers). I've always found electrostatic headphones to be the best thing for picking out minute audible differences, so that would have been my choice there.
  
 As for the accusations that the authors "had an agenda".... personally I'm inclined to say that, if they did, I saw no evidence to that effect. (To put it bluntly, while the test produced a "statistically significant" result, it would hardly serve to convince people that there's some sort of significant and obvious difference worth paying for. In fact, it tended more to suggest that the difference was there, but was rather minor and difficult to hear.)
  
 I have to admit that I'm at a bit of a disadvantage here in that classical music isn't what I normally listen to, so I personally would not be the most likely person to notice whether the sounds of specific instruments, or the sound of the ambiance in a real concert hall, were or were not "rendered accurately".


----------



## jcx

> ... whether the sounds of specific instruments, or the sound of the ambiance in a real concert hall, were or were not "rendered accurately".


 
 can't expect recorded music to be "accurate" to start with - the art and illusion starts with microphone choice and positioning and continues throughout the mastering process - pianos are notoriously different sounding in recordings
  
 the most you can say is whether you like one or another set of recording choices - I would expect it to be a bad idea to listen for what recordings deliberately manipulate "to taste" when trying to distinguish digital audio sample rate


----------



## Roly1650

keithemo said:


> Just to be clear - I'm not actually challenging anything one way or the other - because I haven't run a properly controlled test (and, again, even if I personally couldn't hear a difference, that wouldn't prove that nobody can). My main point is that "the established science" may simply not be right. Five hundred years ago, the established science they taught in school was that the Earth was flat,
> >>>>>>>>>>snip snip



Sorry Keith, but this old bollocks always touches a nerve with me and it's amazing how many otherwise knowledgeable people will trot it out. The truth is that the only reason "the established science" was that the earth was flat was at the insistence of the church and in those days if your views ran counter to the church, nasty, painful things tended to happen to you. Excommunication was probably the best you could hope for, at least then the church ignored you!

Science established that the earth was anything other than flat centuries before the church intervened. No Greek writer, after about 500 BC, considered the earth anything other than non-flat, an Egyptian "scientist" had calculated the diameter to within 2% iirc, centuries before that and the Phoenicians, being a shipfaring nation had guessed due to ships disappearing over the horizon. Even the heliocentric, as opposed to the geocentric, solar system had been proposed centuries BC. All that "lost" science was the result of religious belief, which threw Europe into what we now call "The Dark Ages". But of course it's easier to blame religious ignorance on science isn't it? 

A pi** poor example imo.


----------



## limpidglitch

roly1650 said:


> Sorry Keith, but this old bollocks always touches a nerve with me and it's amazing how many otherwise knowledgeable people will trot it out. The truth is that the only reason "the established science" was that the earth was flat was at the insistence of the church and in those days if your views ran counter to the church, nasty, painful things tended to happen to you. Excommunication was probably the best you could hope for, at least then the church ignored you!
> 
> Science established that the earth was anything other than flat centuries before the church intervened. No Greek writer, after about 500 BC, considered the earth anything other than non-flat, an Egyptian "scientist" had calculated the diameter to within 2% iirc, centuries before that and the Phoenicians, being a shipfaring nation had guessed due to ships disappearing over the horizon. Even the heliocentric, as opposed to the geocentric, solar system had been proposed centuries BC. All that "lost" science was the result of religious belief, which threw Europe into what we now call "The Dark Ages". But of course it's easier to blame religious ignorance on science isn't it?
> 
> A pi** poor example imo.


 
  
 Not even the church have ever held the idea of a flat earth as official doctrine, mainly, I suspect, because a spherical earth didn't challenge the idea of mankind's central position in god's creation.
  
 The Egyptian you're thinking of is Eratosthenes, who was ~16% off in his calculations.


----------



## castleofargh

limpidglitch said:


> Not even the church have ever held the idea of a flat earth as official doctrine, mainly, I suspect, because a spherical earth didn't challenge the idea of mankind's central position in god's creation.
> 
> The Egyptian you're thinking of is Eratosthenes, who was ~16% off in his calculations.


 

 16% off, what a noob!
  
  
  
  
 I'm pretty sure I wouldn't be that precise if I had to tell how far my car is parked. humans really kick ass(well some of them).


----------



## KeithEmo

roly1650 said:


> Sorry Keith, but this old bollocks always touches a nerve with me and it's amazing how many otherwise knowledgeable people will trot it out. The truth is that the only reason "the established science" was that the earth was flat was at the insistence of the church and in those days if your views ran counter to the church, nasty, painful things tended to happen to you. Excommunication was probably the best you could hope for, at least then the church ignored you!
> 
> Science established that the earth was anything other than flat centuries before the church intervened. No Greek writer, after about 500 BC, considered the earth anything other than non-flat, an Egyptian "scientist" had calculated the diameter to within 2% iirc, centuries before that and the Phoenicians, being a shipfaring nation had guessed due to ships disappearing over the horizon. Even the heliocentric, as opposed to the geocentric, solar system had been proposed centuries BC. All that "lost" science was the result of religious belief, which threw Europe into what we now call "The Dark Ages". But of course it's easier to blame religious ignorance on science isn't it?
> 
> A pi** poor example imo.


 
  
 Actually I disagree.
  
 If you actually try to find information about "the frequency range of human hearing" you will find that subject mentioned in a lot of books... and most of them seem to agree that everybody else seems to agree that "the commonly accepted range of human hearing is 20 Hz to 20 kHz". At one point I tried to research exactly where that number came from, and I reached a point where most books were simply quoting other books, or saying that "it was commonly accepted". Now, while it may in fact be true, I am always leery of things that "everybody knows", but nobody seems to want to quote the original research to substantiate. (So, while most people in the middle ages "knew" the world was flat because their priest said so, a lot of people today just seem to accept that "it's commonly known that the limit of human hearing is 20 Hz to 20 kHz", which to me seems a lot like the same blind acceptance of presumed authority. While I found plenty of books and references that states the limits of human hearing as "commonly accepted", I entirely failed to find mention of an actual test, performed by an author of one of those books, to confirm this "well known" information.) Considering that, in the last twenty or thirty years, a lot of "commonly known facts" have turned out NOT to be true, I'm not willing to simply accept this one at face value because it's been repeated a lot - for a very long time.
  
 Note that I have run across several references where individuals state that "they have found this to be true" - which is clearly anecdotal; I've also found one reference that "human hearing extends down to 12 Hz under laboratory conditions"; and one or two others claiming to have "detected response to frequencies well above 20 kHz in humans under some circumstances"; and at least one other that claimed to have shown that test subjects heard differences in samples that were band-limited to 20 kHz when compared to those that weren't (that study seemed to show that limiting the bandwidth to 20 kHz caused a shift in the perceived location of some instruments in the sound stage).
  
 In short, I don't think that "fact" rises anywhere near the level of certainty necessary to justify using it to claim that further research is pointless or unnecessary.


----------



## Roly1650

limpidglitch said:


> Not even the church have ever held the idea of a flat earth as official doctrine, mainly, I suspect, because a spherical earth didn't challenge the idea of mankind's central position in god's creation.
> 
> The Egyptian you're thinking of is Eratosthenes, who was ~16% off in his calculations.



It may not have been the official line, but it was certainly used at the grass roots to keep the peasants in line.

As for Eratosthenes, it would appear that impossible to beleve, but we are both right, from Wikipedia:

"An early report on the circumference of the Earth was given by Aristotle at 400,000 stadia.[1] The first scientific estimation of the radius of the Earth was given by Eratosthenes about 240 BC. Estimates of the accuracy of Eratosthenes’s measurement range from within 2% to within 15%."

And:
"He is best known for being the first person to calculate the circumference of the Earth, which he did by applying a measuring system using stadia, a standard unit of measure during that time period. His calculation was remarkably accurate. He was also the first to calculate the tilt of the Earth's axis (again with remarkable accuracy). Additionally, he may have accurately calculated the distance from the Earth to the Sun and invented the leap day.[4] He created the first map of the world incorporating parallels and meridians, based on the available geographical knowledge of the era."

In the mathematical genius category I'd say.


----------



## Roly1650

keithemo said:


> Actually I disagree.
> 
> If you actually try to find information about "the frequency range of human hearing" you will find that subject mentioned in a lot of books... and most of them seem to agree that everybody else seems to agree that "the commonly accepted range of human hearing is 20 Hz to 20 kHz". At one point I tried to research exactly where that number came from, and I reached a point where most books were simply quoting other books, or saying that "it was commonly accepted". Now, while it may in fact be true, I am always leery of things that "everybody knows", but nobody seems to want to quote the original research to substantiate. (So, while most people in the middle ages "knew" the world was flat because their priest said so, a lot of people today just seem to accept that "it's commonly known that the limit of human hearing is 20 Hz to 20 kHz", which to me seems a lot like the same blind acceptance of presumed authority. While I found plenty of books and references that states the limits of human hearing as "commonly accepted", I entirely failed to find mention of an actual test, performed by an author of one of those books, to confirm this "well known" information.) Considering that, in the last twenty or thirty years, a lot of "commonly known facts" have turned out NOT to be true, I'm not willing to simply accept this one at face value because it's been repeated a lot - for a very long time.
> 
> ...



I think there's a fair chance you didn't read my post......


----------



## OddE

limpidglitch said:
			
		

> .
> The Egyptian you're thinking of is Eratosthenes, who was ~16% off in his calculations.




-That depends on what definition of the length of a Stadion you prefer; he may well have been off by slightly under one percent.


----------



## limpidglitch

roly1650 said:


> It may not have been the official line, but it was certainly used at the grass roots to keep the peasants in line.
> 
> As for Eratosthenes, it would appear that impossible to beleve, but we are both right, from Wikipedia:
> 
> ...


 
  
 16% seems much more probable to me, considering the method he used to measure the distance between Alexandria and Aswan(?).
  
 And 16% isn't bad, considering that Columbus was ~25% off in 1492


----------



## Roly1650

limpidglitch said:


> 16% seems much more probable to me, considering method he used to measure the distance between Alexandria and Aswan(?).
> 
> And 16% isn't bad, considering that Columbus was ~25% off in 1492



Yea, but 2% is much more impressive.

Columbus wasn't off 25%, this fecking great lump of land jumped in his way, otherwise he'd have hit India no problem.


----------



## limpidglitch

roly1650 said:


> Yea, but 2% is much more impressive.
> 
> Columbus wasn't off 25%, this fecking great lump of land jumped in his way, otherwise he'd have hit India no problem.


 
  
 If that great lump of land hadn't been there, he's likely to have starved to death.
  
 Currently reading up a bit on al-Biruni. Now there's a guy who knew how to measure the world, 800 years before Humboldt.


----------



## Roly1650

limpidglitch said:


> If that great lump of land hadn't been there, he's likely to have starved to death.
> 
> Currently reading up a bit on al-Biruni. Now there's a guy who knew how to measure the world, 800 years before Humboldt.



Thanks for the link, very interesting. It's amazing how much the sum store of human knowledge was advancing in the rest of the known world while Europe was submerged in the Dark Ages.


----------



## limpidglitch

I found another article by the same author that goes into more detail on his geodesy. Really impressive stuff.


----------



## Roly1650

limpidglitch said:


> I found another article by the same author that goes into more detail on his geodesy. Really impressive stuff.



Thanks again, both articles will be read and digested.


----------



## old tech

keithemo said:


> Just to be clear - I'm not actually challenging anything one way or the other - because I haven't run a properly controlled test (and, again, even if I personally couldn't hear a difference, that wouldn't prove that nobody can). My main point is that "the established science" may simply not be right. Five hundred years ago, the established science they taught in school was that the Earth was flat, and tomatoes were poisonous; now we know better. When I went to high school, they taught in science class that all matter was made up of protons, neutrons, and electrons - which were the smallest indivisible "pieces" of matter; and that model was good enough to bring us nuclear power plants and the fusion bomb; but now we find that notion quaint, and there's an active debate about whether matter is "really" vibrating 11-dimensional energy strings, or a collection of smaller particles called quarks, or something not quite either one. And, the last time I looked, we _still _don't know exactly how the human brain works (and "hearing" occurs in both the ears and the brain).
> 
> Incidentally, for an interesting experiment, go buy yourself one of those new souped up half watt LASER pointers that operates at 720 nm or 840 nm; that's the "invisible infrared" color used by a lot of remote controls; and a LASER puts out a very clean single frequency. Shine the dot somewhere and you will probably find that the "invisible" dot is in fact clearly visible; I can see it quite clearly as a pale pink - and so can most people. So I guess the "science" about IR light being "invisible" is wrong too. (Actually, in order to be visible to most of us, it has to be so bright that it is somewhat dangerous to look at for more than a few seconds, but my point stands - the "commonly accepted fact" is in fact _wrong_. And, in fact, a TV that was actually able to display long-wave IR, and so make the bright sun in the picture of the desert actually feel warm on your face, would - at least to me - have much better fidelity than the one I have now.)
> 
> ...


 
 I agree with you on the point that there is no harm in obtaining a higher than 16/44 res recording, especially given that storage space is cheap. (*with one qualification that very high sample rates eg 192khz may introduce distortions into the playback system).  However doing so, in the belief that the higher res file on its own will result in an improvement in playback fidelity is misguided.
  
 The examples you provided of how claims that the Dynaco Stereo 70 and Koss pro4AA's are prefect show how the science can be wrong is not valid.  That may have been the claim by some audiophiles, but there was no real science backing up that claim.  There are many issues of non-linearity in frequency response which mean that no transducers or room acoustics (in the average home) can be perfect. That is quite different to the hundred years of experiments which demonstrate that the range of human hearing falls between 20 to 20,000hz.  The latter is not merely an audiophile claim (indeed many audiophiles don't accept this, hence the demand for hi res), but accepted by all the science disciplines - try convincing an audioligist that tests patients in their clinics day in day out that this is not the case.
  
 The example of infra red which can be seen, doesn't prove anything either.  The frequency range of human vision is also well understood and that range includes frequencies which some people call infra red.  In reality, it is not infra red in the true meaning of the word but the upper limit of light frequency which we can percieve.  A bit like saying that 17khz audio is "ultra sonic".  For example, you can see the "infra red" light, even if it is faint, on some remote controls but on most you cannot.  The ones you can see are within the human range of vision while those you can't see are just outside it.
  
 I agree that there can be differences in DACs behaving differently at different sample or bit rates.  But there is no technical reason why any competently implemented DAC should have this issue, even if some do have this issue.  Using your speaker analogy, there is no competently designed DAC that is non-linear so its reproduction of the frequency response in the human hearing range going to be much more perfect (broadly defined) than what you would get out of a speaker. In any event, non of this has anything to do with the issue at hand of human hearing limits, these are issues of hardware or software design and implementation.
  
 Lastly, the main issue I have with hi res audio is that it is distracting manufacturers and consumers on what are the real issues of concern with high fidelity and where R&D effort ans marketing should be focused, that is the design of speakers, room acoustics and most importantly, the mastering of the sound material in the first place.


----------



## dprimary

keithemo said:


> Actually I disagree.
> 
> If you actually try to find information about "the frequency range of human hearing" you will find that subject mentioned in a lot of books... and most of them seem to agree that everybody else seems to agree that "the commonly accepted range of human hearing is 20 Hz to 20 kHz". At one point I tried to research exactly where that number came from, and I reached a point where most books were simply quoting other books, or saying that "it was commonly accepted". Now, while it may in fact be true, I am always leery of things that "everybody knows", but nobody seems to want to quote the original research to substantiate. (So, while most people in the middle ages "knew" the world was flat because their priest said so, a lot of people today just seem to accept that "it's commonly known that the limit of human hearing is 20 Hz to 20 kHz", which to me seems a lot like the same blind acceptance of presumed authority. While I found plenty of books and references that states the limits of human hearing as "commonly accepted", I entirely failed to find mention of an actual test, performed by an author of one of those books, to confirm this "well known" information.) Considering that, in the last twenty or thirty years, a lot of "commonly known facts" have turned out NOT to be true, I'm not willing to simply accept this one at face value because it's been repeated a lot - for a very long time.
> 
> ...


 

 How far back did you go?
  
 Francis Galton was researching it in back in 1883 and had whistles made that went to 84k page 26 and page 252 he writes about them. Not much in details of what he found.
 http://www.mugu.com/galton/books/human-faculty/text/galton-1883-human-faculty-v4.pdf
  
 The Evolution of Human Hearing
 Bruce Masterton1, Henry Heffner1 and Richard Ravizza1 J. Acoust. Soc. Am. 45, 966 (1969)
 
Measured the upper limit to be 18KHz with threshold of audibility being 80dBSPL vs 0dBSPL for 4kHz where we are most sensitive.
 
There was tons of research done in the 50's you could spend months going through references back over 100 years.


----------



## dprimary

keithemo said:


> I agree there.... there are many different filter types and options - and they only tested one of them (and proving that a single filter is audible doesn't prove that others are). I would have also liked it if they had included a few more types of test equipment (since they basically used one DAC and one pair of speakers). I've always found electrostatic headphones to be the best thing for picking out minute audible differences, so that would have been my choice there.
> 
> As for the accusations that the authors "had an agenda".... personally I'm inclined to say that, if they did, I saw no evidence to that effect. (To put it bluntly, while the test produced a "statistically significant" result, it would hardly serve to convince people that there's some sort of significant and obvious difference worth paying for. In fact, it tended more to suggest that the difference was there, but was rather minor and difficult to hear.)
> 
> I have to admit that I'm at a bit of a disadvantage here in that classical music isn't what I normally listen to, so I personally would not be the most likely person to notice whether the sounds of specific instruments, or the sound of the ambiance in a real concert hall, were or were not "rendered accurately".


 
 I don't think I implied they had an agenda. The test did prove it well enough to warrant more study. I agree the differences for this and many things in audio are so minor most people would never pay for it. When you need complete concentration in a NC 20 room to hear the difference your market is very small.
  
 The piece of music they used is very well recorded. We could place the position of each member clearly. Every recording engineer in the room was impressed. We only listened to twice since we are trying to troubleshoot an issue with the monitors


----------



## RRod

dprimary said:


> I don't think I implied they had an agenda. The test did prove it well enough to warrant more study. I agree the differences for this and many things in audio are so minor most people would never pay for it. When you need complete concentration in a NC 20 room to hear the difference your market is very small.
> 
> The piece of music they used is very well recorded. We could place the position of each member clearly. Every recording engineer in the room was impressed. We only listened to twice since we are trying to troubleshoot an issue with the monitors


 
  
 What piece is it?


----------



## KeithEmo

old tech said:


> I agree with you on the point that there is no harm in obtaining a higher than 16/44 res recording, especially given that storage space is cheap. (*with one qualification that very high sample rates eg 192khz may introduce distortions into the playback system).  However doing so, in the belief that the higher res file on its own will result in an improvement in playback fidelity is misguided.
> 
> The examples you provided of how claims that the Dynaco Stereo 70 and Koss pro4AA's are prefect show how the science can be wrong is not valid.  That may have been the claim by some audiophiles, but there was no real science backing up that claim.  There are many issues of non-linearity in frequency response which mean that no transducers or room acoustics (in the average home) can be perfect. That is quite different to the hundred years of experiments which demonstrate that the range of human hearing falls between 20 to 20,000hz.  The latter is not merely an audiophile claim (indeed many audiophiles don't accept this, hence the demand for hi res), but accepted by all the science disciplines - try convincing an audioligist that tests patients in their clinics day in day out that this is not the case.
> 
> ...


 
  
 Actually I think the example of a Stereo 70 is absolutely pertinent. First, if you look around, you will see many claims that "there is no audible difference between tube and solid state electronics as long as the frequency response and THD remain below audible limits" - and the Stereo 70 would fit the criteria stated in those claims for "a tube amp that shouldn't sound audibly different from a solid state amp of equivalent power as long as you don't overload either one". However, my real point there was that, when most of the tests most people reference were actually performed, those were both "the latest equipment"... and, back when the Stereo 70 was current, a lot of people did in fact claim that "there was no point in doing any further development because it was plenty good to satisfy the abilities of human hearing" - and most of us no longer consider that claim to be true. In fact, that same claim has been made for tube amplifiers, vinyl recordings, cassette recordings, open reel recordings, and CDs... but opinions of whether it is true or not for each of them have changed over time. Perhaps, twenty years from now, people will look back and say "they were right - and CDs really are good enough to sound perfect within the limits of human hearing", but I'm not convinced about that - at least not yet. (Perhaps, instead, everyone will own a $20 pair of headphones - or some sort of other technology for listening to music entirely - through which the difference between CDs and high-res files is obvious. I read one interesting, but somewhat vague, paper claiming that humans had been confirmed to be able to hear well above 20 kHz - using bone conduction rather than "through the air conduction"- which bypasses the mechanisms of the middle ear - which they claimed was "what limited human hearing to 20 kHz".)
  
 As for DACs, I agree that no well-designed DAC should have high enough noise or distortion, or a frequency response far enough off-flat, that it should be audible. However, their transient responses can vary considerably depending on how their filter is designed, and I don't recall anyone doing any definitive tests about whether that is audible or not. And transient response is generally shown with an oscilloscope trace picture - so there is no single commonly accepted "spec" to compare. (There seems to be general agreement that time errors become audible at some point - but nobody seems to agree on where that line would be.)
  
 I definitely agree with what I consider your most important point - which (to me) is that the single biggest issue with most modern recordings is the mastering itself. Very few modern CDs are produced well enough that they sound anywhere near as good as the format is capable of. I also agree that, for most people, speakers and the acoustics of the room they're located in also probably make a much bigger difference.
  
 I do, however, disagree with what I guess would be the logistics of a few of your other statements.
  
 Assuming that "the music industry" was monolithic, I agree that I would rather see money spent on better production values and mastering than on higher resolution. However, the production industry is _not_ monolithic. The companies selling DACs are not the same companies who are producing albums. And the choice of whether to deliver a given master at 16/44k or 24/192k is merely a matter of picking a different setting (or, at worst, buying one new piece of equipment). In short, I don't see producing content at a higher resolution as "diverting funds from anywhere else". I also believe that the current obsession with high-res content, even if it turned out to be technically meaningless, is still "a step in the right direction", because at the very least it encourages people to pay attention to the technical aspects of the music they're listening to. (Given the choice, I'd rather have consumers wondering whether 24/192k sounds better rather than wondering if 128k MP3 files are "good enough for them". So I see the trend of simply paying attention to the production quality of the music as a good thing.) In other words, perhaps, if people really are paying attention to what the music they're buying sounds like, and are a little more demanding when they are asked to pay extra for a "high-res version", that will in fact encourage the industry to use better production values all around. (But I do agree that it won't help if people start assuming or imagining that the new version is better because it's high-res alone - to the point of ignoring whether it actually sounds good or not.)
  
 I'm also a firm believer in "trickle down technology".... the idea that, if manufacturers of players, and amplifiers, and speakers, work to make their top end products capable of playing flat to 40 kHz, just maybe the end result will be that even their low end products get a tiny bit better - as better technology becomes "the norm". (Maybe, if the DAC vendors get more orders for 24/192k DACs, they'll drop ones that don't even work well at 16/44k from the bottom of their product line, which will mean that you'll end up with a better DAC in the next $20 player you buy - because this year's "cheapest DAC you can buy" will be a little bit better than last year's.)


----------



## KeithEmo

dprimary said:


> How far back did you go?
> 
> Francis Galton was researching it in back in 1883 and had whistles made that went to 84k page 26 and page 252 he writes about them. Not much in details of what he found.
> http://www.mugu.com/galton/books/human-faculty/text/galton-1883-human-faculty-v4.pdf
> ...


 
  
 Actually, I was looking on the Internet for some specific references, and most of what I got when I searched on "limits of human hearing" and such things was references to relatively recent textbooks which, when I actually tracked them down, said things like "it is widely accepted" or "it is commonly believed" or simply referenced some other book in the footnotes... 
  
 And I have two basic problems with this situation.....
  
 First, it's usually bad when people start accepting things "because they're generally known" or "because everybody knows".... If I was someone new to the audiophile world, I could read on this forum how "everybody knows human hearing doesn't extend above 20 kHz", but I could read on a different thread that "everybody knows vinyl sounds better than digital", and in another venue entirely about all sorts of folk remedies that "everybody knows work".... and I'm pretty sure there's a thread pretty nearby where "everybody knows that high-def files sound better". So, without specific references to test results, how should I know which "everybody knows" is in fact correct? (The Internet seems to be a bad influence here - encouraging many people to simply repeat what they hear rather than to even take the effort to track down quotes and sources... and, in this case, the _most_ references you will find repeat the 20 Hz to 20 kHz number. People actually seem to forget that many of the entries on Wikipedia may well have been written by people who don't actually know any more than they do; and repetition has come to be associated with "truth".)
  
 Second, and related, is that we tend to generalize a bit too much... and things that are "generally true" may still not be "always true" (and many people seem to not make that distinction). If we want to make a claim that "most humans can't hear above 20 kHz", then we can safely test a few hundred people, and report those results as being "generally applicable to humans" - however they cannot be claimed to apply to _all_ humans. I don't expect to make it to 100 years old - yet there is a long list of folks who made it to 115 - so you cannot set "the top limit on human life expectancy at 100". (If you want to test whether _anyone_ can hear past 20 kHz, you'd be better off offering a $1000 prize to anyone who can demonstrate that they can hear 22 kHz - by which you avoid having to test everyone on Earth yourself by providing a clear incentive for subjects to self-select. And it would be easy enough to offer a prize to anyone who can hear the difference between a high-res file and the 16/44k equivalent; we could leave it to someone claiming the opposite to produce two otherwise identical files, offer the test subject his or her choice of equipment to listen to them with, and see whether the subject can distinguish them or not. And, if one single person comes forward who can reliably tell the difference, then we have to stop saying that "_nobody_ can".)
  
 (My main problem with the "direction" of this thread is that I believe that most of us are really more concerned with whether we, or perhaps most people, can hear a difference.... yet the question hasn't been phrased that way. I'm reminded of one of the early science experiments looking for "exotic" subatomic particles - involving a massive tank of cleaning solvent and a sensor looking for "the telltale flash" the particular particle would produce. I believe that, in the first several months, after a lot of observations, they detected _ONE_.... which still proved that the particle did in fact exist. I simply haven't seen this level of thoroughness expended looking for any single human who can hear above 20 kHz. If the real object here is to prove that high-res files don't make sense because _most _people are unable to actually hear the difference, then we can do so with a much smaller sample... and a lot less work.) 
  
 Incidentally, thanks for that Galton link - it is most interesting....


----------



## RRod

I don't doubt people exist who can hear 20kHz or maybe even a bit beyond. I'm sure my son hears higher than I do. "People can't hear over 20kHz" is just shorthand for the much longer, but more correct statement that would go something like:
 "A large percentage of adult human beings have an in-lab frequency hearing limit below 20kHz. The amplitude at which they can hear a tone at their limit will be substantially higher than the amplitude needed to hear tones in the main audible range, and thus their effective hearing limit at normal music-listening volumes will be lower than their lab value."
  
 You seem to want absolute truth, but that just isn't in the realm of any statistical procedure to give. If there were, at any point in time, 0.0001% of humans who could benefit from hi-res given their particular music-listening situation, then it's doubtful we're ever going to put in the $$ to figure that out statistically. And even if we did find one of these guys, we're still going to have some false-positive rate on the test, so we can never even be sure. Even this Meridian experiment with filters, if it is all legit, still has some possibility that all the positive results were by chance. The best one can do in a statistical environment is have your own standards on errors and verify to the limits of your ability the assumptions of your test, statistical or otherwise.


----------



## KeithEmo

rrod said:


> I don't doubt people exist who can hear 20kHz or maybe even a bit beyond. I'm sure my son hears higher than I do. "People can't hear over 20kHz" is just shorthand for the much longer, but more correct statement that would go something like:
> "A large percentage of adult human beings have an in-lab frequency hearing limit below 20kHz. The amplitude at which they can hear a tone at their limit will be substantially higher than the amplitude needed to hear tones in the main audible range, and thus their effective hearing limit at normal music-listening volumes will be lower than their lab value."
> 
> You seem to want absolute truth, but that just isn't in the realm of any statistical procedure to give. If there were, at any point in time, 0.0001% of humans who could benefit from hi-res given their particular music-listening situation, then it's doubtful we're ever going to put in the $$ to figure that out statistically. And even if we did find one of these guys, we're still going to have some false-positive rate on the test, so we can never even be sure. Even this Meridian experiment with filters, if it is all legit, still has some possibility that all the positive results were by chance. The best one can do in a statistical environment is have your own standards on errors and verify to the limits of your ability the assumptions of your test, statistical or otherwise.


 
  
 I agree absolutely. (If no difference was audible, you could conclude that neither the sample rate or the filters made an audible difference; but, if a difference is audible, I can't think of a way to eliminate the possibility that it's due to whatever sample rate conversion process was used. And, even if you were to record the same analog original using two of the same recorder, set to record at different sample rates, then how could you rule out the possibility that it performs differently in other ways at each sample rate, or that the DAC you're using does?)
  
 My point is that a lot of people posting on this thread seem to believe that "nobody can hear a difference between different sample rates - because, in order to do so, they would have to be able to hear past 20 kHz, which no human can do". That seems like an absolute statement to me - which we both seem to agree would be impossible to prove.
  
 Once we get past that sticking point, I'm quite prepared to agree with you that _most_ listeners, using _most_ equipment, and listening to _most _samples, are quite unlikely to be able to detect a difference. At which point I will also agree that, as far as an individual is concerned, it's up to them whether they want to test themselves, or buy high-res files on the chance that they can hear a difference, or simply decide whether to "buy the best - just in case" or "not pay extra for something they probably can't hear".
  
 As I've mentioned before, I'm inclined to "buy the best just in case", and I've found that many high-res remasters do sound better - although quite possibly the difference isn't due to the higher sample rate, which, to me, justifies the few extra bucks.


----------



## RRod

keithemo said:


> My point is that a lot of people posting on this thread seem to believe that "nobody can hear a difference between different sample rates - because, in order to do so, they would have to be able to hear past 20 kHz, which no human can do". That seems like an absolute statement to me - which we both seem to agree would be impossible to prove.
> Once we get past that sticking point, I'm quite prepared to agree with you that _most_ listeners, using _most_ equipment, and listening to _most _samples, are quite unlikely to be able to detect a difference. At which point I will also agree that, as far as an individual is concerned, it's up to them whether they want to test themselves, or buy high-res files on the chance that they can hear a difference, or simply decide whether to "buy the best - just in case" or "not pay extra for something they probably can't hear".
> 
> As I've mentioned before, I'm inclined to "buy the best just in case", and I've found that many high-res remasters do sound better - although quite possibly the difference isn't due to the higher sample rate, which, to me, justifies the few extra bucks.


 
  
 I think people are also thinking about audibility in the context of music, and thus being able to hear 20kHZ only at, say, +50dB over where you hear 1kHz is probably not going to cut it.
  
 The issue isn't that hi-res exists, it's the underlying pitch that "oh, NOW we can deliver you good masters because of hi-res," which is bunk.


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## castleofargh

I know I start losing it at 16.5khz, so I feel pretty confident when making claims about my own hearing ^_^. and I use mostly IEMs that roll off like mad before 15khz. and I get some audible noise floor from most amp sections that are far above 16bit quantization noise. so high res is useless to me, I can really make that claim. I lack all the circumstances that might create an audible difference. 
  
 the little problem with keeping an open mind and accepting people who say they can hear a difference( and to be honest for a great number of reasons, I'm sure a lot of people do hear "a" difference), is how the very huge majority of people making the claim happen to have never tested it under controlled conditions, and even better, usually are strongly opposed to blind testing. it gives a bad feel of "officer, I'm not lying, but I refuse to pass a lie detector test!".
 when on the other hand, people saying they can't hear a difference, happen to be a majority of people accepting blind testing and the need of removing bias. that's what makes me believe almost nobody hears a difference. not human hearing, not the number of people on each side, but how the ignorant people tend to rush toward the same side of the argument.  I guess it's a bias of its own for me ^_^.
  
 outside of blind testing, do I believe I hear a difference? absolutely!!!  but then again I feel that music is different after I take a ****. I'm the difference, not the music.
 I often talk about how I EQ something for a minute, to realize only at the end that I had the EQ bypassed, that's the kind of stuff I experience all the time without control, differences from suggestion instead of actual differences.
  I wouldn't trust myself after a sighted evaluation, so how could I trust some random guy on the net? I cannot.
 so I wouldn't say, nobody can tell the difference, because I don't know that. but I can say I have massive trust issues. and rejecting controlled testing is enough for me to believe the guy is a joke. does that make me narrow minded?


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## goodyfresh

castleofargh said:


> I know I start losing it at 16.5khz, so I feel pretty confident when making claims about my own hearing ^_^. and I use mostly IEMs that roll off like mad before 15khz.* and I get some audible noise floor from most amp sections that are far above 16bit quantization noise*. so high res is useless to me, I can really make that claim. I lack all the circumstances that might create an audible difference.
> 
> the little problem with keeping an open mind and accepting people who say they can hear a difference( and to be honest for a great number of reasons, I'm sure a lot of people do hear "a" difference), is how the very huge majority of people making the claim happen to have never tested it under controlled conditions, and even better, usually are strongly opposed to blind testing. it gives a bad feel of "officer, I'm not lying, but I refuse to pass a lie detector test!".
> when on the other hand, people saying they can't hear a difference, happen to be a majority of people accepting blind testing and the need of removing bias. that's what makes me believe almost nobody hears a difference. not human hearing, not the number of people on each side, but how the ignorant people tend to rush toward the same side of the argument.  I guess it's a bias of its own for me ^_^.
> ...


 

 One thing I lvoe about my Fiio X3ii is that it never seems to have ANY audible noise-floor with ANYTHING!   Also, your hearing and mine apparently top out at exactly the same frequency, 16.5Khz!  Haha.

 LMAO music is different after you take a ****, huh?  Maybe due to a resonance in your intestines?  AHAHAHAHA, I can't stop laughing after reading that, man you always manage to crack me up, castleofargh >_<
  
 It's not narrow-minded to not take folks seriously who reject controlled or blind testing.  Those people are the narrow-minded ones who are being ignorant.  Because when it comes to evaluating stuff like how we perceive the physical world around us (I made sure to include the word "physical" in order to preclude any potential consideration of spiritual perceptions, of course) the *only thing we can really rely on is the Scientific Method!  *Anything that is "established" without the use of proper scientific control is absolutely bunk, really.
  
 I've had teh same experience as you with EQ. . .set up a custom EQ curve, then played some music and was like "dude I can totally hear more bass now," or "wow the mids are so much more forward, NICE," only to realize 30 minuets later when I went to change the EQ again that I had it *bypassed the whole time*.  I heard more bass or mids because _I WAS LISTENING FOR MORE BASS OR MIDS.  _Hahahahahahaha.  The power of suggestion and placebo is very, very strong when it comes to the area of human hearing.  "ERMAHGERD HI-RES SOUNDS LIKE, SO MUCH BETTER, MAAAAN!"  Yeah, sure, when you KNOW it's high-res!  I'm not taking a hit off the communal high-res hookah/bong/blunt/whatever, sorry guys.  And neither is castleofargh, apparently *high five*


----------



## old tech

rrod said:


> I think people are also thinking about audibility in the context of music, and thus being able to hear 20kHZ only at, say, +50dB over where you hear 1kHz is probably not going to cut it.
> 
> The issue isn't that hi-res exists, it's the underlying pitch that "oh, NOW we can deliver you good masters because of hi-res," which is bunk.


 
 I think that is the main point.  I have no doubt that there are young people who can hear above 20khz.  The last time I saw an audiologist he told me of one kid who could hear up to 23khz, but two things stand out - they are very young and they are outliers.
  
 Even if you could hear up to 20khz, the sound would be so faint that it would be masked by other music content.  It is the same concept in regard to 16bit vs say 20bit.  The noise floor of 16bit is very low but still audible if you pick a silent passage and turn the stereo up loud (assuming no dithering has been applied).  It is however masked with music content so you wouldn't pick it out under any normal listening conditions.
  
 What is more relevant is our ability to discriminate within the music content as we age.  The effects of masking increases as we get older.  In healthy ears we normally do not notice the deterioration except in subtle ways such as needing to turn up the TV louder to better hear dialogue over background music, or more concentration required to follow a conversation in a noisy background such as in parties.  If we could instantly turn back the clock to how our hearing was when we were say 16, we'd be blown away with the extra clarity and detail in the music. No amount of hi res would compensate for this progressive loss in hearing detail.
  
 I hope that in part this illustrates why I believe that all this attention to hi res audio is futile.  Perhaps not in our time, but I think the biggest breakthrough in hi fidelity will be when genetic engineering allows a regeneration of our hearing to what it was when we were children or very young adults.  Who knows, the bio technology may even allow us to hear like dogs and then appreciate 24/96 playback.


----------



## old tech

keithemo said:


> Actually I think the example of a Stereo 70 is absolutely pertinent. First, if you look around, you will see many claims that "there is no audible difference between tube and solid state electronics as long as the frequency response and THD remain below audible limits" - and the Stereo 70 would fit the criteria stated in those claims for "a tube amp that shouldn't sound audibly different from a solid state amp of equivalent power as long as you don't overload either one". However, my real point there was that, when most of the tests most people reference were actually performed, those were both "the latest equipment"... and, back when the Stereo 70 was current, a lot of people did in fact claim that "there was no point in doing any further development because it was plenty good to satisfy the abilities of human hearing" - and most of us no longer consider that claim to be true. In fact, that same claim has been made for tube amplifiers, vinyl recordings, cassette recordings, open reel recordings, and CDs... but opinions of whether it is true or not for each of them have changed over time. Perhaps, twenty years from now, people will look back and say "they were right - and CDs really are good enough to sound perfect within the limits of human hearing", but I'm not convinced about that - at least not yet. (Perhaps, instead, everyone will own a $20 pair of headphones - or some sort of other technology for listening to music entirely - through which the difference between CDs and high-res files is obvious. I read one interesting, but somewhat vague, paper claiming that humans had been confirmed to be able to hear well above 20 kHz - using bone conduction rather than "through the air conduction"- which bypasses the mechanisms of the middle ear - which they claimed was "what limited human hearing to 20 kHz".)
> 
> As for DACs, I agree that no well-designed DAC should have high enough noise or distortion, or a frequency response far enough off-flat, that it should be audible. However, their transient responses can vary considerably depending on how their filter is designed, and I don't recall anyone doing any definitive tests about whether that is audible or not. And transient response is generally shown with an oscilloscope trace picture - so there is no single commonly accepted "spec" to compare. (There seems to be general agreement that time errors become audible at some point - but nobody seems to agree on where that line would be.)
> 
> ...


 
 Those claims made about the Stereo 70, valve amps etc may have been made in hi fi mags (most of which are very subjective and emotional) but I doubt that they had the backing of audio science.  That digital audio was developed and refined around the same time is a case in point.  I was around (just) when reel to reels were the benchmark with home audio (with the right tapes) and we whinged about the record player.  They perhaps were the best, or even perfection, for that time but I don't recall people saying that that no improvements were possible.  Even if they did, the measurements would give lie to that.
  
 I have heard of that bone conduction theory and it is just that, a theory without any supporting evidence.  The only study of peer review quality I am aware of that purported to find humans can hear or percieve ultrasonic content, is the Japanese study of the late 90s.  Professor Oohashi and his team designed an experiment which showed that humans can percieve (not hear though) ultrasonic sound.  That would have been a breakthrough finding pointing the way to further research.  Unfortunately, his peers were not able to replicate his findings and his methodology was later found to be flawed.  Oohashi later accepted that the study was flawed, though you still see his paper quoted in some Hi Res and vinylphile forums.
  
 I agree with what you say about the HiFi manufacturing industry not being homogenous, with some specialising in speakers or amps and others in DACs or ancillaries.  That is part of the problem.  A bit like cables, DAC and home playback digital technology has plataued.  To stave it from being commoditsed a whole lot of marketing tripe around hi res, high end DACs etc is employed so people keep spending and upgrading when it is not necessary. This is commercial expediency for products that really have matured. That is the point I was making.  An informed consumer would be priortising their HiFi spend on things that really matter ie speakers, room acoustics and quality mastered material.


----------



## dprimary

Here is the graph of measured hearing for mammals and man. As you can see when you reach the upper limit the sensitivity for all mammals drops dramatically. From the 1969 study. The more I read the old research 20KHz seem to be the outlier and not the norm for anyone over 20.


----------



## arnyk

keithemo said:


> Actually I think the example of a Stereo 70 is absolutely pertinent. First, if you look around, you will see many claims that "there is no audible difference between tube and solid state electronics as long as the frequency response and THD remain below audible limits" - and the Stereo 70 would fit the criteria stated in those claims for "a tube amp that shouldn't sound audibly different from a solid state amp of equivalent power as long as you don't overload either one".


 
  
 Simply not true.
  
 I'm not a relative newbie to this audio thing having owned several Stereo 70s back in the day.
  
 If you load it with pure resistive loads the FR of a Stereo 70 performs better than audible limits, but that's irrelevant to actual use with loudspeakers.
  
http://home.indy.net/~gregdunn/dynaco/components/ST70/hfst70.jpg
  
 Reprints High Fidelity magazine's 1959 technical review which puts its "Damping factor" at 9, which corresponds to a source impedance of approximately 1 ohm or worse (tested @ 8 ohm resistive load at a probable 1 KHz test frequency).
  
 Take a modern speaker that dips to 4 ohms in the audible range and you have an audible frequency response variation.  Take a look at the test results for nonlinear distortion and you have > 0.1 @ < rated output, which is again audible under a critical test.
  
 As a rule you have to find an exceptional, not average tubed amp to compare to SS in order to find "No audible difference".   The Stereo 70 may be the best selling tubed audio amp ever, and its performance wasn't bad for the day, but here's another example of why tubes fell out of favor among audiophiles who prefer sonically accurate performance.


----------



## KeithEmo

Exactly......  but a significant number of people still insist "the difference must be inaudible" based on those not-totally-relevant measurements.
  
 I consider myself to be an "absolute objectivist" - meaning that I do not believe that it's even _possible_ for an audible difference to exist that cannot be measured (if it exists, then it can be measured - because our current technology allows us to make measurements much more precisely, and over a much wider range, than we can hear). However, just because something can be measured doesn't mean that we're currently taking the correct measurements to "see" it.
  
 If you go to the right discussion group, you will still find people who ignore things like real-world speaker loading and insist that "the Stereo 70 must sound the same - because its noise floor and THD are inaudibly low", and those people insist that anyone who claims to hear a difference "must be imagining it" - based on those two single numbers - measured under one specific set of conditions. And they totally ignore all the other differences which are clearly visible when you use different tests - such as a load that simulates an actual speaker instead of a resistor. And, if you go to another forum, you'll find a group of people cheerfully declaring that "all good modern DACs sound exactly alike" - based on frequency response and S/N alone, and ignoring several other measurements which clearly show differences, based on an assumption that none of those other measurements are audible.
  
 My point is that the way our brains work to locate sounds in space, and to "extract" other information from what our ears pick up, is in fact rather complicated - and still not entirely understood. Therefore, I find the claim that "high-res files can't possibly sound different because the only difference is that they have better frequency response - and that difference only exists beyond the limit of audibility", to be on the same level as those claims that the Stereo 70 "couldn't possibly sound different"... both are based on incomplete information being quoted by people who may not understand that the information is in fact incomplete.
  
 True, there have been plenty of tests that show that a typical human being, under typical listening conditions, cannot consciously detect the presence or absence of frequencies above 20 kHz, but that is not at all the same as saying that "frequency response above 20 kHz is useless". In fact, there have been tests that concluded that subjects could detect a difference when an audio signal was bandwidth-limited to 20 kHz, which suggests that there might be something involved besides consciously "hearing" the presence of specific frequencies or not.
  
 (Part of the reason I can easily tell the difference between actual bright sunlight and a TV picture of sunlight is that the TV picture lacks the invisible spectrum of the long wave light frequencies we call "heat", so it's bright but it doesn't feel warm; so I guess that being unable to reproduce those invisible frequencies actually does reduce the "fidelity and accuracy" of a TV picture, and having a camera that would record them, and a TV that could play them, actually would be an improvement in fidelity - even though they are invisible. So I guess the jokes about "how silly it would be to make TVs that could reproduce light frequencies we can't see" are somewhat misguided.)
  
 So far there seem to be a lot of unknowns and undetermined details involved; perhaps the differences detected in those tests were in fact due to the sample rate conversions and filtering used to create the samples being audible; perhaps frequencies above 20 kHz, and very minute differences in timing which require frequency response above 20 kHz to record and reproduce accurately, while not audible as consciously detected sound, do have something to do with how we determine location. Or perhaps the tests results are simply artifacts of the filtering process used to produce the samples being audible. And, to be honest, I'd like to know..... but, more to the point, until we know for sure, it's premature to label high-resolution audio as "a hoax" or "clearly being different only in people's imaginations". I personally consider "anti-snake-oil" (the act of labeling something as snake oil before you know all the facts) to be technically as bad as the opposite (accepting every claim as true). I'd prefer to wait until a bit more information is available before making blanket assumptions like that.
   Quote:


arnyk said:


> Simply not true.
> 
> I'm not a relative newbie to this audio thing having owned several Stereo 70s back in the day.
> 
> ...


----------



## arnyk

Come on Keith, I proved you wrong (again) and all you've got to fall back on is a lot of apparantly unattributed anecdotes.
  
 I think I can come up with a proper attribution for those false claims - you are their author!
  
 I think that it would be good for you to take responsibility for them!


----------



## KeithEmo

arnyk said:


> Come on Keith, I proved you wrong (again) and all you've got to fall back on is a lot of apparantly unattributed anecdotes.
> 
> I think I can come up with a proper attribution for those false claims - you are their author!
> 
> I think that it would be good for you to take responsibility for them!


 
  
 I'm sorry - I seem to be missing something here - exactly what "claims" are you talking about ???
  
 You agree with me that a Stereo 70 in fact does sound quite different in many situations than an "equivalent solid state amp" - even assuming that both are being operated below clipping, and neither is generating what are generally considered to be "audible levels of distortion". We seem to be in agreement there.
  
 And, if you Google the subject, you will also find many discussions, reaching from the distant past to the present, arguing that the sole sonic differences between tubes and solid state are related to overload level and that, as long as you absolutely prevent the amplifier from clipping, you would not be able to hear an audible difference between those two amplifiers. (The subject was frequently discussed in magazines before the Internet became popular, but plenty of articles managed to get copied, referenced, and posted, and the topic is still discussed today; one side claiming that "tubes sound different because of x, y, and z" and the other claiming that, if you avoid overload, they don't sound different at all, and that anyone claiming that they do hear a difference "must be imagining it". And since, back in those days, "everybody knew that distortion below 0.5% or so in inaudible", the fact that the distortion spectra of the various types of amplifiers might be audibly different was generally dismissed as "one of those thing people imagined they were hearing".)
  
 I don't understand what "claims" I've made that you take exception to.....


----------



## El Zilcho

> Part of the reason I can easily tell the difference between actual bright sunlight and a TV picture of sunlight is that the TV picture lacks the invisible spectrum of the long wave light frequencies we call "heat", so it's bright but it doesn't feel warm; so I guess that being unable to reproduce those invisible frequencies actually does reduce the "fidelity and accuracy" of a TV picture, and having a camera that would record them, and a TV that could play them, actually would be an improvement in fidelity - even though they are invisible. So I guess the jokes about "how silly it would be to make TVs that could reproduce light frequencies we can't see" are somewhat misguided.


 
  
 A bit off topic, but I still think it would be totally silly for a TV to emit frequencies beyond the visible spectrum.  Yes, I can feel the heat of the Sun, but I wouldn't want that from my TV.  The last thing I would want is to get a Sunburn from my TV.  Or a painful burn on my eyes from looking at someone welding on TV (or the Sun, for that matter).  In terms of visible fidelity and accuracy, no, those things would not improve it.  Nobody is arguing that the TV is reproducing all the feelings of actually standing in the Sun when the Sun is on screen, it's only reproducing the images we detect with our eyes.  Just as a speaker or headphone is only reproducing the sounds we hear, nothing else.   
  
 IR and UV light in levels accurate to the original source on screen is not something I would want from my television (nor any of the other radiation the Sun emits, nor would I want a video of the ocean to flood my house, nor would I want to smell the stuff Mike Rowe works with), and I wouldn't equate it with "image" fidelity, because it isn't part of the image.  
  
 If there is some heretofore yet unknown impact upon our bodies from sound frequencies outside the audible range (at the dB levels produced by headphones/speakers at normal listening volumes), who is to say they are even desirable?  Honest question:  Are sounds at those frequencies (above 20kHz) even produced in a recording studio?  What's to reproduce or "feel" if they aren't even there in the first place?


----------



## RRod

el zilcho said:


> If there is some heretofore yet unknown impact upon our bodies from sound frequencies outside the audible range (at the dB levels produced by headphones/speakers at normal listening volumes), who is to say they are even desirable?  Honest question:  Are sounds at those frequencies (above 20kHz) even produced in a recording studio?  What's to reproduce or "feel" if they aren't even there in the first place?


 
  
 Frequencies that high are definitely produced and captured (a spectrogram of any legit hi-res track should show that). But certainly by 25kHz we aren't using our normal hearing mechanism to sense them, if indeed we are sensing them at all. Any effect is likely quite small (that's my conjecturing), and it is legitimate to question whether any effect they could be shown to have is in fact "musical."


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## KeithEmo

el zilcho said:


> A bit off topic, but I still think it would be totally silly for a TV to emit frequencies beyond the visible spectrum.  Yes, I can feel the heat of the Sun, but I wouldn't want that from my TV.  The last thing I would want is to get a Sunburn from my TV.  Or a painful burn on my eyes from looking at someone welding on TV (or the Sun, for that matter).  In terms of visible fidelity and accuracy, no, those things would not improve it.  Nobody is arguing that the TV is reproducing all the feelings of actually standing in the Sun when the Sun is on screen, it's only reproducing the images we detect with our eyes.  Just as a speaker or headphone is only reproducing the sounds we hear, nothing else.
> 
> IR and UV light in levels accurate to the original source on screen is not something I would want from my television (nor any of the other radiation the Sun emits, nor would I want a video of the ocean to flood my house, nor would I want to smell the stuff Mike Rowe works with), and I wouldn't equate it with "image" fidelity, because it isn't part of the image.
> 
> If there is some heretofore yet unknown impact upon our bodies from sound frequencies outside the audible range (at the dB levels produced by headphones/speakers at normal listening volumes), who is to say they are even desirable?  Honest question:  Are sounds at those frequencies (above 20kHz) even produced in a recording studio?  What's to reproduce or "feel" if they aren't even there in the first place?


 
  
 I'm inclined to agree with you about IR light - and I wouldn't want to actually feel the shock wave from a bomb blast in the movie I was watching either (although I can imagine someone complaining that that lack of IR was "why watching a sunset on TV just didn't seem real".) Personally, I think I would prefer a sort of compromise, where I would feel a bit of warmth - but with strict safety limits - and an option to disable it entirely.
  
 However, in the context of the current discussion, at least one study _seems_ to have shown that limiting the bandwidth of a high-res audio recording at least sometimes causes subjects to report a shift in the sound stage of the recording. The subjects didn't hear anything "missing" from the version that was band limited, or even claim that it sounded different, but they perceived the various instruments as occupying slightly different positions in the sound stage. The authors of the test suggested that, even though audio frequencies above 20 kHz weren't directly audible as sounds, perhaps some of the phase cues that we use to resolve the location of sounds cannot be accurately reproduced at the lower bandwidth. 
  
 One possible mechanism whereby that might occur is simple cancellation. If, for example, you play the same tone from two speakers near each other, the result will be a cancellation pattern (usually referred to as a comb filter) - and you will hear this pattern as a series of louder and quieter spots as you move your head from left to right. If you delay the sound being sent to one of those speakers by a tiny amount, the location of the nulls and nodes in that pattern will shift. And, under certain circumstances, even moving one of those speakers a fraction of an inch (which corresponds to a very tiny time shift - which would be inaudible by itself) may result in the pattern being shifted significantly more than the distance the speaker was moved. The authors of that particular article suggested that a sample rate of at least 50k would be required in order to ensure that the signal was reproduced accurately enough to ensure that this sort of alteration didn't occur.
  
 Please note that this was just one study, and I'm _NOT_ specifically endorsing the results; in fact it's quite possible that what they experienced was simply an artifact of the sample rate conversion - but I do take it as an indication that "all of the facts may not be in yet".
  
 In terms of technology, some microphones have response that extends well above 20 kHz, but many to not. Ditto for other studio equipment - some yes; some no. You also need to bear in mind that what information can be _resolved_ is not the same as frequency response alone. For example, if I had two microphones which were absolutely identical, and absolutely didn't respond above 20 kHz, I could still record a single instrument using a pair of them and have the sound being received by one of them 1/100,000 of a second sooner than the other - and that 1/100,000 of a second in difference in timing could be measured by measuring the time difference between the zero crossing points - so the information collected by both of the microphones, relying partly on their position relative to each other, could in fact contain information that either microphone separately couldn't record. I could then use an oscilloscope to determine the comparative location of the signal source and the two microphones. Of course, whether our ears can resolve this sort of detail is one of those things which I consider to still be somewhat undetermined so far. Likewise, a software program, designed to simulate echo or room reverberance, or to produce entirely artificial sounds, could generate "fake" information that extended well above 20 kHz. (As a trivial example, if I were to sample cymbals using a good microphone, then play that sample back backwards at 2x speed as a sound effect, then that altered sample would have frequency response that extended twice as high as the original microphone could record - as long as I did all of the processing at an appropriately high sample rate. In that case, you would absolutely require a bandwidth of at least 40 kHz to accurately reproduce either the minute time difference between those two microphones or all of the overtones in the frequency doubled cymbal (again ignoring whether limiting that would produce an audible difference or not).


----------



## Thad-E-Ginathom

Things that emit things are not at all limited to emitting the _human-sensible_ things.
  
 Do TV screens (of various technologies) emit _only_ the human visible part of the spectrum? I'd be surprised if that were the case. But people seem to be sensibly not-bothered about them. We have nobody selling us content, or machines to watch it on, that is supposedly better because it contains more UV or IR, and zero hot arguments about whether or not that makes a difference.


----------



## KeithEmo

thad-e-ginathom said:


> Things that emit things are not at all limited to emitting the _human-sensible_ things.
> 
> Do TV screens (of various technologies) emit _only_ the human visible part of the spectrum? I'd be surprised if that were the case. But people seem to be sensibly not-bothered about them. We have nobody selling us content, or machines to watch it on, that is supposedly better because it contains more UV or IR, and zero hot arguments about whether or not that makes a difference.


 
  
 The standards that apply to video recording itself specify a "color gamut" - which is the range of colors that are covered. That range is slightly different for the different video color standards, and is typically more than we humans can see in certain "color directions" and slightly less for other colors. Most computer monitors and TV screens, in turn, are limited to accurately portraying a significant percentage of the visible color spectrum (usually 80% or so). Some fancy computer monitors are actually rated to deliver 100%, or even slightly over, of the color range officially covered by the video standard - which is still somewhat less than what humans can see in certain directions.
  
 (The "graph" of color is usually portrayed as a warped triangle, with red, green, and blue at the three vertices. The graph of all colors is typically shown, with a triangle superimposed on top of it to show the range covered by a particular screen or video standard - or the range of colors most humans can typically see.)


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## KeithEmo

thad-e-ginathom said:


> Things that emit things are not at all limited to emitting the _human-sensible_ things.
> 
> Do TV screens (of various technologies) emit _only_ the human visible part of the spectrum? I'd be surprised if that were the case. But people seem to be sensibly not-bothered about them. We have nobody selling us content, or machines to watch it on, that is supposedly better because it contains more UV or IR, and zero hot arguments about whether or not that makes a difference.


 
  
 Incidentally, one of the virtues of the new 4k TV standard _other than resolution_ is that it supports a greater range of color resolution. (The normal HD TV standard actually doesn't support full resolution for all colors. For example, the green "color plane", to which our eyes are the most sensitive, is carried at full resolution; while the blue plane, to which we are relatively insensitive, is broadcast at lower resolution.) This has in fact been a "hot topic" in the video forums, as has a new standard called HDR, which supports greater dynamic range of both color and brightness than the basic 4k standard.
  
 (Since everybody knows that the regular video standards don't support the full range of colors we can see, and that computer monitors often cover a wider range, graphic artists often choose whether to pay the significant increase in cost to get a "wide gamut" monitor based on what they're using it for... but there's not much debate about whether the difference is visible since everyone agrees that it is - and under what circumstances.)


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## Thad-E-Ginathom

Keith, thanks. I didn't know the answer. Nor have I progressed beyond CRT technology for TVs, which I rather suspect emits more than visible light?
  
 I seem to remember that I shouldn't keep my CFL desk lamp (well, it was, it has an LED bulb in it now) closer than 12 inches to skin due to UV emission.
  
 Honestly, I should put ??? after almost every word, because I am stumbling in the dark here.
  
 There are probably better ways to have said that more is not always better, sometimes we get more, and know it is irrelevant... etc.
  
 (The only thing I've wondered about debatable stuff on the AV side of things is when I hear people talking about "blacker blacks." Isn't _black_, ultimately, all pixels off? I imagine they are seeing greater contrast. Or a gold and blue dress. Or something...)


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## goodyfresh

*But hearing is not like vision, at all*.  So such comparisons are silly.  Our vision is FAR more precise than our hearing.  Under the right conditions, the human eye and visual cortex can fully perceive a light source emitting only a few photons per second.  Seriously.  And our ability to resolve different shades of color is far more precise than our ability to resolve audio frequencies.


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## Don Hills

thad-e-ginathom said:


> Keith, thanks. I didn't know the answer. Nor have I progressed beyond CRT technology for TVs, which I rather suspect emits more than visible light?
> ...


 
  
 Standard CRT blue phosphors definitely emit near ultraviolet light, causing significant fluorescence in UV-fluorescing objects.


----------



## Thad-E-Ginathom

goodyfresh said:


> *But hearing is not like vision, at all*.  So such comparisons are silly.  Our vision is FAR more precise than our hearing.  Under the right conditions, the human eye and visual cortex can fully perceive a light source emitting only a few photons per second.  Seriously.  And our ability to resolve different shades of color is far more precise than our ability to resolve audio frequencies.


 
  
 And, only the fact that "high resolution" has, therefore, been accepted in the words of video as a legitimate idea, has allowed it to be palmed off on the audio world.
  
 Listening to music is supposed to relax the mind, not deaden it.


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## OddE

don hills said:


> Standard CRT blue phosphors definitely emit near ultraviolet light, causing significant fluorescence in UV-fluorescing objects.


 
  
 -Very much so. One of my colleagues at work loves to tell about the 20" CRT he was issued in the late eighties or so, to assist in his CAD work. Cost more than a half-decent car at the time, and emitted enough UV to give you a sunburn in the winter months when your skin didn't get exposed to much of the stuff from the sun.


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## arnyk

keithemo said:


> I'm sorry - I seem to be missing something here - exactly what "claims" are you talking about ???


 
  
 Since your posts typically only quote yourself, we must assume that everything in them is what you claim is true.
  
 For example, your claims about the Dyna 70 that I debunked.


----------



## KeithEmo

arnyk said:


> Since your posts typically only quote yourself, we must assume that everything in them is what you claim is true.
> 
> For example, your claims about the Dyna 70 that I debunked.


 
  
 My friend, you need to read more carefully.
  
 I didn't make any claims _AT ALL_ about the Stereo 70.
  
 What I said was that I've read a lot of articles claiming that we shouldn't be able to tell the difference between "a high quality tube amp and a solid state amp of equivalent power - if neither is driven into overload" - and that the Stereo 70 was often cited as being "a good tube amplifier", and so was used as an example. I think we can both agree that the Stereo 70 is a "typical good consumer tube amp of that era". (My point being that, back then, people were claiming that the distortion specs "proved" that there was no audible difference between amplifiers, which we now agree isn't true; and that, right now, we have people claiming that frequency response specs can "prove" that higher sample rates "can't possibly" sound audibly different from 16/44k. I see that as a parallel demonstrating people's tendency to make claims, and then support them with the numbers that are handy, whether those numbers justify the claim or not.) 
  
 Since I'm _NOT_ agreeing with the claims in those articles, and I've never agreed with the claim that "all amplifiers sound the same", I see no reason to cite any particular articles claiming that (or even to look them up).


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## arnyk

keithemo said:


> My friend, you need to read more carefully.
> 
> I didn't make any claims _AT ALL_ about the Stereo 70.
> 
> ...


 
  
 If your claims are not properly attributed to someone else, then sole authorship defaults to you, and the claim is therefore your own invention.


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## jcx

no one is claiming all amplifiers sound alike on distortion specs alone - frequency response is far more audible with limits fairly well established and in fact our sensitivity to frequency response variations very much smaller that most "subjectivist" control/match in listening
  
 amplifier output impedance causes additional frequency response changes with varying Z load - like most dynamic drivers
  
 while by itself you could just call the Carver Stereophile Challenge an anecdote there are lots of supporting Psychoacoustics results, Richard Clark's $10k challenge too
  
  
 the Carver Stereophile Challenge conditions, results need explaining by people wanting to use tube/ss "known audible difference" as a point in a argument
  
 Carver nulled his $600 SS amp against Stereophile's choice of "SOTA tube amp" and "idiosyncratic speakers" - largely with added output Z and FR trim of feedback - Stereophile's Golden Eared professional reviewers in their own room, their choice of music failed to distinguish the amps blind and level matched


----------



## goodyfresh

thad-e-ginathom said:


> And, only the fact that "high resolution" has, therefore, been accepted in the words of video as a legitimate idea, has allowed it to be palmed off on the audio world.
> 
> Listening to music is supposed to relax the mind, not deaden it.


 

 Yup.

 I guess the point here is that high-res video has a point, but high-res audio doesn't.  Our eyes can clearly see a difference between images and video with 24-bit vs. 32-bit colors and brightness levels, for example, but the idea of being able to hear a difference between 24-bit and 32-bit audio is just absurd.


----------



## OddE

goodyfresh said:


> (...) but the idea of being able to hear a difference between 24-bit and 32-bit audio is just absurd.


 
  
 -Oh, that's easy. The full utilization of 24 bits of dynamic range will leave you deaf in an instant.
  
 32 bits will leave you dead in an instant.
  
 See? Easy peasy.


----------



## goodyfresh

odde said:


> -Oh, that's easy. The full utilization of 24 bits of dynamic range will leave you deaf in an instant.
> 
> 32 bits will leave you dead in an instant.
> 
> See? Easy peasy.


 
 144 dB and 192 dB of dynamic-range, respectively.  So yeah, pretty much what you said 
	

	
	
		
		

		
			





 

 Let's keep in mind, guys, that the vast majority of music has no more than 20dB of dynamic range.
  
 That being said, bit-depth determines more than just the dynamic range of music.  Ever tried listening to an 8-bit music file?  If so, you should have been able to perceive that just because 8-bit can reproduce almost 50dB of dynamic range (more than enough for any and all music) does not mean it won't sound like crap.  That's because it also determines thingss like the level of the noise-floor relative to the main signal, amplitude of harmonic distortion, etc.


----------



## RRod

goodyfresh said:


> 144 dB and 192 dB of dynamic-range, respectively.  So yeah, pretty much what you said
> 
> 
> 
> ...


 
  
 I have a few metal tracks that I can take down to 8bit with no issues, but like you said it doesn't take much dynamism at all to get those 8-bit artifacts to pop through. It's still fun to look at DR/loudness range and try to predict if a track will work that low


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## goodyfresh

rrod said:


> I have a few metal tracks that I can take down to 8bit with no issues, but like you said it doesn't take much dynamism at all to get those 8-bit artifacts to pop through. It's still fun to look at DR/loudness range and try to predict if a track will work that low


 
 Glad to know I'm not the only guy who's nerdy enough to spend freetime playing around with a converter (dBpoweramp in my case), converting things down to 8-bit and then seeing what I can and what I can't hear a difference on 
	

	
	
		
		

		
		
	


	



  
 So seriously though.  24-bit and 32-bit.  Over 140 and 190 dB of dynamic-range, REALLY?  As you said, fully-utilizing that wouldn't just be stupid, it would be physically dangerous, and in the case of the over 190dB range for 32-bit, actually DEADLY.  As in, sound on that level would literally be the same thing as (the sound from) shockwave from a fricking nuke going off in your face, it would reduce human flesh, blood, and bone to an indeterminable pile of mush, lmao.  It wouldn't be the same as experiencing the shockwave itself, but would be like the sound from it, and the effect ont he human body would pretty much be the same, i.e. it'd get ripped to shreds.
  
 http://www.gcaudio.com/resources/howtos/loudness.html
  
 "140dB--Loudest sound recommended for exposure WITH hearing-protection."
 "180dB--Death of hearing tissue."
 "194dB--LOUDEST SOUND POSSIBLE!"
  
 Loudest sound possible means just that. . .sound at that level would create waves with such a large amplitude that their troughs would become actual VACUUM, in the atmosphere at average pressure at sea-level here on earth.  Any "sound" beyond 194dB in our atmosphere at sea-level would no longer actually be a soundwave, but rather an actual shockwave, in terms of the physics of what would be happening.  So that yes, sound in the range of 190 to 194dB would have the same general effect on living tissue as the shockwave from a large explosion.  Meaning that 32-bits of dynamic range could literally reduce a human being to nothing more than little teeny bits and pieces.  Lol.


----------



## KeithEmo

goodyfresh said:


> Glad to know I'm not the only guy who's nerdy enough to spend freetime playing around with a converter (dBpoweramp in my case), converting things down to 8-bit and then seeing what I can and what I can't hear a difference on
> 
> 
> 
> ...


 
  
 You're right - there is almost certainly no need for a dynamic range even as wide as that offered by 24 bits of depth in a recording that you're simply going to play as is. However, there are two "exceptions". First off, you need to differentiate between overall dynamic range and "local" dynamic range" (meaning dynamic range over a short time). In the case of listening to music, what that means is that we've all been assuming that you're going to turn the music on and let it play - so you're going set the level to where the loudest parts are comfortable (or, at least, non-lethal). However, not everyone listens to music that way all the time. So, for example, if you were to start listening to your favorite classical CD, with the level set so that the loudest parts were quite loud, the 16 bit CD audio would have plenty of dynamic range. But, if you manually turn it up 30 dB on the quietest spots "to hear them better", then the noise floor, or even digital artifacts, may be audible at that point - because, by raising the gain by 30 dB, you've also raised the noise floor by 30 dB (which makes the "effective S/N ratio" of your CD at that moment about 65 dB). So, if you're the kind of person who "turns it up to hear the details in the quiet spots", then using 24 bits, or even 32 bits, will prevent this from happening.
  
 The other case is really the same thing - but when you're making a recording or editing it. If you've ever made your own recordings, then you know that you can't always predict exactly how loud things will be in advance. In a real studio setting, with time for sound checks, this isn't usually a huge problem. However, in less formal settings, it's usually very difficult to predict what's going to happen, and not at all uncommon to find yourself diving for a level control when things get loud (assuming you prefer to avoid limiters and compressors). Room acoustics change - a lot - when the audience arrives, and suddenly finding yourself recording from the fifth row instead of the twentieth can make a huge difference. You may also find that things get a lot louder when the commentator stops talking and the music starts.
  
 If you've ever tried to record using a cassette deck, or a digital recorder that only supports 16 bit audio, then you know how difficult it can be to select a recording level low enough that you're positive you won't get any overload, yet high enough to avoid the noise floor of your recording equipment. Even though most micrphone preamps don't really have 96 dB of dynamic range anyway, between the various level controls on the mic preamp, the input pads on your equipment, and the unknowns of your source, having a recorder that operates at 24 bits makes it a lot easier to pick a level that's safely below even the remote possibility of overload, yet safely above the noise floor. (This is partly because you have more adjustment range to use, and partly because, even though portable recorders really don't have much more than 90 dB of "real" dynamic range, ones designated as 24 bit devices usually are quieter than 16 bit ones.)
  
 The same is a lot more true during editing - where different tracks may be boosted or cut in level - sometimes drastically. With editing, you also have the issue that any noise or artifacts that are present may be "magnified" when you add tracks together, and many types of processing actually boost the audibility of certain types of artifacts, as well as introduce new artifacts and mathematical rounding errors of their own. If you're mixing multiple tracks, recorded at different levels, and applying different processing to each, it's not unusual to raise and lower the level many times, and over a very wide range, over the entire process. And it's pretty obvious what's going to happen to the noise floor if you take a drum recording, recorded at a relatively low level to begin with, and boost the treble by 15 dB to "sharpen it up a bit". (I'm sure you've heard otherwise good sounding commercial recordings where the background noise jumped in level when a certain instrument or performer joined the mix; and various types of dynamic processing are well know to introduce "breathing" if the noise floor is even slightly audible before you apply them.) Making sure that the noise floor is _FAR_ below audibility provides a safety margin against this. This is why most audio editing software actually converts the audio to be edited to 32 bits, or even 64 bits, and operates on it  at that bit depth. 
  
 This isn't entirely limited to "process editing" either. For example, if you play a two-channel recording in simulated surround using Dolby PLIIx, you may find that certain artifacts and types of noise are exaggerated - because the decoding process "pulls" those sounds to the rear channels and raises their level. This was a major problem in the latter days of SQ4 four channel - where the typical decoder would specifically "misidentify" background noise and "place" it in the rear channels at boosted and varying levels - the result was that what sounded like relatively innocuous tape hiss or record surface noise in the front channels might end up sounding like a very audible and annoying desert sand storm in the rear channels. And, in modern equipment, you may find that your room correction system has "eaten up" 10 or 12 dB of dynamic range by boosting the high frequencies to compensate by a slight high frequency droop in your speakers, or a little extra absorption in your listening room.


----------



## RRod

keithemo said:


> You're right - there is almost certainly no need for a dynamic range even as wide as that offered by 24 bits of depth in a recording that you're simply going to play as is. However, there are two "exceptions". First off, you need to differentiate between overall dynamic range and "local" dynamic range" (meaning dynamic range over a short time). In the case of listening to music, what that means is that we've all been assuming that you're going to turn the music on and let it play - so you're going set the level to where the loudest parts are comfortable (or, at least, non-lethal). However, not everyone listens to music that way all the time. So, for example, if you were to start listening to your favorite classical CD, with the level set so that the loudest parts were quite loud, the 16 bit CD audio would have plenty of dynamic range. But, if you manually turn it up 30 dB on the quietest spots "to hear them better", then the noise floor, or even digital artifacts, may be audible at that point - because, by raising the gain by 30 dB, you've also raised the noise floor by 30 dB (which makes the "effective S/N ratio" of your CD at that moment about 65 dB). So, if you're the kind of person who "turns it up to hear the details in the quiet spots", then using 24 bits, or even 32 bits, will prevent this from happening.


 
  
 Wait, so now we're justifying 24-bits by an effect we can get from dynamic compression? I find that the only reason I turn up the volume to hear quiet parts is because they've fallen near my room noise, in which case the artifacts you talk about would have to punch through the same room noise.


----------



## KeithEmo

rrod said:


> Wait, so now we're justifying 24-bits by an effect we can get from dynamic compression? I find that the only reason I turn up the volume to hear quiet parts is because they've fallen near my room noise, in which case the artifacts you talk about would have to punch through the same room noise.


 
  
 For the sake of discussion, let's talk about a cassette, which might typically have a S/N ratio of 60 dB; and let's assume that the cassette is "well recorded", so the loudest sound on it really is recorded at 0 dB (actually an unrealistic situation for a cassette), and I'm listening to my tape at a nice comfortable level. Now, I happen to notice that there are very quiet voices in the background; those voices are at a level of -30 dB, which is only 30 dB above the noise floor on my cassette. Now, when the music stops, I crank the volume up 30 dB to make those voices loud enough to make out. And, at this point, with that 30 dB of extra gain, which boosts both the noise floor and the voices equally, my S/N is only 30 dB.
  
 However, if that recording was on a CD, which has a S/N of about 96 dB, and the voices are at the same -30 dB level, they are still 66 dB above the lower noise floor of the CD  (because the voices are at the same level but the noise floor is much lower). And, when I boost the level by 30 dB so I can hear the voices clearly, at that point I will still have a much more listenable S/N of 65 dB. (But, at a S/N of 65 dB, the noise will still be audible. However, with the -140 dB S/N of a 24 bit file, it would not.)
  
 The fact is that we humans actually _CAN_ hear a dynamic range of somewhere around 90 dB or 100 dB - if you allow for some time lag. If I walk into a well-designed concert hall, before the crowd arrives, I might actually be able to drop a dime and hear it hit the floor; and, when the full orchestra comes in and starts to play, at the loudest parts they may well be 100 dB louder that the sound of that dime hitting the floor; and I can hear both. It may take a minute or two for my ears to adjust between hearing one and hearing the other, but they could certainly both be included on a recording.... or that recording could contain both crashing crescendos from the full orchestra, interspersed by tiny tinkle taps of a 3" silver hammer on a tiny bell, not to mention the assorted breathing, chair bumping, and sounds of fingers contacting strings common in live recordings. (I'm sitting here in a very quiet office right now, and I can hear my own breath, and the sound it makes if I rub two fingers together a foot in front of me. I'll leave it to you to calculate the dynamic range it would take to accurately record both the sound of my fingers rubbing together and the slamming of an office door.) 
  
 It also so happen that we humans have evolved to be much more sensitive to certain sounds than others. For example, I personally wouldn't know if a cello sounded "perfect" or "a little bit off", but, like most of us, I am quite sensitive to the way human voice, and even breathing, sound - so I notice it if the intake of breath between words in a song sounds wrong. Likewise, many people are very sensitive to the low level sounds that make up "room ambiance", which are what allow us to recognize different rooms, or to hear with our eyes closed whether there are two people or a half dozen in our living room. Different people vary widely in terms of how sensitive they are to this sort of thing - which is one reason why I would "prefer to be safe than sorry" when it comes to accurately recording and reproducing "acoustic information". (I haven't heard very many recordings of a cymbal where I honestly couldn't tell with my eyes closed whether I was listening to a recording or a real cymbal - there in the room with me.... try it some time. To me this means that obviously _something_ is still not quite prefect in the process.)


----------



## goodyfresh

rrod said:


> Wait, so now we're justifying 24-bits* by an effect we can get from dynamic compression*? I find that the only reason I turn up the volume to hear quiet parts is because they've fallen near my room noise, in which case the artifacts you talk about would have to punch through the same room noise.


 
 Dynamic compression is absolute crap and I hate it   Haha.


----------



## RRod

keithemo said:


> For the sake of discussion, let's talk about a cassette, which might typically have a S/N ratio of 60 dB; and let's assume that the cassette is "well recorded", so the loudest sound on it really is recorded at 0 dB (actually an unrealistic situation for a cassette), and I'm listening to my tape at a nice comfortable level. Now, I happen to notice that there are very quiet voices in the background; those voices are at a level of -30 dB, which is only 30 dB above the noise floor on my cassette. Now, when the music stops, I crank the volume up 30 dB to make those voices loud enough to make out. And, at this point, with that 30 dB of extra gain, which boosts both the noise floor and the voices equally, my S/N is only 30 dB.
> 
> However, if that recording was on a CD, which has a S/N of about 96 dB, and the voices are at the same -30 dB level, they are still 66 dB above the lower noise floor of the CD  (because the voices are at the same level but the noise floor is much lower). And, when I boost the level by 30 dB so I can hear the voices clearly, at that point I will still have a much more listenable S/N of 65 dB. (But, at a S/N of 65 dB, the noise will still be audible. However, with the -140 dB S/N of a 24 bit file, it would not.)
> 
> ...


 
  
 Yes I get what you meant about hearing noise; I was pointing out realities of listening to music in actual normal environments. Going from fingers rubbing to a door slamming indeed takes a lot of dynamic range, but you are assuming that we actually hear the door slamming in a perfect way, when we in fact have mechanisms to protect us from sudden loud sounds (so how loud are we actually hearing the door slam)?
  
 Still it remains that listening to a recording by pumping up the volume at certain spots isn't how we listen to actual live music, where you just have to take what you are given. And there's the issue of timbre as well: a cello playing piano but with the volume cranked up does not sound like a cello playing forte with the volume at actually listening levels. So I don't really take to arguments about noise that require we "break the rules", as it were. I have no doubt many "passed" ABX tests online dealing with dynamism have been obtained by jacking up the volume.
  
 I was just listening to an album that is one of the more dynamic (high DR and loudness range) in my collection, and indeed if I jack up the very quietest parts I can hear noise. But when O Fortuna comes blaring back in I'm reaching for the volume, and that's in my quiet back room with sealed cans on. So really, how much more range do you want?
  
 The cymbal example seems to be leaping a bit from dynamic range to just sound in general. Many recordings will have the drums/cymbals manipulated separately from the rest of the instruments, which to me is probably a bigger culprit in why they might sound "off". But that's not a sample/bit spec issue.


----------



## RRod

goodyfresh said:


> Dynamic compression is absolute crap and I hate it   Haha.


 
  
 If done moderately it's quite useful, especially when you have a toddler running around who takes quite a few dB of dynamic range from the room


----------



## goodyfresh

rrod said:


> If done moderately it's quite useful, especially when you have a toddler running around who takes quite a few dB of dynamic range from the room


 

 OH well you didn't mention the part about the toddler before!  That changes EVERYTHING   Haha.


----------



## KeithEmo

rrod said:


> Yes I get what you meant about hearing noise; I was pointing out realities of listening to music in actual normal environments. Going from fingers rubbing to a door slamming indeed takes a lot of dynamic range, but you are assuming that we actually hear the door slamming in a perfect way, when we in fact have mechanisms to protect us from sudden loud sounds (so how loud are we actually hearing the door slam)?
> 
> Still it remains that listening to a recording by pumping up the volume at certain spots isn't how we listen to actual live music, where you just have to take what you are given. And there's the issue of timbre as well: a cello playing piano but with the volume cranked up does not sound like a cello playing forte with the volume at actually listening levels. So I don't really take to arguments about noise that require we "break the rules", as it were. I have no doubt many "passed" ABX tests online dealing with dynamism have been obtained by jacking up the volume.
> 
> ...


 
  
 I guess that all depends on what you expect from a recording.
  
 If I actually had "an all access pass" to a concert, I really might wander into the venue while it was still empty, sit down, and hear the sounds from the air conditioning system, and the scuff of the musicians' shoes as they sat down, and the rustle of papers being arranged on the music stands. At this point the "gain" of my ears would be at it maximum sensitivity - and at that moment I could probably hear that tiny sound made by my fingers rubbing together. Then, as the music started, my ears themselves would automatically reduce their gain so the loud parts weren't painful - at which point I wouldn't be able to hear my own breathing, or my fingers rubbing together. (I would be able to experience that huge dynamic range because of the automatic level control built into my ears.) However, if I had my recorder turned on from the moment I walked in, and I wanted to actually record and reproduce that entire experience, then my recorder and my playback system would in fact need to be able to reproduce that entire dynamic range as well.
  
 To look at it another way, the 0 dB to 140 dB SPL scale is specified as being the range from the quietest sound a human being can hear to the threshold of pain, so, arguably, in order to properly reproduce "every sound that a human can hear at the proper level", a recording (and the playback system that goes with it) really should be able to cover that range. And, if it doesn't, then there is room for improvement. In fact, as with most design criteria, I would prefer a safety margin, so my recording and playback system should have at least a little bit _more_ range than that. Cymbals are difficult to reproduce for a number of reasons, one of which is that they have a very wide dynamic range (the initial burst of energy when the drumstick hits the cymbal is massively powerful). Very few recording and reproduction chains can reproduce this successfully, so virtually every recording has at least some limiting and compression applied to the cymbal tracks, and the same holds true for the sharper drums like snare drums. (It also seems difficult to reproduce wire brush cymbals without their sound "blurring" in such a way that they sound like the simple burst of white noise produced by a steam valve, and I do hear significant differences in how different DACs reproduce them with some speakers - however, since I have one or two very good 16/44k recordings which seem to reproduce them perfectly, it obviously doesn't have anything to do with extended frequency response.)


----------



## RRod

keithemo said:


> I guess that all depends on what you expect from a recording.
> 
> If I actually had "an all access pass" to a concert, I really might wander into the venue while it was still empty, sit down, and hear the sounds from the air conditioning system, and the scuff of the musicians' shoes as they sat down, and the rustle of papers being arranged on the music stands. At this point the "gain" of my ears would be at it maximum sensitivity - and at that moment I could probably hear that tiny sound made by my fingers rubbing together. Then, as the music started, my ears themselves would automatically reduce their gain so the loud parts weren't painful - at which point I wouldn't be able to hear my own breathing, or my fingers rubbing together. (I would be able to experience that huge dynamic range because of the automatic level control built into my ears.) However, if I had my recorder turned on from the moment I walked in, and I wanted to actually record and reproduce that entire experience, then my recorder and my playback system would in fact need to be able to reproduce that entire dynamic range as well.
> 
> To look at it another way, the 0 dB to 140 dB SPL scale is specified as being the range from the quietest sound a human being can hear to the threshold of pain, so, arguably, in order to properly reproduce "every sound that a human can hear at the proper level", a recording (and the playback system that goes with it) really should be able to cover that range. And, if it doesn't, then there is room for improvement. In fact, as with most design criteria, I would prefer a safety margin, so my recording and playback system should have at least a little bit _more_ range than that. Cymbals are difficult to reproduce for a number of reasons, one of which is that they have a very wide dynamic range (the initial burst of energy when the drumstick hits the cymbal is massively powerful). Very few recording and reproduction chains can reproduce this successfully, so virtually every recording has at least some limiting and compression applied to the cymbal tracks, and the same holds true for the sharper drums like snare drums. (It also seems difficult to reproduce wire brush cymbals without their sound "blurring" in such a way that they sound like the simple burst of white noise produced by a steam valve, and I do hear significant differences in how different DACs reproduce them with some speakers - however, since I have one or two very good 16/44k recordings which seem to reproduce them perfectly, it obviously doesn't have anything to do with extended frequency response.)


 
  
 We need to decide if we're talking music or the gamut of hearing; I don't think anyone wants their music to have a 140dB dynamic range. I mean really, imagine surfing a 24-bit audio internet with cans that could deliver 24-bits sitting on your ears and tell me you actually want this. Besides, if I recorded my whole day there's no way I'd need 140dB for it. Even if I went to a metal concert tonight, I wouldn't need it, since I'm never in an environment that's much quieter than a library.
  
 Wanting a safety margin is of course sensible, and hence why you won't see much contention on here for recording at 24bit. But saying it's necessary for delivery isn't on as solid ground, especially with noise-shaped dithering out there.


----------



## KeithEmo

rrod said:


> We need to decide if we're talking music or the gamut of hearing; I don't think anyone wants their music to have a 140dB dynamic range. I mean really, imagine surfing a 24-bit audio internet with cans that could deliver 24-bits sitting on your ears and tell me you actually want this. Besides, if I recorded my whole day there's no way I'd need 140dB for it. Even if I went to a metal concert tonight, I wouldn't need it, since I'm never in an environment that's much quieter than a library.
> 
> Wanting a safety margin is of course sensible, and hence why you won't see much contention on here for recording at 24bit. But saying it's necessary for delivery isn't on as solid ground, especially with noise-shaped dithering out there.


 
  
 I agree - I don't think  anything past 16 bits is _NECESSARY_ for most people, in most situations, either.... by which I mean specifically that regular music, well recorded and properly mastered, can almost certainly sound good enough when recorded at 16 bits that there would be no audible difference by using 24 bits instead.
  
 I guess the difference is that I'm inclined to see safety margin as a necessity in and of itself. (If you had asked me, as an engineer, to design a system to reliably record a certain range of frequencies and a certain dynamic range, I would absolutely have set parameters somewhere around 50% _OUTSIDE_ the required range - and not touching it. In other words, if you had requested a frequency range of what humans can hear, and a dynamic rage of 90+ dB, I probably would have chosen 24/96k as "being able to deliver that with a reasonable safety margin".)
  
 I can understand how that didn't happen with CDs - because the constraints of the available technology barely allowed the inventors to squeeze an hour of music onto the requisite sized disc with a bandwidth of 20 Hz to 20 kHz. However, now that we have virtually unlimited bandwidth and dynamic range available, it seems a shame not to add a reasonable safety margin - just in case - simply to reduce costs by a (very) few cents. (It's sort of the same reason why, if I needed a two quarts of water, I'd probably fetch a full gallon jug, and then pour out the excess, rather than risk getting exactly two quarts and coming up short.)


----------



## RRod

keithemo said:


> I agree - I don't think  anything past 16 bits is _NECESSARY_ for most people, in most situations, either.... by which I mean specifically that regular music, well recorded and properly mastered, can almost certainly sound good enough when recorded at 16 bits that there would be no audible difference by using 24 bits instead.
> 
> I guess the difference is that I'm inclined to see safety margin as a necessity in and of itself. (If you had asked me, as an engineer, to design a system to reliably record a certain range of frequencies and a certain dynamic range, I would absolutely have set parameters somewhere around 50% _OUTSIDE_ the required range - and not touching it. In other words, if you had requested a frequency range of what humans can hear, and a dynamic rage of 90+ dB, I probably would have chosen 24/96k as "being able to deliver that with a reasonable safety margin".)
> 
> I can understand how that didn't happen with CDs - because the constraints of the available technology barely allowed the inventors to squeeze an hour of music onto the requisite sized disc with a bandwidth of 20 Hz to 20 kHz. However, now that we have virtually unlimited bandwidth and dynamic range available, it seems a shame not to add a reasonable safety margin - just in case - simply to reduce costs by a (very) few cents. (It's sort of the same reason why, if I needed a two quarts of water, I'd probably fetch a full gallon jug, and then pour out the excess, rather than risk getting exactly two quarts and coming up short.)


 
  
 The safety margin has existed on the recording side for many years now (when did the first 20-bit recordings start to happen?) And on the delivery side you have things like dithering and oversampling that have also been around for a while to help make things easier. And all of this was still not using up all that 16-bits could offer on the playback side, because having that much dynamic range in a recording is pretty much super annoying.
  
 Here's a screenshot of one of my more dynamic albums, recorded in 1985:

  
 I did the following:
 .found the softest section
 .gained it down 15dB without dither
 .gained it back and listened to the difference; nothing there to hear
 .put this back-to-back with the loudest section of the piece
  
Here's the result. I literally have to max out my volume to get that quiet section where I want it (in isolation), and there's no way I want the loud section to play at that level. The only option I have to possibly enjoy this would be to move to a quieter room, and I'm already at 37dBA in here. So yes, maybe in a 20dB room I might detect some differences from the gain, but that is not where about 99.999% of people listen to music (my out-of-the-air figure 
  
 And this is Mahler, fergawdssake. For much more normally dynamicked music, 16bits *is* already a safety margin. Now if recording engineers want even more safety for tracks like above, great. And if they don't want to bother with making 16-bit masters then fine. But don't charge me any more for it, which is really the brass tacks here.


----------



## nick_charles

rrod said:


> And this is Mahler, fergawdssake. For much more normally dynamicked music, 16bits *is* already a safety margin. Now if recording engineers want even more safety for tracks like above, great. And if they don't want to bother with making 16-bit masters then fine. But don't charge me any more for it, which is really the brass tacks here.


 
  
 LOL - Mahler is my go to for dynamic range demonstrations - even Mahler does not go much above 60db DR (10 bits give or take...)


----------



## RRod

nick_charles said:


> LOL - Mahler is my go to for dynamic range demonstrations - even Mahler does not go much above 60db DR (10 bits give or take...)


 
  
 Yeah, -65 to -70dbFS is the best RMS I've seen in the stuff I have.


----------



## castleofargh

rrod said:


> nick_charles said:
> 
> 
> > LOL - Mahler is my go to for dynamic range demonstrations - even Mahler does not go much above 60db DR (10 bits give or take...)
> ...


 

 it's easy to create something with more, it's also very easy to notice how annoying it is to listen to it. if it's instantaneous I tend to fail to abx the file compared to only the loudest song most of the time. and if I go for changes of loudness from calm moments to loud moments, then I can't hear half the song or I spend my time adjusting the volume from too loud to too quiet. if that's the ultimate purpose of highres sign me off.
 in fact I already don't listen to classical on my portable gears or in a car for that very reason, too much dynamic that would force me to listen too loud for my own good if I wanted to still get the quiet parts. for that job, stupidly compressed pop ironically ideal to me.


----------



## Thad-E-Ginathom

castleofargh said:


> ... ... ... I already don't listen to classical on my portable gears or in a car for that very reason, too much dynamic that would force me to listen too loud for my own good if I wanted to still get the quiet parts. for that job, stupidly compressed pop ironically ideal to me.


 
  
 Not only should they give us the tone controls back as standard but...
  
 Well, I was going to say that they should give us a compression knob too to cover these situations. But there used to be the "Loudness" button. I suppose that did something entirely different, in boosting the bass and treble, but it had the effect of making more of the music audible at low volumes.
  
 Like many of us, I do not want artificially compressed-to-hell music forced upon me, but that does not mean there is anything wrong with the tool. I was messing with this stuff on my PC the other day, and found I could compensate my poor hearing more effectively than I could with an Equaliser.
  
 Not _sure_ what I did though. The tool was the Calf Multiband Limiter.  Just messing around. It's fun, and sometimes useful


----------



## old tech

Another thing I find hard to understand is that the best 24track analogue recorders are generally accepted to be equivalent to about13bits. Yet, many analogue recordings are highly regarded and certainly good enough for well implemented CDs. So why so much discussion about possible playback benefits from 24bit compared with 16, when 16 already is higher than most master tapes?

That's got me thinking, and this is genuine question, if as one example the Led Zep original masters were made by a 13bit recorder, how does it become 24bit as a HD Tracks download?


----------



## OddE

old tech said:


> That's got me thinking, and this is genuine question, if as one example the Led Zep original masters were made by a 13bit recorder, how does it become 24bit as a HD Tracks download?


 
  
 -It leaves 11 bits to bring out every last nuance of the tape hiss...


----------



## arnyk

old tech said:


> Another thing I find hard to understand is that the best 24track analogue recorders are generally accepted to be equivalent to about13bits. Yet, many analogue recordings are highly regarded and certainly good enough for well implemented CDs. So why so much discussion about possible playback benefits from 24bit compared with 16, when 16 already is higher than most master tapes?
> 
> That's got me thinking, and this is genuine question, if as one example the Led Zep original masters were made by a 13bit recorder, how does it become 24bit as a HD Tracks download?


 
  
 If you do perceptual testing, you find that 13 bits is a pretty good fit with most real world recordings.  There is some need for careful level setting.
  
 16 bits is actually an overkill format and includes a reasonable margin for imprecise level setting.
  
 The rest is for people who either don't want to exercise care with level setting,  judge sound quality by means of sighted evaluations and/or like spec sheets with impressive-sounding numbers.
  
 Upsampling from 13 to > 13 bits is usually accomplished by  adding trailing zeroes to each sample in order to make the data words as long as is desired.


----------



## RRod

old tech said:


> So why so much discussion about possible playback benefits from 24bit compared with 16, when 16 already is higher than most master tapes?


 
  
 Because people are convinced of the myth of "microdynamics"; that is, that they are hearing the smaller gradations in quantization levels at 24bits compared to 16bits, regardless of the dynamic range and recording level of the music.


----------



## interpolate

This thread is still going? Oh well, stranger things have happened at sea.


----------



## goodyfresh

interpolate said:


> This thread is still going? Oh well, stranger things have happened at sea.


 






 People will never agree.  Some people will never accept the idea that blind-testing is the only true way to figure out the answers to questions like these, while other people (myself being one of them) will never accept the idea of subjective impressions based on non-blind testing being the least bit valid.


----------



## KeithEmo

I'm inclined to agree with you in general - however, I don't think anyone is charging more _because_ they're using 24 bit, or 32 bits, or whatever. The fact is that, if your master is already 24 bits, or 32 bits (which most programs I know use internally anyway), then you have it already. All you have to do is slap it directly into a file for distribution instead of re-saving it at 16 bits. And, while there was a space issue with CDs, that issue doesn't exist with downloads. Back when power steering was new, it cost extra, and, when I was young, you always paid a significant premium for air conditioning on a car - now both are standard. From that perspective "high-res files" are simply the latest "new and improved" version. And, when you think about it, a vinyl album that cost $10 or $15 in 1975 was a lot more expensive in real dollars than a $25 high-res download now. There's always going to be a "new" version, and a "premium" version, and at the moment high-res audio files fill that niche. (And are you really sure that the "premium" dishwasher detergent that costs 50% more is really 50% better? I really doubt it.)
  
 And, please, don't be misled by any blather about "bandwidth being expensive", and how high-res music files take up a lot of space to store on the server. The pay-per-view movie that you buy for $3 on cable takes up as much storage space and bandwidth as twenty five or thirty 24/96k albums. Sure, at some level, if the iTunes store was selling all high-res downloads instead of compressed ones, they'd have to buy bigger servers and faster Internet connections, but the _percentage_ of their operating budget that accounts for is minimal. Likewise, as I mentioned, every serious mastering program I know of operates internally at 24 or 32 bits, so nobody needs new equipment to record it. The reality is that the only one who pays more for high-res music is the consumer, and that's mostly simply because they're willing to do so. And, just like DVDs, and Blu-Ray discs, I would expect that, as soon as most downloads are 24/96, the price on those will drop back to "normal". (You may end up paying $10 for them or, more likely, just like restaurant prices, the prices won't actually drop, but they'll remain the same for another ten years as the value of the dollar drops "past" them.) In the end, you're going to end up paying whatever the market will bear for your music... and it has very little to do with cost.
  
 There are lots of other audio gadgets where you can pay a lot more for "high end" versions that really aren't, and for _VERY_ expensive tweaks that don't do anything at all, which is why, to me, whether high-res files really sound better doesn't seem like that much of a deal.... in fact, it's about as important as whether my alloy wheels really work any better than the punched steel hub caps available on the "regular" version of my car.
  
 Quote:


rrod said:


> The safety margin has existed on the recording side for many years now (when did the first 20-bit recordings start to happen?) And on the delivery side you have things like dithering and oversampling that have also been around for a while to help make things easier. And all of this was still not using up all that 16-bits could offer on the playback side, because having that much dynamic range in a recording is pretty much super annoying.
> 
> Here's a screenshot of one of my more dynamic albums, recorded in 1985:
> 
> ...


----------



## RRod

keithemo said:


> I'm inclined to agree with you in general - however, I don't think anyone is charging more _because_ they're using 24 bit, or 32 bits, or whatever. The fact is that, if your master is already 24 bits, or 32 bits (which most programs I know use internally anyway), then you have it already. All you have to do is slap it directly into a file for distribution instead of re-saving it at 16 bits. And, while there was a space issue with CDs, that issue doesn't exist with downloads. Back when power steering was new, it cost extra, and, when I was young, you always paid a significant premium for air conditioning on a car - now both are standard. From that perspective "high-res files" are simply the latest "new and improved" version. And, when you think about it, a vinyl album that cost $10 or $15 in 1975 was a lot more expensive in real dollars than a $25 high-res download now. There's always going to be a "new" version, and a "premium" version, and at the moment high-res audio files fill that niche. (And are you really sure that the "premium" dishwasher detergent that costs 50% more is really 50% better? I really doubt it.)
> 
> And, please, don't be misled by any blather about "bandwidth being expensive", and how high-res music files take up a lot of space to store on the server. The pay-per-view movie that you buy for $3 on cable takes up as much storage space and bandwidth as twenty five or thirty 24/96k albums. Sure, at some level, if the iTunes store was selling all high-res downloads instead of compressed ones, they'd have to buy bigger servers and faster Internet connections, but the _percentage_ of their operating budget that accounts for is minimal. Likewise, as I mentioned, every serious mastering program I know of operates internally at 24 or 32 bits, so nobody needs new equipment to record it. The reality is that the only one who pays more for high-res music is the consumer, and that's mostly simply because they're willing to do so. And, just like DVDs, and Blu-Ray discs, I would expect that, as soon as most downloads are 24/96, the price on those will drop back to "normal". (You may end up paying $10 for them or, more likely, just like restaurant prices, the prices won't actually drop, but they'll remain the same for another ten years as the value of the dollar drops "past" them.) In the end, you're going to end up paying whatever the market will bear for your music... and it has very little to do with cost.
> the
> There are lots of other audio gadgets where you can pay a lot more for "high end" versions that really aren't, and for _VERY_ expensive tweaks that don't do anything at all, which is why, to me, whether high-res files really sound better doesn't seem like that much of a deal.... in fact, it's about as important as whether my alloy wheels really work any better than the punched steel hub caps available on the "regular" version of my car.


 
  
 They're definitely using the higher specs to justify extra cost*. And like you said, now they don't even have to bother with Redbook mastering stuff. Add this to the whole audibility issue, and the only real thing they should charge more money for is a new mix/master with actual work done (a musical nose job, if you will). But there have been remasters released on CD before, and many of them cost just what a CD costs, not $25+ for a single-disc album. Once again, the car analogies don't work. Don't try to tell a Texas boy that hi-res is like getting air conditioning in your car 
	

	
	
		
		

		
		
	


	



  
 I agree that bandwidth issues aren't compelling. And I agree that prices are somewhat the fault of consumers, but that's partly because many of the consumers default to just believing that hi-res makes an audible difference, and the companies are happy to "confirm" that however they can. DVD made a visible difference from VHS, and Blu-ray made a visible difference from DVD, and it's easy to tell. That isn't the case with hi-res, so once again such between-senses analogies break down.
  
 *as an example, there is a $7 difference between the 24/96 and 24/192 versions of "Kind of Blue" on the HDTracks site.


----------



## KeithEmo

rrod said:


> They're definitely using the higher specs to justify extra cost*. And like you said, now they don't even have to bother with Redbook mastering stuff. Add this to the whole audibility issue, and the only real thing they should charge more money for is a new mix/master with actual work done (a musical nose job, if you will). But there have been remasters released on CD before, and many of them cost just what a CD costs, not $25+ for a single-disc album. Once again, the car analogies don't work. Don't try to tell a Texas boy that hi-res is like getting air conditioning in your car
> 
> 
> 
> ...


 
  
 We seem to be having two entirely different conversations here.
  
 First off, I actually do think that the car analogy works quite well. A lot of people spend a lot of money on various car "upgrade options" of dubious value. (Do you really think that half of the people who buy "the performance package" actually know if it in fact does improve performance? Do you think even half that many know which items in the package do what - or if they in fact do anything at all?) My point was that, in 1980, air conditioning was an expensive accessory in a car; today it's standard. Today 24/192k is a premium format; in ten years _it_ will be standard. A while ago, VHS was the standard, and DVD was the premium upgrade; in five years, Blu-Ray will be "the old standard" and 4k will cost $5 more - and, in five more years, 4k will be the standard. (And, while there have in fact been actual improvements at each step along the way, that's almost incidental to the process.)
  
 Second, while I suppose you might have some faint claim that someone who spends $25 for their _FIRST_ 24/192k album was misled, the most that's going to cost them is $10 to $15. After that, if they continue to buy high-def music because they really hear a difference, or because they imagine they hear a difference, or even just so they can impress their friends, how can you possibly say that they're doing so because they were misled or cheated? Nobody's holding a gun to their heads. Unless they're incredibly stupid, if they bought that first album and didn't hear any difference, and don't find any other justification, then they aren't going to be buying a second one. You may be able to argue that "they were tricked into buying" the _FIRST_ one but, after that, it's all them. In fact, most of the places that sell high-res albums offer at least one or two free sample downloads... so even the first purchase should have been made _AFTER_ they listened and decided for themselves if they heard a difference or not. What you're doing is a lot like accusing a company of selling overpriced expensive wine - when they offer a free taste to anyone planning to buy it.
  
 (Are you suggesting that people shouldn't be allowed to decide for themselves? 
	

	
	
		
		

		
		
	


	




 )


----------



## RRod

keithemo said:


> We seem to be having two entirely different conversations here.
> 
> First off, I actually do think that the car analogy works quite well. A lot of people spend a lot of money on various car "upgrade options" of dubious value. (Do you really think that half of the people who buy "the performance package" actually know if it in fact does improve performance? Do you think even half that many know which items in the package do what - or if they in fact do anything at all?) My point was that, in 1980, air conditioning was an expensive accessory in a car; today it's standard. Today 24/192k is a premium format; in ten years _it_ will be standard. A while ago, VHS was the standard, and DVD was the premium upgrade; in five years, Blu-Ray will be "the old standard" and 4k will cost $5 more - and, in five more years, 4k will be the standard. (And, while there have in fact been actual improvements at each step along the way, that's almost incidental to the process.)
> 
> ...


 
  
 You are leaving out the part where it's trivial to know if your car has AC versus knowing if you're hearing hi-res or not. Just because AC is standard today doesn't mean that back when it first came out people were like "I'm unsure if this vehicle has air conditioning, but here is my money anyway."
  
 On the second part, you are ignoring the part where people are being misled by the claims of hi-res. If you tell me my car will be cooler with AC than without in the heat of summer, you are not lying to me or in any way, shape, or form stretching the truth. You're also missing the part where people convince themselves they hear things to make their world-view work, so just saying "well they're stupid if they buy it again" is making things too simple. "Hey, this company that ostensibly knows about audio says this stuff sounds better, so I guess it sounds better. Here's more $$" ← don't tell me this doesn't happen.
  
 That companies are typically d***s and that consumers only do so much digging into literature are hardly things to debate, I'll agree.


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## Roly1650

keithemo said:


> We seem to be having two entirely different conversations here.
> 
> First off, I actually do think that the car analogy works quite well. A lot of people spend a lot of money on various car "upgrade options" of dubious value. (Do you really think that half of the people who buy "the performance package" actually know if it in fact does improve performance? Do you think even half that many know which items in the package do what - or if they in fact do anything at all?) My point was that, in 1980, air conditioning was an expensive accessory in a car; today it's standard. Today 24/192k is a premium format; in ten years _it_ will be standard. A while ago, VHS was the standard, and DVD was the premium upgrade; in five years, Blu-Ray will be "the old standard" and 4k will cost $5 more - and, in five more years, 4k will be the standard. (And, while there have in fact been actual improvements at each step along the way, that's almost incidental to the process.)
> 
> ...



If the download service tells you that the file is 24/192 before you download and after the download you know the content is no better than 16/44, what's that? Fraud would seem to be a word that springs to mind, but the download service is protected, because the file is 24/192 even though the content isn't. You don't think there should be protections for this rip-off?


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## Ruben123

keithemo said:


> We seem to be having two entirely different conversations here.
> 
> First off, I actually do think that the car analogy works quite well. A lot of people spend a lot of money on various car "upgrade options" of dubious value. (Do you really think that half of the people who buy "the performance package" actually know if it in fact does improve performance? Do you think even half that many know which items in the package do what - or if they in fact do anything at all?) My point was that, in 1980, air conditioning was an expensive accessory in a car; today it's standard. Today 24/192k is a premium format; in ten years _it_ will be standard. A while ago, VHS was the standard, and DVD was the premium upgrade; in five years, Blu-Ray will be "the old standard" and 4k will cost $5 more - and, in five more years, 4k will be the standard. (And, while there have in fact been actual improvements at each step along the way, that's almost incidental to the process.)
> 
> ...


 

 But airco at least offers something one can perceive. Any higher than CD will (almost?????) no one be able to differentiate from hi-res.


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## KeithEmo

rrod said:


> You are leaving out the part where it's trivial to know if your car has AC versus knowing if you're hearing hi-res or not. Just because AC is standard today doesn't mean that back when it first came out people were like "I'm unsure if this vehicle has air conditioning, but here is my money anyway."
> 
> On the second part, you are ignoring the part where people are being misled by the claims of hi-res. If you tell me my car will be cooler with AC than without in the heat of summer, you are not lying to me or in any way, shape, or form stretching the truth. You're also missing the part where people convince themselves they hear things to make their world-view work, so just saying "well they're stupid if they buy it again" is making things too simple. "Hey, this company that ostensibly knows about audio says this stuff sounds better, so I guess it sounds better. Here's more $$" ← don't tell me this doesn't happen.
> 
> That companies are typically d***s and that consumers only do so much digging into literature are hardly things to debate, I'll agree.


 
  
 As for the first part - I was just using that as an example of how every "new feature" (whether real or imagined) costs extra when it first appears, but quickly becomes "standard". Today you pay extra for CD quality music as compared to AAC or MP3 compressed content - and there is a "market price" for both "ordinary music" and "audiophile quality music"... and I see no reason to believe that this basic fact will ever change. Therefore, eventually, high-res files will fill the niche of "audiophile music" and be priced accordingly... but the price for that niche will level off at "what the market will bear" as it always does.
  
 I'm also simplifying the second point to actual reality. I agree entirely; people are very much influenced by their expectations. However, if you want to go that far, then let's take it all the way. If someone buys a high-res file because he or she thinks it will sound better; and then, when they play it, they actually _believe_ that it sounds better, and so enjoy it more, then haven't they in fact gotten their money's worth? Is an expensive restaurant a "cheat" because, thanks to the wonderful ambiance, you imagine the food tastes better? In fact, if someone paid $5 more for that high-res download, and actually enjoyed it $5 more because they deluded themselves into thinking they heard a difference, then aren't _YOU_ depriving them of that $5 of extra value by pointing out to them that they only imagined the difference? 
  
 I say that, if people _CHOOSE_ to base their worldview on what other people tell them, to the total exclusions of including their own personal experience, then they deserve what they get - to live in the world as other people imagine it to be.


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## RRod

keithemo said:


> As for the first part - I was just using that as an example of how every "new feature" (whether real or imagined) costs extra when it first appears, but quickly becomes "standard". Today you pay extra for CD quality music as compared to AAC or MP3 compressed content - and there is a "market price" for both "ordinary music" and "audiophile quality music"... and I see no reason to believe that this basic fact will ever change. Therefore, eventually, high-res files will fill the niche of "audiophile music" and be priced accordingly... but the price for that niche will level off at "what the market will bear" as it always does.
> 
> I'm also simplifying the second point to actual reality. I agree entirely; people are very much influenced by their expectations. However, if you want to go that far, then let's take it all the way. If someone buys a high-res file because he or she thinks it will sound better; and then, when they play it, they actually _believe_ that it sounds better, and so enjoy it more, then haven't they in fact gotten their money's worth? Is an expensive restaurant a "cheat" because, thanks to the wonderful ambiance, you imagine the food tastes better? In fact, if someone paid $5 more for that high-res download, and actually enjoyed it $5 more because they deluded themselves into thinking they heard a difference, then aren't _YOU_ depriving them of that $5 of extra value by pointing out to them that they only imagined the difference?
> 
> I say that, if people _CHOOSE_ to base their worldview on what other people tell them, to the total exclusions of including their own personal experience, then they deserve what they get - to live in the world as other people imagine it to be.


 
  
 On the first point: we're talking now about a paradigm where "audiophile quality music" is based on differences that are audibly small if at all existent. That's quite different from earlier paradigms were "audiophile quality" meant "has been worked over with special care to make a new mix", as in productions by the likes of MoFi (at least on the content end; hardware is its own can of worms). So yes, while the existence of an audiophile market persists, they way in which is differentiates itself from the "ordinary" market has changed. For instance, you can get AAC of the new Wilson mix of Aqualung for $10 on iTunes, but in hi-res for $18 on HDTracks. So what seems to be separating "ordinary" from "audiophile" in this case is $8 and content that most people probably can't hear, including audiophiles.
  
 On the second point: A restaurant with a wonderful ambiance at least has a wonderful ambiance. Our situation would be more like eating at the same restaurant, but having the chef tell you that he's using a new brand of iodized salt. And yes, if someone wants to pay $5 more for that, so be it. My pointing out to them that the salt probably blind-tastes the same as other salt isn't depriving them, it's giving them information upon which to base a decision. People can call that "crapping" all they want.


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## KeithEmo

roly1650 said:


> If the download service tells you that the file is 24/192 before you download and after the download you know the content is no better than 16/44, what's that? Fraud would seem to be a word that springs to mind, but the download service is protected, because the file is 24/192 even though the content isn't. You don't think there should be protections for this rip-off?


 
  
 It seems to me like what you're really claiming is that you personally see no benefit to the file being 24/192k - which is a whole different matter.
  
 If they said it was a 24/192k file and it wasn't, then that would be fraud. But the fact is that that's _NOT_ what we're talking about here. What they're selling you is a 24/192k file, with a frequency response flat from 10 Hz to 90 kHz, and a dynamic range of about 140 dB. As long as they didn't lie about any of the particulars, then it's up to _YOU_ to decide whether it suits your needs or not. Just because _you_ can't hear a difference does not mean that no difference exists. And, even if I can't hear a difference either, that _still_ doesn't mean that there isn't any. Perhaps I really do want to test whether my pet bat (who can hear to at least 50 kHz) responds better to Mozart than to Beethoven; or perhaps I'm using my AP test set to analyze the upper harmonics of a violin. The point is that, as long as they didn't misrepresent what the product _IS_, then they didn't defraud you.
  
 Common table salt is probably about 99% pure. But, somewhere out there, there's a chemical supply company that will happily sell you chemical sodium chloride that's pure to 99.999999%, and I'm sure they charge a lot more for it than your local supermarket charges for table salt. Would they be defrauding you if they sold it to you for $1000 a pound? No. In fact, if your local gourmet shop decided to sell it as "gourmet super purity salt", then they wouldn't be defrauding you either - because that's precisely what it is. And, even if they said "some of our gourmet customers insist that it tastes better", they _STILL_ wouldn't be defrauding you. And that's true because they have told you exactly what you're buying, which makes it your responsibility to decide whether you need it or not. (And, even if they said "it tastes better", that _still_ wouldn't be fraud - because "tastes better" is a matter of _OPINION_ and not fact. (Perhaps they really do think it tastes better - and, as long as they never claimed that xxx participants could pick it out in a blind taste test, which then turned out to not be true, then they haven't lied. I've seen lots of TV commercials claiming that this or that snack food is "delicious", and then decided, when I tasted it, that I didn't think it tasted good at all. However, I don't think there was any fraud involved.)
  
 There is a bit of as grey area - which has happened occasionally with high-res file vendors - where a real plain old 16/44k file was upsampled to 24/192k and sold as such. Now, in that case, you might have a case for claiming that they misrepresented the product, because it technically does not in fact legitimately "have the characteristics of a 24/192k file", and they have also clearly represented that it does. If so, then they've probably made some false advertising claims, and certainly a few misleading ones, which may be legally actionable.
  
 There is in fact a most powerful protection you can invoke to keep this from happening to you.....
  
 You should use your brain, do your homework, and decide for yourself whether a 24/192k file has any benefit _TO YOU_ and, if not, then you shouldn't buy them. You should also read all the fine print and make sure that they actually do claim, in writing, that the file was converted from the original master to 24/192k, and hasn't been processed in such a way as to negate the benefits of doing so. (Much like, if you do decide to buy some of that 99.999999% purity sodium chloride, you should make sure that the company you buy it from is prepared to certify that it has been handled properly to maintain its purity and avoid contamination.... and that they do so in such a way that their claim is legally binding - so you can sue them if you find out they're lying.)


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## KeithEmo

rrod said:


> On the first point: we're talking now about a paradigm where "audiophile quality music" is based on differences that are audibly small if at all existent. That's quite different from earlier paradigms were "audiophile quality" meant "has been worked over with special care to make a new mix", as in productions by the likes of MoFi (at least on the content end; hardware is its own can of worms). So yes, while the existence of an audiophile market persists, they way in which is differentiates itself from the "ordinary" market has changed. For instance, you can get AAC of the new Wilson mix of Aqualung for $10 on iTunes, but in hi-res for $18 on HDTracks. So what seems to be separating "ordinary" from "audiophile" in this case is $8 and content that most people probably can't hear, including audiophiles.
> 
> On the second point: A restaurant with a wonderful ambiance at least has a wonderful ambiance. Our situation would be more like eating at the same restaurant, but having the chef tell you that he's using a new brand of iodized salt. And yes, if someone wants to pay $5 more for that, so be it. My pointing out to them that the salt probably blind-tastes the same as other salt isn't depriving them, it's giving them information upon which to base a decision. People can call that "crapping" all they want.


 
  
 I agree entirely.... but it's a matter for the individual considering the purchase to find out what "audiophile quality" means, and then decide whether that matters to them or not. And, again, I have no sympathy if they simply decide to take the lazy way out and assume that it's better because a bunch of other people say it is.... or because it has a new and much cooler name.... Personally, I don't especially like driving vans, and calling them "sport utility vehicles" hasn't changed my mind; but I suspect that a lot of families who wouldn't have even considered buying a van now own a sport utility vehicle.... which is what marketing is all about. (And a lot of people with brand new "audiophile music players" will buy "audiophile music files" to go with them.)
  
 I'm not in the least suggesting that you shouldn't do your best to inform people of the facts, and even of your opinions.... Nor am I suggesting that it's at all unreasonable to say that, if they want to convince an informed audience that their product is in fact better, then the vendors selling these products would be well served to produce some test results to justify their claims. I'm just saying that the guys on the other side of the table aren't lying, or cheating, or committing fraud. (The reality is that more people are likely to be influenced by seeing a bunch of "cool people" using a product on TV, or finding out that it's endorsed by a popular celebrity, than are likely to be influenced by actual facts; that's just a fact of modern life.)


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## Roly1650

keithemo said:


> It seems to me like what you're really claiming is that you personally see no benefit to the file being 24/192k - which is a whole different matter.
> 
> If they said it was a 24/192k file and it wasn't, then that would be fraud. But the fact is that that's _NOT_ what we're talking about here. What they're selling you is a 24/192k file, with a frequency response flat from 10 Hz to 90 kHz, and a dynamic range of about 140 dB. As long as they didn't lie about any of the particulars, then it's up to _YOU_ to decide whether it suits your needs or not. Just because _you_ can't hear a difference does not mean that no difference exists. And, even if I can't hear a difference either, that _still_ doesn't mean that there isn't any. Perhaps I really do want to test whether my pet bat (who can hear to at least 50 kHz) responds better to Mozart than to Beethoven; or perhaps I'm using my AP test set to analyze the upper harmonics of a violin. The point is that, as long as they didn't misrepresent what the product _IS_, then they didn't defraud you.
> 
> ...



That's not what I'm claiming at all. What I can state with 100% certainty, having used my brain and done my homework, that none of the 5 or 6 albums I've downloaded from HDTracks are what they claim they are. They are no better than 16/44, while claiming to be 24/96 or 24/192. The excuse is that they are at the mercy of the record companies and only pass on what they are given, but the 2 Chesky Records I downloaded were the same. Chesky owns HDTracks, so who's kidding who? Upsampled is what it is.
Can they be had for fraud, which is what it is? I think a good lawyer would argue that they are 24/96/192, so where's the fraud? 
If you want to kid yourself that you're getting hi rez files when you download from HDTracks, don't let me burst your bubble, I ceased giving them my money a few years back. I suspect that Pono Music is exactly the same. It seems to be forgotten that to remaster even the most popular of the back catalog is a mammoth task, taking years. Does anybody really think the labels are going to do that? Real simplistic to think that's gonna happen.


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## Roly1650

I should add, that the Chesky Records albums are superb recordings, but they're 16/44 whatever HDTracks might claim to the contrary. How do I know? I can down/up sample and the files null, I can look at the spectrum and see there's nothing above about 21kHz. which isn't also about 100 dB down.


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## KeithEmo

roly1650 said:


> That's not what I'm claiming at all. What I can state with 100% certainty, having used my brain and done my homework, that none of the 5 or 6 albums I've downloaded from HDTracks are what they claim they are. They are no better than 16/44, while claiming to be 24/96 or 24/192. The excuse is that they are at the mercy of the record companies and only pass on what they are given, but the 2 Chesky Records I downloaded were the same. Chesky owns HDTracks, so who's kidding who? Upsampled is what it is.
> Can they be had for fraud, which is what it is? I think a good lawyer would argue that they are 24/96/192, so where's the fraud?
> If you want to kid yourself that you're getting hi rez files when you download from HDTracks, don't let me burst your bubble, I ceased giving them my money a few years back. I suspect that Pono Music is exactly the same. It seems to be forgotten that to remaster even the most popular of the back catalog is a mammoth task, taking years. Does anybody really think the labels are going to do that? Real simplistic to think that's gonna happen.


 
  
 I would say that, if you can _PROVE_ that the albums you were sold really were _NOT_ converted from analog to digital at the sample rate they claim, then you would have a case.
  
 However, it sound to me like you're confusing "remastering" and "remixing" and "re-converting".
  
 I agree with you that remixing an entire album is often a major project. The 24/192k re-releases of the Grateful Dead studio albums were re-mixed and restored. (There is a two or three page description of everything that was done to remaster the various albums included in the set on HDTracks website.) And, as a result, they do in fact sound _VERY_ different.
  
 When the majority of "high-res remasters" are created, they simply take the master tapes that were converted at 16/44k to make the original CDs and perform the conversion process again at 24/192k to produce a new 24/192k version. Assuming that you have access to the master tapes, and a proper A/D converter, this process shouldn't take more than a few hours. So, if an album is "simply remastered to 24/192k", then the only differences you should specifically expect to hear are those associated with the audible differences between the sample rates (which many people claim not to hear at all).
  
 Unfortunately, the word "remastering" is used to mean lots of different things. At this stage, the engineer may in fact also adjust the mix, or the EQ, or perform other "acts of restoration" - but those are a separate thing, and should be described. However, unless any specific claims to the contrary are made, what you should expect is simply a new conversion, starting from the same original master tape, and ending up at a new digital version at a higher sample rate.
  
 Note that this is _NOT_ the same as "upsampling" - which means actually taking the 44k digital version and converting it to a higher sample rate file directly. When an album is remastered, the conversion is performed at a higher sample rate, and contains more information at higher frequencies - which may or may not be audible - and may or may not consist of anything more than high frequency tape noise from the master tape.


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## sonitus mirus

Not sure how the original reel-to-reel tapes used in the Grateful Dead's studio recordings could magically make use of 24-bit audio.  16 bits was already a substantial overkill.  I've heard both the HD files and the lossy versions, and they both sound excellent.  I can't hear a difference, but I'm sure it is my inferior ears or inexpensive equipment I use.


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## Roly1650

T





keithemo said:


> I would say that, if you can _PROVE_ that the albums you were sold really were _NOT_ converted from analog to digital at the sample rate they claim, then you would have a case.
> 
> However, it sound to me like you're confusing "remastering" and "remixing" and "re-converting".
> 
> ...




Not in the least bit confused, reel in the condescension a tad......

Your "a few hours each" would equate, if your lucky, to 1,000 albums/person/year, which sounds like a drop in the bucket. The few data points I have, as opposed to ancedotal blah, blah, suggests that the labels are upsampling the cd master in the majority of cases. The thought of them employing rooms full of people re-recording, in real time, from the original tape is dreaming, imo. Their attitude is likely to be, "we did this crap already for the cd, just upsample for now, we can come back to it if needs be".

Sure, they'll provide the funding for a remaster of the "important" back catalog, if they see a chance of commercial success. At that stage, dependant on access to the multi-track master, an engineer can decide to become artist or the artists themselves can have an epithany and want to change things. But the key is access to the original master, some artists have retained ownership of these and denied use by third parties, some may not exist anymore and others are too far gone.


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## KeithEmo

sonitus mirus said:


> Not sure how the original reel-to-reel tapes used in the Grateful Dead's studio recordings could magically make use of 24-bit audio.  16 bits was already a substantial overkill.  I've heard both the HD files and the lossy versions, and they both sound excellent.  I can't hear a difference, but I'm sure it is my inferior ears or inexpensive equipment I use.


 
  
 I have no idea if the 24 bits makes any difference or not; and, for that matter, I haven't down-sampled them to 16/44k to see if that would hurt either. However, in this latest iteration, the re-masters have also been completely re-mixed, and, at least according to the literature, they actually adjusted the time stability, and did some other significant restoration on some of the albums as well. The re-mixing is pretty obvious and, at least to my ears, all for the better (which I don't always think is the case). In short, I think the improvements they made justified the price of buying the set again, so I'm not really concerned whether any of that improvement was due to it's being issued at 24/192k or not.
  
 As for lossy files..... I've been around through the invention and evolution of MP3 files, and I've done the whole "let's see which encoder and which setting works best with _this_ song" routine, and I simply don't have the time to waste. (Because of the way MP3 encoding works, even though the decoding process is standard, different encoders are allowed to make different decisions about what to discard, which means that, at the same exact settings, the output of two different encoders often sounds different. This means that, if you're determined to use lossy compression, it makes sense to encode each song with several different encoders and then decide which version you prefer - and, with luck, one of them won't sound noticeably worse than the original. Nowadays, storage space is cheap, and my time isn't, so it's not worth it to me to take the time to find out if I could store some of my music in a little less space.)
  
 Just for the record, I will say that I absolutely have bought 24/192k albums that didn't sound any better than the 16/44k original (I'm sure they were simply a re-conversion at a higher sample rate rather than a re-mix; and, either the sample rate really doesn't matter, or that particular master tape simply wasn't good enough for it to make a difference). However, this has always been the situation with "remasters" and "reissues" (some are better, some worse, and many not noticeably different), so I don't see this as specific to "high-res" reissues.


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## KeithEmo

You could be right - although, especially in this particular discussion, "anecdotal evidence" seems to be frowned upon.
  
 As someone who does understand the actual technology, I also have trouble with a lot of the anecdotal claims themselves. For example, many of the early studio mastering tape recorders simply didn't have frequency response past 20 kHz, so there won't be anything on the master tapes above there. If that's the case, then I can convert that tape to digital at 16/44k, or at 24/192k, and probably not end up with any more information (other than a more accurate rendition of the ultrasonic tape hiss and, perhaps, a cleaner recording of the residual bias frequency). However, while this suggests that encoding that particular tape at anything above 16/44k is "pointless and useless", it still doesn't mean that it wasn't done. There is some relatively simple software that can determine with pretty good accuracy, by looking for specific anomalies, whether an uncompressed audio file was "sourced" from an MP3 file. However, reliably determining whether a high-res file was really upsampled from a CD, or was simply made from a master tape that was of limited quality to begin with, is a little bit more difficult.
  
 Several years ago, the "high-res remaster" of a certain popular album was found to be an upsampling of an ordinary CD (I don't know how it was determined for sure). There was a proverbial "big stink" about it, apologies were made, along with the claim that it was "an error", and a new proper version was provided to everyone who had bought the bogus one. And I'm sure that it damaged sales figures for the companies who carried that label. I would have to suspect that, if any of the major labels was provably and consistently really selling simple upsampled copies, someone would have found out, made accusations, and published a major expose on the subject.
  
 Of course, another possibly controversial area is where a source, which could include an original 16/44k file, is "fixed" and then re-encoded at 24/96k or 24/192k. One example of this is the current version of Dolby's latest "professional encoder" - which is very commonly used to encode the audio tracks on Blu-Ray discs. One of the options this encoder offers is to re-encode content that was originally at 48k, using special filters to "repair" some of the ringing caused by the digital filters used in the original encoding process (they use digital processing to shift pre-ringing into mathematically equivalent, but less audible, post-ringing). Interestingly, the output of the encoder with this option enabled _MUST_ be taken at 24/96k (and they specify that the benefits of the improvement will be lost if you reduce that back to 48k). In the context of this discussion, however, what you have is a 96k version of a file, which began life as a 48k file, but has been processed in such a way that the result, which is claimed by Dolby to sound significantly better, can only be stored at 96k or higher. In other words, at least according to Dolby, you have an "upsampled copy" _which is_ _better than the original_ - because of the improvement their processing has wrought. (So, at least according to Dolby labs, an upsampled CD really _could_ be different - and better - than the original.)  
  
 You can find white papers about it on Dolby labs' website here:
 http://www.dolby.com/us/en/technologies/dolby-truehd-encoder-white-paper.pdf
 http://www.dolby.com/us/en/technologies/dolby-truehd-lossless-audio-performance-white-paper.pdf
  
  
 Quote:


roly1650 said:


> T
> Not in the least bit confused, reel in the condescension a tad......
> 
> Your "a few hours each" would equate, if your lucky, to 1,000 albums/person/year, which sounds like a drop in the bucket. The few data points I have, as opposed to ancedotal blah, blah, suggests that the labels are upsampling the cd master in the majority of cases. The thought of them employing rooms full of people re-recording, in real time, from the original tape is dreaming, imo. Their attitude is likely to be, "we did this crap already for the cd, just upsample for now, we can come back to it if needs be".
> ...


----------



## sonitus mirus

keithemo said:


> I have no idea if the 24 bits makes any difference or not; and, for that matter, I haven't down-sampled them to 16/44k to see if that would hurt either. However, in this latest iteration, the re-masters have also been completely re-mixed, and, at least according to the literature, they actually adjusted the time stability, and did some other significant restoration on some of the albums as well. The re-mixing is pretty obvious and, at least to my ears, all for the better (which I don't always think is the case). In short, I think the improvements they made justified the price of buying the set again, so I'm not really concerned whether any of that improvement was due to it's being issued at 24/192k or not.
> 
> As for lossy files..... I've been around through the invention and evolution of MP3 files, and I've done the whole "let's see which encoder and which setting works best with _this_ song" routine, and I simply don't have the time to waste. (Because of the way MP3 encoding works, even though the decoding process is standard, different encoders are allowed to make different decisions about what to discard, which means that, at the same exact settings, the output of two different encoders often sounds different. This means that, if you're determined to use lossy compression, it makes sense to encode each song with several different encoders and then decide which version you prefer - and, with luck, one of them won't sound noticeably worse than the original. Nowadays, storage space is cheap, and my time isn't, so it's not worth it to me to take the time to find out if I could store some of my music in a little less space.)
> 
> Just for the record, I will say that I absolutely have bought 24/192k albums that didn't sound any better than the 16/44k original (I'm sure they were simply a re-conversion at a higher sample rate rather than a re-mix; and, either the sample rate really doesn't matter, or that particular master tape simply wasn't good enough for it to make a difference). However, this has always been the situation with "remasters" and "reissues" (some are better, some worse, and many not noticeably different), so I don't see this as specific to "high-res" reissues.


 
  
 I'm not really struggling with the lossy conversion process at all.  I use the Lame encoder and have the command line parameters set at a quality level that I have, to this point, been unable to hear a difference in an ABX test with any music CD that I have ripped or HD file that I have converted.  My listening goal is to try and have a library that includes practically everything.  I want the equivalent of a huge Tower Records store filled with millions of albums at my fingertips.  I get this with the music streaming services, and I don't believe I am sacrificing sound quality for the convenience.  I've done a great deal of testing to prove this to myself.  In this process, I have also become quite skeptical of the HD audio industry and those that rave about it without also providing any type of verification other than suggesting that they trust their own ears.
  
 I was able to test the HD (24/96) version against the streaming MP3 files that Google Music subscription service uses.  (Lame version 3.98 320 CBR)
  
 I wanted to make sure I was not missing anything, and I could not hear any difference.  If there is a difference, it is more subtle than any differences found in one of the Philips Golden Ear challenge tests, which I can pass, albeit with considerable effort.


----------



## KeithEmo

I think the problem is that audiophiles tend to like to exaggerate.... the differences we're talking about are much smaller than, say, the differences between two different brands of speakers.
  
 Back when MP3 files were new, I compared the output of various MP3 decoders to the original (which was a CD; this was before high-res files). My conclusion was that, if I used a reasonable level - like 256 VBR or 320k CBR, on about 80% of 85% of them I didn't hear any difference at all; on another 10% or 15% I could hear a slight difference, if I listened carefully, but it wasn't huge; and, on a few specific recordings, the MP3 versions sounded somewhere between very different and absolutely awful.
  
 Which makes perfect sense if you realize that the MP3 format was designed based on the concept of "psychoacoustic masking" - which is a fancy way of saying that, if there's a loud sound, it keeps you from hearing the details of quieter sounds that are near it in frequency and time, so you can store those quieter sounds with much less resolution and probably not notice the difference. And the way the encoder decides what information it's safe to store at lower resolution is based on a whole bunch of research about exactly how this masking effect works. As a result, it really does work well with most music (because most music "fits the model"); however, with certain specific pieces of music which don't fit the model, it doesn't work well. (It's easy enough to deliberately construct a test signal where the difference will be obvious.)
  
 Also, since this model is based partly on what different people do and do not notice, different people react differently to the result. I personally find MP3 versions of vocals to sound very close to the original, but that cymbals often sound very wrong. Other people say that it tends to make the attack transient on piano notes sound odd, but I don't tend to notice that very much. And most people find that "electronic music", made up of pure tones, often gets altered quite a bit by the process. The effects also vary based on the specific frequencies present in the content, and on how long they're present, (so you may end up with a recording that sounds perfect - except for a few "odd spots" where there are obvious artifacts).
  
  
  
  


sonitus mirus said:


> I'm not really struggling with the lossy conversion process at all.  I use the Lame encoder and have the command line parameters set at a quality level that I have, to this point, been unable to hear a difference in an ABX test with any music CD that I have ripped or HD file that I have converted.  My listening goal is to try and have a library that includes practically everything.  I want the equivalent of a huge Tower Records store filled with millions of albums at my fingertips.  I get this with the music streaming services, and I don't believe I am sacrificing sound quality for the convenience.  I've done a great deal of testing to prove this to myself.  In this process, I have also become quite skeptical of the HD audio industry and those that rave about it without also providing any type of verification other than suggesting that they trust their own ears.
> 
> I was able to test the HD (24/96) version against the streaming MP3 files that Google Music subscription service uses.  (Lame version 3.98 320 CBR)
> 
> I wanted to make sure I was not missing anything, and I could not hear any difference.  If there is a difference, it is more subtle than any differences found in one of the Philips Golden Ear challenge tests, which I can pass, albeit with considerable effort.


----------



## sonitus mirus

keithemo said:


>


 
  
 Yes, your description of MP3 comparisons match what I typically find, though I am only able to hear these differences at much lower bitrates. I also listen at very low volume levels for extended periods of time, rather than a shorter, louder session, so I probably can get away with lossy formats easier than most.  I generally turn up the volume level when I attempt to ABX, but I have a difficult time hearing artifacts, regardless.


----------



## RRod

> I wanted to make sure I was not missing anything, and I could not hear any difference.  If there is a difference, it is more subtle than any differences found in one of the Philips Golden Ear challenge tests, which I can pass, albeit with considerable effort.


 
  
 Among the tests in the Golden Ear challenge, the frequency band and mp3 compression tests are often cited as the hardest. And the compression test doesn't go beyond 192 if memory serves. (edit: it goes up to 160 only).


keithemo said:


> Also, since this model is based partly on what different people do and do not notice, different people react differently to the result. I personally find MP3 versions of vocals to sound very close to the original, but that cymbals often sound very wrong. Other people say that it tends to make the attack transient on piano notes sound odd, but I don't tend to notice that very much. And most people find that "electronic music", made up of pure tones, often gets altered quite a bit by the process. The effects also vary based on the specific frequencies present in the content, and on how long they're present, (so you may end up with a recording that sounds perfect - except for a few "odd spots" where there are obvious artifacts).


 
  
 I think "very wrong" and "obvious artifacts" are quite a strong word choices unless you've tested things blindly. As I said above, the Golden Ears test is pretty hard, and indeed I've read many people focusing in on the transients to eek (eek, mind you) out the differences at bit rates lower than 320kbps.


----------



## Bmullock

Why do we assume that our ears are the only receptors to sound. Other parts of the body may accept higher frequencies and process them for the brain. Back to double bling crossover studies of a reasonable statistical sample if you want to get anywhere near the truth and remove selling hype!


----------



## castleofargh

bmullock said:


> Why do we assume that our ears are the only receptors to sound. Other parts of the body may accept higher frequencies and process them for the brain. Back to double bling crossover studies of a reasonable statistical sample if you want to get anywhere near the truth and remove selling hype!


 

 it's not about a maybe, we know the skin can feel ultrasound at very loud volumes. but we also know the ultrasonic content of the songs by looking at it, you can see that it's never even as loud as the audible range. so it's unlikely that it matters. and when it gets loud it's way up high and usually the result of noise shaping instead of actual music(and we better hope that we can't perceive it, else some DSD stuff would have made people cry).
 to that you can add the simple fact that when mastered, the sound engineer is but another guy, and so it's not like he heard, tuned, and mixed the ultrasounds like he did the audible range. so at best he most likely applied some attenuation, or didn't even care for what happened to the ultrasonic content. so the chances that it ends up having any meaning that correlate with the music are small, and that's being optimistic.
  
 I believe there is a difference between being open minded, and being concerned about useless stuff. if anything, it's the low frequencies that could be a game changer outside of the ears. I believe I heard Floyd Tool talk about how having something physically vibrating at low frequencies, even unrelated to the music content, had the effect of greatly improving the sense of realistic bass. for headphones that's something we might want to keep on looking. but ultrasounds... instruments have little ultrasound content, we don't perceive them unless very loud, and when very loud, because of how much energy they have, it's bad for our ears. so I say, away with ultrasounds!


----------



## CloudeKr

Oh come on this thread is degrading to a fight between audionerds with their barrages of logically flawed analogies and sciencenerds with practically no respect. It's the same thing with Apple products. People get that they're overpriced overmarketed and sensibily less practical but they buy them because they look good feel good and you got that special feel. Upwards from MP3, there's no difference in real quality that can be noticeably picked up, but cmon it feels good to have a dedicated kit playing music at that fidelity level. Don't ruin theists' days by saying that their gods don't exist, don't ruin audiophiles' by shoving these articles up their ass. It's true, logical, scientific, but not the least sensible and decent.

You'd expect half the people on head-fi to be like "wow youre right imma refund all my flac tracks and sell all my dedicated high fidelity players"


----------



## castleofargh

cloudekr said:


> Oh come on this thread is degrading to a fight between audionerds with their barrages of logically flawed analogies and sciencenerds with practically no respect. It's the same thing with Apple products. People get that they're overpriced overmarketed and sensibily less practical but they buy them because they look good feel good and you got that special feel. Upwards from MP3, there's no difference in real quality that can be noticeably picked up, but cmon it feels good to have a dedicated kit playing music at that fidelity level. Don't ruin theists' days by saying that their gods don't exist, don't ruin audiophiles' by shoving these articles up their ass. It's true, logical, scientific, but not the least sensible and decent.
> 
> You'd expect half the people on head-fi to be like "wow youre right imma refund all my flac tracks and sell all my dedicated high fidelity players"


 

 not sure it's all related. as you can see I'm one who couldn't care less for highres albums unless they cost the same as a 16bit version, but at the same time I find ludicrous to still buy old 16bit DACs with the cheap yet great stuff we now have. the improvement can go beyond playing highres files(if only for oversampling purpose or for a better handling of volume setting).
 so I'm really not one to spit on progress. but ultrasounds... that's not progress we need, but human mutation.


----------



## CloudeKr

castleofargh said:


> not sure it's all related. as you can see I'm one who couldn't care less for highres albums unless they cost the same as a 16bit version, but at the same time I find ludicrous to still buy old 16bit DACs with the cheap yet great stuff we now have. the improvement can go beyond playing highres files(if only for oversampling purpose or for a better handling of volume setting).
> so I'm really not one to spit on progress. but ultrasounds... that's not progress we need, but human mutation.



I didn't bother reading through individual posts. I just picked up the preemble and also noted this guy defending this hifi with an analogy of table salt. So I'm stating that in general. Cheers.


----------



## arnyk

keithemo said:


> I say that, if people _CHOOSE_ to base their worldview on what other people tell them, to the total exclusions of including their own personal experience, then they deserve what they get - to live in the world as other people imagine it to be.


 
  
 I say that I know an excluded middle argument when I see one.


----------



## KeithEmo

Actually my words were chosen quite intentionally. When I say "obvious artifacts" I'm talking about things like when something that plays as a distinct tone, which stops and starts suddenly, on the WAV file, has an obvious whanging reverb tail on the MP3, or when a wire brush cymbal "disappears" entirely into a long hissy blur, or when there's an obvious harmonic content that turns a pure tone into something else. I'm not talking about imagining that things sound "slightly off", or about some obvious and ongoing alteration; I'm talking about, when I play something that I'm very familiar with, like Dark Side of the Moon, over the course of a half hour or hour recording, hearing one or two spots with distinct "clinkers" - which are repeatable with a given encoded version. (Since individual encoders make slightly different choices, this will be different for copies encoded at the same bit rate using different encoders, or even the same encoder with different settings.)
  
 Incidentally, when comparing files at different bit rates, it is a reasonable argument (whether entirely true or not) that the differences "shouldn't" be audible because they occur outside what are "generally accepted" to be the limits of human audibility. The exact opposite situation holds for MP3 files. If you look at a spectral analysis of an MP3 file, or even at an oscilloscope image of the waveform, you will generally find that the MP3 file is very obviously different from the original - and that a lot of those differences fall squarely in the range of frequencies and distortion limits where they are clearly "audible" to humans; MP3 relies of the way our brain works to basically "hide" the huge errors "where we aren't listening" - and, considering how much damage is actually involved, does a remarkably good job. The problems arise for two reasons. First, because the model of what things will and will not be noticeable varies from person to person, and the encoder must use assumptions based on what "most" people will or won't notice or mind. And, second, because, since the algorithms are "purely machine based", they tend to miss "special cases" - where our brains are extra sensitive to certain types of errors in certain sounds. (Much like you probably wouldn't notice if a picture of your car had a distinct error, and showed five wheel lugs on each wheel instead of four, but, if a picture of your friends showed each with three eyes, it would be a glaring error. And, yes, the way MP3 encoding works _DOES_ allow for errors of that magnitude - although they're supposed to be hidden where they won't be noticed.)
  
  
 Quote:


rrod said:


> Among the tests in the Golden Ear challenge, the frequency band and mp3 compression tests are often cited as the hardest. And the compression test doesn't go beyond 192 if memory serves. (edit: it goes up to 160 only).
> 
> I think "very wrong" and "obvious artifacts" are quite a strong word choices unless you've tested things blindly. As I said above, the Golden Ears test is pretty hard, and indeed I've read many people focusing in on the transients to eek (eek, mind you) out the differences at bit rates lower than 320kbps.


----------



## RRod

keithemo said:


>


 
  
 Yes I know how MP3 encoding works. I also know that at 320kpbs the differences are quite low in amplitude, even for your much beloved cymbals, and that even they can be impossible to ABX for many people at that high a rate (and even harder for 256 AAC). Tell me how that works out to "obvious artifacts" in the context of actually listening to music? If you've tested this stuff blind (and without touching the volume) then fine, otherwise…
  
 And why do you keep bringing up different mp3 encoders? There's maybe a handful used, and their performance differences up at 320kpbs aren't much to talk about. It really just sounds like concern trolling. The question is why not use AAC or Vorbis anyway.


----------



## arnyk

keithemo said:


> Actually my words were chosen quite intentionally. When I say "obvious artifacts" I'm talking about things like when something that plays as a distinct tone, which stops and starts suddenly, on the WAV file, has an obvious whanging reverb tail on the MP3, or when a wire brush cymbal "disappears" entirely into a long hissy blur, or when there's an obvious harmonic content that turns a pure tone into something else. I'm not talking about imagining that things sound "slightly off", or about some obvious and ongoing alteration; I'm talking about, when I play something that I'm very familiar with, like Dark Side of the Moon, over the course of a half hour or hour recording, hearing one or two spots with distinct "clinkers" - which are repeatable with a given encoded version. (Since individual encoders make slightly different choices, this will be different for copies encoded at the same bit rate using different encoders, or even the same encoder with different settings.)


 
  
  
 Well Keith, that seems to pretty answer any any questions I might have had about whether or not you've actually done any good reliable listening tests related to any of the "Obvious artfacts" that you seem to go on and on about.


----------



## goodyfresh

castleofargh said:


> it's not about a maybe, we know the skin can feel ultrasound at very loud volumes. but we also know the ultrasonic content of the songs by looking at it, you can see that it's never even as loud as the audible range. so it's unlikely that it matters. and when it gets loud it's way up high and usually the result of noise shaping instead of actual music(and we better hope that we can't perceive it, else some DSD stuff would have made people cry).
> to that you can add the simple fact that when mastered, the sound engineer is but another guy, and so it's not like he heard, tuned, and mixed the ultrasounds like he did the audible range. so at best he most likely applied some attenuation, or didn't even care for what happened to the ultrasonic content. so the chances that it ends up having any meaning that correlate with the music are small, and that's being optimistic.
> 
> I believe there is a difference between being open minded, and being concerned about useless stuff. *if anything, it's the low frequencies that could be a game changer outside of the ears.* I believe I heard Floyd Tool talk about how having something physically vibrating at low frequencies, even unrelated to the music content, had the effect of greatly improving the sense of realistic bass. for headphones that's something we might want to keep on looking. but ultrasounds... instruments have little ultrasound content, we don't perceive them unless very loud, and when very loud, because of how much energy they have, it's bad for our ears. so I say, away with ultrasounds!


 
 I find it silly, as well, when people worry about such silly, useless things.

 And you are right about low frequencies.  It's a known fact that infrasounds (frequencies below 20Hz) actually have profound effects on the brainwaves and perception of human beings.  Ever heard of "the hum" which is in certain areas?  Or heard of places where people get an unexplainable feeling of dread, or other types of warped perceptions (even, in extreme cases, some very mild hallucinations) due to the presence of infrasounds in the area?  It's widely thought that a good portion of ghost-sightings and reported hauntings could actually be due to hallucinations caused by infrasounds (although these certainly couldn't account for more "vivid" so-called "apparitions," whatever they might be, meh).  So it's definitely true that vibrations of our skeleton and the bones of our inner-ears, caused by frequencies below 20Hz, can greatly effect human perception.  I have a feeling that's part of why the bass sounds so visceral ("felt" as much as heard) from my Sony MDR-1A headphones, despite the fact that while certainly bass-boosted, they are not basshead headphones. . .the bass from them sounds a good deal more visceral than from many other headphones that are far more bass-boosted.  The likely reason?  The MDR-1A has some kind of aluminum-coated LCP driver tech that apparently gives it INSANE levels of sub-bass extension all the way down to 4Hz (yes, the membrane is capable of accurately reproducing sounds at the level of only four vibrations per second).  In songs that are mastered to be bass-heavy, the bass is just. . .so PHYSICAL from those headphones.

 Ultrasounds, though. . .there is NO actual evidence that they have any meaningful impact on human perceptions except at EXTREMELY loud volumes.


----------



## KeithEmo

Actually
  
 https://www.soundonsound.com/sos/apr12/articles/lost-in-translation.htm
  
  


arnyk said:


> Well Keith, that seems to pretty answer any any questions I might have had about whether or not you've actually done any good reliable listening tests related to any of the "Obvious artfacts" that you seem to go on and on about.


 
  
 You're quite right.... and I have also entirely avoided doing any proper double-blind tests to ensure that the blue pen in my pocket is really a different color than the black one (I could be imagining the difference), and I trust my own eyes that cool white bulbs really are less yellow than warm white ones. There really is a point where something becomes obvious enough that actual testing is sort of a waste of time. One minute folks seem to be claiming that the subjective differences people claim to hear between 44k files and 192k files must be all in their imagination because the differences fall outside the range of audibility (so the specs are more important than subjective experience); the next minute, they're claiming that the major, and clearly measurable, differences in MP3 files - which do fall mostly within the audible range of frequencies and amplitudes - are also meaningless (so, this time, the subjective experience is more important than the numbers)... because psychoacoustics has assured us that, even though the differences should _physically_ be audible, they aren't _subjectively_ audible because of masking effects.
  
 (Since the whole science of psychoacoustics is really just based on statistical analysis of what people do and do not claim to hear, I'm confused as to why we should assume that the people who claim they can't hear all the measurable flaws in an MP3 file should be considered to be "credible", but the people who claim they hear differences between different sample rates should be ignored. It all starts to look to me like some people simply prefer to believe whatever "proof" agrees with what they already believe, and prefer to discount any "evidence" that proves the opposite. Could it possibly be that people only imagine that MP3 files sound OK - for the same reasons that they imagine they can hear the difference between 44k files and 192k files? or could it possibly be that they both have a point?   
	

	
	
		
		

		
			





 )
  
 Incidentally, here's a link to an article that describes, in precise detail, all of the flaws introduced by the MP3 encoding process - and what to listen for to hear them (in case anyone actually does want to listen for themselves):
  
 https://www.soundonsound.com/sos/apr12/articles/lost-in-translation.htm


----------



## RRod

keithemo said:


> (Since the whole science of psychoacoustics is really just based on statistical analysis of what people do and do not claim to hear, I'm confused as to why we should assume that the people who claim they can't hear all the measurable flaws in an MP3 file should be considered to be "credible", but the people who claim they hear differences between different sample rates should be ignored. It all starts to look to me like some people simply prefer to believe whatever "proof" agrees with what they already believe, and prefer to discount any "evidence" that proves the opposite. Could it possibly be that people only imagine that MP3 files sound OK - for the same reasons that they imagine they can hear the difference between 44k files and 192k files? or could it possibly be that they both have a point?
> 
> 
> 
> ...


 
  
  
 We've already done that ourselves, Keith, what don't you get about that? The differences between mp3 and WAV are clearly audible when *isolated*, that's trivially true. Anyone can make a difference file and verify that. The point is to hear them while music is playing at the volume where you actually listen to the music. You yourself are saying this but don't seem to listen to it.


----------



## sonitus mirus

keithemo said:


> Actually
> 
> https://www.soundonsound.com/sos/apr12/articles/lost-in-translation.htm
> 
> ...


 
  
 Nobody should seriously consider it to be credible evidence when someone claims they can't hear a difference between any lossy or 16-bit files and an equivalent 24-bit HD file.
  
 Anyone claiming that there is an obvious difference should be able to provide some proof, as the the tools and methodology exists to accomplish this with a reasonable degree of credibility.  Trusting one's own ears is simply just piling on to the mostly worthless anecdotes.


----------



## Thad-E-Ginathom

cloudekr said:


> Oh come on this thread is degrading to a fight between audionerds with their barrages of logically flawed analogies and sciencenerds with practically no respect. ... ... ...


 
  
 Are we _supposed_ to respect logically-flawed analogies? Sorry, nobody told me.
  
 Second best thing to do is to page down through the barrages, which I've done on the last few visits. _Best_ thing to do is to leave this thread now.  I don't think it offers anything much any longer, other than repeated head/wall collisions.
  
 <Unsubscribe>


----------



## KeithEmo

rrod said:


> We've already done that ourselves, Keith, what don't you get about that? The differences between mp3 and WAV are clearly audible when *isolated*, that's trivially true. Anyone can make a difference file and verify that. The point is to hear them while music is playing at the volume where you actually listen to the music. You yourself are saying this but don't seem to listen to it.


 
  
 What I don't get is how you can believe that 44k files and 192k files must logically sound the same - because no major quantifiable differences exist between them in what is agreed by most people to be the range of human hearing; yet you also seem to think that, even though there _ARE_ major differences between MP3 files and uncompressed files that _DO_ fall squarely inside the range of human hearing, you still think that they should also logically sound the same and the major flaws can safely be ignored.
  
 To put it bluntly, I want to see as much proof that MP3 files sound the same, as the amount of proof you seem to want to see that various sample rates sound different - and for the same reasons. (The same logic that suggests that different sample rates shouldn't sound audibly different suggests that MP3 files should - assuming that we are agreeing that flaws outside the range of human hearing don't matter, and flaws that fall inside that range do matter. Since, when you analyze an MP3 file, there are significant differences inside the range of human audibility, logic suggests to me that those differences should be very audible - and any claim to the contrary seems to require "extraordinary proof"... lest we just assume that people are only imagining that they don't hear a difference. And, while I've seen plenty of tests showing that MP3 files sound "acceptably good", I see very few tests claiming to prove that there is _NO_ audible difference. I've never heard of anyone even trying to determine the THD+N rating of an MP3 file - but I think we can all agree that it would fall _AT LEAST_ in the full-digit percentages - since "eliminating 50% of the information" equates to 50% THD+N.)
  
 I'm also willing to concede that 320k MP3 files will "sound audibly the same to most people most of the time" - but, as a matter of science, we're talking absolutes here.
  
 Incidentally, just for fun, I just made up a test file with 0.5% THD - where that 0.5% THD is _VERY_ clearly audible - even at reasonably low listening levels (so much for the claim that 0.5% THD is inaudible). If anyone wants to test that truism out for themselves, I'll be glad to post the instructions about how to do so for yourself.


----------



## sonitus mirus

keithemo said:


> Incidentally, just for fun, I just made up a test file with 0.5% THD - where that 0.5% THD is _VERY_ clearly audible - even at reasonably low listening levels (so much for the claim that 0.5% THD is inaudible). If anyone wants to test that truism out for themselves, I'll be glad to post the instructions about how to do so for yourself.


 
  
 I consider .5% to be high, but you didn't bother to post any relevant details about how this measurement was obtained.
  
 I have an inexpensive Peavey USB DAC that has the following THD specifications, and this is worse than anything else I have purchased in the last decade.
  
 THD + Noise: < .015% @ 1 kHz, -9 dBu full scale output into 10k Ohm


----------



## RRod

keithemo said:


> What I don't get is how you can believe that 44k files and 192k files must logically sound the same - because no major quantifiable differences exist between them in what is agreed by most people to be the range of human hearing; yet you also seem to think that, even though there _ARE_ major differences between MP3 files and uncompressed files that _DO_ fall squarely inside the range of human hearing, you still think that they should also logically sound the same and the major flaws can safely be ignored.
> 
> To put it bluntly, I want to see as much proof that MP3 files sound the same, as the amount of proof you seem to want to see that various sample rates sound different - and for the same reasons. (The same logic that suggests that different sample rates shouldn't sound audibly different suggests that MP3 files should - assuming that we are agreeing that flaws outside the range of human hearing don't matter, and flaws that fall inside that range do matter. Since, when you analyze an MP3 file, there are significant differences inside the range of human audibility, logic suggests to me that those differences should be very audible - and any claim to the contrary seems to require "extraordinary proof"... lest we just assume that people are only imagining that they don't hear a difference. And, while I've seen plenty of tests showing that MP3 files sound "acceptably good", I see very few tests claiming to prove that there is _NO_ audible difference. I've never heard of anyone even trying to determine the THD+N rating of an MP3 file - but I think we can all agree that it would fall _AT LEAST_ in the full-digit percentages - since "eliminating 50% of the information" equates to 50% THD+N.)
> 
> ...


 
  
 You're shifting goalposts. We took exception to your claim of "obvious" differences. That differences exist has never been in question, and that some people can spot differences in blind testing with certain material and with abilities like quick-switching at hand is exactly what codec developers depend on to make improvements. That doesn't change the fact that the second people have to do things blind at high bit rates, things get hard, especially with normal musical material, which is exactly where you have claimed to hear obvious differences.
  
 Your reasoning in:
 "when you analyze an MP3 file, there are significant differences inside the range of human audibility, logic suggests to me that those differences should be very audible"
 is flawed. If I reduce bit depth (24 to 16, whatever) at 44100, the differences will be within the range of human audibility. But those differences might not be audible within the context of the music. Try reducing any of your songs to 15 bits and tell me you can hear a difference in a blind test. The same thing goes for MP3; various perceptual phenomena like masking can prevent differences from being audible in the context of the file as a whole, even though the differences are audible in isolation.
  
 The 50% information thing is not an argument. Zipping a file reduces the file size by some percentage, but none of the information is lost. FLAC can get 50% or more compression and return the exact waveform.


----------



## castleofargh

keithemo said:


> rrod said:
> 
> 
> > We've already done that ourselves, Keith, what don't you get about that? The differences between mp3 and WAV are clearly audible when *isolated*, that's trivially true. Anyone can make a difference file and verify that. The point is to hear them while music is playing at the volume where you actually listen to the music. You yourself are saying this but don't seem to listen to it.
> ...


 

 0.5% THD that's about 45db below music, so not at all inaudible depending on what it is and what music content we have.
 just nulling MP3 320 and flac you usually get a good 10db below that, enough IMO to go from audible to slightly audible. and then you have to account for how mp3 tries to take advantage of masking when deciding to create such "loud" differences in the file. the parts that are not assumed to be masked would show differences much lower.
  
  
 some page before you mentioned doing a test when mp3 was still new. well the first iterations of mp3 weren't transparent at all IMO(in fact at the time I used ogg that sounded better to me). the old 320 kind of felt like a 192 now for me.
 it's really not all that easy to pass an abx with 320mp3 nowadays. not perfect transparency, but real damn close, at least for my abilities.


----------



## zareliman

rrod said:


> Yes I know how MP3 encoding works. I also know that at 320kpbs the differences are quite low in amplitude, even for your much beloved cymbals, and that even they can be impossible to ABX for many people at that high a rate (and even harder for 256 AAC). Tell me how that works out to "obvious artifacts" in the context of actually listening to music? If you've tested this stuff blind (and without touching the volume) then fine, otherwise…
> 
> And why do you keep bringing up different mp3 encoders? There's maybe a handful used, and their performance differences up at 320kpbs aren't much to talk about. It really just sounds like concern trolling. The question is why not use AAC or Vorbis anyway.


 
 Do you have a source for those ABX studies for lossy formats ?
 I'm trying to find the best lossy formats and ABX test myself. 256 AAC is more accurate to human ears than 320 mp3 or ~300 VBR OGG ?


----------



## RRod

zareliman said:


> Do you have a source for those ABX studies for lossy formats ?
> I'm trying to find the best lossy formats and ABX test myself. 256 AAC is more accurate to human ears than 320 mp3 or ~300 VBR OGG ?


 
  
 See here for various results. Here are results from the last test in that list (note they are using variable bit rates). As far as data from the horse's mouth, I have yet to e-mail Apple; I doubt I'd get much


----------



## KeithEmo

rrod said:


> You're shifting goalposts. We took exception to your claim of "obvious" differences. That differences exist has never been in question, and that some people can spot differences in blind testing with certain material and with abilities like quick-switching at hand is exactly what codec developers depend on to make improvements. That doesn't change the fact that the second people have to do things blind at high bit rates, things get hard, especially with normal musical material, which is exactly where you have claimed to hear obvious differences.
> 
> Your reasoning in:
> "when you analyze an MP3 file, there are significant differences inside the range of human audibility, logic suggests to me that those differences should be very audible"
> ...


 
  
 I'm inclined to agree with you on your first point - and I frequently don't notice a difference between a CD and a 320 VBR MP3 file either.
  
 The problem with perceptual masking is that it is based on generalities - which sometimes fail to hold true in specific situations. For example, it may well be true that in most music a loud sound at 2400 Hz will mask the presence of other sounds within 10% above that frequency and at levels 10 dB or more lower. However, that generality may not hold true with two distinct discrete tones; and, if the presence of that second tone is what makes a certain instrument sound like it should, then its absence could make a difference.
  
 Another issue that they mentioned in that article I linked to is that sometimes there are cumulative effects. For example, if you had a recording of a background of pink noise, with several discrete tones added, the MP3 algorithm would omit the background noise in areas of the spectrum where it was masked by the tones. However, if those tones were all at higher frequencies, then those omissions, while individually inaudible, might collectively add up to a reduction in the total high frequency content of the mix (in which case you wouldn't hear anything specific missing, but the entire mix would sound "less bright"). With algorithms which perform complex alterations on the sound, like an MP3 encoder, you always have to be on the lookout for equally complex interactions and "side effects".
  
 Your last statement is irrelevant to the current discussion. Zip and FLAC are lossless compression technologies, which means that you really do get back exactly what you put in. In direct contrast, the MP3 encoding process deliberately discards significant portions of the content, and the resulting output is _NOT_ the same as the original. At its highest compression setting, FLAC is said to be within 5% or 10% of the best lossless compression possible - which means that any size reduction that a given MP3 file achieves past that is at the cost of discarding actual information which cannot be gotten back once it's gone. If you look at an MP3 file on a spectrum analyzer or even an oscilloscope, the difference from the original is pretty obvious - the only question is how audible it is.


----------



## arnyk

keithemo said:


> Actually
> 
> https://www.soundonsound.com/sos/apr12/articles/lost-in-translation.htm
> 
> You're quite right.... and I have also entirely avoided doing any proper double-blind tests to ensure that the blue pen in my pocket is really a different color than the black one (I could be imagining the difference), and I trust my own eyes that cool white bulbs really are less yellow than warm white ones.


 
 Pretty typical of the responses we get from people who are in complete and total denial about their ear's insensitivity to small differences, right down to conflating visual and audible differences and grossly overestimating the size of common audible differences.


----------



## RRod

keithemo said:


> I'm inclined to agree with you on your first point - and I frequently don't notice a difference between a CD and a 320 VBR MP3 file either.
> 
> The problem with perceptual masking is that it is based on generalities - which sometimes fail to hold true in specific situations. For example, it may well be true that in most music a loud sound at 2400 Hz will mask the presence of other sounds within 10% above that frequency and at levels 10 dB or more lower. However, that generality may not hold true with two distinct discrete tones; and, if the presence of that second tone is what makes a certain instrument sound like it should, then its absence could make a difference.
> 
> ...


 
  
 The thing is the artifacts you talk about decrease in magnitude as you increase the bit-rate of the codec, so while you can call them a feature of lossy encoding, the devil is in the details. There are sections in difference files in 320mp3 where there has basically been no compression due to the decision of the perceptual engine. That will pretty much never happen at 128kpbs, though often that's the bit-rate people are referring to when they use the broad brush of "MP3." It's still really easy to listen to the difference file then switch over to the actual encoded track to see just how well you can notice differences, and just as easy to do an actual blind comparison to the original.
  
 My point on zip and flac is that you can't just assume compression means data loss. You were claiming 50% data loss, which would be equivalent to quantizing to 8-bits, yet somehow 320kbps MP3 sounds better than 8-bit WAV (note that mp3 uses Huffman coding as a last step). And I've already mentioned looking at difference files and their spectrograms a jillion times in the course of this thread, so let's just take that for a given at this point.


----------



## CloudeKr




----------



## arnyk

keithemo said:


> The problem with perceptual masking is that it is based on generalities - which sometimes fail to hold true in specific situations.


 
  
 Yet another global generalization with no reliable evidence to support it.
  
 The fundamental error in it is that unlike Mr. Emo's evidence-free global assertion, not all perceptual masking is the same.
  
 I'd like to see Mr. Emo list the generalities that perceptual masking is based on. He cites them, but is he just spreading fear and doubt?
  
 The general problem with Mr. Emos postings  is that they are based on generalities - which as he himself says sometimes fail to hold true in specific situations. If his postings are wrong for reasons that he cites, why believe them?
  
 There's a logical fallacy behind this - the well known adage that an exception does not break a rule. This is especially true if the exception is not stated, and it isn't stated here.


----------



## KeithEmo

rrod said:


> The thing is the artifacts you talk about decrease in magnitude as you increase the bit-rate of the codec, so while you can call them a feature of lossy encoding, the devil is in the details. There are sections in difference files in 320mp3 where there has basically been no compression due to the decision of the perceptual engine. That will pretty much never happen at 128kpbs, though often that's the bit-rate people are referring to when they use the broad brush of "MP3." It's still really easy to listen to the difference file then switch over to the actual encoded track to see just how well you can notice differences, and just as easy to do an actual blind comparison to the original.
> 
> My point on zip and flac is that you can't just assume compression means data loss. You were claiming 50% data loss, which would be equivalent to quantizing to 8-bits, yet somehow 320kbps MP3 sounds better than 8-bit WAV (note that mp3 uses Huffman coding as a last step). And I've already mentioned looking at difference files and their spectrograms a jillion times in the course of this thread, so let's just take that for a given at this point.


 
  
 In that case I agree with you 100%...
  
 I just wanted to avoid any confusion anyone else might have about the fact that lossy compression is intended to _NOT_ retain all the information. (If you ZIP a file, or compress it using FLAC, when you unzip it or play it, you _absolutely, positively_, will get back _exactly_ what you started with. So, for example, you can convert a CD into a FLAC, convert it back, and have _exactly_ the bits you started with; you _cannot_ do this with an MP3 file. If you use JPG to compress an image, or MP3 to compress audio, you _absolutely, positively_ will _NOT_ end up with exactly what you started with. We can discuss whether there is a visible or audible difference or not, or whether the analog output after you convert it is _significantly_ different, but the bits will definitely be changed from what you started with.) The fact that MP3's sound as good as they do is due to very careful selection of what to discard based on what we humans tend to notice (much as a Hollywood back lot set looks very good - as long as you don't walk around the propped up facades and see that there are no buildings behind them).
  
 Maybe other people don't look at it that way, but I personally tend to put MP3 and other lossy compression options in an entirely different category than things like different sample rates - probably because of where the process occurs. To me, when you're deciding whether to buy the 44k version of something or the 192k version, you are deciding whether to spend more money on more information that may or may not be significant or useful to you (in visual terms that would be equivalent to deciding whether you really want to pay extra for an ultra-high-resolution copy of an image that you plan to print out at a size of only six inches square). However, while I might share a photocopy of a picture with a friend, or record a copy of an otherwise totally unavailable song from the radio, or download an MP3 copy of it, I wouldn't consider hanging a photocopy on the wall, or putting that tape or MP3 file in my music collection, unless I had exhausted all my options for getting "a real copy". (And, if you were to buy a book online, represented simply as being "a book", you probably wouldn't be too happy if what you received was a pile of photocopied pages - even if they were _good_ photocopies.)
  
 Perhaps some of that viewpoint is simply due to my "historical" bias... During most of the time I purchased music, albums - and then CDs - were "the normal version". And, over most of that time period, converting a file to MP3 was something you did after you bought the CD - with the specific understanding that you were creating a reduced-quality copy because you needed to use it where size was more of a priority than quality. You took the CD or album you owned, and made a cassette or an MP3 copy of it to play in the car, or on a portable player. (You also might make a tape of a friend's album, or record the song off the radio, but only if you couldn't get a "real copy", and with the understanding that the copy wouldn't be as good as the original.)
  
 Also, historically, if you read the original literature from back when MP3 was a new format, you will find that even the inventors never envisioned it as being "audibly equal" to uncompressed formats. Rather, all of the original claims revolved around "a huge reduction in size without too much loss of quality". The whole idea that MP3, or AAC, files actually sounded as good as the original, at least to many people, only started to take hold when vendors started selling downloads in that format. (With the obvious dual motivation that compressed files take up less storage space and download bandwidth at the store, and that many of those downloaders were buying them to play on portable players anyway, and preferred to purchase their files already in the correct format for that purpose.)
  
 (So, if you want to note that it is the stores selling high-res files who expend the most effort convincing people that they're better, then it's only fair to note that the ones who started claiming that AAC and MP3 files were "just as good as uncompressed files" were the stores hoping to move product in that format, and who therefore had an obvious agenda to convince people that their product was good. Another obvious facet of that agenda is DRM - aka "copy protection". Uncompressed WAV files are virtually impossible to copy-protect, while the lossy compression format adopted by Apple in the iTunes store happens to have very powerful copy-protection features - which many people have suggested might have been "the real reason behind the choice". )


----------



## RRod

I tend to view the paradigm of lossy audio as simply an extension of the paradigm of lossless audio, only having a more complicated perceptual model (a constant 16/44.1 across a whole track is a pretty hammer-and-nail solution). "Cut at 20kHz since audiological evidence shows that to be the limit of human hearing" is a perceptual model. Noise-shaped dither is another perceptual model. Even the choice of bit depth itself could be viewed as perceptually-based, since it is based on human and environmental limits.
  
 JPG is a good comparison, and in fact has similar behavior to MP3: there comes a compression level where the artifacts are easily discernible, such that no real proper test is needed. But at the lowest compression level any casual observation will almost certainly fail to spot differences with anything but contrived examples.
  
 I agree that a paradigm of "give the consumer the original" is good (I was miffed my wedding photographer would only give us JPGs and not the RAWs). But for something like a streaming service, likely to be used often in sub-optimal listening conditions, I can't see why people make such an enormous fuss. Of course, the mantra of hi-res audio is that we should be *adding* to Redbook, not subtracting, but the fact that hi-rate lossy codecs can sound as good as they do should be a major question mark to such a mantra.
  
 I have no doubt that Apple has a monetary agenda for pushing the transparency of AAC, but I would remind you that just a few days back you rejected the notion that a paper by Meridian might be suspect, even though said paper was a prelude to their release of a hi-res codec. The truth is I don't trust either of them, which is why actual proper studies by third parties are important, and why I encourage people to do testing within the means they have on their own equipment.


----------



## KeithEmo

Since perceptual encoding in general, and MP3 in specific, have been around for quite a long time, you can generally find a chapter about how both work in most reasonably recent audio engineering textbooks. You will also find a truly amazing number of posts and websites on the subject - including papers written by the folks who invented it. Since I'm not especially interested in writing yet another article on the subject - and you undoubtedly won't believe me anyway - I'll leave you to do your own homework.
  
 You will find that, while most authorities agree that both frequency and temporal masking occur, there is a lot of disagreement on the details - and, further, that individual MP3 encoders use different mathematical algorithms - each based on their own specific interpretation of the process. (Each encoder is based on a specific interpretation of a specific version of "the model" - all of which are slightly different.)
  
 Some useful things to Google include "perceptual encoding" and "how MP3 encoding works" and perhaps "psychoacoustics".
  
 After you read a few of those, I'll be glad to participate in an intelligent discussion of the strengths, weaknesses, and flaws of MP3 encoding.
  
 The following article, while it doesn't go into a lot of detail on the process itself, covers the basics in reasonable detail...
   http://www.soundonsound.com/sos/may00/articles/mp3.htm
  
 Incidentally, triteness aside, if a "rule" has an exception, then either the rule is incorrect, or it is incomplete....
 I prefer to keep such approximations separate from actual facts whenever possible.
 (And I have no interest in using a method to compress my music "that audibly sounds absolutely perfect - except when it doesn't".)
  
 Quote:


arnyk said:


> Yet another global generalization with no reliable evidence to support it.
> 
> The fundamental error in it is that unlike Mr. Emo's evidence-free global assertion, not all perceptual masking is the same.
> 
> ...


----------



## sonitus mirus

There have been significant improvements to the MP3 encoders and there are mostly only 2 major MP3 encoders that are widely used today. (Lame or Fraunhofer)  Yes, there are limitations, but most of this is at lower bitrates, and at higher bitrates it is exceedingly rare to find any qualifying evidence to suggest anyone can hear obvious or even subtle differences.  With iTunes encoded AAC at 256 kbps, I have yet to see any reliable test results showing that someone can hear a difference between AAC 256 and Red Book.
  
 You will find lots of information on early MP3 encoders and their design, such as the link you provided from 2000.  Unless you are including the bitrates and the actual encoder version being used, your information, though technically accurate, can be highly misleading.
  
 MP3 and other lossy formats were brought into this discussion about 24-bit audio being "worthless" to emphasize the ridiculousness of some of the industry marketing claims and audiophile reviews.  If this is too distracting, why not bring the focus back to lossless 16-bit or CDs and just forget about lossy files completely with regards to this discussion?


----------



## KeithEmo

rrod said:


> I tend to view the paradigm of lossy audio as simply an extension of the paradigm of lossless audio, only having a more complicated perceptual model (a constant 16/44.1 across a whole track is a pretty hammer-and-nail solution). "Cut at 20kHz since audiological evidence shows that to be the limit of human hearing" is a perceptual model. Noise-shaped dither is another perceptual model. Even the choice of bit depth itself could be viewed as perceptually-based, since it is based on human and environmental limits.
> 
> JPG is a good comparison, and in fact has similar behavior to MP3: there comes a compression level where the artifacts are easily discernible, such that no real proper test is needed. But at the lowest compression level any casual observation will almost certainly fail to spot differences with anything but contrived examples.
> 
> ...


 
  
 I think we do actually mostly agree here.... but I would tend to extend the "I don't understand why most people make a fuss" to the extra requirements for high-res audio as well. We live in a world where people consider themselves "disadvantaged" if their Internet connection isn't fast enough to stream HD video - and next year that will move up to 4k. Compared to the storage space and transmission bandwidth for streaming HD video, the storage and download bandwidth requirements for MP3 and 44k are both negligible. And, as for making a fuss, a lot of the market does seem to be satisfied with iTunes and Spotify, both of which use lossy compression, so it's really only an audiophile minority who's making any fuss at all. (However, being audiophiles, some of us would still rather have the better quality copy - whether we can actually hear the difference or not.)
  
 I also agree that, strictly speaking, the price usually charged for high-res downloads isn't justified by cost. (My cable company can apparently store a full HD movie on their server, allow me to download it at full resolution to watch it, as many times as I want to over the next month, for a $3 on-demand fee; clearly the additional cost of letting me download a one-hour album once at 24/192k instead of 16/44k must be much less than that.) But, since we live in a capitalist society, that isn't a problem per-se - because price is set by supply and demand rather than cost. (So the $5 or $10 premium for high-res files has a lot to do with demand, and the market, and very little to do with cost.)
   
As for Meridian's paper - everybody has an agenda, so I don't necessarily accept anyone's conclusions at face value, but there's a difference between slanting your conclusions to fit your own agenda and outright falsifying data. I find it more interesting that, in their "official description", Meridian refers to their new CODEC as "high-resolution", and the term "lossless" is used quite a few times in the technical description of how the process works, while the actual process is in fact lossy. (They store all the information that would be contained in a normal 24/44k recording losslessly, then process the information that falls between the range covered by 24/44k and that covered by 24/192k, losslessly compressing and keeping the information in ranges they consider "significant", while discarding any content in some other areas that they consider to "contain only noise and no useful audio information" to reduce file size. (SInce they do in fact discard some content, the process overall is _NOT_ lossless.)

  
 Since the CODEC is lossless "up to 44k", and also contains information outside that range that certainly qualifies it as being "high resolution", it should logically perform at least as well as 24/44k, and potentially also offer many of the benefits of higher-resolution files (assuming such exist). However, referring to it as a "lossless high resolution CODEC" would technically be incorrect. (Note that Meridian has carefully avoided referring to it that way, but the way they word their explanations certainly could be interpreted as encouraging other folks to make that error.)


----------



## sonitus mirus

keithemo said:


> I think we do actually mostly agree here.... but I would tend to extend the "I don't understand why most people make a fuss" to the extra requirements for high-res audio as well. We live in a world where people consider themselves "disadvantaged" if their Internet connection isn't fast enough to stream HD video - and next year that will move up to 4k. Compared to the storage space and transmission bandwidth for streaming HD video, the storage and download bandwidth requirements for MP3 and 44k are both negligible. And, as for making a fuss, a lot of the market does seem to be satisfied with iTunes and Spotify, both of which use lossy compression, so it's really only an audiophile minority who's making any fuss at all. (However, being audiophiles, some of us would still rather have the better quality copy - whether we can actually hear the difference or not.)
> 
> I also agree that, strictly speaking, the price usually charged for high-res downloads isn't justified by cost. (My cable company can apparently store a full HD movie on their server, allow me to download it at full resolution to watch it, as many times as I want to over the next month, for a $3 on-demand fee; clearly the additional cost of letting me download a one-hour album once at 24/192k instead of 16/44k must be much less than that.) But, since we live in a capitalist society, that isn't a problem per-se - because price is set by supply and demand rather than cost. (So the $5 or $10 premium for high-res files has a lot to do with demand, and the market, and very little to do with cost.)
> 
> ...


 
  
 Meridian's new codec will include DRM.  It would be a proprietary format that would require licensing fees for anyone to be able to use it.  Any device would have to include a certified chipset and consumers would be thrown back into the proverbial dark ages again.  Even the master of all things convenient and techy is against DRM, as Apple has removed this from all of their AAC files.
  
 While I personally don't believe the new Meridian format makes any real differences in audio quality to what already exists, it is more technically advanced, but the greatest advantages are not beneficial to the consumer.  There isn't any traditional competition to keep things in check.  Distribution is already completely locked down, and I don't want the flexibility of how and where I use my media to be under such a burdensome control by the same conglomerates.


----------



## KeithEmo

sonitus mirus said:


> Meridian's new codec will include DRM.  It would be a proprietary format that would require licensing fees for anyone to be able to use it.  Any device would have to include a certified chipset and consumers would be thrown back into the proverbial dark ages again.  Even the master of all things convenient and techy is against DRM, as Apple has removed this from all of their AAC files.
> 
> While I personally don't believe the new Meridian format makes any real differences in audio quality to what already exists, it is more technically advanced, but the greatest advantages are not beneficial to the consumer.  There isn't any traditional competition to keep things in check.  Distribution is already completely locked down, and I don't want the flexibility of how and where I use my media to be under such a burdensome control by the same conglomerates.


 
  
 I do agree - that DRM is a pain in the butt - and well worth making a fuss about.
  
 (Unfortunately, these days, there's no way to get _commercial_ acceptance for a new compression format unless it includes DRM. It will be interesting to see how they position it. The file size is still much bigger than other lossy compression formats, and they seem to be positioning it (technically) as "all of the audible sound quality of 24/192 with a file size similar to 16/44k". I can see this as a benefit to streaming services, who will "talk up" the higher quality, but otherwise a very hard claim to justify. They've also floated a few claims that it will actually sound _better_ than "regular high res files" - which seems like it will be an uphill climb to prove. (Presumably, you'll be able to buy high-res albums in the new format, and they're hoping that you'll believe that there's at least a slight difference in quality, and that you'll ignore the DRM. Since the industry in general seems convinced that DRM at least cuts down on illegal copying, it seems like that will provide them with excellent incentive to help Meridian sell it to customers.)
  
 It will also be interesting to see if the new Meridian _encoder_ ever makes its way to consumer products (if it will let you encode your own 24/192k FLACs, and give you a file size similar to a standard CD-quality WAV, then some people will embrace it for that reason).


----------



## OddE

keithemo said:


> Since the industry in general seems convinced that DRM at least cuts down on illegal copying, it seems like that will provide them with excellent incentive to help Meridian sell it to customers.)


 
  
 -One would believe that the recording industry had learned from past fubaring and had come to realize by now that DRM is a bit like shooting yourself in the foot. With a thermonuclear warhead. Twice.
  
 Like nag screens on DVD/blu-rays, DRM is a hassle only for your paying customers; the pirates aren't bothered by it.
  
 Since when did annoying your paying customers while giving the pirates a demonstrably better (and cheaper!) product make good business sense?
  
 Sigh.


----------



## KeithEmo

odde said:


> -One would believe that the recording industry had learned from past fubaring and had come to realize by now that DRM is a bit like shooting yourself in the foot. With a thermonuclear warhead. Twice.
> 
> Like nag screens on DVD/blu-rays, DRM is a hassle only for your paying customers; the pirates aren't bothered by it.
> 
> ...


 
  
 Interesting question..... I sure wish they'd stop putting those previews on DVDs and Blu-Rays - the ones you can't always skip.
 (And DVDs and Blu-Rays _DO_ also have DRM anti-copy systems on them - they haven't stopped doing that either.)


----------



## OddE

keithemo said:


> (And DVDs and Blu-Rays _DO_ also have DRM anti-copy systems on them - they haven't stopped doing that either.)


 
  
 -My point exactly; any DRM device is bound to fail sooner rather than later - and as far as being a consumer goes, the very best you can hope for with a DRMed product is that it will not be an inconvenience.
  
 Heck, back in the day when various DRM schemes were employed on plastic discs looking an awful lot like red book CDs, I found myself on several occasions having to either source a de-DRMed copy online or stripping the disc of the DRM and making a red book-compliant copy in order to be able to listen to a title I had paid good money for.
  
 For all intents and purposes, the presence of an anti-copy scheme forced me to copy the discs, costing both time and money - and, in some jurisdictions, thankfully not the one I live in, turn me into a criminal in the process.
  
 And don't even get me started on the whole DIVX debacle...
  
 Basically, any business model which is centered on protecting you from your paying customers is a doomed one, and it amazes me that the lesson has not yet sunk in.


----------



## castleofargh

I remember downloading torrents of stuff I owned several times because it was such bother to rip it to put on my DAPs. that's how messed up DRMs are, it makes you regret paying for something.


----------



## goodyfresh

castleofargh said:


> I remember downloading torrents of stuff I owned several times because it was such bother to rip it to put on my DAPs. that's how messed up DRMs are, it makes you regret paying for something.


 

 Wow, so when I wasn't looking this turned into a debate about Digital Rights Management instead of one about sound science?  Hahaha 
  
 There's all kinds of other policies going on these days in that regard.  LIke all the music download services which have an EULA in which people have to basically agree that they DON'T ACTUALLY OWN THE MUSIC they are paying to download, so that if they ever lose the copy  they downloaded before, they have to pay all over to download it again.
  
 Myself, my own super-extensive CD collection is back home somewhere in a box in my mom's basement in Maryland, and I gave up on redownloading things with iTunes or any other download service a long time ago after my umpteenth hard-drive over the years died on me, so the vast majority of music I now possess is stuff that I bought in the past, but then didn't feel like buying all over again (since that is stupid and unfair) and thus proceeded to _*torrent*_, instead.  And, not gonna lie, when I am told abuot a band or musician that "you should totally check out, dude," I don't just go and buy their stuff. . .I torrent it first to see if it will be worth actually going and paying for. If any of you are copyright lawyers, then whatever, go ahead and lynch me.  Lmao.


----------



## OddE

goodyfresh said:


> There's all kinds of other policies going on these days in that regard.  LIke all the music download services which have an EULA in which people have to basically agree that they DON'T ACTUALLY OWN THE MUSIC they are paying to download, so that if they ever lose the copy  they downloaded before, they have to pay all over to download it again.


 
  
 -I am quite possibly mistaken, as I only own about a dozen digital downloads (Mostly of albums I have since purchased on CD or vinyl - basically, albums I WANTED to listen to ASAP upon release, and my local record store didn't have them on release day...) - but don't most download services offer at least a limited number of re-downloads once you login to your account? (I just logged onto cdon.com where I bought an album five years ago - it states that I have four downloads left of that title, 320kbps mp3) I may just be lucky, though - consumer protection laws in Norway are pretty strongly in favour of the customer.)
  


goodyfresh said:


> Myself, my own super-extensive CD collection is back home somewhere in a box in my mom's basement in Maryland, and I gave up on redownloading things with iTunes or any other download service a long time ago after my umpteenth hard-drive over the years died on me,


 
  
 -The solution to that problem is enterprise-grade disks, (hardware) RAID and NAS storage. (For audio purposes, storage is effectively free nowadays - for the equivalent of $600, I have redundant off-site storage for (approx) 6,000 CDs with cloud backup as an extra insurance. I have no intent of ever ripping my CDs again if I can help it.


----------



## goodyfresh

odde said:


> -I am quite possibly mistaken, as I only own about a dozen digital downloads (Mostly of albums I have since purchased on CD or vinyl - basically, albums I WANTED to listen to ASAP upon release, and my local record store didn't have them on release day...) - but don't most download services offer at least a* limited number of re-downloads* once you login to your account? (I just logged onto cdon.com where I bought an album five years ago - it states that I have four downloads left of that title, 320kbps mp3) I may just be lucky, though - consumer protection laws in Norway are pretty strongly in favour of the customer.)
> 
> 
> -The solution to that problem is enterprise-grade disks, (hardware) RAID and NAS storage. (For audio purposes, storage is effectively free nowadays - for the equivalent of $600, I have redundant off-site storage for (approx) 6,000 CDs with cloud backup as an extra insurance. I have no intent of ever ripping my CDs again if I can help it.


 
 And that "limited number" is where the issue lies.  You have BOUGHT that music.  You should have a right to UNLIMITED re-downloads of it!  Sheesh.  Lol.  And also yeah, the consumer protection laws here in the United States are absolute crap and are completely in favor of the corporations.  You should see what the iTunes user license agreement is like here in the U.S.  You basically have to sign over your very soul, lol.


----------



## OddE

goodyfresh said:


> And that "limited number" is where the issue lies.  You have BOUGHT that music.  You should have a right to UNLIMITED re-downloads of it!




-I don't agree, actually. I've bought it, and should be expected to look after it - after all, if I'd lost the actual CD, very few people would expect the record company to give me a new one (though that is exactly what GD Music does, or at least did - I once lost a Dick's Picks CD and got a new one for a token fee.) 

Arguably, the download deal leaves me better off (if I wasn't such a sucker for shelves full of music, that is!) - I can lose the file four times, and still the seller will give me a new copy, even several years after original purchase! 

(Note that the files in question are DRM-free, so I can have a gazillion backup copies if I like.) 

Anyway, I've managed to derail this thread from its most recent derailment; perhaps we should start a new one titled "DRM: Spawn of Satan" or something like it...


----------



## goodyfresh

odde said:


> -I don't agree, actually. I've bought it, and should be expected to look after it - after all, if I'd lost the actual CD, very few people would expect the record company to give me a new one (though that is exactly what GD Music does, or at least did - I once lost a Dick's Picks CD and got a new one for a token fee.)
> 
> Arguably, the download deal leaves me better off (if I wasn't such a sucker for shelves full of music, that is!) - I can lose the file four times, and still the seller will give me a new copy, even several years after original purchase!
> 
> ...


 

 The difference between losing files and losing a physical CD is that the former can happen through no fault of one's own, due to the fact that storage such as hard-drives can fail completely unpredictably.


----------



## RRod

goodyfresh said:


> The difference between losing files and losing a physical CD is that the former can happen through no fault of one's own, due to the fact that storage such as hard-drives can fail completely unpredictably.


 
  
 Basically there are a lot of people for whom "don't lose the CD" is a much easier concept than "maintain a backup." It's easy to forget that not everyone is a nerd, especially folks who didn't have cell phones in utero.


----------



## OddE

goodyfresh said:


> The difference between losing files and losing a physical CD is that the former can happen through no fault of one's own, due to the fact that storage such as hard-drives can fail completely unpredictably.


 
  
 -Which is why you maintain (at least!) an additional copy on an independent medium, preferably located off-site. (While a small inconvenience, I find it a price I am more than willing to pay to ensure both my music collection and, more importantly, my photos are safely kept through any calamity short of full-scale nuclear war - in which case my albums and photos would probably be the least of my worries...)


----------



## goodyfresh

odde said:


> -Which is why you maintain (at least!) an additional copy on an independent medium, preferably located off-site. (While a small inconvenience, I find it a price I am more than willing to pay to ensure both my music collection and, more importantly, my photos are safely kept through any calamity short of full-scale nuclear war - in which case my albums and photos would probably be the least of my worries...)


 

 I don't currently have access to almost 200Gb external storage-space, which is what it would cost me to back up my entire music library.  Just saying.


----------



## OddE

goodyfresh said:


> I don't currently have access to almost 200Gb external storage-space, which is what it would cost me to back up my entire music library.  Just saying.


 
  
 -Fair enough, though I would consider investing in a $70 500GB external hard drive for peace of mind. (Price checked at a Norwegian retailer; price includes 25% sales tax)
  
 Anyway, we've derailed the derailment of the original derailment, so I guess I had better call it a night.
  
 And, in case there was any doubt - I much prefer my CDs and LPs to digital downloads, but IMHO downloads can be a useful option if done right (No DRM, multiple, ideally unlimited re-downloads)


----------



## Roly1650

goodyfresh said:


> I don't currently have access to almost 200Gb external storage-space, which is what it would cost me to back up my entire music library.  Just saying.



Our local Walmart has been selling a 1tb, usb drive for $20. Formatted for Mac, but no big deal to reformat for any os. You may have to try a few Walmart's and move things around on the shelf to find them, but they're there.


----------



## castleofargh

doing backups should be taught and explained every time somebody buys a computer or a cellphone. it sucks that we almost all had to learn the hard way after losing important stuff(pictures, music, work, porn stash...).


----------



## goodyfresh

castleofargh said:


> doing backups should be taught and explained every time somebody buys a computer or a cellphone. it sucks that we almost all had to learn the hard way after losing important stuff(pictures, music, work, *porn stash*...).


 
 That last being the most important of all!!!!!


----------



## OddE

castleofargh said:


> doing backups should be taught and explained every time somebody buys a computer or a cellphone. it sucks that we almost all had to learn the hard way after losing important stuff(pictures, music, work, *porn stash*...).


 
  
 -Thankfully the Internet serves as a giant backup service for such materials. 
  
 (I remember back in the day when peer-to-peer file sharing was just taking off, and we (a couple of fellow students and I) tried out what we called hardcore backup. We encrypted, tarred and gzipped an archive, renamed it 'BritneySuckingAndF***ing.mpg or something to that effect, then proceeded to upload it to some filesharing service - probably Napster.
  
 Result? Lots of copies floating around; people downloaded, found they didn't have a codec to view it, shrugged and let it stay on the disk to improve their share ratio...)
  
 Heck, we should be able to sue anybody holding a patent on cloud-based backup on grounds of prior art...


----------



## sonitus mirus

castleofargh said:


> doing backups should be taught and explained every time somebody buys a computer or a cellphone. it sucks that we almost all had to learn the hard way after losing important stuff(pictures, music, work, porn stash...).


 
 I have taken the cloud route with just about everything.  I still have a few thumb drives around that I move folders of stuff onto every now and then, but mostly I'm all offsite and remotely saving everything.
  
 I've been streaming music for several years now.  My photos are saved and accessible online.  Games are connected to my Steam account and available to download to any computer.  Important documents are similarly saved and made available on the cloud.  Videos are on YouTube account.  Movie collection is available online via Vudu or M-GO.  Even my browser settings and favorites are saved and instantly available.  
  
 I could toss my computers into the ocean and build/buy new ones and be right back where I am today in just a few hours.  I made specific choices so that I could make this possible.  It has been wonderful.  Unless the infrastructure goes down, I'm good, though if that ever does occur, I'll be concerned with other things.
  
 Back to our regular bickering about 24-bit audio, which is mostly useless with regards to audio quality improvements.


----------



## goodyfresh

odde said:


> -Thankfully the Internet serves as a giant backup service for such materials.
> 
> (I remember back in the day when peer-to-peer file sharing was just taking off, and we (a couple of fellow students and I) tried out what we called hardcore backup. We encrypted, tarred and gzipped an archive, renamed it '*BritneySuckingAndF***ing.mpg or something to that effect*, then proceeded to upload it to some filesharing service - probably Napster.
> 
> ...


 

 LMAO you're hilarious dude.


----------



## hogger129

I'm happy with a CD.  I would like to see online retailers like iTunes, Amazon and Google start selling CD-quality lossless downloads.  The storage space is available for most people and the bandwidth is there, unlike 10 years ago when lossy downloads came on the scene.


----------



## RRod

hogger129 said:


> I'm happy with a CD.  I would like to see online retailers like iTunes, Amazon and Google start selling CD-quality lossless downloads.  The storage space is available for most people and the bandwidth is there, unlike 10 years ago when lossy downloads came on the scene.


 
  
 The probably don't see much need for that:
 .From a sound standpoint there is little-to-nothing to gain from moving up from 256aac or 320mp3 to FLAC
 .From a bottom-line standpoint, any audiophile desire is really a niche market compared to the latest Taylor Swift release
 .From a control standpoint, they probably prefer to not let you download the original file content
  
 What I'd like to be able to do is pay a bit more for extra content: album art, album notes, scores, etc. Music is losing so much of its context by just being dumped into huge online libraries.


----------



## KeithEmo

I think this subject does actually tie in to the main subject of this thread... and that tie is is through the very basic question of whether, when you "buy" an album, or a CD, or a download, it seems somewhat vague exactly what you are in fact buying. Back in the vinyl days, before tapes were even very good, it was clear that you were buying a piece of plastic with music on it... and the two were inextricably linked. That all changed once tape recorders became good enough that you could actually make a listenable copy of an album. (And all of this has remained pretty much the same, but simply become more relevant, as copying has become more perfect and so much easier.)
  
 First, let's get one thing out of the way: The actual _physical_ cost of a download is a few cents (by which I mean - what it costs when you divide the cost of the server space holding the file by the number of people that download it, then add the cost of the bandwidth each download costs), and I can easily have CDs mass produced, with labels and jewel cases, for about $1. So this is the actual physical cost of music distribution.
  
 So, let's try and figure out where the "value" of that album lies.... Let's assume that I bought my favorite album in vinyl - which probably cost me somewhere around $15. Now, I decide I really want it on CD; so I pop down to the store and buy a CD. Oddly, even though I've already paid for both a plastic disc, and for the music recorded on it, they expect me to pay exactly the same price for that CD as some other guy who hasn't already paid for either one. I don't seem to be getting any sort of discount "because I already own the music, so all I'm paying for is the plastic". And, if I buy it as a download, for yet a third round, again they expect me to pay full price. It sort of seems like I'm paying over and over again for the right to listen to the same piece of music, doesn't it. (Or like they're saying that the right to listen to the music is worth pretty much nothing. After all, if the right to listen cost $13 and the plastic disc was $2 then, if I broke the CD, I could sweep the plastic shards into an envelope, send it in to prove that I already owned the right to listen, and have them send me a new piece of plastic to store it on for some reasonable price - say $2.) However, if the plastic breaks, now they're telling me the license to listen to that music was worth nothing - because they don't let me "transfer" that license to a new piece of plastic.
  
 What's really funny is that, even though - for all of that paragraph - they seemed to be telling me that it was the plastic that "held all the value", if I buy a download, they seem to expect me to pay for it all over again; and if I were to "steal" a copy by copying the data, or give a copy to a friend, they act like the right to listen to the music, which they seemed to value at nothing a few seconds ago, is now the biggest part of the value. Wouldn't it make more sense if, once I paid for the right to listen to that particular music, that was considered to be a separate item... in which case, if I already owned the CD, or even the vinyl album, I should be able to trade in that particular piece of plastic for a different one for only the difference in cost, or trade it in for the download for nothing....
  
 It really seems like the _FAIR_ way to do this would be to simply count the two items involved separately. If I buy a download for $12, I should get a certificate (or my name should be entered into a database somewhere) that says I paid $10 for the "music album" itself, and $2 for the download service. And, if I buy a CD, I get a certificate for the $10 "music album", and a receipt for the $5 piece of plastic. But, and here's the difference, if I already own the "music album" because I bought it as a download, then I should be able to present my proof of ownership, and get a new copy on CD for that $5 "service charge", and, if I break the CD, I should be able to replace only the plastic, again for a reasonable service charge. 
  
 Now, to bring all this "home" to this thread, if music was treated like this, then the cost of "upgrading" to a high-res version of an album you already own would be much more reasonable, which would make a huge difference in terms of "whether it was worth it". For example, assuming that I paid $15 for my CD, and $10 of that was for the actual license for the "music album", then $5 would be a reasonable service charge for selling me a different copy of the music I already own. And, if the music was remastered, considering that the same master tapes were used, and the artist presumably got royalties or payments at the point where I originally bought it, it would also be only reasonably to charge me _only for the service of remastering the album_. (So, maybe, if the original album actually cost "$5 for the music; $5 for the mixing and mastering, and $5 for the piece of plastic", then it would be fair to pay an additional $5 for the _NEW_ mixing and mastering, and another $5 for the _SERVICE_ of packaging and delivering that music in a new high-res format.) If the industry followed this idea, then buying a "simple high-res reissue" of a CD or download you already own might cost a very reasonable $5, and buying a "remastered version - in high-res" - might cost $10... and I think that, if the prices were that reasonable, most of us here wouldn't even be arguing about whether the extra cost was justified.
  
 (And, of course, another benefit would be that you would always have that upgrade path - you could always decide later to pay the "extra bump charge" to "upgrade" your CD-quality download file to a high-res version later if you decided to. Personally, I suspect that doing it this way could in fact be very profitable to the music industry overall - just imagine how many more people would be willing to pay a reasonable upgrade fee than are now willing to buy a lot of music over again... and imagine "five packs of upgrade coupons" for "stocking stuffers", and special promo deals where you get "1 free album upgrade to high-res when you buy five regular albums".)
  
   Quote:


odde said:


> -I don't agree, actually. I've bought it, and should be expected to look after it - after all, if I'd lost the actual CD, very few people would expect the record company to give me a new one (though that is exactly what GD Music does, or at least did - I once lost a Dick's Picks CD and got a new one for a token fee.)
> 
> Arguably, the download deal leaves me better off (if I wasn't such a sucker for shelves full of music, that is!) - I can lose the file four times, and still the seller will give me a new copy, even several years after original purchase!
> 
> ...


----------



## KeithEmo

I agree entirely... Back "in the beginning", a lot of customers had dial up connections, so the amount of time they saved by downloading an MP3 instead of a FLAC was significant, and nobody would have wanted to buy a $400 iPod classic that held "100 albums" ("10,000 of your favorite songs" sounded a lot more impressive). Also, to put it bluntly, nobody cared (so, if Apple could save 2 cents a song by using AAC 128 instead of FLAC, it still added up to a few bucks - and, I assume, somebody did their homework and figured out that most of their customers at that time _really_ didn't care one way or the other... and, let's be honest, most of their customers _still_ don't care.)
  
 However, this late in the game, a lot of the problem has almost certainly been the simple fact that, once someone like the iTunes store has millions of songs already selling in a certain format, going back and reissuing all of them is a major single effort - and, as such, quite expensive. (Don't assume that it's as simple as doing a batch conversion - someone has to manage the process, update all those files, dig out all the original masters and make sure they're good enough not to embarrass themselves in the new format, and update all those shopping cart items. Spending $5 a song adds up if you're talking about 1,000,000 songs. And, collectively, the storage space and bandwidth do add up as well. Remember the story about the guy who saved his company millions by suggesting that they put one less olive in each of the millions of jars of olives they sell...)
  
 (I still find it rather sad that iTunes took the time and effort to go back and reissue a lot of music in a higher-quality format, but still used AAC256 - instead of "taking the step" and going to lossless. It's also somewhat interesting to wonder what their future plans are.... If you look at the "guidelines for mastering content for the iTunes store", they still recommend doing so at 24/96...
 hmmmm.... )
  
  
 Quote:


hogger129 said:


> I'm happy with a CD.  I would like to see online retailers like iTunes, Amazon and Google start selling CD-quality lossless downloads.  The storage space is available for most people and the bandwidth is there, unlike 10 years ago when lossy downloads came on the scene.


----------



## sonitus mirus

Part of the reason Amazon only offers MP3 downloads is because they do not want to cannibalize their own CD sales.  They have warehouses filled with CDs, as they take up very little space and can be delivered the same day or overnight for pennies.  When you purchase a CD, the MP3 files become immediately available for download and use.   
  
 The other reason is the record labels' refusal to allow lossless versions to be sold at the same quality level as the CD.  They will make HD versions available to be sold at a premium cost for select vendors, but you won't see a lossless copy made available for many of the major music labels.  At least they are not going to allow it without implementing some form of DRM to control distribution and to create an artificial revenue stream just because they can.


----------



## interpolate

All good points well stated.


----------



## Joeybgood

My uninformed butt is trying to understand something. HighRes DL sites(HDTracks specifically) offer all these classic cds that were recorded in the 50s,60s 70s at different levels of 24bit. Since these albums were not likely recorded in 24bit, how could upsampling/inflating  them to this resolution level make any bit of difference? I mean, all this space is filled with what? Certainly not actual music data that originated from the artist when it was recorded. correct? Is it just meaningless filler that they are adding and then asking much higher prices for them? Please be so kind as to help me understand what it is that these sites are offering here. Regards, Joey


----------



## RRod

joeybgood said:


> My uninformed butt is trying to understand something. HighRes DL sites(HDTracks specifically) offer all these classic cds that were recorded in the 50s,60s 70s at different levels of 24bit. Since these albums were not likely recorded in 24bit, how could upsampling/inflating  them to this resolution level make any bit of difference? I mean, all this space is filled with what? Certainly not actual music data that originated from the artist when it was recorded. correct? Is it just meaningless filler that they are adding and then asking much higher prices for them? Please be so kind as to help me understand what it is that these sites are offering here. Regards, Joey


 
  
 If you truncate these tracks down to 16 bits, pad back to 24 bits, and do a difference, all you'll have left is whitish noise. So they're selling noise, basically. Below is the difference spectrogram for a 24-bit test track, truncated down to 16bits and padded back up. As you can see, mostly noise at about -115dB, with a little burst of stuff at the end around -110dB (file peak is -6.74dB).
  

  
 And this is a classical track actually recorded at 24-bits. Stuff from back-in-the-day will just have louder noise. What they'll try to sell you on is that since 24-bits has less rounding error than 16 bits, you'll hear "micro-details" in the music better. I guess noise counts as detail these days.


----------



## Joeybgood

rrod said:


> If you truncate these tracks down to 16 bits, pad back to 24 bits, and do a difference, all you'll have left is whitish noise. So they're selling noise, basically. Below is the difference spectrogram for a 24-bit test track, truncated down to 16bits and padded back up. As you can see, mostly noise at about -115dB, with a little burst of stuff at the end around -110dB (file peak is -6.74dB).
> 
> 
> 
> And this is a classical track actually recorded at 24-bits. Stuff from back-in-the-day will just have louder noise. What they'll try to sell you on is that since 24-bits has less rounding error than 16 bits, you'll hear "micro-details" in the music better. I guess noise counts as detail these days.


 
  


rrod said:


> If you truncate these tracks down to 16 bits, pad back to 24 bits, and do a difference, all you'll have left is whitish noise. So they're selling noise, basically. Below is the difference spectrogram for a 24-bit test track, truncated down to 16bits and padded back up. As you can see, mostly noise at about -115dB, with a little burst of stuff at the end around -110dB (file peak is -6.74dB).
> 
> 
> 
> And this is a classical track actually recorded at 24-bits. Stuff from back-in-the-day will just have louder noise. What they'll try to sell you on is that since 24-bits has less rounding error than 16 bits, you'll hear "micro-details" in the music better. I guess noise counts as detail these days.


 
 Pretty much as I expected. thanks much!


----------



## icebear

joeybgood said:


> *My uninformed butt* is trying to understand something. HighRes DL sites(HDTracks specifically) offer all these classic cds that were recorded in the 50s,60s 70s at different levels of 24bit. Since these albums were not likely recorded in 24bit, how could upsampling/inflating  them to this resolution level make any bit of difference?....


 
 Hilarious, LOL 
	

	
	
		
		

		
		
	


	




 .... it's very simple:
 After high rez marketing hype made a lot of consumers buy new DACs capable of 24/96 to 192 or even 32/384, DSD 1234 xyz or what have you, there had to be software on offer to feed all these new gadgets and make the respective LED's light up or displays tell the resolution. And then people were hearing that they got their money's worth
	

	
	
		
		

		
		
	


	




.
  
 The classic recordings you mention do sound spectacular already on CD (living stereo, living presence etc.). These are analog recordings and given the age of the tapes, the sound that was captured is just astounding. Unless the orig. master tapes are newly A/D transferred with a higher resolution, then the old "CD master" are just upsampled to make the DAC display say 24/96. Everybody should be able to decide if that's worth paying for or stick with "lowly RB" at 16/44. The age of these tapes makes it very questionable if renewed transfer will give better result than what has been done 20 or 30 years ago. A new mastering of the orig. transfer files might help in certain cases when producers have been over eager to make the CD sound "obviously better than analog". The age of early digititis
	

	
	
		
		

		
		
	


	




. The tapes don't really improve with age. At some point another playback will destroy them completely. What hasn't been archived into a new format already, might be lost at some point.


----------



## castleofargh

well the easy way to keep archives was to copy onto newer tapes, but that was not helping the resolution for sure.


----------



## Joeybgood

icebear said:


> Hilarious, LOL
> 
> 
> 
> ...


 
 gotcha.. tks!


----------



## twsmith

As a general comment -- I have a small number of CDs that I also downloaded as hi-res (DSD, PCM) files from various sources.  I perhaps naively concluded that the high res version would sound much better.  I was initially surprised by what I heard but having read through this thread, I shouldn't have.   In almost all instances, I found no (or very little) differences between the 44/16 and higher rez files, and what differences I did hear I attributed to the use of possibly a different master or (vice versa) maybe slight compression of the CD version.   What was even more interesting was that there were several instances where the CD version actually sounded better than the downloaded "hi-res" file -- go figure.   This is not to say that there may be some value in downloading a hi-res file, simply because the mastering may truly be better.   Some of these older CDs (as pointed out here) may have been mastered from analogue tapes using ADC converters that would now be considered obsolete, and then you add in the fact that engineers didn't know how to equalize the masters properly for the digital format.   In some cases, these older CDs have been remastered as new CDs and it is here that I find absolutely no difference in SQ between the CD and hi-res version.  As many have commented, there may be good reasons for studios to use 24 bit and sampling higher the 48 kHz in the initial production and mastering process, but if the downsampling to 44/16 is done properly, I see no advantage to purchasing a higher res version.   Actually my biggest beef with hi-res downloads is the limited repertoire available in this format, especially for those of us whose musical tastes deviate from the norm.


----------



## sonitus mirus

I'm all for making the CD obsolete and moving all media to 24/96 or better.  What I don't want is for a premium price tag to be placed on whatever format is decided upon.  I know the goal is to try and get everyone to repurchase all of their music again, and it was a great ride for the music industry for several decades, but the Millennials have made it clear that they don't care about anything but convenience.  Being caught in the middle, and armed with the knowledge we have, a few of us are taking advantage of the situation and finding cheap, convenient ways to enjoy tons of music at a great quality for practically nothing.
  
 Shuffling through a few thousand LPs and CDs at a local shopping mall's record store 30 years ago, I would have never dreamed of the scenarios we have today with music availability and sound quality.  It's insane...and absolutely wonderful.


----------



## RRod

Well said, sm. The paradigm of the engineers using high specs and the end-user using well-made lossy files is actually quite remarkable. They get more leeway for getting the recording done right, and we get smaller files that sound just as good and that we can stream easily and download as needed.
  
 People like to say "oh file sizes don't matter." But consider this scenario: I want to play music constantly at work (where I can't just stream all day or guarantee hard-drive space), 8 hours a day, 5 days a week, ~4 weeks a month. If we suppose that streaming FLAC/ALAC gets you ½ file sizes, then uncompressed 24/192 content would require about 2.6TB of bandwidth for the month. Streaming 256 AAC, on the other hand, would require 147GB. Which one do you think is more viable via mobile streaming + DAP storage? So yes, if there is no audible difference between the 2 why the heck are we so concerned to have the former, especially if we're talking streaming?


----------



## sonitus mirus

When it comes to streaming music, I want the smallest file size that offers transparency.   I stream at work, on my commute, at home, visiting friends or family, at airports, at hotels for business or travel, and practically anywhere I have to park my butt for an extended time.  For me, it has not been about storage needs or even cost.  Granted, I live in Ashburn, Virginia, a cushy suburb of Washington DC, and our internet infrastructure is extremely robust in this region, so others may not have a similar experience, but a working internet connection is more reliable than tap water.  Except for one time when the power was out for a few hours from a nearby lightning strike, I have had constant access to the web and my streaming music for over 5 years.


----------



## XenHeadFi

rrod said:


> So yes, if there is no audible difference between the 2 why the heck are we so concerned to have the former, especially if we're talking streaming?


 
 In reference to streaming, I want a codec with the smallest data rate (file size) that is transparent. 256kbit AAC or 64kbit MPQuantum512LAFU, doesn't really matter. Of course when we really get unlimited wireless data (100% wifi coverage), then it won't matter.
  
 For my own reference and storage, I want lossless. I would rather not transcode from lossy to lossy. Flashbacks to the horrors of magnetic media (Cassettes and VHS/Beta).


----------



## goodyfresh

> *Flashbacks to the horrors of magnetic media (Cassettes and VHS/Beta).*


 
 OH GOD DON'T REMIND ME ABOUT THE HORRORS OF CASSETTES!  *breaks out in a cold sweat* 
	

	
	
		
		

		
		
	


	




 

 Speaking of old-fashioned technological horrors and monstrosities, anybody remember 28 and 56K dial-up internet?  LMAO!


----------



## OddE

goodyfresh said:


> Speaking of old-fashioned technological horrors and monstrosities, anybody remember 28 and 56K dial-up internet?  LMAO!




Oh, we used to DREAM of 28k... (My first modem, which in all fairness never saw a http request pass through it, was a 75 baud acoustically coupled monstrosity. 

You could actually read the data as it was being downloaded. 

Curiously, streaming music was not very widespread in the early eighties. (A couple of months later I got a slightly less despair-inducing 2400bps modem. Blistering speeds!


----------



## RRod

xenheadfi said:


> In reference to streaming, I want a codec with the smallest data rate (file size) that is transparent. 256kbit AAC or 64kbit MPQuantum512LAFU, doesn't really matter. Of course when we really get unlimited wireless data (100% wifi coverage), then it won't matter.
> 
> For my own reference and storage, I want lossless. I would rather not transcode from lossy to lossy. Flashbacks to the horrors of magnetic media (Cassettes and VHS/Beta).


 
  
 Back when lossy was, let's admit it, pretty bad, it sure made a lot of sense to keep a lossless version around, as it let you transcode to the latest, greatest codec. Now that the codecs have gotten so transparent, the need for transcoding is diminishing. I keep all my FLAC rips around too, but when I really think about it, I have nary a device that doesn't support VBR AAC. My hanging on is partially a combination of habit and mistrust of the industry, fed by things like the loudness war and the current hi-res money-grab.


----------



## XenHeadFi

rrod said:


> My hanging on is partially a combination of habit and mistrust of the industry, fed by things like *the loudness war* and the current hi-res money-grab.


 
 Loudness War is by far the most important reason why I keep my CD's and FLACs. Trying to find an old master of an album or track is just a stupendous pain in the *****. Pretty much invisible unless you look for a "mark": a new copyright date, a second difference in length, etc. Kinda like how Enya's "Book of Days" was silently "upgraded" to "Far and Away Theme", I grabbed the entire stack of used CD's, took them to the counter and played each one to find the one with "Book of Days" on it. They had a sticker, but it was on the plastic wrap, which was long gone by the time I even thought about picking up the album.
  
 That is what makes the Loudness War so insidious; the changes are nearly invisible until you listen to it. Maybe that's just a new cover for an old album or may be its the old cover on a secretly boosted CD. Evil. Then comes streaming...you are at the mercy of the streaming master (most here in Sound Science know bit-rate is usually swamped by the quality of the master), with no recourse except to not play that song from that service.
  
 Dream format - 24-bit/48khz lossless (compress those zeros!), DRM-free. Dunno if multichannel sound would need more bits at a higher frequency or they just add data tracks so 24/48 is way more than enough.


----------



## RRod

xenheadfi said:


> Loudness War is by far the most important reason why I keep my CD's and FLACs. Trying to find an old master of an album or track is just a stupendous pain in the *****. Pretty much invisible unless you look for a "mark": a new copyright date, a second difference in length, etc. Kinda like how Enya's "Book of Days" was silently "upgraded" to "Far and Away Theme", I grabbed the entire stack of used CD's, took them to the counter and played each one to find the one with "Book of Days" on it. They had a sticker, but it was on the plastic wrap, which was long gone by the time I even thought about picking up the album.
> 
> That is what makes the Loudness War so insidious; the changes are nearly invisible until you listen to it. Maybe that's just a new cover for an old album or may be its the old cover on a secretly boosted CD. Evil. Then comes streaming...you are at the mercy of the streaming master (most here in Sound Science know bit-rate is usually swamped by the quality of the master), with no recourse except to not play that song from that service.
> 
> Dream format - 24-bit/48khz lossless (compress those zeros!), DRM-free. Dunno if multichannel sound would need more bits at a higher frequency or they just add data tracks so 24/48 is way more than enough.


 
  
 Personally I'd love if some compressed ambisonic-like format became a standard; combine that with the new standards for HRTF formatting and you could have yourself an adaptable system that used minimal bandwidth. Arguing over bits/sample specs, to me, is missing what audio truly could be doing.
  
 I've used discogs + loudness-database to hunt down non-loudness masters, and yeah it's a painus-in-the-anus. Thankfully as a classical guy I am usually outside of such issues, but there seems to be some creep happening. Part of the problem is the seemingly large % of people these days for whom music just isn't a top priority, so I can't entirely blame the industry I guess for just giving them loud, loud, loud. And like you said, it's not always plainly evident that anything is wrong unless you have the side-by-side.


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## KeithEmo

There are several reasons why this is going to be very slow to happen....
  
 One reason is "marginal cost"... An actual CD (with the box and the label) costs abut a buck to make in quantity. So, by _NOT_ giving you the CD, they save about a buck. The problem is that most people "see" the CD as having more value than that; a lot of people who pay $12 for a CD wouldn't be happy paying $11 for the download; they would expect it for less. In other words, people "see" $12 for a CD - which includes a jewel case and all the rest - as being more of a bargain than an $11 download, and so are more willing to buy it. (This isn't just a matter of "well, I'm just going to rip it anyway, so who cares"; it's more a matter of the fact that, to most people, getting an actual box, with an actual disc, and a printed label, "feels like you're getting more than a substanceless download".)
  
 Another factor is "piracy". Regardless of what the actual numbers are, a certain number of people who buy CDs are never going to RIP them - or share them with their friends; this means that physical CDs reduce music bootlegging - at least a little bit - that way. And, believe it or not, having the actual piece of plastic, and the booklet, really are the reason why some people buy CDs - which also means that some people who might not see a huge moral distinction between buying a download copy and "sharing" or "borrowing" one from a friend might still be willing to buy the CD to get the "whole experience" including the booklet and the jewel case. (And, yes, a few old timers still like to sit there and read those liner notes - _ON PAPER_ - while they're listening to the album.)
  
 The single biggest force opposing the "physical CD mindset" is probably Amazon Prime (with Amazon Prime, they pay the shipping instead of the customer; which gives them several dollars worth of incentive to get you to buy the download instead, which gives them more incentive to sell it to you at a lower price).
  
 Personally, I think one thing that could be done to encourage this would be to provide nice "digital extras" with the download copy. With the download, you should at least receive a nice high-quality scan of the album cover and liner notes. They could even take this further and, much like when you buy a DVD of an old movie, give you "digital extras" like extra notes, or more information, or alternative album cover artwork.... Simply, they should give you something that you _DON'T_ get with the CD. With the video market, it seems to be pretty well accepted that one way to sell re-issues and remasters is to provide additional content (directors' cuts; interviews; making of's....; bonus tracks; etc); one of the biggest topics of discussion in DVD and Blu-Ray reviews is the extra content included with this or that version....
   Quote:


hogger129 said:


> I'm happy with a CD.  I would like to see online retailers like iTunes, Amazon and Google start selling CD-quality lossless downloads.  The storage space is available for most people and the bandwidth is there, unlike 10 years ago when lossy downloads came on the scene.


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## KeithEmo

Unfortunately, the answer isn't totally simple..... and not quite as simple as a lot of people seem to prefer to think.....
  
 The digital masters from which many early CDs were made probably were only themselves created at 16/44k or 24/44k.
 Simply converting those digital masters to a higher resolution digital file has no real benefit.
 However, most of the content on those early CDs was originally recorded on analog master tape, so re-converting it at a higher resolution _might_ be an improvement.
 Likewise, because some of the early converters were somewhat flawed, even a modern conversion at the same sample rate _might_ be an improvement.
 And, yes, it's possible that converting an old digital file to a higher resolution _might_ bring improvements - if additional processing or restoration was involved. 
  
 The important point there is that those are all _POSSIBLE_ improvements; the information you need to determine whether any of that possible improvement exists in a particular situation, you pretty much need to know the same information as you would about any other re-master (basically, you need to know how good the original master was, how good the original issue version was, and what the differences are between the original and the re-master). You certainly should _NOT_ assume that the quality is better simply because it's "a high-res reissue".
  
 Most modern music is mastered at higher sample rates (most video sound tracks are at 48k or higher; most audio at 24/96k or higher). But, again, whether there's anything present on those masters that would result in better sound at their native resolution rather than after being down-sampled to 44k depends on a lot of things. Without getting into the argument of whether you can hear it if it's there or not, some master recordings (digital or analog) contain content that can be reproduced more perfectly at 96k than at 44k, while others do not. (And, if there's absolutely no useful information there that can't be reproduced perfectly at 44k, then there's no benefit to be had by using a higher sample rate.)
  
 Quote:


joeybgood said:


> My uninformed butt is trying to understand something. HighRes DL sites(HDTracks specifically) offer all these classic cds that were recorded in the 50s,60s 70s at different levels of 24bit. Since these albums were not likely recorded in 24bit, how could upsampling/inflating  them to this resolution level make any bit of difference? I mean, all this space is filled with what? Certainly not actual music data that originated from the artist when it was recorded. correct? Is it just meaningless filler that they are adding and then asking much higher prices for them? Please be so kind as to help me understand what it is that these sites are offering here. Regards, Joey


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## RRod

keithemo said:


>


 
  
 He was asking about bit depth and you seem to be addressing sample rate. I task you to find a 24-bit master made from analog tape that doesn't null to noise when truncated down to 16 bits.


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## KeithEmo

Analog tapes themselves aren't that quiet - which is something that many people seem to forget.
 Back in those days, a good master tape machine had a better S/N than vinyl (hopefully), which was plenty good enough. 
 (I'm equally dubious about whether any of those early A/D converters had 24 bits of resolution anyway - you need a S/N of 96 dB to get 16 bits of real resolution.)
  
 However, it is still technically possible for someone to produce a re-mix that is much quieter than the original master mix, or even than the original tracks themselves, with sufficiently aggressive processing and "restoration" - so I can't say it's impossible that somewhere there is a re-master of an original tape master that would "justify" 24 bits - but I'm not holding my breath. (Generally, when you do that much processing, I find the results to have so many serious artifacts that they don't sound good at all.)
  
 That also brings up another interesting thing that people seem to forget:
 Just as with any other re-mix or re-master, it's quite possible for a "high-res remaster" to sound _worse_ than the original release.





  
  
 Quote:


rrod said:


> He was asking about bit depth and you seem to be addressing sample rate. I task you to find a 24-bit master made from analog tape that doesn't null to noise when truncated down to 16 bits.


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## RRod

keithemo said:


>


 
  
 Even today I don't think we have a true 24-bit ADC or DAC. As far as the mastering, if the process made the track quieter, one only need to gain the track up to 0 dBFS for the max peak before converting to 16-bit. I can take a 24-bit track and gain it down to only take up the lowest 16 bits; if I were to then truncate the file to 16-bit, I would essentially end up with an 8-bit file, which of course is insufficient for most music. But I would also call that bad mastering


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## KeithEmo

I do have an interesting question for you there.....
 Why do you insist on streaming music all the time?
  
 It seems to me that today most people seem to believe that there's some sort of "hard line" between streaming and simply playing music.
  
 Personally, I have a large collection of albums that I own, and which I play quite often.
 I also sometimes listen to streaming music - or even ... gasp... radio.
 And, when I listen to my own albums, I simply play them directly - which uses no bandwidth whatsoever.
  
 My point is that, especially if you're in a situation where you pay for your bandwidth, then it only makes sense to stream music when you specifically want to listen to music that you don't already own. (In other words, just like with radio in the old days, I see streaming music as a supplement to playing my own music - rather than a replacement for it.)
  
 (I would even go as far as to say that I am perfectly happy streaming music for casual listening - even using high quality lossy compression - then obtaining and actually owning a normal-res or high-res copy of only the music that I especially like. I simply don't see why it should be an either/or proposition, nor do I see why I should sacrifice the best quality in order to accommodate the limitations of a data service...  when I can play albums I already own, at whatever resolution I like, as often as I like, and for as long as I like, for free. And, as an added benefit, I can listen to the music I own any time and any place I want to - even where I don't have an Internet connection or reliable cell phone service. And, in fact, since I don't use that huge amount of data, I can also live quite happily with a cheaper data plan.)
  
  
 Quote:


rrod said:


> Well said, sm. The paradigm of the engineers using high specs and the end-user using well-made lossy files is actually quite remarkable. They get more leeway for getting the recording done right, and we get smaller files that sound just as good and that we can stream easily and download as needed.
> 
> People like to say "oh file sizes don't matter." But consider this scenario: I want to play music constantly at work (where I can't just stream all day or guarantee hard-drive space), 8 hours a day, 5 days a week, ~4 weeks a month. If we suppose that streaming FLAC/ALAC gets you ½ file sizes, then uncompressed 24/192 content would require about 2.6TB of bandwidth for the month. Streaming 256 AAC, on the other hand, would require 147GB. Which one do you think is more viable via mobile streaming + DAP storage? So yes, if there is no audible difference between the 2 why the heck are we so concerned to have the former, especially if we're talking streaming?


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## RRod

Don't forget the "at work" part. We have bare-bones PCs, purely intended to log into a virtual desktop. Not much hard drive space to use, so I can't simply put my entire collection on there in any format. They are pretty lax on music streaming, though, but that's only because I'm not streaming 2TB a month. At home, of course, I can just load up my ripped FLACs, but there's really little reason to even rip anything anymore, since it will already be available via streaming in a format that is audibly transparent to my ears. My MO is still to buy albums that I like, but the truth is that any cataclysmic event that would manage to destroy the internet would probably also destroy my power…


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## OddE

rrod said:


> Even today I don't think we have a true 24-bit ADC or DAC.


 
  
 -Nope. The thermal noise power at room temperature over a bandwidth of 20kHz is on the order of -130dBm or so, so assuming a 600 ohm load and everything else is perfect, the noise floor will be in the 22nd bit.
  
 In real-world applications, obviously we'll have other noise sources which will bring the noise floor further up - so, let's say you maybe get 20 bits of 'real' resolution on a very good day.


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## sonitus mirus

keithemo said:


>


 
  
 For me, it is not a matter of insisting on streaming music all of the time, it is just always available to stream.  The bandwidth is inexpensive and I am nearly always able to use WiFi of direct connections to the internet.  If I was unable to stream music when I wanted to listen to songs, I'd try to find a different solution.  
  
 I use Google's streaming service that I pay $7.99/month (promo rate locked in for early subscriber) to use.  I play my music on anything that has the ability to use the Chrome browser, the iOS app, or the Android app.  At home I have a couple of Chromebooks that I use almost exclusively as "jukeboxes" that can play the entire Google catalog of over 30 millions songs seamlessly with a thousand or so of my own music that is not otherwise made available to stream.  I can play full albums, single songs, genre-specific radios, and radio created by artist, album, or single songs.  I am able to maintain a library of my music that is displayed by genre, artist, album, or song with album covers and wiki entries of the artist and album in many cases.  When I shuffle my library or play any radio, not only are the subscription songs played, but my own music can also be included if the songs are related to the type of radio I selected.
  
 At this point, between the vast library of music available to stream and the music that I added that was not available, I have absolutely every album and song that I am currently aware of that I would ever want to hear.  I explore quite a bit when it comes to music, and I try to listen to a little bit of everything.  When I find something that sounds good, I often add the full album to my library so that any of the songs on it could pop up when I shuffle my library.  For me, it is like a dream come true, only I could never have dreamed this up 20-30 years ago.  So, it is even better than a dream come true. 
	

	
	
		
		

		
		
	


	



  
 Now, if I wanted to listen to music and I could not get to it by streaming, I'd be frustrated and look for a different option.  To this point, I have not had any troubles, so I see no reason to move on to anything else.  I don't listen to any of my CDs directly anymore.  I rip a CD to MP3 and upload it into Google. 
  
 I don't choose to stream everything all the time, it just works for me all the time, so I do it.  It is awesome.


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## interpolate

The fastest speed I ever got from a dial-up modem was around 48K (capped by the parity error checking). U.S. Robotics one with answering machine facility.
  
 Surprisingly this company still exists in that capacity.


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## KeithEmo

Back when phone lines were "discrete lines" rather than IP-based, the theoretical limit for a single line was 64k; "56k modems" were adaptive, so you got slightly different speeds depending on line conditions, but their theoretical maximum was somewhere around 52k (because of a sort of technicality, which "used up" some of the theoretically possible data bandwidth, they couldn't actually do 56k - and I never saw one do full 52k except in a lab setting). As I recall, US Robotics was considered by most people to be the top premium brand for such devices back in those days.
  
 Quote:


interpolate said:


> The fastest speed I ever got from a dial-up modem was around 48K (capped by the parity error checking). U.S. Robotics one with answering machine facility.
> 
> Surprisingly this company still exists in that capacity.


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## KeithEmo

I'd have to say that, for me, it's more a matter of "the principle of the thing" or "my comfort zone" rather than totally practical considerations.
  
 For one thing, I like to own the music I really care about, and to actually be in possession of it - whether it's in the form of plastic, or of a computer file, which I can back up. Not that it's especially likely, but there have been cases where Kindle books "disappeared" from the collections of people who had "purchased" them when they were found to be illegal to sell due to copyright or other issues. There was also a video service a very long time ago called DivX (not to be confused with the video format of the same name). With this service, you received a disc for free, then paid for a license for either a single viewing, or "lifetime ownership". However, one day the company went out of business, and everyone who thought they owned videos found that they now owned unplayable discs - and their "video collection" had magically disappeared. I am simply not sufficiently confident that a given service will remain in business, or that a given album won't be "pulled" due to some sort of legal or contractual dispute, to trust someone else to "keep track of" my music. (It would ruin my entire week if my favorite song were to disappear one day, or my streaming service were to go under, and I couldn't find another one which had all of my favorite music. Companies - and services - tend to come and go... and I simply don't like to take that risk with my music collection.)
  
 Also, to be honest, I haven't found any form of lossy compression so far that I consider to be "absolutely inaudible at all times". Therefore, I insist on at least CD quality content for serious listening (I'm perfectly willing to use MP3 or Ogg Vorbis for casual listening and "background music"). It just seems more reliable to RIP my own CDs - where I can confirm that the RIPs really are perfect and haven't been modified or compromised along the way - than to trust someone else to do it and get it right. (Streaming over the Internet is also virtually always subject to occasional dropped samples and such.) I also don't especially favor listening to music from a phone, or using earbuds, so that level of portability isn't a consideration for me.
  
 However, as long as it works for you, that's obviously what counts (and streaming music seems to work for lots of people).
  
   Quote:


sonitus mirus said:


> For me, it is not a matter of insisting on streaming music all of the time, it is just always available to stream.  The bandwidth is inexpensive and I am nearly always able to use WiFi of direct connections to the internet.  If I was unable to stream music when I wanted to listen to songs, I'd try to find a different solution.
> 
> I use Google's streaming service that I pay $7.99/month (promo rate locked in for early subscriber) to use.  I play my music on anything that has the ability to use the Chrome browser, the iOS app, or the Android app.  At home I have a couple of Chromebooks that I use almost exclusively as "jukeboxes" that can play the entire Google catalog of over 30 millions songs seamlessly with a thousand or so of my own music that is not otherwise made available to stream.  I can play full albums, single songs, genre-specific radios, and radio created by artist, album, or single songs.  I am able to maintain a library of my music that is displayed by genre, artist, album, or song with album covers and wiki entries of the artist and album in many cases.  When I shuffle my library or play any radio, not only are the subscription songs played, but my own music can also be included if the songs are related to the type of radio I selected.
> 
> ...


----------



## OddE

keithemo said:


> Back when phone lines were "discrete lines" rather than IP-based, the theoretical limit for a single line was 64k; "56k modems" were adaptive, so you got slightly different speeds depending on line conditions, but their theoretical maximum was somewhere around 52k (because of a sort of technicality, which "used up" some of the theoretically possible data bandwidth, they couldn't actually do 56k - and I never saw one do full 52k except in a lab setting). As I recall, US Robotics was considered by most people to be the top premium brand for such devices back in those days.


 
  
 -Once again, Claude Shannon is our man; the Shannon-Hartley channel capacity theorem states (basically, any signal processing buff, please bear with me) that in the presence of noise, there is a finite limit to the number of bits per second which can be transmitted without error in a band-limited channel.
  
 (The idea is that as there's noise (and bandwidth constraints), you cannot have an infinite number of signal levels - you'll need some margin to reliably determine which symbol you received - and, hence, you also have a finite capacity. Bummer.)
  
 The capacity (in bits/second) is the available bandwidth times the binary logarithm of (1+ SNR), or C=B*log2(1+S/N) for the algebraically inclined.
  
 So, doing the numbers in my head, to get 56k over a 3kHz phone circuit,  you'll need an SNR on the order of 50dB or so, slightly more, I'd say.
  
 Edit: And 50dB sounds like way more than you'd expect from your average phone circuit, hence the problems with obtaining the theoretical bandwidth; IIRC the last modem I used at my parents' prior to the arrival of broadband got me on the order of 36-38k on a good day - uploading it got pretty close to the maximum 33.6kbps.


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## reginalb

keithemo said:


> For one thing, I like to own the music I really care about, and to actually be in possession of it - whether it's in the form of plastic, or of a computer file, which I can back up. Not that it's especially likely, but there have been cases where Kindle books "disappeared" from the collections of people who had "purchased" them when they were found to be illegal to sell due to copyright or other issues. There was also a video service a very long time ago called DivX (not to be confused with the video format of the same name). With this service, you received a disc for free, then paid for a license for either a single viewing, or "lifetime ownership". However, one day the company went out of business, and everyone who thought they owned videos found that they now owned unplayable discs - and their "video collection" had magically disappeared. I am simply not sufficiently confident that a given service will remain in business, or that a given album won't be "pulled" due to some sort of legal or contractual dispute, to trust someone else to "keep track of" my music. (It would ruin my entire week if my favorite song were to disappear one day, or my streaming service were to go under, and I couldn't find another one which had all of my favorite music. Companies - and services - tend to come and go... and I simply don't like to take that risk with my music collection.)


 
  
 Obviously different strokes, but I would say a few things in reply to this:
  
 The likelihood that Google goes out of business is pretty darn small, as you correctly point out, the more likely scenario are individual albums being pulled for copyright reasons. But I ask you: So what? Just buy that album, and upload it to your Play Music library (or keep it directly on your device). How could it possibly ruin your week? You'd just go pick it up elsewhere and move on with your life. I am also an All Access subscriber at $7.99/month, that's less than $100 per year to have access to every album that Google gets access to. I used to spend $100 in a sitting buying music, now I get more, for FAR less money. 
  
 The DivX comparison really isn't a fair one, either. I haven't bought physical media, and if Play Music shutters, I will just switch to a different service, there are plenty to choose from, all the while saving a lot of money.
  
 And the curated stations are just a wonderful way to discover music. Obviously, you don't have to use the service, it's not for everyone, but your reasoning seems to veer towards Luddism.


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## sonitus mirus

> I'd have to say that, for me, it's more a matter of "the principle of the thing" or "my comfort zone" rather than totally practical considerations.
> 
> For one thing, I like to own the music I really care about, and to actually be in possession of it - whether it's in the form of plastic, or of a computer file, which I can back up. Not that it's especially likely, but there have been cases where Kindle books "disappeared" from the collections of people who had "purchased" them when they were found to be illegal to sell due to copyright or other issues. There was also a video service a very long time ago called DivX (not to be confused with the video format of the same name). With this service, you received a disc for free, then paid for a license for either a single viewing, or "lifetime ownership". However, one day the company went out of business, and everyone who thought they owned videos found that they now owned unplayable discs - and their "video collection" had magically disappeared. I am simply not sufficiently confident that a given service will remain in business, or that a given album won't be "pulled" due to some sort of legal or contractual dispute, to trust someone else to "keep track of" my music. (It would ruin my entire week if my favorite song were to disappear one day, or my streaming service were to go under, and I couldn't find another one which had all of my favorite music. Companies - and services - tend to come and go... and I simply don't like to take that risk with my music collection.)
> 
> ...


 
  
 I certainly own my own music as well.  I have a collection of my favorites that go back to when CDs were first put on the market.  I even have a collection of records stored away.  But for about a quarter per day, I have the privilege of being able to enjoy over 30 million songs if I have the time and the ability to do so; which, as it turns out, is quite frequently.  Again, I am not looking specifically for a solution to stream music all the time, it is simply always available and the most convenient method for me right now.
  
 I've been a subscriber to some form of music subscription service for over 12 years now.  I started with Rhapsody (thought it was Rap City when someone first told me about it), then joined MOG and Spotify and kept both for a few years before settling for Google Music All Access when it came out.  I believe that I have tried all of the available services for at least a trial.  I even have an account with Tidal that I have activated for a few months at a time.  With Google, I have thoroughly tested the audio quality against many of my own CDs and even a few HD tracks and I have come to the conclusion that I am unable to hear any differences.  It is always available to me, and any music that is not available through the subscription service can easily be uploaded to be included as part of the Google Music interface as if it were simply just another album or artist.
  
 If it goes away tomorrow it would be a shame.  I have a Python script that I run every once in a while that grabs the Artist - Album in my Google favorites section and lists these in a spreadsheet.  I will always have this list, and if I need to purchase some of this music later of rebuild this library on a new/better service, I will at least have the list to assist with this effort.  I've been through this before when changing subscription services in the past, and every time there have been 3rd-party tools available to help make this process rather simple.
  
 What is serious listening to you?  For me, it is relaxing to music while using my computer to research information about the artist, song, lyrics, or practically anything that interests me while listening.  I might look at anything from photos or album art to an online map's street view showing an old house that Robert Plant use to stay in for a summer.  Sometimes I enjoy the music with a strong cup of coffee, and other times I listen while enjoying a cold beer.  Occasionally I'll just sit there and stare into the vast nothingness as the music transcends me to a dreamlike state of euphoria and bliss.  Okay, maybe that is taking it a bit too far, but I have had those moments where I got lost in the music for a minute or longer.  Other than the many listening tests that I have done, this is about as serious as I get when listening to music.  
  
 From my experience, when you swallow your own saliva or take a breath, that action probably has a greater impact on what you are hearing compared to any differences between a well-encoded lossy file and that same song played directly from a CD.  No lossy format will probably ever be absolutely inaudible at all times, in the same sense that no clock or watch will ever be absolutely perfect with its ability to keep time.  At some point, though, the clock or the lossy file becomes incredibly useful and nearly perfect for its intended application.


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## RRod

sonitus mirus said:


> What is serious listening to you?  For me, it is relaxing to music while using my computer to research information about the artist, song, lyrics, or practically anything that interests me while listening.  I might look at anything from photos or album art to an online map's street view showing an old house that Robert Plant use to say in for a summer.  Sometimes I enjoy the music with a strong cup of coffee, and other times I listen while enjoying a cold beer.  Occasionally I'll just sit there and stare into the vast nothingness as the music transcends me to a dreamlike state of euphoria and bliss.  Okay, maybe that is taking it a bit too far, but I have had those moments where I got lost in the music for a minute or longer.  Other than the many listening tests that I have done, this is about as serious as I get when listening to music.
> 
> From my experience, when you swallow your own saliva or take a breath, that action probably has a greater impact on what you are hearing compared to any differences between a well-encoded lossy file and that same song played directly from a CD.  No lossy format will probably ever be absolutely inaudible at all times, in the same sense that no clock or watch will ever be absolutely perfect with its ability to keep time.  At some point, though, the clock or the lossy file becomes incredibly useful and nearly perfect for its intended application.


 
  
 I do get tired of the sentiment from the "other side" that none of us sciencey types are "serious" music listeners (not that Keith meant this). I'll sit down and listen to some of the best-recorded classical music out there, score in hand, comparing oboe parts between recordings. But you know, since I don't instantly think hi-res will fix the problems with Zeppelin IV I must be some kind of idiot.
  
 You should re-brand your saliva comment as a motto or something, it's too good. "AAC: Less annoying than phlegm." (or more transparent than phelgm…)


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## sonitus mirus

rrod said:


> I do get tired of the sentiment from the "other side" that none of us sciencey types are "serious" music listeners (not that Keith meant this). I'll sit down and listen to some of the best-recorded classical music out there, score in hand, comparing oboe parts between recordings. But you know, since I don't instantly think hi-res will fix the problems with Zeppelin IV I must be some kind of idiot.
> 
> You should re-brand your saliva comment as a motto or something, it's too good. "AAC: Less annoying than phlegm."


 
  
 I enjoy Keith's participation in many of the discussions I have read.  I was genuinely interested in some feedback on what others consider to be serious listening sessions when compared to having music on in the background while working or just listening to fill the silence.


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## castleofargh

I'm biased with beds and love everything that can be done in one, my biggest music moments are when I'm lying down with my eyes closed in an otherwise almost perfectly silent dark room. those are the most serious, music moments I get. I feel like I can notice anything.
  
 next in line would be like RRod said, when trying to decide what interpretation of a classical piece I prefer(don't you ever wish you could Frankenstein the songs and use the recording engineer of A, with the chorus of B, and the tempo of C?).
  
 else I don't really pay attention to music, and if I do it's usually to go sing along, or air guitar like a mad man for one particular song that grabbed my attention. I don't think that can count as serious listening even though I'm mighty concentrated ^_^.
  
  
  
  
 about breathing and swallowing, I feel exactly the same and do no worry about minute quality changes in sound for the same kind of reasons. most of my music experience is done with IEMs when I'm on the go, so it doesn't start well for a perfect music experience. I can't stand to walk with silicone tip on my IEMs, because of the loud "thump" that goes with each step. how could I be concerned about stuff down -60db on mp3 at that moment?
 when there is some wind and the IEM is vented, you feel like you get a storm going on in your ears. and that's not even accounting for all the situations with trains, planes, subway, cars on the street... I couldn't care less about having a better sound, what I need is a better isolation.
 when I'm home listening with some better gear and get noises from the street...
 there is always something louder than the limits of my format. probably why I value so much my music at night when all is calm and quiet. but even then, I don't feel I'm improving sound when going to flac, not like I feel improvement by changing my source and getting rid of some background hiss, or when I have found a proper EQ for my headphone, or found a knob position on my amp that doesn't have much of a channel imbalance. those are my hifi moments.


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## KeithEmo

I guess it all depends on what you listen to.
  
 There is a certain album that I rather liked years ago... when it was only available on vinyl. It has since gone out of print, and was reissued once on CD about twenty years - in France. I now find myself in a situation where I simply cannot obtain it (the last one sold on eBay went for over $100, and that was years ago, and I haven't seen one since). I really don't want to imagine that happening with a significant number of other albums I enjoy - perhaps if someone were to lose or renegotiate a licensing deal. (I don't know if it's still the case, but, for some time, _NONE_ of the Beatles albums were available on Spotify - because they hadn't reached a mutually agreeable licensing agreement with the owner. And whole blocks of movies and TV shows "disappear" when the license owner decides to hoard them to push up market values.)
  
 I'd be happy to pay Google $100 for that one single album - but that option isn't available... so they cannot meet my need. (And the fact that they'll happily give me 30 million songs that I'm not interested in doesn't really make me feel differently about that. It's handy, but it doesn't solve the original problem. I do in fact subscribe to a streaming service, and I use it, in addition to my personal collection. Personally, I've never been especially excited by the choices made for me by "curators", but I can see how that's an individual thing.... but my point was that streaming  doesn't satisfy _ALL_ of my needs... and, in fact, the whole "model" of simply "renting" but not "owning" music never will... which is why I would hate to see it replace other methods of owning music to the point that they disappear.)
  
 The other "problem" I have with even some of the download services is more insidious. In the past several years, several groups I really like have decided to release albums on the iTunes store. Unfortunately, in practical terms, this means that those albums are _ONLY_ available in low-quality lossy format. I find this a great loss - because it means that those albums are simply not available in good quality - from anywhere - at any price. (And, yes, I consider streaming services, which usually cater to the "lowest common denominator", as being partly to blame when that occurs.)
  
 Quote:


reginalb said:


> Obviously different strokes, but I would say a few things in reply to this:
> 
> The likelihood that Google goes out of business is pretty darn small, as you correctly point out, the more likely scenario are individual albums being pulled for copyright reasons. But I ask you: So what? Just buy that album, and upload it to your Play Music library (or keep it directly on your device). How could it possibly ruin your week? You'd just go pick it up elsewhere and move on with your life. I am also an All Access subscriber at $7.99/month, that's less than $100 per year to have access to every album that Google gets access to. I used to spend $100 in a sitting buying music, now I get more, for FAR less money.
> 
> ...


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## reginalb

keithemo said:


>


 
  
 What album was it? If you're worried about quality, 320k MP3 is pretty darn high, as is 256k AAC which iTunes offers. And by pretty high, I mean transparent to most, if not all, humans. But then there's Tidal lossless. Now, if they don't have a large portion of the music you like, that's clearly a different story, but with their enormous selections, these services are really an amazing advancement for music lovers.
  
 Now I'm working on collecting speakers and what not to set up Google Cast throughout my place - Sonos has always seemed overpriced to me. While standalone cast speakers (like those from Sony) are close in price to the equivalents from Sonos - the Sonos receiver has always seemed the most insane to me in terms of price. I mean, just to receive your stream, and run it through my great HT system costs $350? The $35 Chromecast audio really sealed that one for me. 
  
 Any way, my point is that far from just being good for the lowest common denominator, I think these are extra great advancements for the biggest lovers of music.


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## Don Hills

odde said:


> ...
> So, doing the numbers in my head, to get 56k over a 3kHz phone circuit,  you'll need an SNR on the order of 50dB or so, slightly more, I'd say.
> ...


 
  
 The real bandwidth limit is because the analogue signal from the modem is digitised by the telco's "digital modem" to 64K bits/sec (sampling rate 8 KHz, 8 bits/sample). The 56K theoretical limit is because every 6th LSB is "stolen" for in-band signalling. This doesn't matter much for speech, it just reduces the S/N ratio slightly. But since there is no way to tell which bits were stolen, the LSBs can't be "trusted" so have to be discarded. 8 K samples/sec x 7 bits = 56 K bits/sec. (In practice, limited further by FCC restrictions and line length.)


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## sonitus mirus

I don't know if the Beatles are allowed on any streaming music service, which is part of the reason I used their albums in my screen captures I posted earlier.  I have both the stereo and mono box sets uploaded into Google Music.  There probably isn't a single solution that will fully satisfy every possible situation for every person.  For me, when I want to listen to music, I want to have the full library that is available to me through Google Music, and I want the sound quality to be at least as good as a CD to my ears.  I have had my own special versions of music that I had difficulty in procuring and probably spent way too much money to obtain.  Now, though, once I am able to get this music I can upload it into Google Music to enjoy with the other millions of songs that are available.  
  
 I admit that I really wish Google was using AAC rather than MP3.  I tried Apple's music subscription service, but their process takes the liberty of matching my own ripped music to whatever version they might have available.  This was a huge problem when I attempted to upload both a mono and stereo version of the Beatles' albums, as iTunes only had the stereo version and converted all of the mono tracks to the stereo equivalent, complete with the ridiculously hard-panned tracks on some of the earlier releases that I don't enjoy with headphones. (I'm ok with this with my speaker setup)  With Google, my uploaded Lame-encoded vbr -0 files are maintained in the exact format that I ripped them into, and they do not alter the file in any way.  When I download my own music back to a PC from Google, the audio file properties are maintained in the exact format that they were originally ripped from the CD.


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## OddE

don hills said:


> The real bandwidth limit is because the analogue signal from the modem is digitised by the telco's "digital modem" to 64K bits/sec (sampling rate 8 KHz, 8 bits/sample). The 56K theoretical limit is because every 6th LSB is "stolen" for in-band signalling. This doesn't matter much for speech, it just reduces the S/N ratio slightly. But since there is no way to tell which bits were stolen, the LSBs can't be "trusted" so have to be discarded. 8 K samples/sec x 7 bits = 56 K bits/sec. (In practice, limited further by FCC restrictions and line length.)


 
  
 -Sure, but this is probably (it has been quite a few years since I dabbled in telecom) is one of those chicken-and-egg-situations - Shannon-Hartley is a hard limit to the amount of data which can be passed through a channel; they could've sampled it at 24/192 and still the bandwidth limitation imposed by the analog front-end would negate any benefit; you'd still only get the maximum bit rate determined by bandwidth and noise. (So, qualified guess - 8kHz/8bit was chosen because it would be enough to capture anything which could possibly be present on an analog line.)
  
 Hm. I guess it is time to dig out my old, dog-eared copy of Tanenbaum; your post in a couple of sentences showed me that I'd forgotten lots. (Not that the intricacies of telecom networks is something I run into often in the subsea/offshore business, but I don't like forgetting things I've once known...)


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## KeithEmo

The album is Streetlight Shine by The Shirts. (It is in the same genre as Blondie, and was popular enough at the time to be reissued several times on vinyl. Many data sources claim that it was never issued on CD, but it was in fact issued on CD in France - I believe on the Harvest label. However, I've never been able to locate a copy of the CD for sale; one historical search suggested that a copy had sold on eBay for $110 years before.) I do in fact have a copy in 320k MP3 - which does sound pretty good - and, in fact, quite "acceptable" but, since I like the album quite a bit, I would like to pick up a full quality copy eventually.
  
 Different people clearly look at what they're listening to differently. When I'm listening casually, like in the car or at work, I'm perfectly satisfied to listen to what's on the radio, or from a "curated playlist" of things I "might like". However, when I sit down to do serious listening, I usually want to hear a particular version of a particular album - and, in that situation, the fact that I have a choice of 30 million _other_ songs doesn't really interest me. Even though I keep all of my music on a hard drive, and play it by computer, _mentally_ I still tend to think of it as flipping through my collection and picking out a song to play - and the possibility that a specific favorite song might "go missing from my collection" worries me. I most certainly agree that it's great to have easy access to such a huge variety of music, and the chance to hear new music without buying it, including relatively rare songs that never get played on the radio, is also very nice. (Therefore, as I said, I very much like streaming music services - I just don't want to see them succeed to the point where they entirely replace "direct ownership".)
  
 I would also agree that modern lossy formats (AAC256 or 320k MP3) usually sound quite good, but, when I do a direct A/B with a CD, I do sometimes notice differences. A few times I've heard something I never noticed before when listening to a song then, when I pulled out the original CD, found that it in fact wasn't there - and so was some sort of artifact of the MP3 process. Once that happens, like noticing a bug on your windshield, it distracts me from enjoying the album until I eliminate it - and I wonder what else isn't "quite as it should be". To me, I would rather use ten times as much space to store my entire collection, than risk this happening on one out of a hundred albums I play; I just consider it the cost of minimizing the chance of being annoyed. But that's just me and I can well understand how other people might not find it to be such a big deal.
  
 The problem I find with most streaming services and even 'cast devices is a lack of concrete information.
  
 If you were buying a copy of a classic book, you probably wouldn't be willing to buy a copy from a company who said: "All of the books we sell read just as well as the original versions, but we're not promising that we didn't change a few words here and there". Yet, to me, that's exactly what's happening when someone claims that a copy of a file "sounds audibly the same as the original but may not be bit perfect". If the file or stream is bit-perfect, then all of the words are in fact the same; if not, then we're both acknowledging that words have been changed, and I'm being asked to trust someone else that I'll be pleased with the result. Since what I'm looking for is in fact the original, I find this an unreasonable situation, and I prefer to avoid it. (I _KNOW_ that the original will read/sound precisely like the original; since it's not all that hard to get a perfect copy of the original, why should I risk anything else?)
  
 I've tried to find out specifically whether certain streaming devices are in fact bit-perfect; and, often, I was totally unable to do so. (A few years ago one of the major vendors offered a cute - and quite economical - streaming device, which was claimed to offer "CD quality". Yet, when I succeeded in actually talking to the head engineer of the US division of the manufacturer, even he was unable to say for sure whether it altered the data stream or not - because the programming which might or might not have preserved the accuracy of the data was outsourced.) Clearly, to many vendors, this isn't a priority. (And what you're paying for with higher-priced solutions like Sonos is often simply a promise or commitment that they are delivering a certain level of quality; as opposed to being able to hope that the lower cost solution does equally well, but being unable to know for sure.)
  
 Of course, you do have to always avoid products that are simply marketed using magic words like "audiophile quality" - but don't have the technical performance to go with them.
  
 An excellent analogy would be color calibration with color printers.
  
 The reality is that a typical $100 ink jet printer actually prints colors quite accurately. Therefore, if you buy such a printer, and set the color controls to "automatic", your photos will probably come out looking quite good. However, if you're a professional printer, or even an amateur who is "serious" about printing color prints and photos, you still buy a printer that's capable of actually being calibrated, and then you buy a calibrator, and you take the time to use it. The reason is that this allows you to know for an absolute fact that your colors are correct, rather than just assume that they are, or guess that they're "close enough". On a given day, a properly calibrated printer may not give you a print that looks any better than that $100 consumer printer; what you're paying extra for is the certainty that it always will, and the ability to avoid wondering or worrying about it. Obviously, this is worth it to some folks and not others.
  
 Quote:


reginalb said:


> What album was it? If you're worried about quality, 320k MP3 is pretty darn high, as is 256k AAC which iTunes offers. And by pretty high, I mean transparent to most, if not all, humans. But then there's Tidal lossless. Now, if they don't have a large portion of the music you like, that's clearly a different story, but with their enormous selections, these services are really an amazing advancement for music lovers.
> 
> Now I'm working on collecting speakers and what not to set up Google Cast throughout my place - Sonos has always seemed overpriced to me. While standalone cast speakers (like those from Sony) are close in price to the equivalents from Sonos - the Sonos receiver has always seemed the most insane to me in terms of price. I mean, just to receive your stream, and run it through my great HT system costs $350? The $35 Chromecast audio really sealed that one for me.
> 
> Any way, my point is that far from just being good for the lowest common denominator, I think these are extra great advancements for the biggest lovers of music.


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## Don Hills

odde said:


> ... this is probably ... one of those chicken-and-egg-situations - ...
> 
> Hm. I guess it is time to dig out my old, dog-eared copy of Tanenbaum; your post in a couple of sentences showed me that I'd forgotten lots. (Not that the intricacies of telecom networks is something I run into often in the subsea/offshore business, but I don't like forgetting things I've once known...)


 
  
 Indeed, chicken and egg. The choice of 8 KHz sample rate was based on the previous 4 KHz spacing of channels in analogue carrier systems, itself based on what was practical to implement in analogue technology at the time.
  
 I've probably forgotten more than I ever learned, too.  I started my career as a telegraph and data engineer. We still had Morse code sets in the storeroom for emergencies...


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## KeithEmo

Indeed - and that determines the bandwidth limit for the channel itself.
  
 However, specifically with standard modems, the type of modulation used converts bit patterns to "symbols", which are then transmitted. That 56k theoretical limit is further reduced to 52k because, of the "symbol constellation" that is used to encode information in the standard modulation scheme, a few of the symbols are "disallowed". (I forget, but it probably had something to do with too many ones or zeros next to each other being difficult for the hardware to handle).
  
 Back when such things were routinely used, there also used to be digital services available that directly used the phone company's digital channels. For home use, in most places, you could get ISDN service (Integrated Services Digital Network??... it's been a while). With ISDN, you got an actual 64k digital channel, and the "dialing" was also handled digitally - which gave you some extra features as well. (It required a special connection to bypass the line card out on the pole, and a special modem, which also weren't cheap.) And, since channels were handled in pairs, and the physical line coming into the d-mark carried two channels, many ISDN modems could "bond" two channels to give you a single 128k "virtual" digital channel. It was very nice, and _VERY_ reliable, and usually very fast - if you could afford it. (The "dialling and connecting" process took under a second if the connection was machine-to-machine.)
  
 Back when this was current (up to the 1970's or 1980's) a 128k bonded ISDN line cost between $100 and $200 a month, if you could get the service in your area, and a T1 line, which gave you 1.544 mBits/second, was about $1500 a month in New York... which is why I always find it entertaining when people whine about "_ONLY_ getting 20 or 30 mBits for $50 a month"; in 1975, a connection that gave you the same bandwidth as a typical cable modem would have cost between $10k and $20k a month. (You would have needed _TWO T3 lines_ to get 50 mBits of bandwidth.)
  
 Quote:


don hills said:


> The real bandwidth limit is because the analogue signal from the modem is digitised by the telco's "digital modem" to 64K bits/sec (sampling rate 8 KHz, 8 bits/sample). The 56K theoretical limit is because every 6th LSB is "stolen" for in-band signalling. This doesn't matter much for speech, it just reduces the S/N ratio slightly. But since there is no way to tell which bits were stolen, the LSBs can't be "trusted" so have to be discarded. 8 K samples/sec x 7 bits = 56 K bits/sec. (In practice, limited further by FCC restrictions and line length.)


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## KeithEmo

I agree with your comment about MP3 vs AAC - but to me it goes further. Bandwidth and storage are cheap; but, even beyond that, they are both readily _available_. Therefore, there's no legitimate reason why a service like iTunes can't offer both lossy and lossless formats - even if they were to charge more for the extra bandwidth and space required by the lossless files (then it would be up to the individual to decide whether to pay the extra charge - rather than the higher-quality files simply not being available). The reason Apple "took the liberty" of matching your uploaded files to ones they already had in their system is because they placed their convenience, and saving themselves a few cents, _ABOVE_ providing you the service you wanted. (And it was the convenience aspect of it that prevented them from offering you the option of using your version of the files anyway for two cents more per song.)
  
 In recent years, a few of the bands I like have chosen to "publish songs or albums on iTunes". What this means is that those particular albums are _ONLY_ available in lossy format. (I can't get a CD quality copy for a little more, and I don't get to make my own decision about whether to pay the difference; in short, the question of what constitutes "acceptable quality" has been reduced to that lowest common denominator. Luckily, many bands still offer to sell full quality versions directly - but not all.)
  
 I also find it interesting that Apple, who claims to be "on the musicians' side", and to champion good quality music, launched the iTunes store only offering 128k AAC DRM-protected versions of what they sold (apparently they'd decided on behalf of their customers that it was "good enough"). Then, when it became obvious that it _wasn't_ good enough for many of their customers, they raised that to 256k AAC without the DRM - for a higher price, and an "upgrade fee" for people who had already purchased a given album (apparently, at that point, AAC 256 was "good enough"). Interestingly, in a paper I read a while ago, where Apple described "how musicians should master music for release on the iTunes store", Apple recommended using 24/96k lossless. (Clearly they're making sure they'll be prepared when there's enough of a market to buy everything again at 96k....)
  
 Honestly, I don't mind paying extra for a higher tier of service, or a better quality product, and I also have no problem when other folks decide that they'd prefer not to, but I do resent when someone else "decides" on my behalf that "I don't need it", and then it becomes entirely unavailable based on that decision. (Over my lifetime, I've seen many products that I liked cease to be available "because not enough people bought them". I never flew in a Concorde, and quite possibly never would have, but I still find it somewhat depressing that most human beings have now _LOST_ the ability to make supersonic passenger flights. It worries me that, if Google's service using MP3 files were to become popular enough, we might actually lose the ability to buy lossless files. At the very least it will help provide "supporting statistics" to "prove that most people don't hear or care about the difference". The market does tend to "settle" at a point where most - but not all - customers are satisfied with a given service or product. Luckily, bandwidth and storage space have become so cheap that they provide little incentive to go in that direction.... )
   Quote:


sonitus mirus said:


> I don't know if the Beatles are allowed on any streaming music service, which is part of the reason I used their albums in my screen captures I posted earlier.  I have both the stereo and mono box sets uploaded into Google Music.  There probably isn't a single solution that will fully satisfy every possible situation for every person.  For me, when I want to listen to music, I want to have the full library that is available to me through Google Music, and I want the sound quality to be at least as good as a CD to my ears.  I have had my own special versions of music that I had difficulty in procuring and probably spent way too much money to obtain.  Now, though, once I am able to get this music I can upload it into Google Music to enjoy with the other millions of songs that are available.
> 
> I admit that I really wish Google was using AAC rather than MP3.  I tried Apple's music subscription service, but their process takes the liberty of matching my own ripped music to whatever version they might have available.  This was a huge problem when I attempted to upload both a mono and stereo version of the Beatles' albums, as iTunes only had the stereo version and converted all of the mono tracks to the stereo equivalent, complete with the ridiculously hard-panned tracks on some of the earlier releases that I don't enjoy with headphones. (I'm ok with this with my speaker setup)  With Google, my uploaded Lame-encoded vbr -0 files are maintained in the exact format that I ripped them into, and they do not alter the file in any way.  When I download my own music back to a PC from Google, the audio file properties are maintained in the exact format that they were originally ripped from the CD.


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## reginalb

keithemo said:


> The reason Apple "took the liberty" of matching your uploaded files to ones they already had in their system is because they placed their convenience, and saving themselves a few cents,_ABOVE_ providing you the service you wanted.


 
  
 This is not accurate. Firstly, given the number of users, the savings are probably in the millions, not a few cents. They can't serve every individual's whims, they have to make a decision, and in doing so make a cost-benefit analysis. There are alternatives already in the market, so they want to fill a niche in a way that both saves them money, ensuring they can make a profit, and will be a service that as many people as possible are happy with. Not everyone will be, and there are _other _services for those that aren't.
  


keithemo said:


> I can't get a CD quality copy for a little more, and I don't get to make my own decision about whether to pay the difference; in short, the question of what constitutes "acceptable quality" has been reduced to that lowest common denominator.


  

 Again, we aren't the "lowest common denominator." I have 0 doubt that you would never be able to pick out a 256k VBR AAC encoded by Apple vs. your FLAC, or Redbook wav. The format is transparent if you aren't a cat.
  
 Quote:


keithemo said:


> I also find it interesting that Apple, who claims to be "on the musicians' side", and to champion good quality music, launched the iTunes store only offering 128k AAC DRM-protected versions of what they sold (apparently they'd decided on behalf of their customers that it was "good enough"). Then, when it became obvious that it _wasn't_ good enough for many of their customers, they raised that to 256k AAC without the DRM



  
 It was competition (from Amazon) that did this. The market that you lament is what forced Apple's hand. 
  
 Quote:


keithemo said:


> I can't get a CD quality copy for a little more, and I don't get to make my own decision about whether to pay the difference; in short, the question of what constitutes "acceptable quality" has been reduced to that lowest common denominator.



  
 The market has pushed towards useless high-res formats, and it's unlikely to change any time too soon, so you're probably safe. But again, I resent your repeatedly calling anyone that disagrees with you (based on silly things like facts) the lowest common denominator. Contrary to that, I love music, and have spent a considerable sum of money building both a hi, and head fi system, so that I can listen to music in high quality both at home and on the road. But I also understand where diminishing returns set in, and so I'm not about to break the bank. 
  
 By the way, I looked up that Shirts album, you don't appear to be alone in your quest to find it on CD. Perhaps one day the label that owns the rights will release it. I do hope it's not in MP3 or AAC, though.


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## sonitus mirus

I would definitely love to have the option to purchase lossless files of music that did not carry the price and format of HD.  Just give me a FLAC/ALAC file at 16/44.1 copied exactly from the master used for the CD pressings.  I also love the idea of Tidal's lossless streaming service, and I would happily pay twice as much or even more to have a similar library size as any of the major lossy services AND the option to upload my own music.  The biggest joy of Google Music for me is the library system and the user interface.  It's not perfect, but it is the best music player for me.
  
 I'm sure we are safe from having lossy subscription services taking over as the only option.  I just hope the music labels don't get lazy and only provide lossy versions of their catalogs with a few HD exceptions for the more popular stuff.  I could see them limiting the production of CDs to the point where they are as difficult to come by as vinyl, and if they don't bother making RB-quality lossless versions available, we are going to be left with only AAC/MP3 versions with a relatively small number of HD choices selling for a lot more than what the cost of a CD currently runs.  If that does happen, look for used CD sales to rise and perhaps even a CD rental service to prop up.  Oh, and of course pirating would certainly increase and be used to justify some sort of DRM that will only impact those of us that try and abide by the rules.


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## KeithEmo

I'm going to address your points in order.....
  
 1) The savings _PER SONG_ are a matter of a few cents... If I'm paying $1 for a song download, I really don't care that, by giving me a lower quality copy, and saving two cents by doing so, _THEY_ get to save a lot of money. (There is an old story, which may or may not even be true, about a marketing exec who saved a company that sells bottled olives tens of thousands of dollars, by suggesting that they put one less olive in each bottle they ship. This made him a hero in the accounting department. However, where do you draw the line? How about removing two olives? How about three?)
  
 2) I'm glad that _YOU_ know for a fact what _I_ can and cannot hear. I've never tried AAC myself, other than to listen to casually, but I did try several "good" MP3 decoders, using sample rates up to 320k VBR, and every one of them produced at least some audible artifacts on some songs - when compared directly against the original.  
  
 2) "Lowest common denominator" isn't some sort of an insult... in this context it is simply the proper name for the lowest level or amount of anything that seems to meet everyone's requirements. In fact, by definition, the lowest common denominator is what is in effect "good enough to do the job perfectly"; if you really see no reason to pursue technical excellence for its own sake - and past that point, and you believe that there's no reason for anyone else to want FLAC files instead of AAC files because you don't hear any difference, then you should take it as a compliment (the lowest common denominator is in fact the most efficient solution).
  
 3) I'm sort of confused about your point re Amazon vs Apple. If Apple's 128k AAC files really were audibly perfect, and people actually believed that, then no amount of competition from anybody else _SHOULD_ have forced them to switch to higher quality ones. And, the converse of that, if Amazon was in fact providing _BETTER_ quality files, then Apples 128k ones were _NOT_ good enough.  
  
 Obviously what happened was that, when Apple was "the only game in town", there was no question about whether the quality level they chose was "good enough" or not - since there were no other options. (At that point you could either download "128k files or nothing".) They chose a quality level that they believed most of their customers would be willing to buy - trading quality for convenience - and, at that point, their choice wasn't unreasonable. Then, when Amazon entered the market, it actually became a competitive market - where consumers were free to choose which product to buy and what they were willing to pay for it... and the market "showed" that it was in fact willing to pay more for high quality files... at which point Apple had to offer a truly competitive product.
  
 My only "problem" with the situation was that they were doing exactly what many people here are now accusing the purveyors of high-res files of doing.... Rather than actually determine what was in fact audibly perfect, they instead decided what quality they were willing to sell, based on _OTHER_ considerations, then launched a campaign to _CONVINCE_ people that it was good enough to suit their needs. (As far as I can tell, you _ARE_ agreeing that the 128k AAC files they were selling originally are _NOT_ audibly perfect, right?) 
  
 4) I don't lament the market at all. What I lament is a market where the market factors drive some of the options to extinction. In other words, I think we're darned lucky that Apple's monopoly wasn't so well established that, by the time Amazon became interested, it wasn't worth their entering the market. If that had happened, Apple would still be selling inferior 128k files, and we wouldn't have the option of buying better ones....   And, if that shift reached its logical conclusion, the next ten albums I buy might _ONLY_ be available in that format - and wouldn't _THAT_ be sad? As I mentioned, I have never flown on a Concorde, and probably never would have - yet I still lament that the option is no longer available to me - or anyone else.
  
 (I seem to recall a lot of "interested audiophiles" commenting at the time that those 128k AAC files didn't sound very good. However, the response to that apparently was that, since most of Apple's customers thought they sounded good enough, it wasn't worth their while to bother with the small percentage of the market that disagreed. And, to put it bluntly, that was almost certainly a sound business decision at the time, but it didn't do _ME_ much good - since I was part of the minority they weren't bothering to please.)
  
 5) On that last one we agree exactly (and, as I said, until then, I do have an 320k VBR MP3 version - and it isn't all that bad 
	

	
	
		
		

		
		
	


	




 )
  
 Quote:


reginalb said:


> This is not accurate. Firstly, given the number of users, the savings are probably in the millions, not a few cents. They can't serve every individual's whims, they have to make a decision, and in doing so make a cost-benefit analysis. There are alternatives already in the market, so they want to fill a niche in a way that both saves them money, ensuring they can make a profit, and will be a service that as many people as possible are happy with. Not everyone will be, and there are _other _services for those that aren't.
> 
> Again, we aren't the "lowest common denominator." I have 0 doubt that you would never be able to pick out a 256k VBR AAC encoded by Apple vs. your FLAC, or Redbook wav. The format is transparent if you aren't a cat.
> 
> ...


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## sonitus mirus

> 2) I'm glad that _YOU_ know for a fact what _I_ can and cannot hear. I've never tried AAC myself, other than to listen to casually, but I did try several "good" MP3 decoders, using sample rates up to 320k VBR, and every one of them produced at least some audible artifacts on some songs - when compared directly against the original.


 
  
 It certainly is not a fact that anyone knows what you can or cannot hear, but I really don't believe this at all.  To me, these are simply empty claims that you believe to be true.  I'd be interested in learning a bit more about the tests that you conducted, if only from your best memories of them.  What were these "good" encoders?  How long ago were these tests conducted.  What type of music were your testing?  Did you use any type of ABX tool?  I get that you don't have any interest at all in lossy formats, but I don't believe it is fair to justify your position with unfounded evidence promoted merely by speculation and gut feelings.


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## reginalb

keithemo said:


> "Lowest common denominator" isn't some sort of an insult


 
  
 I'll take your point here, audiophiles do tend to look down their nose at objectivists. You may not, and obviously this is a metaphorical phrase, I've never known it to be used in the way you describe. I could have just mistaken your meaning.


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## KeithEmo

The most recent time I payed specific attention to lossy formats was about seven or eight years ago. I tried both the LAME encoder (I'm pretty sure that hasn't actually changed much if at all since then - they don't update that much lately) and some other well-regarded one (I think it was the Fraunhofer). Since I was trying to hear "how good it could do" I used their "highest quality setting" - by their description - which was VBR, independent channels, and at either 256k or 320k. The album I tried it with was one of the CD versions of Dark Side of the Moon. I don't have any doubt that many of the MP3 files you find floating around the Internet are poorly or improperly encoded, and many are also from badly ripped or otherwise obtained sources, so they aren't a fair test - but I was curious about "the state of the art" in lossy compression. Therefore I started with a bit-perfect rip of what I considered to be a good CD version of the album. (I've always found Dark Side of the Moon to be a good test album - because I'm quite familiar with it, and because it seems to have a lot of music and sound effects that are difficult to reproduce.)
  
 The differences I'm talking about now are _NOT_ subtle differences in sound quality. What I'm talking about are the sort of things where you go: "Gee, I don't remembering hearing a boinging noise there before" or "I never noticed ringing on that note before", then you go back and listen to the same time code on the original and the mysterious noise isn't there. I personally suspect that this was due mainly to situations where the version of the masking model used by the encoder was slightly oversimplified, but I suppose it could just as well have been an actual mathematical error in the encoder. However, my point is that I'm not talking about "subtle differences in sound quality", but distinct errors, which could have been either outright errors produced by the encoder, or alterations in timing or dynamics which allowed something not audible on the original to be heard on the MP3 version... sort of like looking at a spot of dirt that is present on a print but not on the original. Because of the few, but annoying, distinct errors I heard, I didn't bother to go back and listen for more subtle differences. (It's pointless to say that the errors were in fact visible on a "digital oscilloscope view" of the file - which they were, because MP3 files, even when they do sound audibly perfect, tend to show very different looking waveforms on a scope display.)
  
 Now, in all fairness, I will concede that I wasn't auditioning the format in terms of a balance between quality, storage space, and convenience. Nor was I considering it in general terms of "audible quality". I was strictly trying to determine whether it was audibly absolutely perfect - or not. My background is computers, so I tend to look at data storage in absolutes - the copy is either perfect or it isn't. If I store a billion pages of text and pictures in an archive, with a million pictures, and a single letter, on a single page, or a single pixel, on a single picture, comes back wrong, then that archive is _NOT_ perfect - and, as a "lossless storage format", it fails. (And I can store my entire music library in a ZIP or RAR archive, and every single bit will in fact come back correctly.) Therefore, as soon as I noticed one "clinker" that I could confirm, I discounted the entire format as not being reliably perfect, which answered my question.
  
 Therefore, here are the factors I consider....
  
 1) On at least a few samples I have been able to identify specific glaring and obvious errors. I will concede that these may in fact be rare, may only occur with certain source content, and may even be due to some flaw in the encoder rather than a flaw in the theory behind it. The fact remains that I cannot rely on the encoded version to be perfect.
  
 2) Taking 1) into account, the provenance of files produced with any form of lossy compression is too uncertain. I can easily confirm that a RIP of a CD is a bit-perfect copy of the original. Likewise, I can easily confirm that a given copy of the file is absolutely identical to that original. Both of those tests are fast, simple, and 100% unambiguous. However, while I can confirm that a given copy of an MP3 file is identical to an original copy of that same MP3 file, there is no simple way to confirm that a given specific MP3 file sounds audibly identical to the original lossless source file. (Assuming that the encoder is capable of delivering an audibly perfect encoded version, I would have to either do a comprehensive ABX test on _EACH SONG_ after I encode it, or "trust" the encoder and the person operating it to have "gotten it right this time".)
  
 3) Taking both of those a step further yet, if I were to purchase and download a lossy file (let's say an MP3), then, since I have no way of comparing it to the original myself, I'm both trusting that the encoder was capable of producing an audibly perfect result, and that the person who created the file using that encoder chose the proper settings to do so. 
  
 4) And, to be fair to your statement, knowing that the encoding process discards a significant portion of the audio information, based on the interpretation of a very complex psychoacoustic model by the author of each specific encoder, and that the result looks visibly different from the original on as oscilloscope trace or spectrum analysis, I will admit a strong bias towards not _EXPECTING_ that much alteration of the signal to be totally inaudible. (However, I have seen enough "totally effective" optical illusions, and other demonstrations of how easily our senses can sometimes be fooled, to admit to the possibility that it may in fact work that well.)
  
 5) Finally, to bring all this back to a value judgement, I personally don't consider the storage space and bandwidth saved by lossy compression to be significant at today's prices for storage space and bandwidth. Therefore, at least to me, it adds far too much extra effort and uncertainty, with far too little benefit in return, for me to even consider it. I'd rather spend a few extra cents to store a copy of the actual original file than save a few cents on a copy that might be as good (assuming you're right - and assuming that the encoding was done correctly). To me, the value in making the file a little bit smaller comes nowhere close to justifying the cost and risk involved in doing so.   
  
 That is my position - just FYI - and I don't feel a need to justify it beyond that.
  
 I'm not at all suggesting that anyone who finds that lossy compression satisfies their needs shouldn't use it.
 (In fact, when I'm using a portable player with limited storage capacity, I use 320k VBR MP3 myself.)
  
 Quote:


sonitus mirus said:


> It certainly is not a fact that anyone knows what you can or cannot hear, but I really don't believe this at all.  To me, these are simply empty claims that you believe to be true.  I'd be interested in learning a bit more about the tests that you conducted, if only from your best memories of them.  What were these "good" encoders?  How long ago were these tests conducted.  What type of music were your testing?  Did you use any type of ABX tool?  I get that you don't have any interest at all in lossy formats, but I don't believe it is fair to justify your position with unfounded evidence promoted merely by speculation and gut feelings.


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## RRod

> The differences I'm talking about now are _NOT_ subtle differences in sound quality. What I'm talking about are the sort of things where you go: "Gee, I don't remembering hearing a boinging noise there before" or "I never noticed ringing on that note before", then you go back and listen to the same time code on the original and the mysterious noise isn't there. I personally suspect that this was due mainly to situations where the version of the masking model used by the encoder was slightly oversimplified, but I suppose it could just as well have been an actual mathematical error in the encoder. However, my point is that I'm not talking about "subtle differences in sound quality", but distinct errors, which could have been either outright errors produced by the encoder, or alterations in timing or dynamics which allowed something not audible on the original to be heard on the MP3 version... sort of like looking at a spot of dirt that is present on a print but not on the original. Because of the few, but annoying, distinct errors I heard, I didn't bother to go back and listen for more subtle differences. (It's pointless to say that the errors were in fact visible on a "digital oscilloscope view" of the file - which they were, because MP3 files, even when they do sound audibly perfect, tend to show very different looking waveforms on a scope display.)


 
  
 The differences from 320kbps should be subtle. If they aren't, then something is wrong somewhere. Can you possibly give a source for DSoTM that you like, and a particular part of a track where you heard such differences?


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## KeithEmo

rrod said:


> The differences from 320kbps should be subtle. If they aren't, then something is wrong somewhere. Can you possibly give a source for DSoTM that you like, and a particular part of a track where you heard such differences?


 
  
 I agree with you there... there should not be obvious errors unless something is wrong.
  
 Considering that this was six or seven years ago, and I have at least a half dozen different CD versions of DSOTM, I can't say for sure which version that was, or where the problems cropped up. (Since I wasn't writing a paper on the subject I really didn't keep track).


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## sonitus mirus

keithemo said:


> I agree with you there... there should not be obvious errors unless something is wrong.


 
  
 It was implied that your testing methodology was faulty in order to achieve your claimed results, not that there is something wrong with the MP3 format that creates obvious, audible "boinging".  It still seems unbelievable to me, and it is certainly unsubstantiated, but many will latch on to similar comments and take them to be evidence supporting the idea that AAC and MP3 is an icky format reserved to the lowest common denominator of music lovers.


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## KeithEmo

sonitus mirus said:


> It was implied that your testing methodology was faulty in order to achieve your claimed results, not that there is something wrong with the MP3 format that creates obvious, audible "boinging".  It still seems unbelievable to me, and it is certainly unsubstantiated, but many will latch on to similar comments and take them to be evidence supporting the idea that AAC and MP3 is an icky format reserved to the lowest common denominator of music lovers.


 
  
 Well, then, just so nobody gives MP3 a bad rap .......
  
 What we're talking about here is a few obvious errors - on a single specific source file (that I specifically know of). There are a few things that everyone here needs to remember. First, the whole premise of MP3 encoding is that, when signals of certain spectral content and amplitude are present, they will mask specific other signals from being heard; and, therefore, we can omit or delete the masked sounds and their absence will not be audible; and doing so may save us a lot of space.
  
 The basic models on which this premise is based are widely known, and widely accepted to be valid. However, each individual MP3 encoder is allowed to interpret them in its own way - within certain limitations - so different encoders may produce slightly different results - even with the same source files and when the same settings are used. (The decoding process is standardized, so the same encoded file _should_ always play the same on different decoders.) However, since the encoder entails complex processing being performed by a computer program, it is quite possible that, even if the models themselves are perfectly valid, a given encoder may interpret them incorrectly, or may simply contain mathematical or other programming errors. Therefore, even if you are absolutely certain that the theory is valid, it is still quite possible that a given encoder may sometimes produce incorrect results.
  
 Since, unlike lossless compression methods, the whole goal of MP3 encoding is to discard large blocks of content which are presumed to be inaudible, there's no point in trying to analyze an encoded file for "accuracy" - we already know that it will be very different from the original.
 Also, since two different encoders may produce very different results, in terms of data content, from the same source file and settings, we can't compare the output files produced by two different encoders and expect them to match. Therefore, the only way to "test" an MP3 file is to compare it _audibly_ to the original, and see if we hear differences or not.
  
 (My personal choice is to err on the side of caution. For me there are simply too many unknowns. I don't like unknowns and variables, and the MP3 encoding process has far too many of them to make me comfortable. Perhaps I found the single source file, and the single spot in that file, where that specific encoder got it wrong... or perhaps not. And perhaps I really am imagining the other less obvious differences I think I've heard in other files. However, since I place little value on the size reduction achieved by MP3 files, and a lot of value on absolute certainty, I simply don't consider it worth pursuing the issue one way or the other. However, for someone who _DOES_ value the size savings, I would certainly encourage them to investigate for themselves. _DO_ your own tests, preferably double blind or some other form of ABX test, and decide for yourself. )


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## reginalb

keithemo said:


> Well, then, just so nobody gives MP3 a bad rap .......
> 
> What we're talking about here is a few obvious errors - on a single specific source file (that I specifically know of). There are a few things that everyone here needs to remember. First, the whole premise of MP3 encoding is that, when signals of certain spectral content and amplitude are present, they will mask specific other signals from being heard; and, therefore, we can omit or delete the masked sounds and their absence will not be audible; and doing so may save us a lot of space.
> 
> ...


 
  
 For me, the biggest issue is the pushing of formats that are declared to be better, even though there is plenty of evidence that they are not, and charging a lot more for them. I mean, for the reasons you outline, if Tidal lossless were the same price as Play Music All Access, I'd jump all over it. But taking advantage of information asymmetries to con people out of money is a lousy thing to do, and a lot of the sites service up "Hi-Rezzzzz" are doing just that.


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## sonitus mirus

keithemo said:


> Well, then, just so nobody gives MP3 a bad rap .......


 
  
 I do understand your perspective on the matter, and I admit to having an almost unhealthy propensity for supporting the underdog, which is how I see MP3/AAC in the audio world.  Your patience with my thread-derailing rants is greatly appreciated.


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## XenHeadFi

keithemo said:


> I also find it interesting that Apple, who claims to be "on the musicians' side", and to champion good quality music, launched the iTunes store only offering 128k AAC DRM-protected versions of what they sold (apparently they'd decided on behalf of their customers that it was "good enough"). Then, when it became obvious that it _wasn't_ good enough for many of their customers, they raised that to 256k AAC without the DRM - for a higher price, and an "upgrade fee" for people who had already purchased a given album (apparently, at that point, AAC 256 was "good enough"). Interestingly, in a paper I read a while ago, where Apple described "how musicians should master music for release on the iTunes store", Apple recommended using 24/96k lossless. (Clearly they're making sure they'll be prepared when there's enough of a market to buy everything again at 96k....)


 
 This is a bit of revisionist history. Apple had to negotiate with the RIAA and the labels to get the ability to sell those tracks, competing directly with the labels themselves who were just starting streaming and "download" services (96k odd audio format WMA? or even RealPlayerAudio? if I remember correctly). Settling on 128k AAC with FairPlay was what Apple had to do to get the labels to sign off. Remember, iPods were NOT dominant at that time. Apple was still a computer company and not a lifestyle company. Even at that time FairPlay was considered the best DRM compromise compared to the much more draconian DRM the labels were using for their own services. A few marginalized sellers had MP3s. MP3s at the time were coming from Napster/LimeWire/etc (ie. pirated wares. ARRRR!). BitTorrent wasn't even popular, yet.


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## RRod

keithemo said:


> Well, then, just so nobody gives MP3 a bad rap .......


 
  
 Except on every other forum of this site…
  
 And I'm still not sure what you think, Keith. Before it was "there are obvious errors." And then you agreed with "difference should be subtle." All the talk about encoder differences was maybe a big deal back in 1995, but today I would hope we would at least know if LAME completely screws up encoding DSotM compared to Fraunhofer.


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## reginalb

rrod said:


> Except on every other forum of this site…
> 
> And I'm still not sure what you think, Keith. Before it was "there are obvious errors." And then you agreed with "difference should be subtle." All the talk about encoder differences was maybe a big deal back in 1995, but today I would hope we would at least know if LAME completely screws up encoding DSotM compared to Fraunhofer.


 
  
 I am going to try to sum up what I've taken of it, Keith can feel free to correct me: AAC and MP3 are probably generally pretty transparent. But we know that it's lossy, it pulls audible stuff out of the music, based on some principles wherein the encoder has decided that people probably can't actually hear that sound (that it's masked). In some cases, there might be differences that are audible, and he has the resources to have and play his library in a lossless format, so why not? It's a fair point, I have found that the benefit of Play Music All Access far outweighs some potential theoretical cost (which I see as approaching zero). Different strokes.


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## RRod

reginalb said:


> I am going to try to sum up what I've taken of it, Keith can feel free to correct me: AAC and MP3 are probably generally pretty transparent. But we know that it's lossy, it pulls audible stuff out of the music, based on some principles wherein the encoder has decided that people probably can't actually hear that sound (that it's masked). In some cases, there might be differences that are audible, and he has the resources to have and play his library in a lossless format, so why not? It's a fair point, I have found that the benefit of Play Music All Access far outweighs some potential theoretical cost (which I see as approaching zero). Different strokes.


 
  
 I was more expressing frustration about a lack of example. I mean if someone can't actually recall the part of a track that made them skeptical of lossy encoding, why even bring it up? It's a consternating argumentation style:
 "I heard this thing one time."
 "What was it?"
 "Don't remember, but generally I'm just skeptical that stuff might happen."
 "Why?"
 "I heard this thing one time."
 "What was it?"
 … (turn into Philip Glass opera).


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## limpidglitch

rrod said:


> I was more expressing frustration about a lack of example. I mean if someone can't actually recall the part of a track that made them skeptical of lossy encoding, why even bring it up? It's a consternating argumentation style:
> "I heard this thing one time."
> "What was it?"
> "Don't remember, but generally I'm just skeptical that stuff might happen."
> ...


 
  
 Selective Dissociative Amnesia: When past psychological trauma leads to episodic memory loss, but you can still remember certain aspects when advantageous to your argument in a petty dispute.


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## RRod

limpidglitch said:


> Selective Dissociative Amnesia: When past psychological trauma leads to episodic memory loss, but you can still remember certain aspects when advantageous to your argument in a petty dispute.


 
  
 It's all fine if someone can't recall specifics from way back, but when those outcome both contradicts the current state of things and other assertions made by the same poster, things just get frustrating.
  
 FWIW, and risking going a bit off topic, I made 3 clips from a track that has some castanets, which I've seen several times stated as something mp3 just can't get quite right; enjoy:
 https://drive.google.com/file/d/0BwmVtb5IwniEV0tkVFM0TUg5M2s/view?usp=sharing


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## sonitus mirus

rrod said:


> It's all fine if someone can't recall specifics from way back, but when those outcome both contradicts the current state of things and other assertions made by the same poster, things just get frustrating.
> 
> FWIW, and risking going a bit off topic, I made 3 clips from a track that has some castanets, which I've seen several times stated as something mp3 just can't get quite right; enjoy:
> https://drive.google.com/file/d/0BwmVtb5IwniEV0tkVFM0TUg5M2s/view?usp=sharing


 
 Does this mean that I was incorrect on a 50/50 guess 10 times in a row?  Isn't that in itself quite a feat?
  
 foo_abx 2.0.1 report
 foobar2000 v1.3.7
 2015-10-24 14:51:09
 File A: a.wav
 SHA1: b207ef9ed6a3a2fb3dbaa6de746b8ca5597ed4bc
 Gain adjustment: -0.45 dB
 File B: b.wav
 SHA1: e294c8ff718ca1dd9a65cf2d3f56dfdec7a46f86
 Gain adjustment: -0.04 dB
 Output:
 WASAPI (event) : Speakers (Realtek High Definition Audio), 24-bit
 Crossfading: NO
 14:51:09 : Test started.
 14:51:39 : 00/01
 14:51:58 : 00/02
 14:52:07 : 00/03
 14:52:49 : 00/04
 14:52:57 : 00/05
 14:53:18 : 00/06
 14:53:29 : 00/07
 14:53:41 : 00/08
 14:53:59 : 00/09
 14:54:13 : 00/10
 14:54:13 : Test finished.
  ---------- 
 Total: 0/10
 Probability that you were guessing: 100.0%
  -- signature -- 
 a33bfcc076ed10c109ad9e049b151eb79a8dd53f


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## RRod

sonitus mirus said:


> Does this mean that I was incorrect on a 50/50 guess 10 times in a row?  Isn't that in itself quite a feat?


 
  
 One-sided p-value for the win!
  
 I did make one of the samples ABXable versus the other two (easy to verify if one does nasty false Hobbits things like making a spectrogram or seeing which file compresses the least as a FLAC).


----------



## sonitus mirus

rrod said:


> One-sided p-value for the win!
> 
> I did make one of the samples ABXable versus the other two (easy to verify if one does nasty false Hobbits things like making a spectrogram or seeing which file compresses the least as a FLAC).


 
  
 I was using these wonderful audiophile speakers, so if someone else is unable to miss everything, they probably need to spend even less.
  
 http://www.amazon.com/ARCTIC-S111-Silver-Computer-Multimedia/dp/B003XE3X8Q


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## KeithEmo

You've got it exactly.
  
 Back when 128k MP3 was what everybody was using, most people, including myself, agreed that it was far from audibly perfect. In fact, there were even "meta-encoders" that would let you encode an entire album using three different encoders, then decide which version of each track to keep. Then, as MP3 technology improved over the years, every once in a while I would "spot check" the latest encoder to see if it had finally reached a point where it was audibly perfect to me. At this point I already had my music collection as either the original CDs or in lossless files, so it was more a matter of curiosity than of serious consideration. Whenever a new encoder came out that claimed to finally be "audibly perfect", I'd try it out, usually with a few of my "trickier to encode" files, and see if I could "catch it out". (I don't necessarily consider this an absolutely fair trial because I would usually use the vendor's recommended "good quality" setting, but I certainly didn't expend the effort to try several different options, and I didn't note what problems occurred other than that, on at least one of the files I tried, there was at least one audible flaw.)
  
 I will also admit to a strong "aesthetic" bias against deliberately causing or adding significant flaws to my music. It's sort of like if the person building my house were to ask whether I'd like all the doorways straight, or whether I'd prefer to have them off kilter - "but I promise you wont notice it, nobody ever does" - for a few cents less. Given that choice, I would choose to have them perfect - unless there was a major difference in price, or some other difference that gave the cockeyed construction some other advantage.
  
 Even if I were to do an elaborate double-blind test and find out that the latest 320k MP3 encoder could produce files that I couldn't tell from perfect with my current headphones, I'd still always wonder if the difference (which we all agree is there) might be audible with the next set I buy. And, even if I found out that fifty of my favorite songs were audibly perfect, if I heard an odd noise on song #51, I would always wonder if it was really there in the master, or was an artifact. Since lossy encoding relies on "audible masking", and I can't simply check that the file is bit-perfect, there's no test I can run quickly and easily to confirm that this isn't the fact; therefore, I'm stuck either "trusting" that each encoded file is correct, performing an A/B test for each one, or wondering. Since there is no major incentive for me to use lossy compressed files, the costs simply outweigh the benefits.
  
 (If I was being sent on a five year mission to Mars, and only had room to take 5 gB of my favorite tunes, I just might settle for MP3s... but that simply isn't the case. 
	

	
	
		
		

		
			





 )
  
 However, the real point isn't whether_ I _can hear a difference (I'm sure I don't have either the best or worst hearing in the world); it's up to the person whose music we're talking about to find out whether they can hear a difference; and, if they can, if the savings in size is worth it to them; and, if they can't hear a difference, if they're willing to stick with lossless to make absolutely sure that they won't have to worry about it later. Personally, I'm from a background where, whether it's a song or an image, _if it's something you care about_, you always get and keep the best possible quality copy you can afford. (So, for all the albums I really care about, I have at least the original CD or a bit-perfect copy of it; although, for some albums I only listen to occasionally, I do in fact only have an MP3 copy.) 
  
 Quote:


reginalb said:


> I am going to try to sum up what I've taken of it, Keith can feel free to correct me: AAC and MP3 are probably generally pretty transparent. But we know that it's lossy, it pulls audible stuff out of the music, based on some principles wherein the encoder has decided that people probably can't actually hear that sound (that it's masked). In some cases, there might be differences that are audible, and he has the resources to have and play his library in a lossless format, so why not? It's a fair point, I have found that the benefit of Play Music All Access far outweighs some potential theoretical cost (which I see as approaching zero). Different strokes.


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## sonitus mirus

All good, but I maintain that Google Music is great for serious listening. It is perfect for portable use with inexpensive earbuds and more than capable to use with expensive, higher quality gear that most people probably could not afford.  The music is not missing anything audibly detectable or including audible artifacts that would detract from the enjoyment of the music.  The difference are not easy to identify and nearly impossible to hear in most cases without significant training and the means to be able to quickly switch between test data.  Those few individuals that have made statistical test data available that show differences to be identified have always claimed that the differences were subtle and difficult hear.
  
 There is a lot of misinformation spreading about and speculations with nothing to support it.  Usually the people making the claims have never done any type of proper ABX test, and oftentimes it is discovered that there is some flaw in the testing methodology that renders any supposed positive test to be useless.
  
 If someone has a different opinion, they can offer some proof to contradict my position, which admittedly only applies to me through exhaustive double blind testing.


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## KeithEmo

rrod said:


> Except on every other forum of this site…
> 
> And I'm still not sure what you think, Keith. Before it was "there are obvious errors." And then you agreed with "difference should be subtle." All the talk about encoder differences was maybe a big deal back in 1995, but today I would hope we would at least know if LAME completely screws up encoding DSotM compared to Fraunhofer.


 
  
 OK, here's exactly what I think.....
  
 1) MP3 is a lossy format - which means that information (_inside the range of audible frequencies_) is discarded; nobody disputes this. I find this "aesthetically offensive" because, as someone who fancies themselves to be an audiophile, my goal is not to reduce my music to the absolute minimum quality at which the flaws will be just barely inaudible. I dislike this on general principles; to me it's sort of like a piece of vintage humor that suggested that "the only parts of your car that you need to paint are the hood, the trunk, and the tops of the fenders - because those are the only parts you can see from the drivers' seat". (And, even if you could prove to me that the MP3 version of every file I currently own was audibly indistinguishable from the original on my current system, since I know that the encoder version is technically inferior, I would always wonder if the next file I encoded might be the one that wasn't perfect - which means that, in order to have absolute confidence, I would have to test every one. The problem there is that it's simple to test a lossless file to ensure that it's a perfect copy of the original; whereas it isn't simple at all to confirm that an MP3 file is "audibly identical".)
  
 2) The whole basis of MP3 (and other perceptual lossy encoding) is a model of how audible masking works (how the presence of certain sounds can prevent us from hearing other sounds, which allows us to delete those sounds that were masked from our hearing without producing an audible difference). This model is quite complex, and includes factors of relative amplitude, absolute sound pressure level, spectral content, and time - and, therefore, I'm not convinced that it covers all possible contingencies perfectly.
  
 3) Each MP3 encoder is an independent program, written by a human programmer, and intended to discard content that is inaudible - based on that particular programmer's interpretation of that model. This means that each encoder is subject to simplification, math and programming errors, and the judgment of its creator. So, even if the model were perfect, that doesn't even suggest that a given encoder implements it perfectly
  
 4) In general, lossy formats have been developed with the express intent of "saving storage space and/or bandwidth while preserving a level of quality acceptable to most customers". As far as I recall, it was always accepted that MP3 files at low bit rates (like 128k) sounded "obviously inferior", and, while higher bit rate MP3s and AAC files were considered to be "better", they were never considered to be "perfect" - and so no critical testing was done. (In other words, tests were done to determine "whether most people in the market heard a noticeable difference" - but such testing was uncritical - with a goal of determining whether the majority of people heard a noticeable difference rather than whether even the slightest difference existed.)
  
 5) In my particular experience, a "properly encoded MP3 file" can sound quite good. So I'm quite willing to believe that the occasional "funny noises" I've heard are due to either a simple error on the part of the particular encoder with a certain input, or to a very specific situation where either the masking theory "missed something" or the encoder implemented it inadequately. In other words, all I claim is that I noticed errors - bu I don't claim that the format itself is at fault. However, as long as I am hearing those occasional funny noises, I see no reason to perform more careful testing to detect the presence or absence of more subtle errors,
  
 As for the fact that most folks on other forums generally agree with my opinion - I don't find it as surprising. MP3 and other lossy formats were never promoted - at least until recently - as anything beyond a compromise between quality and file size. (Twenty years ago, when storage space and bandwidth were far more expensive than they are now, such a compromise may have made sense. However, today, while new technology in lossy compression may have reduced the amount of compromise using them entails, drops in the price of storage and bandwidth have eliminated the need for any sort of compromise for most people, so why bother?)


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## RRod

The problem with the middling "might as well position" is that it supports a culture of "more is always better," which is just wasteful. The simple fact is that I could secretly replace your entire collection with some high rate CBR or VBR AAC, Vorbis, Opus, or even MP3 and you wouldn't be the wiser save for the file extensions. Same goes for my collection, I'll admit readily.
  
 You keep bringing up various MP3 encoders, which is just a red herring. Let's pick LAME and just go with what it can do, an example of which I just posted for a sample that should, from what people would tell you, bring it to its knees. The end-goal of the codec designers is also totally irrelevant to how the final product sounds at high rates. I don't care if the design purpose around AAC was to make code that Jobs thought was beautiful when viewed in Garamond; how does it sound?
  
 It's of course all fine that you do what you want with your collection, but what we should be hashing out on here is where things *should* be going. Like sonitus above, the quality of modern streaming sites has utterly convinced me that things like higher PCM formats are just missing the point entirely. We should be making sure that the content on these sites is well-recorded and mastered, not that they stream 24/192.


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## KeithEmo

xenheadfi said:


> This is a bit of revisionist history. Apple had to negotiate with the RIAA and the labels to get the ability to sell those tracks, competing directly with the labels themselves who were just starting streaming and "download" services (96k odd audio format WMA? or even RealPlayerAudio? if I remember correctly). Settling on 128k AAC with FairPlay was what Apple had to do to get the labels to sign off. Remember, iPods were NOT dominant at that time. Apple was still a computer company and not a lifestyle company. Even at that time FairPlay was considered the best DRM compromise compared to the much more draconian DRM the labels were using for their own services. A few marginalized sellers had MP3s. MP3s at the time were coming from Napster/LimeWire/etc (ie. pirated wares. ARRRR!). BitTorrent wasn't even popular, yet.


 
  
 Nothing revisionist about it - but I agree that I didn't discuss _all_ the factors involved.
  
 The whole RIAA/licensing/DRM deal has always been a hodge-podge of compromise between inconveniencing legitimate customers and (presumably) preventing or reducing piracy.
  
 Remember that, while using draconian DRM on download services, the vast majority of CDs sold by record companies did not and still do not include any sort of DRM at all. (The few attempts to use DRM on CDs caused far more problems than they were worth, and weren't widely implemented.) Apparently the record companies were terribly worried about casual piracy of downloaded music files, but not nearly so worried about people ripping and sharing CD content. (One of the "virtues" of AAC is that it includes DRM that is at least effective enough to prevent casual copying. I also have little doubt that part of the reason Apple was able to strike a favorable deal with the RIAA was that they positioned their service more like a radio broadcast than a CD sale... and that part of that positioning was the assurance that, even if the files they were selling for download were to have the DRM removed, much like a recording off the radio, they "wouldn't be of a quality that was competitive with real CDs".)
  
 Since you mention that iPods were not dominant at the time, then I really need to point out that, by choosing a proprietary format, with DRM that not only prevented casual copying but prevented files from being played on any other device, Apple did an excellent job of forcing anyone who bought files from the iTunes store to buy an iPod to play them on 
	

	
	
		
		

		
		
	


	




. (It always amazed me that anyone would be willing to purchase music that could only ever be played on one brand of device.)


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## KeithEmo

I kind of agree with you there.... but I guess I have a little less sympathy for the "victims".
  
 We are continually bombarded with advertising claims that range from specious to downright silly - and I don't see this as being much different. I generally don't believe that last years laundry detergent would really remove dried in blood and grass stains from my white shirt, and I sort of doubt that this year's "ultra new and improved" version will do a much better job - although it's possible. Likewise, I doubt that I'll become an all-star quarterback if I buy those $249 sneakers; and I'm not even sure I'll get a date with this years top swimsuit model if I buy that sports car.
  
 This years hottest new sports car does go faster than last year's model - even though it probably won't make me into Mario Andretti; and those high-res files can store frequencies up to 42 kHz - whether you can hear them or not. And, just like some new cars really do drive better than last year's model, some of those high-res reissues do in fact sound better than the original CD.
  
 As I see it, the problem is that many audiophiles seem to be especially susceptible to pseudo-scientific arguments, and just plain bad science (presumably because they imagine they understand it better than they really do). We also seem to be more willing than some other types of people to forget that everyone has an agenda - and to figure out what that means regarding the products we see and read about. (A company who sells $500 audio cables may not be any more "inherently dishonest" than one who sells $5 cables, but they sure have a lot more incentive to convince you that there's a difference, right?)
  
 So, of course I expect the companies who hope to make a nice living out of selling high-res downloads to claim that they're better. After all, they're in business to sell downloads, and not to educate or to inform. (And, if I was in that business, while I wouldn't lie, I would certainly say something like: "They have a wider frequency response, and a better S/N ratio, and many people say they sound a lot better".)
  
 In other words, I agree with you that it's sort of rotten to inflate the virtues of any product - but, in this case, it doesn't make them any worse than 90% of the other companies who put commercials on TV or radio... and it's just a part of life. (I'd also like to think that nobody is going to starve their kids, or empty their retirement fund, to buy a few high-res downloads.... ).
  
  
 Quote:


reginalb said:


> For me, the biggest issue is the pushing of formats that are declared to be better, even though there is plenty of evidence that they are not, and charging a lot more for them. I mean, for the reasons you outline, if Tidal lossless were the same price as Play Music All Access, I'd jump all over it. But taking advantage of information asymmetries to con people out of money is a lousy thing to do, and a lot of the sites service up "Hi-Rezzzzz" are doing just that.


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## sonitus mirus

keithemo said:


>


 
  
 I like the car analogy.
  
 Let's say that the tire and track conditions represent human hearing capabilities, and there is a limit to how big the tires can be and the effectiveness of the tire compound.  Once a car reaches a specific power level with the right gearing, there is only so much performance that can be gained.  MP3/AAC already provide adequate power to shred the tires, and no matter how much more power is applied, the car simply cannot accelerate any faster out to the governor-limited speed of 155 mph.  Sure, perhaps a very light, professional driver may be able to eek out a little more performance than a typical driver, but this would be challenging and require special skills and near-perfect conditions in order to repeat this feat.  With the governor removed, the top speed may be a bit higher for a car with gobs of overpower, but what person could actually make use of a difference between 235 mph or 255 mph in real world driving conditions?
  
 Cars don't have a limit to the tire sizes and compounds being used, so this would never hold true.  But the human hearing limitations is often ignored when discussing the advantages of 24-bit music or sample rates that reach into the ultrasound realm.
  
 I suppose DSD would be a jet-propelled car that had no tire restrictions in the acceleration test, but the analogy can only go so far.


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## reginalb

keithemo said:


> As I see it, the problem is that many audiophiles seem to be especially susceptible to pseudo-scientific arguments, and just plain bad science (presumably because they imagine they understand it better than they really do). We also seem to be more willing than some other types of people to forget that everyone has an agenda - and to figure out what that means regarding the products we see and read about. (A company who sells $500 audio cables may not be any more "inherently dishonest" than one who sells $5 cables, but they sure have a lot more incentive to convince you that there's a difference, right?)


 
  
 I don't disagree with much of that post (if any). I guess the one thing that bugs me in the realm of audio contrasted to the other industries wherein I am interested in the gear, is that it seems that the enthusiasts in this one seem to know less and less as they do more research. Or more appropriately, they know so much that isn't so. I don't see this in other interests, for example, I'm a SCUBA diver. And as a gear-head, I have an unhealthy obsession with SCUBA gear. And while you get similar marketing, and similar publications that are funded by the companies doing the marketing, and some dive shops are pretty unscrupulous, you only have to spend a little time digging to find both Dive shops and enthusiasts online that will just maim and destroy the similar pseudo-scientific arguments made by the marketers, and give really good advice.
  
 I am also in to photography, and at the Fred Miranda forums, there are fandbois, sure, but just as many people that love to hate on their own gear of choice, and will shred the manufacturers for small mistakes that would be defended to no end on Head-Fi. I mean, I do a TON of research when I buy things, and when I first decided to get "good" headphones, I did tons and tons of reading online, and Hifiman players were nearly universally praised as making your music sound SO much better. I mean, there are certainly some online communities that will take audio companies to task, but it seems that in Head and Hi Fi, these are the exceptions, where in other industries, they are the mainstream. 
  
 I guess I don't understand why audiophiles seem more susceptible to claims of the pseudo-scientific variety, whilst also taking a downright hostile view to any attempt to remove bias from testing. Perhaps my thermometer is off, and this industry is similar to others, but I don't think so. Another way to look at it, and why I feel bad for some when it comes to head-fi: most people don't do tons of research themselves, but instead rely on recommendations of trusted compatriots with specialized knowledge in some field when they're making a purchase. I feel that with audio equipment, that trusted compatriot is fare more likely to just blow marketing smoke at them and give bad advice than in other industries, _even if _that person has done enormous amounts of research in to the industry.


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## KeithEmo

I agree with you - but only up to a point.
  
 I'll bet that, whatever your favorite drink, I could dilute it 5% with water, and you wouldn't taste the difference (and neither would I). But, if we do that today, we could probably get away with adding another 5% water tomorrow, and, pretty soon, it would be pretty watered down. The opposite of a "more is always better" culture is the opposite slippery slope of "we can always give up just a little more and nobody will notice". 
  
 As someone with an engineering background, I tend to think in terms of safety margins. If I wanted to measure the frequency response of an amplifier that was claimed to reach from 20 Hz to 20 kHz, I wouldn't measure it with a meter whose response was accurate from 20 hz to 20 kHz; I would use a meter that was accurate from 5 Hz to 100 kHz; that way I wouldn't have to worry if my measurements were approaching the limits of the accuracy of my equipment, or if the particular meter I was using was a few percent short of its specified performance. Therefore, if our goal is to store music, and we are quite certain that most people can hear everything between 20 Hz and 20 kHz, then I would want to ensure a reasonable safety margin by using equipment with a range of from 10 Hz to 50 kHz or so. (I can see squeezing that last bit of space or performance out of things as a reasonable compromise only if there is a significant amount of cost involved. Maybe that was the case when flash memory cost $1 per mB, and it was a squeak to fit one hour of music at 16/44k onto a CD disc; but the last hard drive I bought cost $129 for 4 tB - which works out to about 5 cents per album in lossless format, and a savings of about two cents per album by using MP3 instead... which is hardly a significant savings.)
  
 (I wouldn't pay extra for an audio amplifier that was flat to 5 mHz; but I would pay extra for one that was flat to 50 kHz instead of one that only promised to be flat to 20.0 kHz.)
  
 I also agree with you that, in many cases, I wouldn't notice the difference whether a song I was listening to was a 320k MP3 file or a FLAC. However, I'm quite certain that I do at least sometimes hear a distinct difference, especially when comparing versions side by side, and pretty sure that I hear a less obvious difference at other times - and I have heard distinct errors from time to time. And, again, I can always easily confirm that a lossless file remains "perfect", but verifying that an MP3 file is "audibly perfect" is a lot less certain - and takes a lot more effort. When I consider those negatives against the few cents worth of storage space or bandwidth that I would stand to gain, to me it's obvious that lossy files aren't worth the bother to consider.
  
 I disagree entirely about multiple encoders, and differences between them, being "a red herring". I'm simply throwing out the possibility that, if/when the LAME encoder does occasionally produce an error, it might simply be a programming error rather than a flaw in the basic model involved. (I don't recall ever seeing a computer program that didn't have at least one minor flaw; I see no reason to believe that the LAME encoder is the first.)  
 I definitely agree with you that poor quality recording and mastering (and the deliberate choice to add excessive compression and boost) are far greater issues than the delivery format used... However, since these are for the most part outside my control, I'm forced to settle for doing my best to avoid introducing further degradation during the reproduction process.
  
 I have no idea what other people might think "should bring the MP3 CODEC to its knees"; I have my own ideas on that score, but I haven't tried them, so I'm not going to describe them here.
 (If I get bored I may make up a few test files just to see; if I do, I'll report the results.)
  
  
 Quote:


rrod said:


> The problem with the middling "might as well position" is that it supports a culture of "more is always better," which is just wasteful. The simple fact is that I could secretly replace your entire collection with some high rate CBR or VBR AAC, Vorbis, Opus, or even MP3 and you wouldn't be the wiser save for the file extensions. Same goes for my collection, I'll admit readily.
> 
> You keep bringing up various MP3 encoders, which is just a red herring. Let's pick LAME and just go with that it can do, an example of which I just posted for a sample that should, from what people would tell you, bring it to its knees. The end-goal of the codec designers is also totally irrelevant to how the final product sounds at high rates. I don't care if the design purpose around AAC was to make code that Jobs thought was beautiful when viewed in Garamond; how does it sound?
> 
> It's of course all fine that you do what you want with your collection, but what we should be hashing out on here is where things *should* be going. Like sonitus above, the quality of modern streaming sites has utterly convinced me that things like higher PCM formats are just missing the point entirely. We should be making sure that the content on these sites is well-recorded and mastered, not that they stream 24/192.


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## RRod

keithemo said:


> I'll bet that, whatever your favorite drink, I could dilute it 5% with water, and you wouldn't taste the difference (and neither would I). But, if we do that today, we could probably get away with adding another 5% water tomorrow, and, pretty soon, it would be pretty watered down. The opposite of a "more is always better" culture is the opposite slippery slope of "we can always give up just a little more and nobody will notice".


 
  
 Except there will come a point where it's obvious you've been watering down my scotch, and I would be able tell so in a blind tasting. It's not like codec developers are just willy-nilly throwing away bits; there are listening tests involved.
  
  


keithemo said:


> As someone with an engineering background, I tend to think in terms of safety margins. If I wanted to measure the frequency response of an amplifier that was claimed to reach from 20 Hz to 20 kHz, I wouldn't measure it with a meter whose response was accurate from 20 hz to 20 kHz; I would use a meter that was accurate from 5 Hz to 100 kHz; that way I wouldn't have to worry if my measurements were approaching the limits of the accuracy of my equipment, or if the particular meter I was using was a few percent short of its specified performance. Therefore, if our goal is to store music, and we are quite certain that most people can hear everything between 20 Hz and 20 kHz, then I would want to ensure a reasonable safety margin by using equipment with a range of from 10 Hz to 50 kHz or so. (I can see squeezing that last bit of space or performance out of things as a reasonable compromise only if there is a significant amount of cost involved. Maybe that was the case when flash memory cost $1 per mB, and it was a squeak to fit one hour of music at 16/44k onto a CD disc; but the last hard drive I bought cost $129 for 4 tB - which works out to about 5 cents per album in lossless format, and a savings of about two cents per album by using MP3 instead... which is hardly a significant savings.)


 
  
 But it's a perfectly fine paradigm to have the safety margins be *for* the engineers. They record at 24-bits to have an easier time setting volumes and to have more leeway for mixing; they record at 192kHz so that they can use analog filters with gentler roll-off and svelte digital filters to go back down to 44.1. The content providers have the final raw data, and deliver it to us in the latest, greatest lossy codec. If the codec gets improved, they roll-out the improvements to us invisibly. Greed of course prevents a perfect system from evolving, but there's no logical reason a paradigm of lossy delivery can't be a high-fidelity paradigm.
  


keithemo said:


>





> I disagree entirely about multiple encoders, and differences between them, being "a red herring". I'm simply throwing out the possibility that, if/when the LAME encoder does occasionally produce an error, it might simply be a programming error rather than a flaw in the basic model involved. (I don't recall ever seeing a computer program that didn't have at least one minor flaw; I see no reason to believe that the LAME encoder is the first.)


 
  
 But you're back on a hypothetical. I have yet to hear an MP3 using the latest LAME where there was some weird blip that wasn't your typical MP3 encoding artifact (pre-echo, etc), and I only hear those at about the 128kbps level. I have little reason to believe that there will be some music track where the data just happens to pop up some coding error where LAME puts a "wha-wha-whaaaaaaaa" into the music.


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## KeithEmo

Quote:


rrod said:


> Except there will come a point where it's obvious you've been watering down my scotch, and I would be able tell so in a blind tasting. It's not like codec developers are just willy-nilly throwing away bits; there are listening tests involved.
> 
> 
> 
> ...


 
  
 But isn't it easier just to keep the bottle sealed than to wonder?
 (If I go to the store to buy a bottle of Scotch, I'd rather have the sealed one that hasn't been watered down at all - wouldn't you?)
  
 I have several huge problems with your proposition.
  
 First, I guess you're assuming a streaming paradigm, while I am not. I remember what happened when Apple "improved their CODEC" with the iTunes store; they told everyone that they would have to pay an upgrade fee to buy new copies of everything they'd already bought if they wanted the benefits of that new improved CODEC. It wasn't at all invisible - or free. (I'm guessing they'll eventually move up to lossless - at which point everyone will get to pay yet again. I can see why the vendor would like that model, but I don't like it much. I'm also guessing that a lot of people who bought those original 128k AAC files thinking they were "the same as a CD" weren't especially thrilled either.)
  
 Second, why would we need to wait for the CODEC to evolve, and why would an improvement need to be rolled out. We already have a perfect CODEC - lossless. And we already know that lossless gives us perfect quality (at least as compared to the original). So why would we want to go through an evolving series of compromises that, if we're lucky, in the end will bring us back to where we are now? Sure, a lossy paradigm can be high fidelity, but a lossless one is _HIGHER_ fidelity (and, at the very best, you can hope that the difference isn't noticeable), so why not just start there?
  
 I'm sorry, I'd just like the good one to begin with please.....
  
 And, again, back to your metaphor.....
 Why should I even wonder whether I notice the water or not, or how much water it takes before I notice it?
 I'd like my bottle with the seal still unbroken, thanks.....
 After all, I'm buying Scotch - not water.


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## sonitus mirus

I've only ever purchased lossy music if it was the only option in a pinch, or when looking to use the files in one of my ABX tests.  I'm ok with renting lossy stuff, and to me I don't hear a difference, but I want the real deal if I am going to purchase to own.  The success of iTunes and other lossy music stores is the convenience and the option to purchase single songs rather than an entire album.
  
 That's not my cup of tea.  I still feel a bit guilty when I shuffle through my song list rather than listening to an entire album.  All the "cool" people in my past insisted that an album should be listened to in its entirety.  I don't adhere to that philosophy so much anymore, but I do find myself always adding a complete album to my Google streaming library rather than simply adding the particular song that captured my attention.  Old habits.


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## reginalb

keithemo said:


> Second, why would we need to wait for the CODEC to evolve, and why would an improvement need to be rolled out. We already have a perfect CODEC - lossless.


 
  
 A perfect codec would be the smallest size file size possible, while still being completely transparent to the listener in question. Therefore, far from perfect, all of the lossless codecs are worse than Apple's AAC (at 256k), Lame MP3 (at 320k), and probably Vorbis Ogg or Opus Ogg.


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## RRod

keithemo said:


> But isn't it easier just to keep the bottle sealed than to wonder?
> (If I go to the store to buy a bottle of Scotch, I'd rather have the sealed one that hasn't been watered down at all - wouldn't you?)
> 
> I have several huge problems with your proposition.
> ...


 
  
 These are basically the same arguments for hi-res versus CD. Lossless is only a "higher" fidelity if you aren't talking audibility, just like 24/192 "beats" Redbook if you use a spectrogram.
  
 As far as Apple, I said above that greed, and more generally corporate nefariousness, can ruin the whole paradigm. But I bet plenty of people can't even ABX 128AAC versus Redbook, yet the "higher is better" mentality means that stuff like doubling the bitrate will always make some amount of money for the company.
  


reginalb said:


> A perfect codec would be the smallest size file size possible, while still being completely transparent to the listener in question. Therefore, far from perfect, all of the lossless codecs are worse than Apple's AAC (at 256k), Lame MP3 (at 320k), and probably Vorbis Ogg or Opus Ogg.


 
  
 Well put. We have to remember that a constant 16/44.1 across a whole file is a hammer-and-nail solution. Many tracks have large chunks that aren't dynamic enough to require 16 bits, and most people over a certain age can't hear up to 20kHz. This is why lossy codecs can work: they ask the question of how many bits and samples you need for a given section of song.


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## limpidglitch

keithemo said:


> But isn't it easier just to keep the bottle sealed than to wonder?
> (If I go to the store to buy a bottle of Scotch, I'd rather have the sealed one that hasn't been watered down at all - wouldn't you?)


 
  
 Hate to break it to you, but your whisky already is watered down.


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## old tech

rrod said:


> These are basically the same arguments for hi-res versus CD. Lossless is only a "higher" fidelity if you aren't talking audibility, just like 24/192 "beats" Redbook if you use a spectrogram.
> 
> As far as Apple, I said above that greed, and more generally corporate nefariousness, can ruin the whole paradigm. But I bet plenty of people can't even ABX 128AAC versus Redbook, yet the "higher is better" mentality means that stuff like doubling the bitrate will always make some amount of money for the company.
> 
> ...


 
  
 That is the key point I think.  Going back to the original purpose of this thread, it we are having these debates whether there are extremely subtle, or any difference at all, to be heard between 320kbts and 16/44, then any argument that we can hear a difference between 16/44 and 24/96 must be psudeoscience territory.


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## KeithEmo

limpidglitch said:


> Hate to break it to you, but your whisky already is watered down.


 
  
 Yes, but hopefully only in the sense that water is added as part of the distilling or mixing process. I consider that to be much the same as the obvious fact that music is generally altered, often significantly, in the mixing and mastering process. Anything that happens before the bottle is sealed at the bottling plant is "part of the creators' artistic intent". But, once the bottle is sealed, and "blessed" with the company and product names, any water that was added after that would be _tampering_. (I'm also guessing that unsealing a bottle, adding water, and resealing it, would be considered to be product fraud.)
  
 Perhaps a better analogy would be art work. With a painting, you can have either the original or a copy; and there is a very "bright line" between them. A copy can be a good copy, or even a great copy, and there are counterfeits so good that even an expert may not notice the difference, but there is still an original - and copies. If you were to buy a Rembrandt for $30 million, you wouldn't be very happy if you were to find out it was a copy, and you would be even less happy if, when you told the gallery where you bought it, their response was "it doesn't matter because it's such a good copy nobody can see the difference". With digital music we actually have an opportunity to buy "an original" (if the data is actually bit-for-bit the same, then it isn't just a copy, it is a clone - in other words, it is a new and identical version of the original, which is different than "something that is just arbitrarily close - a vinyl album is equivalent to an 'artist approved print" whereas a digital copy really is another instance of the 'original' content").
  
 (That analogy is weak because, with the art work, the fact that the painter physically applied that particular paint himself is what makes the painting original; whereas, with music, as long as every bit, and so the numbers they represent, is the same as the original, then it really is "another original" because the bits themselves are transitory, but the original numbers represented by the bits are not.)


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## RRod

Or we could just skip analogies and focus on audibility…


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## KeithEmo

old tech said:


> That is the key point I think.  Going back to the original purpose of this thread, it we are having these debates whether there are extremely subtle, or any difference at all, to be heard between 320kbts and 16/44, then any argument that we can hear a difference between 16/44 and 24/96 must be psudeoscience territory.


 
  
 I don't agree. Let's just say that whether there is an audible difference is "disputed science" - because even "well established science" has frequently been updated when it has been found to be wrong. 
  
 However, even beyond that, the difference between lossy files and lossless files isn't the same as the difference between the same content at different bit rates. Even though many people seem to like to treat things as if there was a "quality continuum" ranging from lossy files to lossless files to high-res files, that is technically a simplification, and, more importantly, not a very accurate one. The difference between files recorded at different sample rates is that the lower sample rates have a more limited bandwidth, and store time-specific events with less temporal accuracy, but both still retain all of the information present in the original event that is inside the generally agreed-upon range of audibility (20 Hz to 20 kHz). In direct contrast, lossy compression discards information that is _INSIDE_ that agreed-upon range of audible frequencies. (In other words, everybody agrees that the information discarded by lossy compression like MP3 is _CLEARLY AUDIBLE_ content. The whole idea is that, even though the discarded content is itself audible on its own, you can't hear it because you are being "blocked" or "masked" from hearing it by other content, and so you won't be able to tell when it's removed.)
  
 (In fact, Meridian's new format is lossy, but claims to retain "some of the important audible differences between standard-res and high-res files". In other words, they claim that their new lossy format is actually of better sound quality than a standard-res lossless CD version. I haven't heard it, so I have no opinion there - and I'm not interested in arguing about their claim.)
  
  
  
  
  
 The claim of the original title of this thread is that there will _NEVER_ be _ANY_ audible difference with high-res files (at least not any difference due strictly to their higher sample rate).


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## RRod

keithemo said:


> I don't agree. Let's just say that whether there is an audible difference is "disputed science" - because even "well established science" has frequently been updated when it has been found to be wrong.
> 
> However, even beyond that, the difference between lossy files and lossless files isn't the same as the difference between the same content at different bit rates. Even though many people seem to like to treat things as if there was a "quality continuum" ranging from lossy files to lossless files to high-res files, that is technically a simplification, and, more importantly, not a very accurate one. The difference between files recorded at different sample rates is that the lower sample rates have a more limited bandwidth, and store time-specific events with less temporal accuracy, but both still retain all of the information present in the original event that is inside the generally agreed-upon range of audibility (20 Hz to 20 kHz). In direct contrast, lossy compression discards information that is _INSIDE_ that agreed-upon range of audible frequencies. (In other words, everybody agrees that the information discarded by lossy compression like MP3 is _CLEARLY AUDIBLE_ content. The whole idea is that, even though the discarded content is itself audible on its own, you can't hear it because you are being "blocked" or "masked" from hearing it by other content, and so you won't be able to tell when it's removed.)


 
  
 I have many tracks, MANY, that I can push down to below 16 bits without an audible difference. Some brickwall stuff I can get down to 8 bits, so there ya go, ½ lossy compression right there. So in fact there are some analogs between lossy compression and moving between PCM specs. Lossy just tries to be gutsy and do its work within frequency bands and sample frames, rather than whole-hog across the file.
  
 The whole "science is sometimes wrong so it must be wrong here" thing always irks me. It's like people on this board would time-travel back to Newton and call him a dumbass for analyzing an apple falling without using relativity.


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## KeithEmo

sonitus mirus said:


> I like the car analogy.
> 
> Let's say that the tire and track conditions represent human hearing capabilities, and there is a limit to how big the tires can be and the effectiveness of the tire compound.  Once a car reaches a specific power level with the right gearing, there is only so much performance that can be gained.  MP3/AAC already provide adequate power to shred the tires, and no matter how much more power is applied, the car simply cannot accelerate any faster out to the governor-limited speed of 155 mph.  Sure, perhaps a very light, professional driver may be able to eek out a little more performance than a typical driver, but this would be challenging and require special skills and near-perfect conditions in order to repeat this feat.  With the governor removed, the top speed may be a bit higher for a car with gobs of overpower, but what person could actually make use of a difference between 235 mph or 255 mph in real world driving conditions?
> 
> ...


 
  
 I like your car analogy a lot.... 
  
 The "problem" I have with many discussions on the subject of "what's audible" is that they claim to be based on "absolute absolutes" - which are things that very rarely occur in nature. The vast majority of humans can't hear sounds above 20 kHz; agreed; but is that true for every single human on the planet? I doubt that either of us will survive to an age of 120; but a reasonably long list of documented humans have; and so you cannot say that it's impossible (even if that list was blank - it would be possible that you or I could be the first). When I went to school, everyone _knew_ that matter was comprised of certain indivisible pieces - protons, neutrons, and electrons - except that _now_ we know that isn't true after all.
  
 As I've said before, I do in fact have many high-res downloads which, at least to me, do sound clearly better than the original versions. However, I don't know for sure whether some, or in fact any, of them sound better _because_ they're high-res, or whether it's because of some mastering or other difference. And, since any type of conversion also introduces slight audible changes, doing what I consider to be an exhaustive test on the subject would be complicated and time consuming. (And, even if I were to find that the difference were not audible to me, that wouldn't prove that nobody else out there can hear it.) So, to me, that question hasn't been sufficiently proven either way. (Just because I don't have "perfect pitch", I can't claim that a more instrument tuner won't be useful to someone who does.)
  
 Lossy compression is different. The whole basis of lossy compression is a model of how certain sounds mask other sounds - in humans. (The basic premise is simple: If Sound A masks Sound B, then you can't hear Sound B; therefore, we can discard Sound B and save space, and you won't hear the difference.) The model itself is based on a relatively straightforward combination of frequency, amplitude, and time. Loud sounds at a given frequency tend to mask quieter sounds of similar frequencies; the effect works "better" if the quieter sound is much quieter, works better the closer together the two are in frequency, and has a certain time factor as well (a sound can mask another sound that occurs before or after it; and, again, how well that works depends on relative level and frequency). The fact that this works pretty well is proven by how good a 320k MP3 usually sounds; but is it absolute, and is it true under all conditions?
  
 To go back to the car analogy; not only do we have the possibility of different sized tires and different compounds, but we need to know how each works on hot days, and cold days, and wet days, and dry days, and on clean asphalt tracks, and on cement tracks, and on cement tracks with a little dirt and sand. But, getting back to audio, we know that "the masking model" seems to define how individual tones mask each other pretty well, but how precisely does it match masking of modulated tones, and masking of decorrelated noise by pure tones, and of pure tones by modulated tones? And how well does a sound in the right channel, and a different sound in the left channel, together mask a third sound in the center? I don't think the research covered all the combinations and permutations there in exhaustive detail. And, even if the research for all that has been done, how well does the simplified version of the model used by a typical MP3 encoder handle all those variations. To be totally honest, until recently most people considered the purpose of lossy encoding to be to achieve high levels of compression with minimal loss of quality - which makes me wonder whether anyone even seriously tried to determine if any encoder really did produce results that were audibly absolutely the same as the original - with all content - under all circumstances.
  
 Your car analogy also brings up another very good point. Since I'm not Mario Andretti, my Nissan Versa works very well for me, and so you could say that, for me, it's "just as good as a Formula 1 racing car"... and I would be foolish to buy the racing car. However, from a strictly factual point of view, you can't reasonably claim that the two are identical in performance. (And, if you're planning to listen to that music on a phone, using $20 ear buds, then the "practical limitation" isn't going to be the limitation of your hearing either...




  
 I don't at all disagree that, when space or bandwidth limitations are factors, MP3 or AAC do an excellent job and are a reasonable compromise. I'm just not convinced that either is "audibly 100.0% perfect. (In fact, I'm personally convinced that they are not.)


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## KeithEmo

sonitus mirus said:


> ....





> I still feel a bit guilty when I shuffle through my song list rather than listening to an entire album.  All the "cool" people in my past insisted that an album should be listened to in its entirety.  I don't adhere to that philosophy so much anymore, but I do find myself always adding a complete album to my Google streaming library rather than simply adding the particular song that captured my attention.  Old habits.


 
  
 I agree with you there. I still tend to think of albums as _ALBUMS_ rather than as individual songs.


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## KeithEmo

rrod said:


> I have many tracks, MANY, that I can push down to below 16 bits without an audible difference. Some brickwall stuff I can get down to 8 bits, so there ya go, ½ lossy compression right there. So in fact there are some analogs between lossy compression and moving between PCM specs. Lossy just tries to be gutsy and do its work within frequency bands and sample frames, rather than whole-hog across the file.
> 
> The whole "science is sometimes wrong so it must be wrong here" thing always irks me. It's like people on this board would time-travel back to Newton and call him a dumbass for analyzing an apple falling without using relativity.


 
  
 I agree entirely....
  
 I think that the perspective I have about types of compression (and a lot of other people do as well) comes from a historical perspective. When I started purchasing music, the "source" was always a CD, and so it was always my choice whether to "save it as is" or "create an MP3 version of it". Therefore, in my head, the CD is "the original" and the lossy version is "a quality-reduced copy".
  
 Part of my feelings on this issue come from my background in photography; where, even though a high-quality JPG file may be visually indistinguishable from an uncompressed version, compression artifacts often appear, and become very visually annoying, if you try to perform any sort of processing on the file. (So I might reasonably claim that, if I were to make a mix disc, or modify that file - perhaps to try and restore some of the dynamic range that was squeezed out during mixing - that the extra dynamic range or frequency response might prove useful.) However, the actual reality is that, even if you were able to provide me 100% proof that I couldn't ever, under any circumstances, hear the difference between 16/44 and 24/96, and I had no plans whatsoever of "playing with" the file, I would _STILL_ buy the higher quality file (again, there being no question that technically it is better). I guess you could say that my goal is to get as close to the original as possible, and deliberately reducing the quality, whether audible or not, is the antithesis of that goal.
  
 To go back to photography again.... No matter what camera I own, there will always be a better one, and the choice of where to stop buying cameras was a matter of budget. However, when I take a picture with a given camera, I still use the highest quality format which the camera I'm using is capable of and, when I save that picture, I save it at its highest quality - just in case it may matter later - and the same mentality holds for me with music. As long as the cost isn't ridiculous, I'd still rather have the best quality version available, and whether I can personally tell the difference "by ear", while important, is actually secondary to the basic fact that "better is simply better".
  
 I also agree with you that any logic claiming that "science _CAN_ be wrong - so it must be here" is silly - and bad logic. (However, the fact that science is sometimes wrong does suggest that we shouldn't blindly accept everything that science supposedly claims.)
  
 However, in this particular case, there is no question about the science. High-res files do store a wider range of frequencies than standard res files; and even more information has gone missing with lossy compression; therefore, according to actual science, and all of the established specs like noise, frequency response, and distortion, high-res files are better than Red Book files, and Red Book files are better than lossy files. The only point in question is whether the differences that we agree exist are _audible_ or not. 
  
 In my _PERSONAL_ perspective, this more or less equates to walking into a store to buy a vase, and being offered a "perfect" one at a certain price, but being offered, at a slightly lower price, "an identical vase that's been broken, but glued back together perfectly, so you absolutely can't see the cracks". Even if I can't see the difference, I would still prefer to have the unbroken vase - for "aesthetic reasons" if you prefer to think about it that way.
  
 And, again, in the case of music, the difference in price is, at least to me, negligible. (I probably wouldn't pay an extra $500 for a 24/192k version of my favorite album, but the $5 extra I'll pay in the current market doesn't seem excessive to me.)


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## reginalb

keithemo said:


> And, again, in the case of music, the difference in price is, at least to me, negligible. (I probably wouldn't pay an extra $500 for a 24/192k version of my favorite album, but the $5 extra I'll pay in the current market doesn't seem excessive to me.)


 
  
 Of course, the actual savings are in the aggregate. Those of us that use a subscription service at $96/year, and have access to everything we could ever care to listen to, save an enormous amount of money. Plus have access to more music (I now am not constrained by budget when my wish list is long, and I have to choose which to buy). So I both spend far less, but get astronomically more for that lower amount of money. Plus I am introduced to new music, through algorithms that are far more accurate than any friend could ever be at picking out what I will like.
  
 You are paying far more than you think you are, and getting far less for it, I think. Perhaps you have such a different taste in music than the 10's of millions of songs that you could have access to that you're better off just buying albums individually, and of course that's your prerogative. But each time you present the tradeoff, you are missing an enormous part of the calculus. 
  
 Since we keep using analogies, which I think are _highly _problematic by the way, I'll also use one. It's like saying that buying a more fuel efficient car only saves me a few cents when I drive to the grocery store, but ignoring the many thousands of other miles that I drive when I calculate the savings. Or that I _could be _driving, but never will because my more expensive vehicle prevents me from ever even dreaming to in the first place.


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## KeithEmo

I disagree with your semantics. If a given CODEC _always_ produces results that a given listener can _never_ distinguish from the original, when used with his choice of source material, and with his choice of equipment, then, in a "local" sense, that CODEC may be perfect _FOR THAT LISTENER_. (I have no problem whatsoever as long as you always describe it as "perfect for you" or "perfect for your application". However, if you want to use the general term "perfect", then it must be perfect for all users, and all applications, under all circumstances.)
  
 The definition of "a perfect CODEC" that I use is "a CODEC that reproduces the original closely enough that I can be absolutely certain that no difference will ever be audible, under any circumstances, by any listener, with any content, using any equipment". I'll even be willing to limit that to human listeners. (I don't recall Apple, who produced that CODEC, ever making a claim as sweeping as that for it.) However, luckily for me, any truly lossless CODEC meets my definition quite easily.  
 Quote:


reginalb said:


> A perfect codec would be the smallest size file size possible, while still being completely transparent to the listener in question. Therefore, far from perfect, all of the lossless codecs are worse than Apple's AAC (at 256k), Lame MP3 (at 320k), and probably Vorbis Ogg or Opus Ogg.


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## reginalb

keithemo said:


> (I don't recall Apple, who produced that CODEC, ever making a claim as sweeping as that for it.)


 
  
 Of course not, that would preclude them from being able to market an even better one for even more money, such as ALAC.


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## KeithEmo

reginalb said:


> Of course, the actual savings are in the aggregate. Those of us that use a subscription service at $96/year, and have access to everything we could ever care to listen to, save an enormous amount of money. Plus have access to more music (I now am not constrained by budget when my wish list is long, and I have to choose which to buy). So I both spend far less, but get astronomically more for that lower amount of money. Plus I am introduced to new music, through algorithms that are far more accurate than any friend could ever be at picking out what I will like.
> 
> You are paying far more than you think you are, and getting far less for it, I think. Perhaps you have such a different taste in music than the 10's of millions of songs that you could have access to that you're better off just buying albums individually, and of course that's your prerogative. But each time you present the tradeoff, you are missing an enormous part of the calculus.
> 
> Since we keep using analogies, which I think are _highly _problematic by the way, I'll also use one. It's like saying that buying a more fuel efficient car only saves me a few cents when I drive to the grocery store, but ignoring the many thousands of other miles that I drive when I calculate the savings. Or that I _could be _driving, but never will because my more expensive vehicle prevents me from ever even dreaming to in the first place.


 
  
 Actually, I'll sort of have to split the difference with you there. I _DO_ use a streaming service, and it's great for casually listening to new things to decide which ones I like. I also listen to music on HD Radio in my car - even though the audio quality is clearly (to me) inferior. Therefore, I have access to try and sample just as many new things as you do.
  
 However, once I find a group or album I really like, then I go and buy a "full quality" copy, so I know it's really at original quality, and I also don't have to worry about things like network outages when I want to listen to it, or whether it might mysteriously become unavailable next week, or whether, even if I am using a "lossless full quality streaming service", a few samples got lost somewhere between their site and my router.
  
 Now, to be fair, I do happen to be the type of person who spends most of my time listening to things I am familiar with and know I like, rather than exploring for new stuff, so I spend most of my time listening to one of the few thousand albums in my "permanent collection". So I'll probably go home tonight, listen to a few old favorites, then spend a half hour or so trying a few new ones I've heard good things about. I must also say that my experience with "recommendations" doesn't jive with yours; I find that I do like songs friends recommend a significant portion of the time, but I rarely like ones suggested by any of the various automated algorithms.
  
 I will also concede that I may have requirements that exceed those of most people. For example, many people might consider it a bargain to pay half as much for files which were audibly indistinguishable from the originals 90% of the time, only noticeably different with very careful listening 8% of the time, and only clearly different 2% of the time. Personally, if I was in that situation, I would feel obliged to carefully test each and every file to make sure it wasn't in "the bad 2%", and the extra time it would take to do so (or the uncertainty of wondering whether every song I played but didn't test was "one of the bad ones") would cost me more in lost fun than the price of simply paying twice as much to avoid the uncertainty. (When I play a physical file on my computer it will actually tell me if a single bit has been lost or altered, and I can even use a program to test the entire collection, all at once, and so easily confirm that every bit on every song is exactly as it should be. You simply can't get that level of assurance with even "lossless" streaming services - and I find it very... reassuring.)
  
 Note that I'm not nearly that picky about the wine I drink, or the quality of the picture on my TV, but I happen to be somewhat obsessed with getting the absolutely best possible experience with my music. I'm not at all suggesting that other people should follow my example in this regard.


----------



## RRod

keithemo said:


> Note that I'm not nearly that picky about the wine I drink, or the quality of the picture on my TV, but I happen to be somewhat obsessed with getting the absolutely best possible experience with my music. I'm not at all suggesting that other people should follow my example in this regard.


 
  
 You don't think the rest of us are obsessed with quality? Hell I just sold off a set of Mahler symphonies I bought because it audibly clipped one time. You think I'd be hanging about with lossy codecs if they were only delivered at low bit rates?


----------



## KeithEmo

reginalb said:


> I don't disagree with much of that post (if any). I guess the one thing that bugs me in the realm of audio contrasted to the other industries wherein I am interested in the gear, is that it seems that the enthusiasts in this one seem to know less and less as they do more research. Or more appropriately, they know so much that isn't so. I don't see this in other interests, for example, I'm a SCUBA diver. And as a gear-head, I have an unhealthy obsession with SCUBA gear. And while you get similar marketing, and similar publications that are funded by the companies doing the marketing, and some dive shops are pretty unscrupulous, you only have to spend a little time digging to find both Dive shops and enthusiasts online that will just maim and destroy the similar pseudo-scientific arguments made by the marketers, and give really good advice.
> 
> I am also in to photography, and at the Fred Miranda forums, there are fandbois, sure, but just as many people that love to hate on their own gear of choice, and will shred the manufacturers for small mistakes that would be defended to no end on Head-Fi. I mean, I do a TON of research when I buy things, and when I first decided to get "good" headphones, I did tons and tons of reading online, and Hifiman players were nearly universally praised as making your music sound SO much better. I mean, there are certainly some online communities that will take audio companies to task, but it seems that in Head and Hi Fi, these are the exceptions, where in other industries, they are the mainstream.
> 
> I guess I don't understand why audiophiles seem more susceptible to claims of the pseudo-scientific variety, whilst also taking a downright hostile view to any attempt to remove bias from testing. Perhaps my thermometer is off, and this industry is similar to others, but I don't think so. Another way to look at it, and why I feel bad for some when it comes to head-fi: most people don't do tons of research themselves, but instead rely on recommendations of trusted compatriots with specialized knowledge in some field when they're making a purchase. I feel that with audio equipment, that trusted compatriot is fare more likely to just blow marketing smoke at them and give bad advice than in other industries, _even if _that person has done enormous amounts of research in to the industry.


 
  
 I agree with you 100%.
  
 The world of photography seems to be dominated, for the most part, by people who have actual knowledge about the technical details of how photography works, and by people who, while they don't understand the technology very well, understand what they don't know, and want to learn. (And by people who don't understand the tech, and don't care to, but are willing to accept what the actual pros tell them.)
  
 If Nikon were to start selling a new and very expensive lens that performed very poorly, odds are that pretty soon someone would publish a review, including the numbers and measurements that proved how bad it was. And, if a few fans spoke up, insisting that, in spite of its poor technical performance, that lens had some intangible virtue that justified its price, they would be expected to provide proof that there was some truth to their claim. After all, it's pretty well known which measurements count in a lens, and how to tell if they're good or bad - so nobody would be trying to convince you that "good resolution - out to the edges - is bad".
  
 In comparison, the audiophile world seems to have a much higher percentage of people who don't understand the technology, but who imagine that they do, and are excessively eager to pass on their (frequently erroneous) knowledge to others. This seems to have lead to a culture where sincere desire has replaced actual knowledge, and where subjective opinions often replace facts. Worse yet, there seems to be a lot of confusion about which is which, no doubt fueled in part by vendors, many of whom make a living off of the uncertainty created by the confusion.
  
 In contrast to the world of photography, if some audio company starts selling a new product which lacks any technical merit, and someone points this out, they will probably be attacked for being "an objectivist", and that attack will be supported by a whole slew of self-proclaimed "subjectivists" who insist that, even with a total lack of any technical facts to support them, "they just like the way it sounds". And, unlike the artistic photographer who will readily admit that he used that old lens with the horrible distortion "for artistic effect", the fans of that product will insist that "their opinion that it sounds good is just as valid as the numbers that show it doesn't do anything at all".

 I think part of the problem is that, unlike photography, a lot of the basic technical facts about audio are actually new. This means that a lot of "the old wisdom" (what you find in twenty year old textbooks about audio) is outdated or irrelevant. This forces newcomers who do want to learn to try to pick the reality from the fiction in fluid and often inaccurate sources of information like the Internet. (Many audiophiles I know would readily admit that they don't know enough about physics to know whether String Theory makes more sense than The Copenhagen Model; yet far too many of them seem quite firmly - and wrongly - convinced that they know the difference between how a D-S DAC and an R2R DAC work, and fully understand the ramifications of that difference - probably based mostly on the claims of a vendor who has a vested interest in promoting one or the other.) It seems to me that, just like astrology sort of flows in to fill a gap in people who are seeking answers about the meaning of life, but can't find them anywhere else, other types of pseudoscience operate much the same way, and the audiophile world is one of the few remaining "frontiers" where this still happens. (In this sense, audiophile-ogy is more of a religion than a science.)
  
 (And, unfortunately, while most colleges offer Physics courses, I don't know of too many relevant short courses on "modern audio technology".)


----------



## sonitus mirus

I own about 30-40% of the music that I have added to my library using Google's service, but I only uploaded those few CDs that were not already available to stream.  I don't bother with the CDs when I can stream the music, and I am confident that the audio quality is identical.  So, if I can enjoy the CD during a serious listening session, than the streaming service is absolutely perfect for doing the same.  If the network goes down, which is practically never around here, I can still listen to music, but my choices will be severely limited.  If an artist or label pulls the plug and they are no longer available to stream and the music is important to me, I can still purchase this later if I don't already own it.
  
 I already have my zombie apocalypse CD collection and a drawer full of batteries to get me through those first few terrifying nights.
  
 Even then, there is nothing perfectly safe from every scenario even with your personal collection.  When the times come that I want to hear a certain song and the network is down or the music has been pulled from the catalog, I can decide if this is occurring frequently enough to move on to some other solution.
  
 There is a difference between some rare, irreplaceable CD and something from Pink Floyd's studio collection or Taylor Swift's streaming holdouts.  Whether the networks is down, the artist is no longer available to stream, or the disc is somehow damaged beyond repair; there is a difference between "that sucks" and "NOOOOOOOO!".  The music that I can only stream with no backup is in my "that sucks" bucket.  Any "NOOOOOOOOO!" CD that I had would be carefully copied and stored in a very safe place.


----------



## KeithEmo

rrod said:


> You don't think the rest of us are obsessed with quality? Hell I just sold off a set of Mahler symphonies I bought because it audibly clipped one time. You think I'd be hanging about with lossy codecs if they were only delivered at low bit rates?


 
  
 Not at all, but I do think we all manifest our concerns differently.
  
 For one thing, you are obviously a classical fan, which means that, for a given Mahler Symphony, there are probably a lot of different performances to choose from. To me, on the one side, this gives you a lot more options - but, on the other side, it introduces a lot more uncertainty. (How can you ever know if you've got the best performance of your favorite symphony or if you've been missing the best one and listening to the second best one... or even the tenth best one?)
  
 In contrast, I tend to listen to pop and rock music, where the reverse situation usually applies - most recent albums have only a single "performance". (And, if you don't like the performance, or the way it was recorded, you're sort of stuck.)
  
 If I were to hear a clip on the CD of my favorite group's latest album, I can't decide to try a different performance - because there is only one release of that album.
  
 Now, let's assume I hear "something a little odd" on that album. If I've got the CD I can confirm whether the copy I'm playing is bit-perfect or not; either by downloading a new copy and comparing them, or by comparing that file against its own checksum. However, if it turns out that the file is perfect, then I'm done... because I already have "the best copy available".
  
 However, what if I was listening to an MP3 version of that album? Well, first I could do the same process - download a new copy and compare it to my current one, or check the file itself with a checksum. However, now what do I do if the odd sound is still there and the file is perfect? The answer is obvious; I buy the CD copy and try to figure out if there is some sort of error with the encoding. (And, yes, it's possible that a lossless file could have an error in it but, since the process of encoding a lossless file usually includes decoding the result and comparing it with the original to verify that there isn't an error, that seems a lot less likely.)
  
 There's a saying when you shoot guns: "Never point a gun at someone you don't plan to shoot; even if you're absolutely certain it's not loaded... because you could be wrong." And that goes double if you're only sure it isn't loaded because they guy who just handed it to you said it wasn't loaded. To me, almost certainly at least partly because I do know how the whole masking process works, and because of a few specific experiences in the past, I will never absolutely "trust" a lossy encoder unless I confirm each encoded file individually and specifically. And, since I require that certainty to be satisfied with a recording, and because I don't have the time or the inclination to do A/B testing on every song I own every time I purchase a new pair of speakers, or a new set of headphones, to confirm that my new equipment doesn't enable me to hear some flaw which I didn't notice before, I find it much simpler and less work to simply retain the actual original file (the cost of the amount of testing I would have to do before I would be comfortable that the lossy file was audibly perfect far outweighs the value of making it smaller). And, even if I actually did trust the MP3 encode process itself 100%, that still only applies to files where the original was of known provenance, and I was in charge of the process. It still wouldn't extend to files passed to me via a streaming service, where I have no control or assurance about what encoder was used, and with what settings, and how carefully the results were verified. I just find it to be much safer to start with a known quantity - and then carefully avoid altering it in unknown ways.
  
 In short, at least to me, the benefits don't come anywhere near outweighing the costs.


----------



## RRod

I don't think anyone can blame someone for hedging in rock/pop, especially with the craptastic mastering efforts of the past decade(s). And you're right about classical: it's nice to have options. And imaging how nice it is to be able to preview those options in full before buying, my recent Mahler problems being the final knife in my back to convince me to get on some streaming service.
  
 The bigger question here, really, is what those of us on the "lossy could work" side would have to do to convince you that everything would be ok if the *only* difference in your content was the codec? The feeling I get right now is that the answer is "nothing, ever, because we can never be sure," in which case I guess we're done here.


----------



## reginalb

keithemo said:


> In short, at least to me, the benefits don't come anywhere near outweighing the costs.


 
  
 I am pretty new to these discussions, and know far less about audio engineering than most people in this section of the forum. I am here to learn, but I have my own biases (towards what I perceive as the best designed studies), which cause me to weigh in on the debate even if I'm not the most knowledgeable. I think that I have enough experience in research in general to give some insight, though.
  
 I think the important thing to remember right now is that it all comes down to perception. We (those that use lossy codecs more readily - I still rip in FLAC for whatever reason, on the very rare occasion that you find me ripping a CD) perceive a lower cost, and much higher benefit to lossy codecs because of the way that we think about them, and the opposite is true for you. You're certainly not an irrational supporter of lossless codecs (I've seen some pretty odd arguments here), and I enjoyed the more philosophical debate here, but like RRod think we've turned up every rock there is to be turned. 
  
 Funny aside, given my position in all of this: I just bought a SACD/Blu-Ray player, and ordered some SACD's, I've wanted a few multichannel recordings for a while, and got a good deal on a Blu-ray player that also plays SACD, so I figured, why not? I think we can all agree, though, that if 24-bit and greater than 48k was not just useless, but bad for music (as the first post suggests) - then that must be doubly true for direct stream digital.


----------



## castleofargh

reading all this, while massively long(even for me), I found many interesting points usually not discussed as to why we pick a "side".
  
 overall in the "let's use high res when we can" group, we have mostly people who want a no compromise listening experience. and how things are really audible on lower resolutions/codecs is just a side story eclipsed by "why take the chance when we can have it all in high res?".
 it does make sens.
  
 the other side, is more in the "why not use more convenient stuff when needed as we don't really hear a difference anyway?". they try, fail to really notice a difference more than maybe once every 10 songs for half a second on max AAC or mp3, and just fail all the time on CD resolution. so the overall feeling is that it's not significant enough to always bother with highres or even lossless 16/44.
 does make sens too.
  
 those are different approaches to the same question, and ultimately we make a choice based on our experience and priorities. from my point of view, no choice is wrong for the guy who finds the drawback minimal. be it size/price, or minute resolution loss.
  
  
 the part RRod mentioned, and where I very much identify with his problem and conclusion, is when you go highres with the idea of getting the best stuff possible, to come home with a crap master that has been upsampled, and a strong feeling of getting probed for appendicitis. it has nothing to do with the file format, it's just some guys who did a poor job. yet still advertised the result as being the superior stuff deserving to be sold at highres price.
  this is the kind of methods from the industry where there is no control, and that happens just too often to call it an accident anymore. we all had a few albums like that, where what we paid extra was pretty much a sox upsampling, or a remaster that should have had the sound engineer fired on the spot.
 this is a very strong component as to why I do not buy highres anymore. it's my citizen way of saying that I dislike being lied to and robbed. because if we don't want something, paying for it to continue isn't the most effective way to make them stop. the industry has only one value, money.
 now when oh miracle, the guys did a great job and the highres(or not) remaster is great, that's what should have an added market value and will make me give my money with a smile as a bonus. not bit depth or fancy formats that in the end do no more than PCM and are just an attempt at making us buy another gear for compatibility.
  
 I would pay more for CD resolution that has good recording and well done mastering(at least not too much brickwalling/clipping...). I won't pay more for a sample rate number. that's just not what makes music sound good.


----------



## KeithEmo

rrod said:


> I don't think anyone can blame someone for hedging in rock/pop, especially with the craptastic mastering efforts of the past decade(s). And you're right about classical: it's nice to have options. And imaging how nice it is to be able to preview those options in full before buying, my recent Mahler problems being the final knife in my back to convince me to get on some streaming service.
> 
> The bigger question here, really, is what those of us on the "lossy could work" side would have to do to convince you that everything would be ok if the *only* difference in your content was the codec? The feeling I get right now is that the answer is "nothing, ever, because we can never be sure," in which case I guess we're done here.


 
  
 I guess that would depend on the specific context we're discussing.
  
 I consider my "needs" to be quite different for "my master copy of an album I really like" and for "reasonably serious listening" and "casual listening". (I consider the copy of the music that I download and keep in my collection as "a master copy" and not as simply "a delivery medium" - which may be the biggest difference in our outlooks.)
   
If we're talking about my only copy of an album that I especially like, then you're probably correct, nothing is going to convince me to deliberately alter it from a perfect and accurate version of the original to save space. (I can buy a CD, RIP the contents, and my ripping program will magically compare the checksum to an online database, which will confirm that my copy of those files is identical to other copies ripped by other people. I can then store that checksum so that I can confirm that any copies I have remain unaltered.) I consider this to be about the minimum level of assurance that "makes me entirely comfortable". (And, at least for the time being, the CD version is still the best available "reference copy" for most albums. And, unlike you, I want my own master copy - I don't trust some service or vendor to "hold it for me and just deliver it to me when I ask".)

   
As I may have mentioned before, my background is in electronics and computers, and I tend to treat my music files just like I would treat any other critical data stored in computer files. I wouldn't settle for having the contents of my checkbook being "close to" what they should be, or even "indistinguishable from correct"; my spreadsheet program tests every file it opens for casual errors, and I keep backups and use data integrity checks to ensure that my data hasn't somehow become corrupted. And I think of my digital music the same way, and treat it the same way, which is why I am so resistant to deliberately introducing minor errors into the data "because they don't matter". There's simply no way I would consider using a checkbook that reported my $1237.49 checking account balance as "about $1250", and, to me, that's what lossy data storage amounts to - even though we may both agree that "about $1250" is absolutely close enough for any practical purpose.

  
 Now, when it comes to more casual listening, like at my desk, or on the way to work, I would probably be satisfied with a high-quality lossy format like MP3. In fact, I have used it in the past when I wanted to store a reasonable amount of music on a portable device with limited storage capacity. The main reason I don't use lossy files for that now is basically that, since I have already "standardized" on lossless files for critical listening, it would require me to make and keep duplicate copies for different purposes; since my master copies are already stored as FLACs, it's simply easier to copy the FLAC files right onto a portable drive as is than it is to convert them. 
  
 I will also concede that, in general, the requirements of "data storage" and "data delivery" are rather different. The Red Book CD playback standard includes three levels of error correction. If an error on the data read from the CD is detected by either of the first two levels, it is corrected perfectly, which means that the data delivered by the drive is exactly the same as the original. However, CD players include a third level - which uses interpolation to fill in gaps that cannot be perfectly corrected with "an approximation" (depending on how much damage is involved, and the content around it, the results range from inaudible to a loud tick or pop - however, in most cases, the CD doesn't stop playing). This sort of thing is necessary in a world where CDs are treated badly, and often become badly damaged. Likewise, because network data is sometimes subject to packet loss or unusual delays, most streaming deliver systems are designed to be "tolerant" of faults. (In contrast, most computer CD drives don't have that third interpolation level of error correction; with most computers, playback will simply stop with an error message if an uncorrectable error is detected, which is why computer CD drives seem "fussier".) Given the choice between an occasional odd noise or tick, or having everything stop, and a red screen popping up that says "Uncorrectable Error Detected - Playback Stopped", most consumers do seem to prefer putting up with a slight flaw now and then.


----------



## sonitus mirus

castleofargh said:


> reading all this, while massively long(even for me), I found many interesting points usually not discussed as to why we pick a "side".
> 
> overall in the "let's use high res when we can" group, we have mostly people who want a no compromise listening experience. and how things are really audible on lower resolutions/codecs is just a side story eclipsed by "why take the chance when we can have it all in high res?".
> it does make sens.
> ...


 
  
 In general I agree, though I would say that if I heard any verifiable differences at all with any of my music at the highest AAC/MP3 quality, I would probably listen mainly via lossless or higher and only use lower quality for background noise or searching for new stuff to purchase.  
  
 I carefully tested a lot of my favorites.  I tested what I thought should be very difficult music for MP3 to create.  I tested what others suggested should be the most difficult music for MP3.  I searched for any sign of a valid ABX test and purchased the same music to test for myself.  In no instance could I hear any differences at all.  My search is not over, though it has become more evident that my results are accurate, and no amount of speculation, empty claims, outright lies or misdirections, ignorance, or references to information that is no longer relevant will convince me otherwise.  I'm still looking for what appears to be valid ABX results from people that do not seem adamant to prove they have awesome ears and audio equipment.  Results that could be tested by others where the methodology could be explained and the files could be examined.  
  
 My ears apparently are good enough to pass the Golden Ears challenge.  My feathers get a bit ruffled when I read about obvious differences and claims that MP3 or AAC are only good enough for casual listening or with lower priced (meant to infer lower quality) equipment.
  
 I say show me what you mean by this, don't just tell me about it.  And so far, bubkis.


----------



## KeithEmo

I agree with you absolutely....
  
 Unfortunately, at least for the moment, buying a high-res file instead of a standard-res one is a lot like buying a bigger box of breakfast cereal..... you may be getting more, or you may just be getting the exact same thing in a bigger box. In fact, you may even be getting an inferior product, but in a prettier box, for a higher price. Sadly, as most manufacturers and advertisers know, most people probably decide which cereal to buy based more on how the box looks in the supermarket, and on how cool the TV commercials are, than on how the cereal actually tastes. Even more sadly, this means that, from their perspective, where the priority is selling more boxes of cereal, it probably makes perfect logical sense to spend more money on the box and the advertising than on the cereal.
  
 Personally, I'm firmly in the "why take a chance" camp, but I'm perfectly happy to acknowledge that convenience is also important, and how different people rate the importance of it vary widely from person to person (and it's certainly true that it doesn't matter how perfect your music collection sounds if it's so inconvenient that you never listen to it). To _ME_ the perfect balance is lossless digital music files played on a computer... but that doesn't mean that's the ideal balance for someone else. 
  
 It would also be nice if everyone, even those who don't actually understand the technology, could at least assume that what they're told is somewhat accurate. (As we were talking about in another post, if you go to buy a camera lens, and pick one from a reputable company that gets good reviews, you probably won't go too far wrong. However, the same can't necessarily be said for audio equipment or for music.) However, at the moment, there's really no substitute for actually knowing enough about the technology to understand what's going on and decide for yourself.
  
 Quote:


castleofargh said:


> reading all this, while massively long(even for me), I found many interesting points usually not discussed as to why we pick a "side".
> 
> overall in the "let's use high res when we can" group, we have mostly people who want a no compromise listening experience. and how things are really audible on lower resolutions/codecs is just a side story eclipsed by "why take the chance when we can have it all in high res?".
> it does make sens.
> ...


----------



## RRod

I guess I'm a communist and aim towards a unified audio solution for all. This feeds my wariness of the "might as well approach", because it literally can never end. Just look at the nonsense of things like DXD and DSD512. Such things invite the never-ending "bigger is better" approach and all the associated dilettantism (300000000 > 10, thus ABX is stupid). A paradigm where you have the whole world of music at your fingertips in formats that are perfect to our ears with a maximum of parsimony is wholly appealing to me. As a start, though, I would just like us to pick a god**** format and start actually trying to address things surround sound and HRTFs more frequently, rather than continually releasing plain old stereo remasters of the same stuff.


reginalb said:


> Funny aside, given my position in all of this: I just bought a SACD/Blu-Ray player, and ordered some SACD's, I've wanted a few multichannel recordings for a while, and got a good deal on a Blu-ray player that also plays SACD, so I figured, why not? I think we can all agree, though, that if 24-bit and greater than 48k was not just useless, but bad for music (as the first post suggests) - then that must be doubly true for direct stream digital.


 
  
 Nothing irks me more than having a collection of SACDs whose multichannel layer is just kind of waiting around for Sony or whoever to just give up and let us decode the stuff without some PS3 hack. I will say that I think the title of the thread is unfortunately combative, as higher specs can certainly be useful for recording.


----------



## castleofargh

sonitus mirus said:


> In general I agree, though I would say that if I heard any verifiable differences at all with any of my music at the highest AAC/MP3 quality, I would probably listen mainly via lossless or higher and only use lower quality for background noise or searching for new stuff to purchase.
> 
> I carefully tested a lot of my favorites.  I tested what I thought should be very difficult music for MP3 to create.  I tested what others suggested should be the most difficult music for MP3.  I searched for any sign of a valid ABX test and purchased the same music to test for myself.  In no instance could I hear any differences at all.  My search is not over, though it has become more evident that my results are accurate, and no amount of speculation, empty claims, outright lies or misdirections, ignorance, or references to information that is no longer relevant will convince me otherwise.  I'm still looking for what appears to be valid ABX results from people that do not seem adamant to prove they have awesome ears and audio equipment.  Results that could be tested by others where the methodology could be explained and the files could be examined.
> 
> ...


 
 well I fail miserably to identify my mp3 file from the lossless, except for a few files where the mp3 decoding seemed to clip the file(I was supposed to look for a track where it happened, but I've forgotten to do so and reverted back to just check the prevent clipping in foobar
	

	
	
		
		

		
			





). but this can be avoided so IDK if that counts as mp3 sounding different? for AAC we've talked about the apple converter that does the job well, so I guess that's a non issue. deep down I'm with you and I really can't stand all the "night and day" propaganda. too many people are making a fool of themselves without knowing it. still we will always have at best, a vast amount of statistic from blind tests all over the world agreeing with us. but that will never really be enough to call someone a liar to his face and be 100% sure to be right about it. so I tend to keep the door open(maybe too much?) when talking about mp3 being audibly different.


----------



## reginalb

rrod said:


> Nothing irks me more than having a collection of SACDs whose multichannel layer is just kind of waiting around for Sony or whoever to just give up and let us decode the stuff without some PS3 hack. I will say that I think the title of the thread is unfortunately combative, as higher specs can certainly be useful for recording.


 
  
 I actually bought a 5-disc SACD changer for dirt cheap, not realizing that copy protection prevents them from outputting the multichannel stuff via optical out, and my receiver (a Pioneer Elite, but not the top of the line Pioneer Elite) doesn't have mulitchannel inputs. So I've got it for sale, and ordered the slightly more expensive (but admittedly much more useful) blu-ray player. 
  
 Could you not use a multichannel ADC to encode the files, though?


----------



## sonitus mirus

castleofargh said:


> well I fail miserably to identify my mp3 file from the lossless, except for a few files where the mp3 decoding seemed to clip the file(I was supposed to look for a track where it happened, but I've forgotten to do so and reverted back to just check the prevent clipping in foobar
> 
> 
> 
> ...


 
  
 I don't know of anyone that is lying and I would not have any evidence to suggest they were a liar, at any rate.  That was just included in my laundry list.
  
 I suppose I want to find that "Eureka!" moment.


----------



## RRod

reginalb said:


> I actually bought a 5-disc SACD changer for dirt cheap, not realizing that copy protection prevents them from outputting the multichannel stuff via optical out, and my receiver (a Pioneer Elite, but not the top of the line Pioneer Elite) doesn't have mulitchannel inputs. So I've got it for sale, and ordered the slightly more expensive (but admittedly much more useful) blu-ray player.
> 
> Could you not use a multichannel ADC to encode the files, though?


 
  
 My poor E-MU can only handle 4 channels I think, and paying for a new ADC just so I can sit around waiting for things to happen in realtime isn't something for which I champ at the bit


----------



## reginalb

rrod said:


> My poor E-MU can only handle 4 channels I think, and paying for a new ADC just so I can sit around waiting for things to happen in realtime isn't something for which I champ at the bit


 
  
 Hahaha, recording in real time takes me back to my minidisc days. I got a pretty early MD recorder for Christmas from my mom one year. She got a deal on it packaged with a Walkman, and got one for me and one for my step-brother. I had no idea what it was, but holy crap did I get miles out of that thing. A friend of mine at the time had some early, I think iriver MP3 player. I used to argue with him that MDs would outlast MP3 players, because the quality was so much better, and I could carry a little pouch with hundreds of songs, while his little MP3 player fit a few crappy little MP3s, and didn't have interchangeable memory. 
  
 A prophet I was not. But recording in real time wasn't too bad, I would set up a CD changer, set it to auto-separate tracks, and walk away. Took me a few weeks to get my whole CD collection on MD, but I did it. I spent a lot of my allowance on CD's so I had a lot of them. I even got an alpine MD head unit for my first car, I was a long time holdout before switching to digital with the third-ish gen ipod. By then I was in college, and that thing was a revelation for a rower in Michigan (winter training involved 7 workouts a week on the rowing machine indoors for like 4 months). 
  
 EDIT: Holy crap, this is it: http://www.amazon.com/Sony-MDSJE320-MiniDisc-Recorder/dp/B00001ZWWI
  
 Silly audiophiles, I need to dig that thing out and sell it.


----------



## OddE

reginalb said:


> Could you not use a multichannel ADC to encode the files, though?


 
  
 -The hardcore way of doing so would be to use a 2-channel ADC and simply run the process three times, then syncing the tracks afterwards. (Come to think of it, easiest way to ensure perfect sync would be to do it five times - ch. 1+2, 2+3, 3+4, 4+5 and 5+6 - that way, you could always sync channels perfectly.)
   
Would probably be more practical to just pop that bleedin' SACD disc in the player, though.


----------



## old tech

reginalb said:


> Funny aside, given my position in all of this: I just bought a SACD/Blu-Ray player, and ordered some SACD's, I've wanted a few multichannel recordings for a while, and got a good deal on a Blu-ray player that also plays SACD, so I figured, why not? I think we can all agree, though, that if 24-bit and greater than 48k was not just useless, but bad for music (as the first post suggests) - then that must be doubly true for direct stream digital.


 
  
 The thing is that you'll probably notice a sound quality improvement with most SACDs.  Where some audiophiles sometimes get it wrong is that they ascribe the better sound to hi res or DSD technology when it is just the music has been mastered better for a more discerning audience.  The label could easily release the same quality mastering on CDs but they usually don't - even the CD layer on most SACDs are mastered differently (usually brickwalled or heavily compressed).  I have many MFSL and original release CDs which sound better than the SACD version which, if  one suscribed to the superior technology theory, wouldn't think was possible.


----------



## reginalb

old tech said:


> The thing is that you'll probably notice a sound quality improvement with most SACDs.  Where some audiophiles sometimes get it wrong is that they ascribe the better sound to hi res or DSD technology when it is just the music has been mastered better for a more discerning audience.  The label could easily release the same quality mastering on CDs but they usually don't - even the CD layer on most SACDs are mastered differently (usually brickwalled or heavily compressed).  I have many MFSL and original release CDs which sound better than the SACD version which, if  one suscribed to the superior technology theory, wouldn't think was possible.


 
  
 This is actually why I've bought some digital downloads from Acoustic Sounds. I like that they disclose who did the master, and I've long suspected that you'll find better masters in the releases aimed at not being listened to on cheap earbuds while walking down the street with tons of ambient noise. They have to raise the quiet sections of a track for that type of situation, or you wouldn't be able to hear those quite sections, without turning up the volume so much that you'd rupture your eardrums when a loud section kicks in (my digital copy of of Ella and Louis has a few moments of pop the in-ears out when the horns kick in because holy crap that hurt)
  
 Wouldn't it be nice if downloads came with 2 masters? Like a high and low ambient noise master? Or if they all just had the low ambient noise master, and you could use the DSP options in something like the bulk transcoder in dbpoweramp to create that? A pipe dream I'm sure, but still.


----------



## sonitus mirus

reginalb said:


> This is actually why I've bought some digital downloads from Acoustic Sounds. I like that they disclose who did the master, and I've long suspected that you'll find better masters in the releases aimed at not being listened to on cheap earbuds while walking down the street with tons of ambient noise. They have to raise the quiet sections of a track for that type of situation, or you wouldn't be able to hear those quite sections, without turning up the volume so much that you'd rupture your eardrums when a loud section kicks in (my digital copy of of Ella and Louis has a few moments of pop the in-ears out when the horns kick in because holy crap that hurt)
> 
> Wouldn't it be nice if downloads came with 2 masters? Like a high and low ambient noise master? Or if they all just had the low ambient noise master, and you could use the DSP options in something like the bulk transcoder in dbpoweramp to create that? A pipe dream I'm sure, but still.


 
  
 I think isolation and noise cancelling are better options over having 2 distinct masters, though not practical for everyone.  It certainly would be nice to have some DSP parameters that could be read and implemented by audio playback software.  It would probably require too much cooperation and money to make a reality.  Maybe the "loudness" option that typically EQs the bass and treble about +6dB is all that is needed for most situations with this type of music?


----------



## reginalb

sonitus mirus said:


> I think isolation and noise cancelling are better options over having 2 distinct masters, though not practical for everyone.  It certainly would be nice to have some DSP parameters that could be read and implemented by audio playback software.  It would probably require too much cooperation and money to make a reality.  Maybe the "loudness" option that typically EQs the bass and treble about +6dB is all that is needed for most situations with this type of music?


 
  
 You're probably right, but most people are going to use the enclosed ear buds when they listen to music. And honestly, I can't stand running outside with too much isolation, but do want to at least kind of hear the music. You're probably right that a simple option on a DAP would do the trick better than my proposition.


----------



## KeithEmo

sonitus mirus said:


> I think isolation and noise cancelling are better options over having 2 distinct masters, though not practical for everyone.  It certainly would be nice to have some DSP parameters that could be read and implemented by audio playback software.  It would probably require too much cooperation and money to make a reality.  Maybe the "loudness" option that typically EQs the bass and treble about +6dB is all that is needed for most situations with this type of music?


 
  
 There actually are some options like that in the audio formats used on most DVDs and Blu-Ray discs. Dolby Volume is intended to make it so you can still hear dialog, even when you're listening in the middle of the night, with everything turned down quiet. The option itself is offered on most home theater pre/pros and AVRs, and, more to the point here, the producer of the disc gets to configure how much compression is applied when the user turns it on.


----------



## hogger129

24-bits and high sample rates are necessary for mastering, not listening.  For listening, CD quality is all you need.  They just need to master CDs and digital downloads the right way.


----------



## coli

I got a high end active monitor setup now, and it is very rare to find a hi-res recording that actually sound better than CD quality. It really depends on how they record/mix/produced the music. K2HD (upsampled 96k/24bit) for example always sounds worse than CD. 
  
 The recording has to be recorded in hi-res, and the subsequent steps must not have any resampling done to it in order to produce something that sounds better than CD. And this is apparently really rare today....
  
 For normal setups though, you won't be able to hear the difference, so it does not matter. CD is the safe choice.


----------



## MacacoDoSom

coli said:


> I got a high end active monitor setup now, and it is very rare to find a hi-res recording that actually sound better than CD quality. It really depends on how they record/mix/produced the music. K2HD (upsampled 96k/24bit) for example always sounds worse than CD.
> 
> The recording has to be recorded in hi-res, and the subsequent steps must not have any resampling done to it in order to produce something that sounds better than CD. And this is apparently really rare today....
> 
> For normal setups though, you won't be able to hear the difference, so it does not matter. CD is the safe choice.


 

 I think you're maybe right, but... can you define 'a high end active monitor setup' ?


----------



## coli

macacodosom said:


> I think you're maybe right, but... can you define 'a high end active monitor setup' ?


 
 Focal SM9 is unbelievable. Consumer setups are just... sad after this.


----------



## MacacoDoSom

coli said:


> macacodosom said:
> 
> 
> > .I think you're maybe right, but... can you define 'a high end active monitor setup' ?
> ...


 

 It's not unbelievable, what I find unbelievable it's the 'audiophile' people that doesn't know about the so called pro market.
 Its a very good professional product...with a price not so high considering...,  there are several other good products/alternatives (much cheaper ones) like that in the market, much cheaper than the half equivalent consumer 'audiophile' products. Maybe that's why they are called consumer....products...
 If you have a pair of SM9's and a small pro audio-interface +/- $300 (once again much cheaper than the consumer DACs with half the features) you will be much better served than a double/triple priced HI-FI ultra super dupper audiophile system you can imagine. (unless you use some tunning dots and atomic cables).
 But let me say it, they are a bit overkill (if there is such thing) for the average home/room. 
 Congratulations... and... happy birthday...


----------



## coli

macacodosom said:


> It's not unbelievable, what I find unbelievable it's the 'audiophile' people that doesn't know about the so called pro market.
> Its a very good professional product...with a price not so high considering...,  there are several other good products/alternatives (much cheaper ones) like that in the market, much cheaper than the half equivalent consumer 'audiophile' products. Maybe that's why they are called consumer....products...
> If you have a pair of SM9's and a small pro audio-interface +/- $300 (once again much cheaper than the consumer DACs with half the features) you will be much better served than a double/triple priced HI-FI ultra super dupper audiophile system you can imagine. (unless you use some tunning dots and atomic cables).
> But let me say it, they are a bit overkill (if there is such thing) for the average home/room.
> Congratulations... and... happy birthday...


 
 It's accurate. Accuracy has both advantage and disadvantage. It's unbelievable how much the Klipsch driver rings after the comparison, but I like the Klipsch sound too


----------



## sonitus mirus

I believe active monitors offer the best price-performance ratio even at the entry level, provided that the room and listening position are appropriate to take advantage of them.


----------



## old tech

sonitus mirus said:


> I believe active monitors offer the best price-performance ratio even at the entry level, provided that the room and listening position are appropriate to take advantage of them.


 
 Could you elaborate on the room and listening position angle?  I have recently set up a pair of AVI's ADM9RS actives with an SVS sub in my living room.  The clarity of the sound is spectacular, particular the mids and mid-highs but it seems to be lacking in the mid-bass area.  The SVS sub does a great job with the lower bass, but there seems to be a hole around the 100 to 300hz range which is frustrating as the clean mids and highs really higlight this defficiency. I have tried different locations for the sub which improves the sound but the hole is still there, and played around with the crossover - but with anything over a 100hz you can then identify the sub.  Any suggestions?


----------



## MacacoDoSom

old tech said:


> sonitus mirus said:
> 
> 
> > I believe active monitors offer the best price-performance ratio even at the entry level, provided that the room and listening position are appropriate to take advantage of them.
> ...


 

 First you should check if the problem comes from the speakers, try to listen to them very close something like 0.8 m to 1 m from your head to each speaker with 0.8 m to 1 m apart from each other (equilateral triangle) with no sub and medium/low volume.
 You should not have that problem, so the problem is with the room. If you still have that hole the problem is with the speakers.....
 If that's not the case you'll have to change your living room... and that can also be a problem...


----------



## old tech

macacodosom said:


> First you should check if the problem comes from the speakers, try to listen to them very close something like 0.8 m to 1 m from your head to each speaker with 0.8 m to 1 m apart from each other (equilateral triangle) with no sub and medium/low volume.
> You should not have that problem, so the problem is with the room. If you still have that hole the problem is with the speakers.....
> If that's not the case you'll have to change your living room... and that can also be a problem...


 
 Thanks for the reply.  I'm fairly sure the problem is with the room acoustics as the sound did not have this issue when set up in my listening room. Unfortunately, even though I spend most of my home time in the living room these days, I can't do too much with it due to the wife factor, and the odd shape of the room limits my options for speaker placement.  I suppose I was just wondering if there are any (relatively) simple acoustic treatment ideas which can target this type of deficit as I'd rather not have to go down the road of a DSP solution.  Probably related to having a lack of mid-bass is that if I play music moderately loud the upper mids, though clear, are a bit overpowering and piercing which again was not an issue in the listening room.  I also have the TV/video hooked up to these speakers and the funny thing is that the lack of mid-bass and piercing upper mids are not very apparent when watching a movie or listening to a music video.


----------



## coli

sonitus mirus said:


> I believe active monitors offer the best price-performance ratio even at the entry level, provided that the room and listening position are appropriate to take advantage of them.


 
 At the entry level, cheap receiver + cheap speaker can be pretty good too. Especially receivers with dynamicEQ. Anything above entry level, and active monitors can not be beaten at all. Makes audiophile company look like scams...


----------



## KeithEmo

old tech said:


> Thanks for the reply.  I'm fairly sure the problem is with the room acoustics as the sound did not have this issue when set up in my listening room. Unfortunately, even though I spend most of my home time in the living room these days, I can't do too much with it due to the wife factor, and the odd shape of the room limits my options for speaker placement.  I suppose I was just wondering if there are any (relatively) simple acoustic treatment ideas which can target this type of deficit as I'd rather not have to go down the road of a DSP solution.  Probably related to having a lack of mid-bass is that if I play music moderately loud the upper mids, though clear, are a bit overpowering and piercing which again was not an issue in the listening room.  I also have the TV/video hooked up to these speakers and the funny thing is that the lack of mid-bass and piercing upper mids are not very apparent when watching a movie or listening to a music video.


 
  
 Sometimes problems in that frequency range can be related to surface-cancellation and boundary effects. If so, then you might be able to find an acoustic solution that isn't too bad on WAF. For example, if you're using stands, then stands of a different height might eliminate (or shift) the problem; or, if you have bare floor directly in front of the speakers, putting a throw rug directly under or in front of them may help to eliminate a cancellation with a reflection from there (it's easy enough to try). You want to try both the points where the sound would make its first reflection off the floor on its way to the listener and directly under the speaker, where there might simply be "a standing wave" between the speaker and the floor - which can cause a cancellation or addition at one frequency.
  
 The range of frequencies over which you're hearing problems have a wavelength between about three and ten feet, which means that any surface between about a foot and a half and ten feet could be related to the problem. (This puts and major surfaces in that range of distances, including the floor, walls, and even the back wall behind the speakers, under consideration. Since you're constrained in what you will be able to do, you might as well try the ones that will be easy to fix first. 
	

	
	
		
		

		
			





 )
  
 The fact that the problem gets worse when you play things more loudly sort of suggests that it is specifically related to "excessive liveness" {shiny/hard surface(s)} somewhere as being a culprit.


----------



## sonitus mirus

coli said:


> At the entry level, cheap receiver + cheap speaker can be pretty good too. Especially receivers with dynamicEQ. Anything above entry level, and active monitors can not be beaten at all. Makes audiophile company look like scams...


 
  
 I saw a recent video where Ethan Winer discusses his home theater set up, and he is using 3 Mackie HR264 monitors as his left, right, and center channel speakers.  You can get the MK2 models for around $500 from some places.  That may not be quite entry level, but it isn't too expensive.  Certainly well within the reach of most consumers.
  
 https://www.youtube.com/watch?v=qu32oisgIq0


----------



## old tech

keithemo said:


> Sometimes problems in that frequency range can be related to surface-cancellation and boundary effects. If so, then you might be able to find an acoustic solution that isn't too bad on WAF. For example, if you're using stands, then stands of a different height might eliminate (or shift) the problem; or, if you have bare floor directly in front of the speakers, putting a throw rug directly under or in front of them may help to eliminate a cancellation with a reflection from there (it's easy enough to try). You want to try both the points where the sound would make its first reflection off the floor on its way to the listener and directly under the speaker, where there might simply be "a standing wave" between the speaker and the floor - which can cause a cancellation or addition at one frequency.
> 
> The range of frequencies over which you're hearing problems have a wavelength between about three and ten feet, which means that any surface between about a foot and a half and ten feet could be related to the problem. (This puts and major surfaces in that range of distances, including the floor, walls, and even the back wall behind the speakers, under consideration. Since you're constrained in what you will be able to do, you might as well try the ones that will be easy to fix first.
> 
> ...


 
 Thanks for the tips Keith.  The room is carpeted with soft furniture, bookshelf etc.  It is also semi-open but the opening is to the side and above speaker height.  The couch is about 3.5m in front of the speakers with a wall and window directly behind it.  I might try and hang a heavier drape over the window.


----------



## Joe Bloggs

old tech said:


> Thanks for the reply.  I'm fairly sure the problem is with the room acoustics as the sound did not have this issue when set up in my listening room. Unfortunately, even though I spend most of my home time in the living room these days, I can't do too much with it due to the wife factor, and the odd shape of the room limits my options for speaker placement.  *I suppose I was just wondering if there are any (relatively) simple acoustic treatment ideas which can target this type of deficit as I'd rather not have to go down the road of a DSP solution.*  Probably related to having a lack of mid-bass is that if I play music moderately loud the upper mids, though clear, are a bit overpowering and piercing which again was not an issue in the listening room.  I also have the TV/video hooked up to these speakers and the funny thing is that the lack of mid-bass and piercing upper mids are not very apparent when watching a movie or listening to a music video.




Why not?  It's not like you have many alternatives given the WAF


----------



## old tech

joe bloggs said:


> Why not?
> 
> 
> 
> ...


 
 It probably is the best and perhaps the only solution but I prefer not to complicate the set-up.  The WAF factor is not just about aesthetics but also being simple to use, given the living room is used mainly for TV/video.  At the moment it is a simple set up only requiring a push of one button on the speaker's remote which also controls volume.  I have a listening room for my other stereo.


----------



## veril459

There has actually been a double blind study on the subject, audiophiles can not distinguish between SACD and regular CD quality (with a success rate greater than a coin toss).
 But there is an interesting conclusion there, the above only applies if the CD quality was properly downsampled from the source material. Turns out plenty of music CD makers do an incredibly bad job resulting in bad quality.
  
 Source: http://drewdaniels.com/audible.pdf


----------



## coli

veril459 said:


> There has actually been a double blind study on the subject, audiophiles can not distinguish between SACD and regular CD quality (with a success rate greater than a coin toss).
> But there is an interesting conclusion there, the above only applies if the CD quality was properly downsampled from the source material. Turns out plenty of music CD makers do an incredibly bad job resulting in bad quality.
> 
> Source: http://drewdaniels.com/audible.pdf


 
 Audiophile setups can not distinguish anything, that's actually the whole point sadly. Only recording studio setups can do it.


----------



## KeithEmo

sonitus mirus said:


> I saw a recent video where Ethan Winer discusses his home theater set up, and he is using 3 Mackie HR264 monitors as his left, right, and center channel speakers.  You can get the MK2 models for around $500 from some places.  That may not be quite entry level, but it isn't too expensive.  Certainly well within the reach of most consumers.
> 
> https://www.youtube.com/watch?v=qu32oisgIq0


 
  
 There are several choices in that price range.... and you should factor in that, by using a powered monitor, you don't need to buy, or find a spot for, an amplifier, and you only need one set of interconnect cables (and no speaker cables).


----------



## KeithEmo

coli said:


> Audiophile setups can not distinguish anything, that's actually the whole point sadly. Only recording studio setups can do it.


 
  
 Let me start by saying that I'm not disagreeing with the basic claim in the article. In fact I'm inclined to agree with the authors of the article - that most high-res recordings that sound better do so simply because they are better recordings. I've never personally heard any difference between a well recorded SACD and the same content after being converted to 88k PCM that couldn't be reasonably attributed to the conversion process itself, and I'm not convinced that there is any.
  
_HOWEVER_, I also don't find the test you cited to be compelling in the general case. The fact that a number of audiophiles were unable to distinguish between two formats in an ABX test, using (unspecified) content, converted using (unspecified) conversion software or hardware, listening on (unspecified) "normal" or (unspecified) "expensive audiophile equipment" hardly seem to me to be compelling evidence that there is no difference that might be audible under other circumstances. (Making the general claim would be like claiming that "nobody can hear when a piano is slightly out of tune" - when the reality is that, even though most people may not notice a given amount of error, there are some few with perfect pitch who can detect it quite easily. I would want to see such a test run on a wide variety of source material, with a variety of recorders, and using a wide variety of speakers, amplifiers, and headphones, before I would consider it to be reasonably representative of "everyone everywhere with all equipment".) Therefore, I would limit my "interpretation" of their results to "several people, using good quality content, and several audiophile quality speakers, were unable to tell the difference" - which is certainly interesting.


----------



## arnyk

keithemo said:


> _HOWEVER_, I also don't find the test you cited to be compelling in the general case. The fact that a number of audiophiles were unable to distinguish between two formats in an ABX test, using (unspecified) content, converted using (unspecified) conversion software or hardware, listening on (unspecified) "normal" or (unspecified) "expensive audiophile equipment" hardly seem to me to be compelling evidence that there is no difference that might be audible under other circumstances. (Making the general claim would be like claiming that "nobody can hear when a piano is slightly out of tune" - when the reality is that, even though most people may not notice a given amount of error, there are some few with perfect pitch who can detect it quite easily. I would want to see such a test run on a wide variety of source material, with a variety of recorders, and using a wide variety of speakers, amplifiers, and headphones, before I would consider it to be reasonably representative of "everyone everywhere with all equipment".) Therefore, I would limit my "interpretation" of their results to "several people, using good quality content, and several audiophile quality speakers, were unable to tell the difference" - which is certainly interesting.


 
  
 If one studies the issue at all, such as reading the article and some well known discussion of it,  one finds that the following is a litany of false claims:
  
 "The fact that a number of audiophiles were unable to distinguish between two formats in an ABX test, using (unspecified) content, converted using (unspecified) conversion software or hardware, listening on (unspecified) "normal" or (unspecified) "expensive audiophile equipment" hardly seem to me to be compelling evidence that there is no difference that might be audible under other circumstances."
  
 For example: http://www.bostonaudiosociety.org/explanation.htm and https://secure.aes.org/forum/pubs/journal/?ID=2
  
 However, the statement that "Audiophile setups cannot distinguish anything" is a very gross generalization and therefore false on its face. There is a strong tendency for audiophiles to be very chauvinistic and often completely wrong about the sound quality and resolving power of their audio systems. But _*all*_ of them?  Unlikely, and IME not true.


----------



## teddytejero

arnyk said:


> If one studies the issue at all, such as reading the article and some well known discussion of it,  one finds that the following is a litany of false claims:
> 
> "The fact that a number of audiophiles were unable to distinguish between two formats in an ABX test, using (unspecified) content, converted using (unspecified) conversion software or hardware, listening on (unspecified) "normal" or (unspecified) "expensive audiophile equipment" hardly seem to me to be compelling evidence that there is no difference that might be audible under other circumstances."
> 
> ...




It's not so that when you can't hear it ... that it is not there when reading about the 'limitations of the human ear'. With my expensive earphones HD-800 I can't 'feel' the difference but when you put me in front of a set of Mark Wilson speakers with adequate amp I can feel it in my gut and feel the vibes through my body. Music is magic and sometimes I feel it is a way to touch the devine. No maths for me ...


----------



## castleofargh

just saw this on another forum
  
  
 Quote:


> Reality is that which, when you stop believing in it, doesn't go away.


 
 P K Dick
  
  
 nicely in context I find ^_^.


----------



## KeithEmo

arnyk said:


> If one studies the issue at all, such as reading the article and some well known discussion of it,  one finds that the following is a litany of false claims:
> 
> "The fact that a number of audiophiles were unable to distinguish between two formats in an ABX test, using (unspecified) content, converted using (unspecified) conversion software or hardware, listening on (unspecified) "normal" or (unspecified) "expensive audiophile equipment" hardly seem to me to be compelling evidence that there is no difference that might be audible under other circumstances."
> 
> ...


 
  
 I don't exactly feel corrected (since I was responding solely to the information in the linked article). _HOWEVER_, I should note that, with the additional documentation, I think they did an excellent job of documenting that, at least under "typical audiophile listening conditions", and with a reasonable sample of source material and equipment, the differences are "minimal at best".
  
 (I actually agree that the test was well run and its results should be considered "significant". I just take exception that we as a society tend to over-generalize... in both directions. The folks selling high-resolution content like to over-generalize that it will _ALWAYS_ sound better; and their opponents tend to overgeneralize that it cannot possibly be so. Based on my personal experience, I would be quite comfortable stating that _SOME_ high-res recordings I own sound significantly better than their CD-quality counterparts on my current equipment; however, I suspect that the difference, when it exists, is often simply the result of better mastering or higher production values, and not because of the high-resolution format... and I can't rule out the possibility that this might always be the reason. I would also state that the majority of CD recordings fail to live up to the quality possible with that format, which makes it difficult to make valid generalized comparisons either way. However, I also doubt that anyone will ever run comprehensive enough tests to be able to state the generalization that "no high-resolution recording will ever sound better than its CD-quality counterpart, when played on any equipment, simply because of the higher quality recording format" - which is what you would have to prove to justify a claim to "there being no benefit at all".)
  
 IN GENERAL, I would say that their test was well thought out, and well documented, within practical limits... and did a pretty good job of showing that, at least with some "typical high end content and equipment", "reducing the quality" of some typical high-resolution recordings to CD quality didn't produce any audible difference - and so the high-resolution format of those recordings was providing no real audible benefit. I would also agree that this strongly suggests that, at least under some circumstances (reasonably typical ones), the claims to the contrary are exaggerated or downright false.
  
 However, I would suggest that a more concise and accurate statement of the results might be....
  
 A controlled test was conducted, using several different test systems comprised of high-end consumer and professional audio equipment, all of which was considered to be "current technology" in 2007. The results of this test showed that, when a variety of high-resolution discs (both SACDs and DVD-As) were played on several high-end consumer and professional sound systems, under normal listening conditions (chosen by the test subjects), the test subjects couldn't reliably determine whether the audio signal had been "passed through a CD-quality record and playback loop" (see details in the original article) - or not. This suggests that, under the conditions tested, and with the content and test equipment used, "reducing" the quality of the high-resolution recordings to "CD quality" produced no audible difference, which further suggests that, under the conditions tested, and using "typical current high-end equipment", the high-resolution recordings offered no audible benefit over "CD quality".
  
 NOTE that my wording accurately states the situation, and the conclusion that no differences were heard using the stated test content and equipment, but correctly neglects to rule out the possibility that differences might be audible with other source content or other equipment (which you can't reasonably rule out without a lot more testing).
  
 (My personal quibble with their test is that the performance of DACs varies considerably, and has progressed since 2007, so I'm not convinced that the disc players they used as sources were "nominally good enough" to be ruled out as a limiting factor. Perhaps, if the content they used were played back on a higher-quality DAC, there would have been details present which would have been audibly lost later in the signal chain. Likewise, while the Quads and Snells are what I would consider to be "very good speakers", I can't rule out the possibility that some of the many other speakers out there might do a better job of making some difference audible. I'm also inclined to feel that, when listening for subtle details, headphones do a better job of revealing differences than loudspeakers - yet they failed to include any headphone listening. This seems like a significant omission - and one that would have been easy to remedy. It occurs to me that they - quite reasonably - were only trying to prove the "typical case" - which they did pretty well.)  
  
 Personally, I would very much like to see a "public challenge", where a content provider, a DAC vendor, a speaker vendor, and an amplifier vendor, would get together and try to provide a sample of high-resolution music that was "so good that it couldn't be reproduced on a CD without audibly obvious degradation". Several makers of DACs and studio ADCs could then attempt to disprove the claim by showing that, when their equipment was inserted into the signal chain, none of the audience could hear a difference.


----------



## sonitus mirus

keithemo said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
  
 Why would anyone be required to disprove the claim?  That makes no sense to me.  I can disprove it now.  I don't hear any difference.  The person making the claim has the responsibility of proving it.  If there is a difference, let them show it and provide all of the details so that others can verify the results.


----------



## KeithEmo

sonitus mirus said:


> Why would anyone be required to disprove the claim?  That makes no sense to me.  I can disprove it now.  I don't hear any difference.  The person making the claim has the responsibility of proving it.  If there is a difference, let them show it and provide all of the details so that others can verify the results.


 
  
 It's a matter of perspective (philosophically).....
  
_YOU_ can make the claim that _YOU_ don't hear any difference.
 And _I _can make the claim that _I_ sometimes do hear a difference.
  
 I am no more obligated to "prove" that I do hear a difference than you are to prove that you don't hear one. However, someone else reading both our statements, and trying to determine which will prove "true" to them in their situation, is probably looking for some sort of objective "truth". And we can safely assume that someone reading about that test is actually doing so because they hope to determine whether _THEY_ can expect to hear a difference or not, and they can only reasonably do that if they are provided with enough details to determine whether the results of that test are likely to agree with their situation or not, and how well they coincide. (As long as the situations are similar, a test conducted using more people, and more variety of equipment and sources, seems more likely to have produced results that will "be right" for more people.)
  
 To me, that test was conducted several years ago, using equipment that, while it was certainly considered to be "good" at the time, almost certainly does _NOT_ perform at the same level as modern equipment; and they also used a limited variety of equipment. I know personally that in 2007 I owned a DAC that was considered to be "high-end", yet I've replaced it since then with one that performs and sounds better. Therefore, even knowing that a high-res file sounded no different than a CD to me, on the DAC I owned in 2007, wouldn't specifically suggest that the same would be true with my current equipment... and with current high-res music sources. (Therefore, while I have no specific reason to disbelieve the results of that test, I also don't have what I consider compelling reasons to assume that they're still true either. Now, if I had exactly the same equipment as that used in the test, I would consider it much more likely that my experiences with it would be much similar.)


----------



## jcx

relative advance from 2007 "good quality" is possible to see in numbers that we expect are far beyond human hearing thresholds
  
 ESS Sabre, some at the margin new releases in AKM top line, Schiit taming the AD5791 would seem to be pretty much the monolithic audio DAC offerings since
  
  
 but as an example no one has advanced with publication quality listening tests jitter thresholds despite the huge amount of ink and hardware pushing the numbers race to 10s of picoseconds
  
 which is particularly odd since its relatively simple to software simulate varying levels, frequency statistic jitters and apply them to test signals or music to do a "staircase" threshold listening test with one of the 10 picosecond jitter DACs
  
 I wouldn't be surprised if the threshold could be extended downwards by factors of several - but we now have 3 orders of magnitude between actual human listening tested jitter thresholds and ad copy, guru assurances of needed performance, current hardware
  
 Ashihara 2005 didn't seem to move the threshold meaningfully vs Benjamin and Gannon


----------



## sonitus mirus

keithemo said:


> It's a matter of perspective (philosophically).....
> 
> _YOU_ can make the claim that _YOU_ don't hear any difference.
> And _I _can make the claim that _I_ sometimes do hear a difference.
> ...


 
  
 I have no way to prove that I can't hear a difference, and I could always fake it; but there are ways to show that a difference can be heard, and this can be verified.  That was the point of my snarky reply that I can't hear a difference without even making an attempt.  My claims would be empty and unverifiable.


----------



## castleofargh

keithemo said:


> It's a matter of perspective (philosophically).....
> 
> _YOU_ can make the claim that _YOU_ don't hear any difference.
> And _I _can make the claim that _I_ sometimes do hear a difference.
> ...


 
 both sides have to try and prove their points! else it's just random people making empty claims. and nobody cares about those guys crying wolf and talking about the end of the world in the subway. proof is what makes it all meaningful.
 except that one cannot prove that he doesn't hear a difference. even if we add in the test some control samples that should be heard to try and trick cheaters, we can never know for sure that the subject will not notice those control sounds, play along and then lie for the rest. we would need to go out of our ways to make something that can never totally become proof. only strong inclinations to believe when the number of people getting the same result increases massively.
 on the other hand, if 1 guy can hear a difference, all he has to do is get statistical significance in any blind test with any gear to absolutely prove he heard a difference. we can later argue the reason for the difference, but not the claim. 
  
 that's a freaking massive difference and I cannot agree about both sides having the same responsibility on the matter. I can't prove that I can't jump over the wall, I can only show that I can fail. while the guy who can do it has the power to demonstrate he can pass over it. it's pretty clear who should bring proof.
 so as an individual who likes to listen to music, nobody give 2 ***** if I can or cannot hear a difference, or jump over a wall. and if I'm honest with myself and want to know, I can test myself all I want and find out(what I do once or twice a year when I get new gears).
   but as a person making the claim for the world to read, the very fact that I can prove it is reason enough for others to expect that I back up my claim and do prove it.
  
 you're trying to present an equilibrium that clearly never existed. objective science only offers readily available tools for proof to one side. the side that suspiciously enough, is so massively composed of people who reject those tools. at some point it stops being a simple matter of choice, and becomes hypocritical.
 if I was so adamant about convincing people that highres and CD audibly sound different, and had the tools to do it, I know what I would do...
	

	
	
		
		

		
			




  
  
 now where I join you, it's about gears. as a personal test, it only matters to me what I can hear with my own equipment, but as proof of an audible difference, it could make sens to use gears that indeed have the best fidelity possible. although I'm not sure any transducer is up for the task. in fact it's a huge point in my reason to doubt the purpose of highres in general. we're trying to improve everything that is already great and pretend to test it with transducers magnitudes below in fidelity. it feels like insisting on giving a V8 engine to a car with square tires.


----------



## KeithEmo

I think you will find that the reason more testing is not done is simply cost....
  
 Real testing, with proper controls, and large samples, costs money. Therefore, it is only going to be sponsored by a company who stands to profit from the results. (Pharmaceutical companies don't ever test drugs that don't stand to make them a profit if the tests are successful.)
  
 Companies who sell expensive interconnect cables (based on people's belief that they make a big difference) have little incentive to risk performing tests that might fail to prove that their product is better; and even a test that showed a slight but insignificant improvement would be a commercial liability. And the companies who sell cheap interconnects sell huge volume, based on people's assumption that their cables are just as good as the expensive ones, so what incentive would they have to pay for tests to confirm that? They'd have to sell a lot of $5 cables to offset the cost of the tests.
  
 Likewise, many studios now sell high-res versions of their standard offerings, and many more plan to in the future - all based on the widely accepted assumption that they're better. The status quo is probably better for their business than running tests that might show that the difference is imaginary, or even that the difference is real, but simply less significant than people now think. Again, there is nobody who would stand to make money by proving that the CD you already have is just fine, so you needn't buy the high-res version, or that you can save money by buying the regular version, so nobody with any incentive to sponsor the tests.
  
 And, even though you might assume that audio magazines and other authorities make their living by providing useful information, the reality is that the ongoing controversy provides for a lot more entertaining editorial copy than would a short summary of the results of a test that answered the question once and for all.  
 Again, with DACs, there are enough reasons why various DACs sound different that the question would require some thorough testing to fully answer it. And, while many companies sell product partly based on claims of superior jitter performance, for which they charge a large premium, there are far fewer who profit by the opposite.... and, also again, companies who sell low-cost DACs with poor performance probably sell mostly on low cost and volume, and so wouldn't sell enough extra product as the result of doing tests to prove that nobody can tell the difference to recoup the costs of doing the tests.
  
 You also need to remember that a lot of the audiophile market is built on superlatives. Just as not everyone who owns a Ferrari has gotten it up past 120 mph, but they're still happy that they own "a performance car", many audiophiles would be quite willing to pay extra for a product with better numbers, even if they weren't certain they would hear the difference - just to "have the best". (Personally, I would cheerfully pay for a higher-res copy of an album I liked, just to have "the best copy available", even if I didn't notice any difference. I've also occasionally paid extra for a certain piece of audio equipment, not because of its actual audio performance, but because it had a nicer cabinet or knobs. However, I'd certainly like to have a clear picture of what I'm spending that extra money for - and I do believe everyone should have the information necessary to make their own informed decision without being misled.)
  
 Quote:


jcx said:


> relative advance from 2007 "good quality" is possible to see in numbers that we expect are far beyond human hearing thresholds
> 
> ESS Sabre, some at the margin new releases in AKM top line, Schiit taming the AD5791 would seem to be pretty much in the monolithic audio DAC offerings since
> 
> ...


----------



## KeithEmo

I agree with most of what you said....
  
 I'm also not uncomfortable when someone says: We tested this with a bunch of people, and a bunch of different equipment; none of them could hear a difference, so you probably won't either. I'm also perfectly happy when someone says that quite a few tests have been run, and so far they all seem to show that very few if any people can hear the difference between high-res files and regular CD quality files a significant percentage of the time.
  
 My problem with this discussion is that it's not _ALL_ "gullible people buying high-res files because the store says they're better" - or, at least, nobody's proven that it is. Most of the online stores that sell high-res files do in fact offer free samples (or, at worst, you'll get to hear for yourself by buying one album). Now, while I agree wholeheartedly that we humans are very suggestible, and so it is in fact possible that the people who think they hear a difference are all imagining it, it's also possible that the people who think they _DON'T_ hear a difference may be the ones with the more vivid imagination. I'm simply pointing out that I don't think the science that's been done so far reaches the level of proving _CONCLUSIVELY_ which is the case.... and I'm quite convinced that the folks who think it has are either misinterpreting the facts (or perhaps they don't fully understand the science of testing and test analysis).
  
 In other words, I would always advise someone to see for themselves if they can hear a difference, and certainly not to spend extra money for a high-res version of something simply because someone told them it sounds better. However, I would also advise them not to assume that the high-res version can't possibly sound better "because science has proven that it doesn't". (I believe that both of those would be setting unfair expectations based on incomplete information.)
  
 (As I've mentioned in other posts, it's been my personal experience that some high-res versions of albums do sound clearly better than their CD quality counterparts - but I simply can't be sure if it's because the higher sample rate really matters, or because the mastering is better, or possibly a combination of both... so, at this point, it seems safer to judge each album release on its own merits. )
  
 Quote:


castleofargh said:


> both sides have to try and prove their points! else it's just random people making empty claims. and nobody cares about those guys crying wolf and talking about the end of the world in the subway. proof is what makes it all meaningful.
> except that one cannot prove that he doesn't hear a difference. even if we add in the test some control samples that should be heard to try and trick cheaters, we can never know for sure that the subject will not notice those control sounds, play along and then lie for the rest. we would need to go out of our ways to make something that can never totally become proof. only strong inclinations to believe when the number of people getting the same result increases massively.
> on the other hand, if 1 guy can hear a difference, all he has to do is get statistical significance in any blind test with any gear to absolutely prove he heard a difference. we can later argue the reason for the difference, but not the claim.
> 
> ...


----------



## L8MDL

Personally, I don't need or care about tests. I buy the highest resolution file available for the music I play. I have not heard ANYONE say the high res versions are WORSE - given the same mastering, of course. In the world of quality, price is no object! I do, however, use cheap interconnects...


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## KeithEmo

I have to agree - and disagree - with you on that one.
  
 One of the main reasons that 16/44k was chosen as the format for CDs was the limited amount of data storage available on a CD. In the engineering world, whether the CD format has "just enough" bandwidth or not is irrelevant - having an extra few hundred percent of safety margin would be nice anyway. Now that downloads are replacing plastic discs, and both storage space and download bandwidth are getting so cheap, one might argue that there's simply no reason (other than artificially inflated sale prices) not to use a higher file resolution. At current prices for a USB hard drive, an album at CD quality uses somewhere between one and two cents worth of storage space; going up to 24/192 raises that to about a dime; neither of which is significant compared to the cost of the album when I buy it.
  
 So, would I have paid an extra $5000 to get a 400 HP turbocharged V8 engine in my Nissan Versa? Definitely not.
  
 But, if someone offered me the upgrade for $50, and it also got better mileage....? Sure, why not.
  
 Therefore, I do believe that high-res files are "the future" - simply because, being technically superior, whether the difference can be heard or not, is still "philosophically" a good thing. To me, the only real question is whether the difference justifies _PAYING_ more. And, to be honest there, I suspect that the market will "shake out" in the next few years, and, by 2025, we'll all be buying super-res 32 bit 768k files for whatever equates to the current price of a CD in 2025 dollars. (Or, perhaps, by then, we'll all be subscribed to streaming services, and won't have any files at all.)
  
  
 Quote:


castleofargh said:


> ...........
> 
> now where I join you, it's about gears. as a personal test, it only matters to me what I can hear with my own equipment, but as proof of an audible difference, it could make sens to use gears that indeed have the best fidelity possible. although I'm not sure any transducer is up for the task. in fact it's a huge point in my reason to doubt the purpose of highres in general. we're trying to improve everything that is already great and pretend to test it with transducers magnitudes below in fidelity. it feels like insisting on giving a V8 engine to a car with square tires.


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## RRod

keithemo said:


> Therefore, I do believe that high-res files are "the future" - simply because, being technically superior, whether the difference can be heard or not, is still "philosophically" a good thing. To me, the only real question is whether the difference justifies PAYING more. And, to be honest there, I suspect that the market will "shake out" in the next few years, and, by 2025, we'll all be buying super-res 32 bit 768k files for whatever equates to the current price of a CD in 2025 dollars. (Or, perhaps, by then, we'll all be subscribed to streaming services, and won't have any files at all.)


 
  
 They really shouldn't be the future, because bits and samples aren't what's wrong with audio today. A return to reasonable mastering and a societal move back towards considering music (and performing arts other than movie acting) important would do more for sound than anything hi-res can offer.


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## sonitus mirus

l8mdl said:


> Personally, I don't need or care about tests. I buy the highest resolution file available for the music I play. I have not heard ANYONE say the high res versions are WORSE - given the same mastering, of course. In the world of quality, price is no object! I do, however, use cheap interconnects...


 
  
 High resolution files could sound worse.
  


> https://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> 192kHz digital music files offer no benefits. They're not quite neutral either; practical fidelity is slightly worse. The ultrasonics are a liability during playback.
> 
> Neither audio transducers nor power amplifiers are free of distortion, and distortion tends to increase rapidly at the lowest and highest frequencies. If the same transducer reproduces ultrasonics along with audible content, any nonlinearity will shift some of the ultrasonic content down into the audible range as an uncontrolled spray of intermodulation distortion products covering the entire audible spectrum. Nonlinearity in a power amplifier will produce the same effect. The effect is very slight, but listening tests have confirmed that both effects can be audible.


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## L8MDL

The Xiph article you quote is from 2012 and has been surpassed by current technology. Any article showing digital music using a "stair-step" image should be questioned. New equipment and transfer techniques have totally eliminated any concerns about high frequency feedback and other problems.


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## RRod

l8mdl said:


> The Xiph article you quote is from 2012 and has been surpassed by current technology. Any article showing digital music using a "stair-step" image should be questioned. New equipment and transfer techniques have totally eliminated any concerns about high frequency feedback and other problems.


 
  
 One point of the article is to show how the stairstep interpretation is wrong…


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## castleofargh

l8mdl said:


> The Xiph article you quote is from 2012 and has been surpassed by current technology. Any article showing digital music using a "stair-step" image should be questioned. New equipment and transfer techniques have totally eliminated any concerns about high frequency feedback and other problems.


 

 stair steps are only mentioned/showed in the article to say how it is a false representation ^_^. maybe read before you judge?


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## L8MDL

Touché! I had read the article long ago and just skimmed it over. The rest of my comment stands, however. Certainly there will be bad hi-res files but they are fast becoming the rare exceptions rather than the rule.


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## arnyk

keithemo said:


> Personally, I would very much like to see a "public challenge", where a content provider, a DAC vendor, a speaker vendor, and an amplifier vendor, would get together and try to provide a sample of high-resolution music that was "so good that it couldn't be reproduced on a CD without audibly obvious degradation". Several makers of DACs and studio ADCs could then attempt to disprove the claim by showing that, when their equipment was inserted into the signal chain, none of the audience could hear a difference.


 
  
It is my contention that the demo above will never happen because producing it requires way too much work and too much altruism. I won't say that in general the people who promote the overpriced gear listed know for sure that they would fail, but I do think that their doubts have probably greatly inhibited their enthusiasm and energy for doing such a thing. 
  
Fact is that people have tried to do DBTs illutsrating that in accordance with their vision of science, and the results have fallen well short of the sort of technical success that a good commercial venture requires. Here is a recent example and the debate that it stimulated: The Meridian typical DAC boondoggle


----------



## arnyk

l8mdl said:


> The Xiph article you quote is from 2012 and has been surpassed by current technology. Any article showing digital music using a "stair-step" image should be questioned. New equipment and transfer techniques have totally eliminated any concerns about high frequency feedback and other problems.


 
  
 Interesting  attempt to invalidate what many find to still be good evidence and still a good read. BTW, it can be found at https://xiph.org/~xiphmont/demo/neil-young.html
  
 The stair step content of the article (under "Sampling fallacies and misconceptions)" is obviously there to show the fallacy of the argument which is still probably being made on some forum, some place. Therefore, discrediting the Xiph article because it mentions this fallacy would be an example of leaping to a conclusion, and is itself invalid.
  
 The article does not mention high frequency feedback at all.  It does mention intermodulation distortion which is something completely different. Thus the reader has provided evidence that the articles biggest fault may be that it surpasses his understanding of the relevant technology.
  
 Also, the intermodulation distortion that was mentioned in the article was mentioned in the context of playback and monitoring equipment, which obviously cannot be addressed by "New equipment and transfer techniques" as they are part of the production process, and mass media producers have no control over the listener's playback equipment choice.


----------



## arnyk

keithemo said:


> (My personal quibble with their test is that the performance of DACs varies considerably, and has progressed since 2007, so I'm not convinced that the disc players they used as sources were "nominally good enough" to be ruled out as a limiting factor. Perhaps, if the content they used were played back on a higher-quality DAC, there would have been details present which would have been audibly lost later in the signal chain. Likewise, while the Quads and Snells are what I would consider to be "very good speakers", I can't rule out the possibility that some of the many other speakers out there might do a better job of making some difference audible. I'm also inclined to feel that, when listening for subtle details, headphones do a better job of revealing differences than loudspeakers - yet they failed to include any headphone listening. This seems like a significant omission - and one that would have been easy to remedy. It occurs to me that they - quite reasonably - were only trying to prove the "typical case" - which they did pretty well.)


 
  
 DACs used in consumer digital player have been tested with DBTs and shown to be sonically transparent ever since the second generation of CD players in the mid-1980s:
  
  "Masters, Ian G. and Clark, D. L., "Do All CD Players Sound the Same?", Stereo Review, pp.50-57 (January 1986)
  
Thus the argument that DAC technical progress since 2007 modifies the relevance of this article: "Audibility of a CD-Standard A/D/A Loop Inserted into *High*-*Resolution Audio* Playback". E. Brad Meyer and David R. Moran. *JAES* 55(9) September 2007 is false, and was probably made based on an incomplete understanding of the performance of modern digital audio gear in general.


----------



## jcx

the comments section discussion of ABX in your link earlier was odd
  
 while not having explored the experimental design literature to that level of detail, in my own use of foorbar2000 ABX plugin it seemed natural to switch A/B to try to learn to discriminate, then A/X, B/X trying to decide same/different, often returning to A/B over several cycles when the difference wasn't obvious
  
  
 another fun point is how audiophile gurus disagree - Schiit's "megaburrito filter" from the few hints given uses a narrower transition band, contrary to many other's recommendations - but we are assured that it is the latest, greatest


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## arnyk

jcx said:


> the comments section discussion of ABX in your link earlier was odd
> 
> while not having explored the experimental design literature to that level of detail, in my own use of foorbar2000 ABX plugin it seemed natural to switch A/B to try to learn to discriminate, then A/X, B/X trying to decide same/different, often returning to A/B over several cycles when the difference wasn't obvious
> 
> ...


 
  
 Without some more complete references, I can't tell what the above means, or even be sure that it was directed to me.


----------



## jcx

sorry should have been more specific:
 Quote:


arnyk said:


> ...Fact is that people have tried to do DBTs illutsrating that in accordance with their vision of science, and the results have fallen well short of the sort of technical success that a good commercial venture requires. Here is a recent example and the debate that it stimulated: The Meridian typical DAC boondoggle


 
 reading the comments to the JAES article


----------



## KeithEmo

rrod said:


> They really shouldn't be the future, because bits and samples aren't what's wrong with audio today. A return to reasonable mastering and a societal move back towards considering music (and performing arts other than movie acting) important would do more for sound than anything hi-res can offer.


 
  
 I agree with you, but there are several "practical" reasons why it won't happen that way:
  
 1) High-res files have a lot of "popular acceptance" - meaning that a lot of people are in fact willing to buy one more remaster of an album they already have if it happens to be in high-res.
  
 2) At the very minimum, it will reach a point where stores will be proudly proclaiming that "we'll give you the high-res version for the same price as our competitor charges for an ordinary CD quality file"; and consumers always jump at the idea that they're getting something better for the same price - or even just a little bit more. (It's just another version of "new and improved".)
  
 3) If it's from an old master, then re-mastering an album actually requires significant effort and cost; re-converting the same analog master tape at a higher bit rate is much easier. And, if the master is digital, then it was almost certainly done at a higher bit rate to begin with, so converting it to a higher rate for delivery, or even simply not converting it and selling copies of the master directly, is also trivial.
  
 4) Many people maintain that the main target audience for music these days actually _LIKES_ music that's overcompressed and sounds poorly mastered - because it sounds better on cheap $10 ear buds and car radios. Because of this, it's possible that we've lost all hope of "the regular version" of anything sounding good.... and, if so, the best we can hope for is that they'll sell us a separate (and more expensive) "audiophile copy" that sounds better.... and, if they do that, then you can bet that it will ALSO be high-res, because that counts as a selling point.
  
 If you look at the high-res albums sold on any popular store - like HDTracks - you'll find that some of them are in fact totally remastered, and many of those sound very good. (I think the Grateful Dead Studio Abum set sounds very good - and quite different from the original. There's also a long description of all the things that were done to "restore, repair, and remaster" the original mix tapes.) What I would be interested to see would be a statistic showing whether albums that are in fact "seriously remastered", and so sound quite different, actually _SELL_ proportionally more copies than ones where the high-res version is indistinguishable from the original. That would give a rough estimate of how many people actually buy the high-res version of an album they already have - because they at least expect an improvement; and how many are simply buying a new album, and choose the high-res version over the regular version because it only costs a little more, and it's "the premium version", but they're not specifically looking for an upgrade from a copy they already have. (It's a well-known marketing fact, for everything from dish detergent to sports cars, that, if you sell a "regular" and an "extra strength" version of a product, or a "regular" and an "XL" model, many people will choose the more expensive "top" or "middle" version, even if they have no specific reason to think they need it or it will work better for them..... just because it seems like it "must" be better than the "regular" version somehow. Therefore, it always makes sense to offer a "basic" version, and a "premium" version.)


----------



## KeithEmo

Under what conditions?
  
 That test showed that several source samples, played on certain disc players, through certain amplifiers and speakers, weren't audibly changed "when they were passed through an additional CD quality audio loop". The problem there is that you can't generalize those results to "everything, everywhere, for everyone". Perhaps there is some difference that's obvious on 10% of the speakers in the world, and totally inaudible on the other 90%, and none of the speakers they chose happen to fall in that critical 10%. Perhaps there's a difference that's easily audible, but only on an acoustic recording of some certain instrument, which wasn't included in their test sample. Or perhaps there's some specific sound that can occur naturally in music, and that some DACs can faithfully reproduce, but that the DACs in the specific disc player they chose to use cannot - in which, if it's already missing or altered by their player, you aren't likely to notice further alteration further down the signal chain. 
  
 My point is simply that testing a few dozen samples, on three or four players, with fifty or a hundred test subjects, isn't sufficient information to make a generalization. I also simply can't agree with you that "every DAC made since the 1980's is audibly transparent" - because I own quite a few DACs, and several of them sound significantly different than others. In fact, a few of them have multiple filter settings, and even those sound slightly different.
  
 I personally suspect that many of the claims made for "things being inaudible" may be over-generalizations.
  
 For example, I've heard, over and over, that "THD less than 0.5% is inaudible". However, one day I created a test tone by starting with a steady 50 Hz tone, and adding to it a 2 kHz tone (that's the 40th harmonic of 50 hz) switching on and off at quarter-second intervals. The level of the 2 kHz "beep" was reduced enough that it was 0.1% of the amplitude of the 50 Hz primary tone. Guess what..... In that specific case, the 0.1% THD was CLEARLY audible as a "beep beep beep" sound ...... so I guess 0.1% THD isn't ALWAYS inaudible.
  
 (The statement that "if the THD of an amplifier is below about .1% it won't be audible" is probably a fair generalization if you limit your "world" to analog linear amplifiers, where it is most unlikely for an amplifier to produce random high-order harmonics without producing much higher levels of low-order harmonics. But it may not be equally valid with digital amplifiers, where some sort of "processing error" might in fact produce just the situation I created with my test file.)
  
 As for the test we're discussing.... We KNOW that there will be measurable differences between SACD files and CD quality files (the SACD files will have a wider frequency response, with a noise floor that rises sharply at ultrasonic frequencies, and a different noise spectrum). So, to be reasonable, if you want to test whether those differences are audible, the first thing you must do is to confirm that your test equipment is capable of presenting them to the test subjects to be heard (or not). And, to put it bluntly, saying that "you used high-end consumer equipment with good specs" doesn't rise to the level of "validating your test equipment". You need to show both that the entire signal chain you used was able to reproduce those differences; and you _ALSO_ need to show that the sample content you used contains them. (If you want to see how well test subjects can distinguish shades of red, then you need to use video equipment that can do so for the tests, and you need to use test patterns that contain the proper test signal.)
  
  
 Quote:


arnyk said:


> DACs used in consumer digital player have been tested with DBTs and shown to be sonically transparent ever since the second generation of CD players in the mid-1980s:
> 
> "Masters, Ian G. and Clark, D. L., "Do All CD Players Sound the Same?", Stereo Review, pp.50-57 (January 1986)
> 
> Thus the argument that DAC technical progress since 2007 modifies the relevance of this article: "Audibility of a CD-Standard A/D/A Loop Inserted into *High*-*Resolution Audio* Playback". E. Brad Meyer and David R. Moran. *JAES* 55(9) September 2007 is false, and was probably made based on an incomplete understanding of the performance of modern digital audio gear in general.


----------



## KeithEmo

I've designed many test protocols for situations other than audio, and, especially if you're trying to determine the smallest difference or value the subject can distinguish (maximum sensitivity), it's almost always best to make the test as simple as possible - and to avoid all extraneous activity or thought. For example, if you want to test how well your subjects can distinguish differences in colors, you make colored tiles of different colors, hold them up in pairs next to each other, and simply ask the subject "Do they look like the same color?" (You don't hold up three tiles and ask them which ones look more like which other ones; that would be testing a more complex "function".)
  
 With this in mind, if you're simply testing whether something is "audibly different or not", then any form of "full ABX testing" is needlessly complicated.
  
 Here's how I would do the test.....
  
 First, again to simplify, let's simply refer to our signals as "Reference" and "X".
 The subject will have a simple way of selecting either the Reference signal or the X signal to listen to.
 (It could be a toggle switch, labelled "Reference" and "X", or a pushbutton that toggles between the choices and an indicator light showing which is currently selected.)
  
 The test run will consist of a series of individual tests.
 For each test, the test set will be configured so that either X is a copy of the Reference signal, or X is the modified signal it's being compared to.
 The test subject will then be allowed to play the test sample, switching between the Reference and X signals as quickly as they like, and as often as they like.
 When they have decided whether they think that the X signal is or is not the same as the Reference signal they will report their choice.
 (If they aren't sure they should be asked to guess. I'm pretty sure that most subjects will find guessing "yes or no" to be less stressful, and to require less thought, than guessing "which something is most like" if they're uncertain. In order to compare the results we get to those expected by simple guessing, we do require each test subject to complete all tests, and answer all of them, so we want to make that as easy as possible for the subject to do.)
  
 By doing it this way we have minimized the requirement for any sort of memory, or for any "cognitive load" associated with deciding upon matches.
 We have simply made the question "Did the sound change when you flipped the switch or not?"
  
 Note that there SHOULD be a very slight audible tick or pause each time the switch is actuated (this will cover up any slight differences between switching between copies of the same sample, and switching between different samples, which might serve as conscious or unconscious cues as to which is occurring).
  
 Obviously, if the signals are really audibly identical, then we would expect results consistent with the subject guessing.
 And, if they statistically do better than we would expect from guessing, then that suggests that there are in fact audible differences.
  
 Note that this test very specifically determines whether there is ANY audible difference between a Reference signal and a test signal.
 It does NOT determine what the difference is, or which signal is better, and does NOT require the test subject to quantify what the difference is.
 (This avoids the possibility that the test subject will think they hear a difference, but be "uncomfortable" reporting a difference that they can't quantify.)
 (Also, interestingly, it also "covers" situations where the subject may not be consciously aware of the difference, but it may still bias their choice.)
  
 Also note that we STILL need to perform the test with a lot of subjects and a variety of equipment. 
 If the test shows that audible differences DO exist, then we have shown both that differences exist AND that our test equipment is able to demonstrate those differences.
 However, if the test shows no audible difference, we still can't know for sure if the null result is due to limitations in our equipment, test samples, or even our test population.
 Therefore, if we get a null result, the test should be repeated many times with different equipment and conditions to rule out that possibility.
 (This could be done in the form of a "challenge", with some sort of prize offered as incentive for vendors or individuals to try it with their own chosen equipment and test samples.)
  
 Also note that this protocol could be implemented with VERY primitive (and even passive) equipment.
 As long as the levels are matched, it doesn't even require computer control or relays.
 Whether the test signal for each test is the Reference signal or X could be set using a manual toggle switch.
 (A simple computer program could print out a random list of the necessary settings for each individual test.)
 (The results should be reasonably valid as long as the test subject can't see the position of the configuration switch.
  However, an automated system would be better, because it would rule out unconscious information leakage from the operator to the test subject.)
  
  
 Quote:


arnyk said:


> Without some more complete references, I can't tell what the above means, or even be sure that it was directed to me.


 
  


jcx said:


> the comments section discussion of ABX in your link earlier was odd
> 
> while not having explored the experimental design literature to that level of detail, in my own use of foorbar2000 ABX plugin it seemed natural to switch A/B to try to learn to discriminate, then A/X, B/X trying to decide same/different, often returning to A/B over several cycles when the difference wasn't obvious
> 
> ...


----------



## arnyk

keithemo said:


> Under what conditions? That test showed that several source samples, played on certain disc players, through certain amplifiers and speakers, weren't audibly changed "when they were passed through an additional CD quality audio loop". The problem there is that you can't generalize those results to "everything, everywhere, for everyone". Perhaps there is some difference that's obvious on 10% of the speakers in the world, and totally inaudible on the other 90%, and none of the speakers they chose happen to fall in that critical 10%. Perhaps there's a difference that's easily audible, but only on an acoustic recording of some certain instrument, which wasn't included in their test sample. Or perhaps there's some specific sound that can occur naturally in music, and that some DACs can faithfully reproduce, but that the DACs in the specific disc player they chose to use cannot - in which, if it's already missing or altered by their player, you aren't likely to notice further alteration further down the signal chain.


 
  
 Under what conditions? Under every condition that we had the resources to test at the time of the review (1986) to test.  There was no cherry picking of people, equipment or media except for maximum sensitivity to small differences, and players that might be expected to sound different or bad. The test conditions were the most sensitive to small differences that were known to be available at the time. This was rather obviously the same goals  that were followed 19 years later by Meyer and Moran.
  
 Let's compare that to the collected writings of hundreds and thousands of audiophiles, including the esteemed KiethEmo. BTW, is KeithEmO short for Kieth@Emotiva (http://emotivalounge.proboards.com/board/57/keiths-corner) ?  Bias, anybody? 
  
 In their evaluations, there appear to be few if any digital music players that sound the same.  If almost all digital music players sound different then finding two players that sound different should be very easy. There should be no reason to test a lot of different players or a lot of listeners because just about every pair we would pick will sound different to just about everybody, if we believe that all those audiophile evaluations were valid.
  
 The debating trick that is being employed is an old one - change the question so as to make things as unfairly difficult as possible for your opponent. You have a track record of claiming without actual proof or evidence except your say so that every CD player can be reasonably be expected to sound different to every reasonable audiophile, while demanding that I actually test every one or a statistically significant sample of them, live and in person.
  
 If your assertions are correct, then I should be able to do very little testing, and if I find no audible differences demonstrate that your hypergenerality is seriously flawed and make my point. This has been done many times in private and no differences were found. It was felt that the well publicized tests in 1986 and 2007 that showed the same results should suffice.


----------



## arnyk

keithemo said:


> I've designed many test protocols for situations other than audio, and, especially if you're trying to determine the smallest difference or value the subject can distinguish (maximum sensitivity), it's almost always best to make the test as simple as possible - and to avoid all extraneous activity or thought. For example, if you want to test how well your subjects can distinguish differences in colors, you make colored tiles of different colors, hold them up in pairs next to each other, and simply ask the subject "Do they look like the same color?" (You don't hold up three tiles and ask them which ones look more like which other ones; that would be testing a more complex "function".) With this in mind, if you're simply testing whether something is "audibly different or not", then any form of "full ABX testing" is needlessly complicated.


 
  
 Unlike you Kieth my designing of test protocols which is very extensive, includes audio testing. Furthermore in the process of developing audio test protocols I devised ABX which is the surely the most discussed testing procedure in the history of audio.
  
 The idea that if you're trying to determine the smallest difference or value the subject can distinguish (maximum sensitivity), it's almost always best to make the test as simple as possible - and to avoid all extraneous activity or thought, is exactly where we started over 30 years ago. 
  
 Only this just wasn't just me, it was a team that was composed of a number of engineers and scientists, BS's and PhD's, including people who were professionally engaged in scientific research. 
  
 Our work was reviewed and enthusiastically approved by the well known research team from the nearby University of Waterloo, Vanderkooy and Lip****z of AES fame. They saw our ABX box and ran right out and built their own. The rest is history, whether you know it or not. 
  
 As you suggest we started with the classic same/different test and then we actually did what it took to make it effective for audio. This was no 2015 retroactive thought experiment simplified for a forum post, this was the real thing done with real high level techies, real equipment UUTs, and real audio systems (many more than just one of each!).  
  
 ABX was the simplest solution that worked with convincing levels of sensitivity.


----------



## KeithEmo

In order.....
  
 1) They apparently chose three disc players for the test, then almost immediately disqualified one because it made an obviously odd noise at one point, and settled on one of the other two. The speakers they chose were two models that were accepted at the time by most people to be "high end audiophile models". I didn't read about any testing done to determine that they were in fact "the most sensitive equipment to small changes available at the time", nor have I seen any data to suggest either why they should "expect that the equipment shouldn't sound bad or different", or to back up that expectation with actual test data. They didn't test "every high-end disc player available" nor "every audiophile speaker". In short, I see a massive collection of assumptions, and an equally large collection of unknowns. If their goal was to prove that the difference was inaudible with "some typical audiophile equipment of the time", then they achieved their goal; but I don't know if there might have been an obvious difference if they'd used some other disc player, or a different speaker or amplifier, and neither do you. 
  
 2) Yes, that's me... (my name on the Emo forums is actually KeithL) 
	

	
	
		
		

		
			





 and it's hardly a secret.... However, since all of our current DACs play both high-res and standard files just fine, I don't see any particular reason why you would think I'd be biased one way or the other on that one. We don't sell high-res music, and, if you read my posts, you'll find that I'm the last person to suggest that anyone should buy a high-res album release for any reason other than because that particular release happens to sound better - for whatever reason.
   3) As for generalities, the fact is that it's very difficult to make valid generalities - because there are so many variables. This is why most actual scientists avoid generalities unless they have a truly massive amount of consistent evidence to back them up. So, if the purpose of that test was to demonstrate that, with typical "audiophile quality equipment", most people couldn't reliably detect a difference between high-res files and CD quality ones, then I have no problem with that... and I agree that it produced results that tend to support that claim. I also have no problem if you want to state that nobody has proven conclusively that there is an audible difference with modern recordings and equipment. (However, neither those statements, nor the results of any test I've seen, proves conclusively and universally that so such difference, audible to any human, using any currently available equipment, exists.... and I can't even how you could frame a test that would show the same for any possible equipment that might go on sale next year.)
  
 4) I never claimed that "every CD player should sound different to every audiophile"; in fact, I've generally avoided making any general claims, because I usually don't have enough data to do so - and I hate to be proven wrong. All I said was that neither you nor anybody else has proven that "all disc players sound audibly identical", or even that most of them do. (I'm even inclined to agree that I personally believe that many of the claims of what people think they hear are based on expectation bias... but that's far from declaring that every single claim that disagrees with what I believe to be the truth "must be false". Even if 90% of them turn out to be wrong, that in no way "proves" that the other 10% aren't right.) You might as well generalize on "how well people can bowl" based on a test of all the members of the Franklin Bowling League, or claim that "humans never live past 110" because nobody you know has a relative that has done so.... which might be very surprising to the occupants of that small town in Russia where people routinely make it past 120.
  
 Would the people who took that test in 1986 have heard obvious differences if they'd used Koss electrostatic headphones, or a different brand of speakers, or a different amplifier, or a different DAC? I don't know... and neither do you... because they didn't test those combinations. And, considering the relatively tiny test population _of equipment_ they used I don't see anywhere near enough information to support reliable broad generalizations.
  
 Yes, resources are almost always a limiting factor, but that just may mean that the limitation prevents you from collecting enough data to make a viable generalization.
  
 And, to be very specific, if they'd first auditioned the top model disc player from the top 20 manufacturers at the time, using a dozen different amplifiers, and the top speaker model from the top twenty speaker vendors, and none of 500 test subjects could detect any difference between any combination of them with their high-res sample files, _THEN_ I would be willing to consider it as a provisional "given" that "all disc players sound the same" and that it was reasonably safe to generalize the results obtained with one of them to the rest. Otherwise, while the test is certainly "interesting", and can reasonably be claimed to support their claims rather than to contradict them, it is hardly "conclusive".... sorry.
  
  
  
 Quote:


arnyk said:


> Under what conditions? Under every condition that we had the resources to test at the time of the review (1986) to test.  There was no cherry picking of people, equipment or media except for maximum sensitivity to small differences, and players that might be expected to sound different or bad. The test conditions were the most sensitive to small differences that were known to be available at the time. This was rather obviously the same goals  that were followed 19 years later by Meyer and Moran.
> 
> Let's compare that to the collected writings of hundreds and thousands of audiophiles, including the esteemed KiethEmo. BTW, is KeithEmO short for Kieth@Emotiva (http://emotivalounge.proboards.com/board/57/keiths-corner) ?  Bias, anybody?
> 
> ...


----------



## old tech

I did an interesting experiment the other day. While i normally do not get into itunes, i heard a lot of good stories about the mastered for itunes albums. I also noticed that these often appear to be the same mastering as the hi res HDTracks releases (often released on itunes in the following week) but in lossy aac and at a fraction of the cost. Anyway a couple of mates and i did some abx tests of the remastered led zep iii comparing the hi res version with the mastered for iitunes version. None of us got better than 52%. Make that of what you will...


----------



## JWolf

keanex said:


> http://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> Read em up boys.




RUBBISH! Use your ears.


----------



## JWolf

bigshot said:


> The limitation is in human ears. It doesn't matter how high a frequency you want your stereo to produce and how wide a dynamic range, it all comes down to whether human ears can hear it.
> 
> Audiophools love to spend lots of money pushing the decimal point further and further to the left and making the frequencies go higher and higher, but at a certain point, it all becomes moot because only bats can hear it.




It's not about needing more dynamic range or a lower noise floor. It's about capturing the audio. 24/96 captures more of the audio than 16/44.1. Even 24/44.1 captures more of the audio. 

With 96 vs 44.1 you don't have to worry about missing samples at the edges of 44.1. You also do capture more of te sound with 24/96 than you do with 16/44.1. When 24/96 is overkill is on these FM radio mastered CDs. Those are the CDs where the DR is compressed with the volume pushed to the point of distorting. With old recordings where the frequency range does not hit 20khz and might only hit 17khz, 24-bit it useful to get more resolution than 16-bit. 

The way I see it, science can say that technically 16/44.1 is enough to capture all the sound. But it's not when you properly listen to a high-res recording.


----------



## OddE

jwolf said:


> It's not about needing more dynamic range or a lower noise floor. It's about capturing the audio. 24/96 captures more of the audio than 16/44.1. Even 24/44.1 captures more of the audio.





> The way I see it, science can say that technically 16/44.1 is enough to capture all the sound. But it's not when you properly listen to a high-res recording.


 
  
 -The only thing 24/44.1 captures more of than 16/44.1 is dynamic range. You do not, repeat not, get finer resolution or anything of the sort. Just plain higher dynamic range.
  
 Also, if the difference between redbook and hi-res is so glaringly obvious, how come the difference seems to disappear whenever someone tries an actual blind test?


----------



## RRod

jwolf said:


> It's not about needing more dynamic range or a lower noise floor. It's about capturing the audio. 24/96 captures more of the audio than 16/44.1. Even 24/44.1 captures more of the audio.
> 
> With 96 vs 44.1 you don't have to worry about missing samples at the edges of 44.1. You also do capture more of te sound with 24/96 than you do with 16/44.1. When 24/96 is overkill is on these FM radio mastered CDs. Those are the CDs where the DR is compressed with the volume pushed to the point of distorting. With old recordings where the frequency range does not hit 20khz and might only hit 17khz, 24-bit it useful to get more resolution than 16-bit.
> 
> The way I see it, science can say that technically 16/44.1 is enough to capture all the sound. But it's not when you properly listen to a high-res recording.


 
  
 What are these "missing samples at the edges of 44.1"?
  
 The view of 24-bits as capturing more resolution is in fact entirely equivalent to its capturing more dynamic range. Any map of 16-bit values to 24-bit by simple integer multiplication will preserve relative volumes. If we use the mapping f(x) = x, then we end up with the viewpoint of 24 bits as having a lots of extra values above the max of 16 bit but with the same step size, allowing for my dynamic range. If we use the mapping f(x) = 256*x (the common way to convert to 24 bits from 16), then we end up with the viewpoint of 24-bits as having finer gradations between steps. The only difference between the two viewpoints (in a perfect world) is where you'd set your volume, or more accurately, where the robot is setting the volume so you can be far the hell away from this experiment.


----------



## KeithEmo

old tech said:


> I did an interesting experiment the other day. While i normally do not get into itunes, i heard a lot of good stories about the mastered for itunes albums. I also noticed that these often appear to be the same mastering as the hi res HDTracks releases (often released on itunes in the following week) but in lossy aac and at a fraction of the cost. Anyway a couple of mates and i did some abx tests of the remastered led zep iii comparing the hi res version with the mastered for iitunes version. None of us got better than 52%. Make that of what you will...


 
  
 For quite some time Apple's official guide to "mastering for iTunes" has recommended mastering at 24/96 if at all possible, as well as including several recommendations that all work out to "master at the best quality you can".
  
 I would also agree with you that, although I didn't do any careful ABX testing, I have the full set of recent Led Zeppelin remasters in high-res, and, in casual listening, I didn't notice any particular improvement with them over the previous versions either. This all highlights an important distinction that some people in this discussion seem to forget when they get excited..... and that is that the following three questions are very different:
  
 1a) Is the "technical superiority" of high-res tracks _EVER_ really audible, under _ANY_ circumstances, with _ANY_ source material, to _ANY_ person, using _ANY_ playback equipment?
  
 1b) Will _MOST_ people, under _MOST_ circumstances, and with _MOST_ source material notice any difference?
  
 1c) Is there a difference that justifies the higher cost?
  
 The first question is a matter of scientific interest, while the second and the third are more "practical and commercial questions".
  
 This same distinction exists with another popular subject today - jitter. Personally, I listened to some "jitter test samples" (which included quite high levels of "simulated jitter"), made from a popular Norah Jones album, and I couldn't detect even relatively high levels of jitter with that album; however, I can hear relatively low levels with different source material; (and, interestingly enough, some people who apparently listen for different details than I do, claim to be able to hear much lower levels than I could with the Norah Jones album.) I personally find more or less pure vocals to be relatively insensitive to all of the various differences between DACs and file resolutions, but to notice even minor differences when I listen to well recorded wire brush cymbals (to me, when everything is good, cymbals actually sound like metal wires on metal, but, when any of several things are "off", they sound more like a burst of steam from a valve - the "metallic sound" seems to get lost).
  
 I suspect that, just like a turntable with a slight speed error would drive a person with perfect pitch nuts, while I wouldn't notice it at all, different people focus on different aspects of the sound, and so something that seems like a glaring issue to one person may not be at all important to another, or may not be "audible" at all. We also need to keep in mind that everything we hear is "filtered" through our brains", so we may literally find something "not audible" if we simply don't happen to pay attention to that particular aspect of the sound - and this could be why, at least sometimes, "trained listeners" hear things that "ordinary folks" don't.
  
 Unfortunately, with music files, there are two conflicting questions/decisions....
  
 On one side, you would hate to pay extra for files that sound exactly the same to you, on your equipment (regardless of whether anyone else hears a difference or not).
  
 And, on the other side, you would hate to buy a new set of headphones someday, or simply "become a more careful listener", and discover that the huge music collection you've bought has to be bought all over again because, with your new equipment, you now notice a shortcoming. (And a lot of people did have that happen when they discovered that their 128k AAC iTunes files really didn't sound as good to them as a higher-resolution version.)
  
 (And, if it were possible to prove, once and for all, that absolutely no difference existed, or that a significant difference existed which could be confirmed, then that decision would be a lot easier.)
 Unfortunately, and much as many people would like it to be so, I simply don't think there is enough evidence yet to say one way or the other in an absolute and general sense.


----------



## KeithEmo

You're quite right... the "only difference" between 16/44k and 24/44k is that the individual samples can have finer gradations at 24 bits than at 16 bits. However, the "fineness of the gradations" is another way of saying that the values are stored more accurately in the 24 bit file, and the inaccuracies in either are really an error between the recorded signal and the original which is quite complex in detail. (You can describe them as "distortion" or as "noise"; the distinction is meaningless.... they are simply errors in the signal. However, if you look at them as "noise", the errors introduced by the limitation in sample resolution have a pattern, and may be correlated to the data in some way, which means that they may well be "less benign" than the old familiar white noise floor in an analog recording. And if, instead, you look at them as "distortion", those errors are usually not harmonically related to the original signal, which suggests that they may also be "more annoying" than plain old harmonic distortion. Therefore, you can't safely assume that "knowledge" about "how much noise or distortion is audible" determined with those other forms will apply to them.) My point there is that these errors, whatever you prefer to call them, are about 96 dB down from a full scale signal at 16 bits, and I wouldn't entirely rule out the possibility that certain types of especially annoying or unpleasant error might not in fact be audible at that level. (Remember that patterns _BELOW_ the noise floor can in fact still be audible, so you can't make any blanket assumptions to the contrary.) To be honest, I can't personally claim to have ever noticed this, but using 24 bits seems worthwhile - just in case - because it pushes these errors down to well below -120 dB, and so offers a better safety margin.
  
 Now, to carry this discussion to the other part.... It is in fact true that there's lots of research to suggest that, _PURELY WHEN EXPRESSED AS BANDWIDTH_, a 44k sample rate can in fact cover the full range of human hearing. What this means is that, if you are measuring audible frequency response, using sine waves, a 44k file should be able to store anything that a human can typically hear. The math also shows that, under similar conditions, even the fine phase relationships required for positioning objects in the sound stage can be reproduced. (And I'm pretty sure that nobody is claiming to hear the difference with a sine wave.) However, the part of the math that most people seem to "miss" is that all of that is true _FOR A SINGLE CONTINUOUS SINE WAVE_. If you compare the output of two signals, one at 44k and the other at 96k, you will find that they will be the same for any continuous sine wave under 20 kHz; but the same will not be true if you use signals that aren't steady-state sine waves. Likewise, you can compare the analog outputs of two DACs with low distortion, and their outputs will be the same with any audible sine wave, but very different with _OTHER_ waveforms. And the reality is that the audibility of some of those other differences hasn't been well studied at all... and is in fact still relatively "unknown territory".
  
 (So, to put it simply, you can't extrapolate the fact that a certain sample rate, or a certain low-distortion DAC, can reproduce a _CONTINUOUS SINE WAVE_ with near-perfect accuracy to suggest that the obvious errors it may introduce to other waveforms will necessarily not be audible.... because, as it turns out, we humans seem to have other listening mechanisms tuned to detecting some of those other specific differences, and the sensitivity of those other mechanisms to these variations has _NOT_ in fact been fully tested at all. One example is that we have very sensitive mechanisms for responding to tiny phase differences, and that response may be altered if you alter the shape of the leading edge of the waveform in question, which might explain how we seem to hear differences between various DAC filters that, while they have virtually no effect on a sine wave, have a major effect on the leading edges of transient waveforms.)
  
 Quote:


rrod said:


> What are these "missing samples at the edges of 44.1"?
> 
> The view of 24-bits as capturing more resolution is in fact entirely equivalent to its capturing more dynamic range. Any map of 16-bit values to 24-bit by simple integer multiplication will preserve relative volumes. If we use the mapping f24(x) = x, then we end up with the viewpoint of 24 bits as having a lots of extra values above the max of 16 bit but with the same step size, allowing for my dynamic range. If we use the mapping f24(x) = 256*x (the common way to convert to 24 bits from 16), then we end up with the viewpoint of 24-bits as having finer gradations between steps. The only difference between the two viewpoints (in a perfect world) is where you'd set your volume, or more accurately, where the robot is setting the volume so you can be far the hell away from this experiment.


----------



## RRod

keithemo said:


> My point there is that these errors, whatever you prefer to call them, are about 96 dB down from a full scale signal at 16 bits, and I wouldn't entirely rule out the possibility that certain types of especially annoying or unpleasant error might not in fact be audible at that level. (Remember that patterns _BELOW_ the noise floor can in fact still be audible, so you can't make any blanket assumptions to the contrary.) To be honest, I can't personally claim to have ever noticed this, but using 24 bits seems worthwhile - just in case - because it pushes these errors down to well below -120 dB, and so offers a better safety margin.


 
  
 Assuming there are no issues with gain (I really can't help someone who ends up recording a 24-bit track into the bottom 8-bits), you'd have to have music that's getting near the LSB and for such a annoying error to actually occur. Seeing as how much stuff people listen to is pushing only 12-bits or so, I'm not too worried. Also note that in the end you end up expressing things as dynamic range/SNR anyway.
  


keithemo said:


> However, the part of the math that most people seem to "miss" is that all of that is true _FOR A SINGLE CONTINUOUS SINE WAVE_. If you compare the output of two signals, one at 44k and the other at 96k, you will find that they will be the same for any continuous sine wave under 20 kHz; but the same will not be true if you use signals that aren't steady-state sine waves. Likewise, you can compare the analog outputs of two DACs with low distortion, and their outputs will be the same with any audible sine wave, but very different with _OTHER_ waveforms. And the reality is that the audibility of some of those other differences hasn't been well studied at all... and is in fact still relatively "unknown territory".


 
  
 If the two DACs have low distortion, then any sum of sines should be pretty near the same as well. Regardless, what 96k gets you over 48k is the ability to capture signals at a higher frequency before aliasing kicks in. I task you to hear anything from any signal that has been high-passed to only include 24kHz to 48kHz.


----------



## sonitus mirus

old tech said:


> I did an interesting experiment the other day. While i normally do not get into itunes, i heard a lot of good stories about the mastered for itunes albums. I also noticed that these often appear to be the same mastering as the hi res HDTracks releases (often released on itunes in the following week) but in lossy aac and at a fraction of the cost. Anyway a couple of mates and i did some abx tests of the remastered led zep iii comparing the hi res version with the mastered for iitunes version. None of us got better than 52%. Make that of what you will...


 
  
 I had similar results between the Google streamed 320kbps mp3 and my own ripped FLAC files from new the Led Zeppelin remasters.  Just me and my older ears, but I spent a few hours attempting to hear any differences with no luck.  I will run a test tonight and see at which bit rate I start hearing differences.


----------



## RRod

sonitus mirus said:


> I had similar results between the Google streamed 320kbps mp3 and my own ripped FLAC files from new the Led Zeppelin remasters.  Just me and my older ears, but I spent a few hours attempting to hear any differences with no luck.  I will run a test tonight and see at which bit rate I start hearing differences.


 
  
 Try it out with Opus too if you don't mind


----------



## hogger129

sonitus mirus said:


> I had similar results between the Google streamed 320kbps mp3 and my own ripped FLAC files from new the Led Zeppelin remasters.  Just me and my older ears, but I spent a few hours attempting to hear any differences with no luck.  I will run a test tonight and see at which bit rate I start hearing differences.


 
  
 What equipment were you listening on?  It's not surprising though.  I fail to hear the difference between LAME-encoded MP3 files at that bitrate and the FLAC source.  I use AAC though for space savings and because I can drop it down to much lower bitrate and still have it sound indistinguishable.


----------



## jcx

I do dislike "hearing below the noise floor" arguments - needs lots of qualifications
  
 hearing is "colored" - the noise floor varies with frequency - and the threshold in quiet, after minutes of accommodation in a anechoic chamber is a limit - you don't hear below that curve
  
 for our special interest in headphone listening it turns out our noise threshold in quiet with accommodation time is up 10 dB worse because of headphone's "microphonics" converting pulse, muscle tremor, any motion, mechanical noise to sound
  
 when listening to music at normal levels your ears aren't operating at threshold in quiet levels, and you aren't often in a room without much higher noise at every conventional audible frequency
  
  
 any modern discussion of the capability of 16/44 should assume dither - since perceptual weighted frequency shaped dither has been pretty universal for decades as a part of the CD release mastering
  
 with noise shaped, weighted dither a 3 kHz tone can be heard 20 dB below -96 dB, "lsb" signal level with 16/44 bit source
  
  
 considering actual noise audibility, dither destroying the quantization correlation (and the problems with Keith's discussion of that too) I suggest writing off his last post entirely


----------



## sonitus mirus

hogger129 said:


> What equipment were you listening on?  It's not surprising though.  I fail to hear the difference between LAME-encoded MP3 files at that bitrate and the FLAC source.  I use AAC though for space savings and because I can drop it down to much lower bitrate and still have it sound indistinguishable.


 
  
 Right now I am using a Modi 2 and Asgard amp with a pair of Lawton-modded Denon D5000 headphones with Mr. Speaker angled Alpha pads.  I have a few other DACs and Amps that I could use, but they don't perform any differently to me.  I was able to successfully pass the Philips Golden Ears challenge with the configuration.
  
 I regularly listen with a pair of Rokit 8 (gen. 3) studio monitors via my PC or Chromebook with this DAC:  http://www.radialeng.com/usbpro.php
  
 Though, I could not complete the Philips Golden Ears challenge using the speaker setup, only the headphones setup, so I don't use the speakers for listening tests.  I don't know if I can't pass the test, but I don't seem to have the patience using the speakers and it seems that the isolation with the headphones eliminates distractions.  I hear dead silence over 14kHz at my normal listening volume levels.  I don't think there is a lot of musical stuff that hurts the sound quality at those frequencies, but if there were noticeable differences up there, I would not be able to tell.


----------



## KeithEmo

You're missing my points, though.....
  
 In the first case, my point is that the "maximum possible S/N" with a 16 bit file is about 96 dB - which would be plenty good for "typical noise" to be inaudible. However, we're all used to noise that consists of either white noise or harmonic distortion - so it's possible that a S/N of 96 dB isn't sufficient to render other, more annoying, noise inaudible. Specifically, many encoding processes generate "data correlated noise" - which varies with the music - and can be much more annoying than white noise. (In reality, virtually all digital recordings produced commercially use dither to "convert" the digital "artifact floor" into true, and innocuous, white noise or something similar - at the sacrifice of about 6 dB of S/N. So a real world CD usually does in fact have "normal" innocuous white noise as its noise floor - at about -90 dB. While I agree that a S/N of 90 dB is probably "good enough", adding a lot more safety margin there by going up to 24 bits seems worthwhile to me.
  
 In the second point, all of the math that describes things like the Nyquist limit is based on the assumption that you are talking about continuous sine waves. This technically means a single sine wave, whose amplitude and frequency _NEVER_ vary, extending from the infinite past to the infinite future. And, by that math, a 44k sample rate can accurately encode a 20 kHz sine wave, and also accurately "retain" even minute phase differences between two such sine waves. You will also find that virtually all modern DACs can reproduce a pure sine wave with very low THD (typically 0.01% or less). 
  
 However, when you look at signals that are _NOT_ pure sine waves, things are quite different. If you look at a pulse - a sound which starts suddenly like what you get when a drumstick hits a cymbal, in the frequency domain, it contains a broad group of frequencies, extending well into the ultrasonic. Then, when you apply the bandwidth limiting necessary to encode that signal at 44k or 96k, the shape of the waveform will change significantly, and the 20 kHz limited waveform (necessary for 44k) and the 40 kHz limited waveform (necessary for 96k) will be different from each other. Then, when you play either through a DAC, the analog output waveforms will also be different. And, furthermore, if you play one of those signals through different DACs, each of which has exemplary flat sine wave frequency response, you will see major differences between _THEM_ because of differences in the response characteristics of their oversampling filters. Because sine waves are continuous, none of these differences will be present with continuous sine waves, but they will be present, and often significant, with non-continuous waveforms.
  
 It is these differences in response to non-continuous waveforms that haven't been fully tested. So, in other words, while most people agree that a 44k sample rate is sufficient to reproduce any useful information present in a sine wave signal up to 20 kHz, some tests have suggested that other information that our brains are able to "extract" from impulse signals, and other non-continuous signals, is lost. (One such test demonstrated that, to at least some test subjects, the positioning of instruments in the sound field "shifted" when the signal was band-limited to 20 kHz, even though none claimed that "they heard anything missing". The conductors of the test theorized that perhaps our brains are able to resolve timing differences between the arrival times of asymmetrical waveforms finer than those that can be accurately reproduced at 44k.)
  
 If you want a "practical exposition" of this, download a datasheet for a DAC with multiple filter choices - like the Wolfson 8741. You'll find that several of the filter choices offer very flat frequency response and low steady-state distortion; yet, when you look at the provided oscilloscope pictures of transients, they are very different. There are several ways in which you can "quantify" ringing such as what you'll see there. If you input a short pulse, and what you get out is a short pulse with a long ringing "tail", you could average all of the energy over time - which would give you a relatively low distortion figure. However, if you look at it in the very short term, after every short pulse, you have a short period where there was no signal going in, but there is significant signal coming out, which you could reasonably characterize as very short bursts of very high distortion. (In other words, if you input a pure sine wave, you get out a pure sine wave with less than 0.01% THD. However, if you put in a series of 1 mS pulses, you get out a series of 1 mS pulses, each with less than 0.1% THD, but each followed by 1 mS of 10% THD.... and nobody I know has run tests on how audible that situation really is.) Suddenly, the question has become very complex, and very difficult to test in detail.
  
 Quote:


rrod said:


> Assuming there are no issues with gain (I really can't help someone who ends up recording a 24-bit track into the bottom 8-bits), you'd have to have music that's getting near the LSB and for such a annoying error to actually occur. Seeing as how much stuff people listen to is pushing only 12-bits or so, I'm not too worried. Also note that in the end you end up expressing things as dynamic range/SNR anyway.
> 
> 
> If the two DACs have low distortion, then any sum of sines should be pretty near the same as well. Regardless, what 96k gets you over 48k is the ability to capture signals at a higher frequency before aliasing kicks in. I task you to hear anything from any signal that has been high-passed to only include 24kHz to 48kHz.


----------



## RRod

keithemo said:


>


 
  
 You worry too much, Keith.


----------



## jcx

ouch, "Fourier Denial" -  the Math is very clearly against your interpretation
  
 please ref any listening test studies - I hate being behind on my reading
  
 if however Kunchur is your source I don't have anything to worry about except more people not able to debunk his Signal Theory errors - all of his tests could have been implemented in 16/44
  


jcx said:


> xionc said:
> 
> 
> > Two words - temporal resolution. link: boson.physics.sc.edu/~kunchur/*temporal*.pdf
> ...


 
  
 in fact Moffat's


baldr said:


> ...Thanks to Dr. Heil, the inventor of the Heil AMT speaker who shared this experiment with me over 40 years ago, Consider this: I am 67 years old – my high end extends to just under 15KHz (not bad for and old fart). I can play back two pulses 200 microseconds in length separated by 20 microseconds and clearly hear two pulses. Not unusual until one considers that 20 microseconds corresponds to a square wave of 50KHz. And then, there is the time domain – home of spatial cues which audio measurement traditionalists ignore. I believe that in the quest for the best sound, an open mind is the most important asset. I will even listen to cables, even though I believe in my heart that all technology about cables is well known. Who knows, even an old fart like me could be surprised.
> ...


 
  
 is laughable too - I created a 400 us single pulse and the twin pulse wav *at 16/44 and did 10/10 myself on 1st pass* - but it didn't sound like 2 clicks to me
  
 then I made the single pulse 420 us long and normalized the amplitude for the same area as the twin pulses - couldn't hear the difference
  
 to me that shows the easily detectible difference was in the frequency content entirely below 22 kHz - the stretched pulse spectra is much closer to the twin
  
 I'd be happy to put the .wav somewhere everyone could try


----------



## RRod

jcx said:


> in fact Moffat's...


 
  
 I just got done reading the AES monograph on HRTF and read Begault's book on the same subject not long ago, and I don't recall them making any deal, at all, about timing issues with 44.1 affecting anything in terms of spatial cues.
  
 As far as Fourier, I didn't want to come off as dismissive to Keith, but I get irked (perhaps you do too) when every single possible mathematical underpinning of Nyquist-Shannon and Whittaker-Shannon is debated as some possible way to justify hi-res phenomena that people can't ever seem to actually hear.


----------



## KeithEmo

I prefer to think the opposite - that I try _NOT_ to worry too much.
  
 Most popular albums get "remastered" or "remixed" every few years, with some of them actual improvements, and others not. Therefore, to me, the fact that some "high-res remasters" are better and others are not isn't at all unexpected, or sinister. (And I don't recall a huge backlash about whether "K2" remasters are really better or not either.) As I see it, it still makes absolute sense to judge whether a given reissue is better or not based on its own merits, which is the way I've always done it and will continue to do.
  
 As such, I consider the question of whether there is any _inherent_ benefit to to high-resolution files to be of academic interest... (But I'm still not going to buy every high-res remaster, even if someone can prove that the format itself is _capable_ of better sound, because some of the individual examples are clearly not particularly better; and I'm not going to _avoid_ buying every one that does sound better, even if someone were able to prove that it couldn't possibly be _because_ it was provided as a high-res file.)
  
   Quote:


rrod said:


> You worry too much, Keith.


----------



## RRod

So you don't worry about ringing of filters or quantization distortion for hi-res formats? Say no and I'll, of course, believe you, but for exactly the same reasons why I don't worry about the same issues for Redbook these days.


----------



## jcx

on balance Keith's long form posts are valuable - I don't want my engineering/numbers commentary to be seen as a pissing match - but his trying to cover so much there will always room for comments, clarifications
  
 and when someone gets a "guru" rep here its likely that non tech but educated intelligent types will make "natural language" inferences that some of us engineers will want to expand/correct
  
  
 2 lsb peak-to-peak triangle probability density function white dither only costs 3 dB S/N, not 6 - and Keith's commentary rolled right past my mention of perceptual weighted, noise shaped dithers
  
 "temporal resolution" relation to digital audio, sample rates and bit depth is really well nailed down in engineering math, signal theory - relating that to human audio perception is more arguable, has soft boundaries to what is known with some certainty and lots that isn't


----------



## castleofargh

for the interest of how things work, I always enjoy reading about jitter, digital and analog filters, LSB, oversampling, nyquist, and music's actual dynamic range. same for the ever growing information we have on psycho acoustic. but I fail to see why they should lead an argument about audibility? shouldn't audibility tests be the valid answer to audibility concerns?
  
 and shouldn't we do our own audibility tests to come up with our own conclusions and choices? as far as I'm concerned, from time to time I notice a difference in the highres file, I convert it to 16/44 and abx it with the cd rip I have(if the differences aren't very obvious. with stuff like "random access memory", ABX really wasn't needed^_^). and sure enough the differences are still very much on the downsampled version. demonstrating that I'm listening to 2 different masters and not to highres vs CD. to me that's important so that we avoid fighting the wrong war.
 based on my tests, clearly my war is on mastering job, and never once was about highres vs cd where trying to ABX the highres file to a downsampled then upsampled file, resulted in guessing statistics for as long as I remember doing it. I am yet to find one album that is the same master but sounds different when played in highres. I've looked long and hard, and listened on a variety of gears throughout the years. nothing, nada, rien.
 so that gets me wondering how many of the people talking about how highres sounds better, are really talking about highres at all?
  
  
 now a guy with something like a pono will hear differences with all downsampled tracks because of how the pono rolls off the trebles when playing 16/44. so while the gear is clearly to blame, it could make sens(somehow) for the person using a pono to listen to highres. or at least for him to upsample his library as a way to avoid the stupidly strong roll off @16/44 if he doesn't want it.
  
 and with different situations we get different reasons to go for, or avoid highres. but actual audibility should be our main audio reason. all other reasons like, fashion, being reassured, misunderstanding the actual limits of 16/44, or just wishing to have the best, those are ok reasons too at an individual level. but they have no relation to what we actually can or cannot hear.


----------



## sonitus mirus

rrod said:


> Try it out with Opus too if you don't mind


 
  
 Done.  Opus is awesome.  
  
 From the new Jimmy Page remasters, I used the intro to Zeppelin's "Rock and Roll" since it had plenty of cymbals and crashes.  I ripped the CD to WAV with dBPoweramp, trimmed the WAV using Audacity, and converted to various lossy formats from the same WAV file using dBPoweramp.  Once all of the lossy conversions were created, I batch converted all of the files to FLAC and zipped them into a single downloadable file at the following link:
  
 https://www.dropbox.com/s/tvipgtirdwfssy9/FLACs.zip?dl=0
  
 I only tried the lowest quality Lame MP3 version vs the lossless, but I did not check any of the others yet.  Hopefully I didn't mess anything up.  I'm only on beer two. 
	

	
	
		
		

		
		
	


	



  
  
 foo_abx 2.0.1 report
 foobar2000 v1.3.7
 2015-12-04 21:39:15
 File A: Z4_RR_LAME_CBR96_16_44100.flac
 SHA1: e339b8a388ad64fec342599ff5892451c9c10fc3
 Gain adjustment: -5.29 dB
 File B: Z4_RR_PCM_16_44100.flac
 SHA1: c457ad5fc7b07c4506f7827f4ab3d210257875b1
 Gain adjustment: -5.76 dB
 Output:
 WASAPI (push) : Speakers (USB Modi Device), 24-bit
 Crossfading: NO
 21:39:15 : Test started.
 21:40:24 : 01/01
 21:40:40 : 02/02
 21:41:11 : 03/03
 21:41:38 : 04/04
 21:42:02 : 05/05
 21:42:17 : 06/06
 21:42:36 : 07/07
 21:42:56 : 08/08
 21:43:19 : 09/09
 21:43:45 : 10/10
 21:44:45 : 11/11
 21:45:09 : 12/12
 21:45:56 : 13/13
 21:46:12 : 14/14
 21:46:49 : 15/15
 21:46:49 : Test finished.
  ---------- 
 Total: 15/15
 Probability that you were guessing: 0.0%
  -- signature -- 
 ab75ac5fc809c1a202d78a2c263a5572a6706299


----------



## old tech

keithemo said:


> For quite some time Apple's official guide to "mastering for iTunes" has recommended mastering at 24/96 if at all possible, as well as including several recommendations that all work out to "master at the best quality you can".
> 
> I would also agree with you that, although I didn't do any careful ABX testing, I have the full set of recent Led Zeppelin remasters in high-res, and, in casual listening, I didn't notice any particular improvement with them over the previous versions either. This all highlights an important distinction that some people in this discussion seem to forget when they get excited..... and that is that the following three questions are ver




That's interesting. While I also don't hear significant differences between most of the original led zep originals and the remasters, the difference between III (and Physical Graffitee) is stark, and much improved.


----------



## RRod

sonitus mirus said:


> From the new Jimmy Page remasters, I used the intro to Zeppelin's "Rock and Roll" since it had plenty of cymbals and crashes.


 
  
 Rock and Roll? Shoot, you should truncate it to 8 bits and test that out too  And if you really don't trust your hearing much past 14k, try taking it down to 8/30 and see how that goes.


----------



## sonitus mirus

rrod said:


> Rock and Roll? Shoot, you should truncate it to 8 bits and test that out too  And if you really don't trust your hearing much past 14k, try taking it down to 8/30 and see how that goes.


 
  
 I was trying to think of a ~30 second sample of a Zeppelin song that would be worthy of testing.  Any recommendations are welcomed.  I have all of the remastered Zep CDs except _In Through the Out Door_ and _Coda_.


----------



## RRod

castleofargh said:


> for the interest of how things work, I always enjoy reading about jitter, digital and analog filters, LSB, oversampling, nyquist, and music's actual dynamic range. same for the ever growing information we have on psycho acoustic. but I fail to see why they should lead an argument about audibility? shouldn't audibility tests be the valid answer to audibility concerns?


 
  
 You run into the "what if" problem pretty quickly when dealing with people, though. What if in the future there's some track where hi-res matters? What if in the future I upgrade gearz and suddenly I can hear the difference? What if we find some new test that I can actually pass? Then they of course come to the inevitable conclusion: we need to keep increasing sample specs forever and ever, because we'll never know! I've found it hard to combat this way of thinking.


----------



## RRod

sonitus mirus said:


> I was trying to think of a ~30 second sample of a Zeppelin song that would be worthy of testing.  Any recommendations are welcomed.  I have all of the remastered Zep CDs except _In Through the Out Door_ and _Coda_.


 
  
 Wrong guy to ask. There's no master I've heard of ZeppIV (as an example) that's ever made me go "wow this is a great sounding album!" It's just like every release of Kind of Blue makes all kinds of claims about great new things but of course can't ever remove the elephant in the room of totally audible hiss and crackly sounds.


----------



## sonitus mirus

rrod said:


> Wrong guy to ask. There's no master I've heard of ZeppIV (as an example) that's ever made me go "wow this is a great sounding album!" It's just like every release of Kind of Blue makes all kinds of claims about great new things but of course can't ever remove the elephant in the room of totally audible hiss and crackly sounds.


 
  
 I agree with you about Zeppelin and blues rock in general.  I don't think it really matters what song is used from any artist.  The results will probably be the same.  I don't know why more people don't conduct their own tests to see where there own limits might reside.


----------



## old tech

sonitus mirus said:


> I was trying to think of a ~30 second sample of a Zeppelin song that would be worthy of testing.  Any recommendations are welcomed.  I have all of the remastered Zep CDs except _In Through the Out Door_ and _Coda_.



I don't have either the remastered ITTOD or Presence as yet. Is Presence a worthwhile improvement?


----------



## jcx

you might look for stuff Jamie has helped with: http://audiophilereview.com/analog/plangent---a-better-way-to-transfer-analog-tape.html
  
https://www.google.com/#q=plangent+process


----------



## Baldr

jcx said:


> in fact Moffat's
> 
> 
> baldr said:
> ...


 
  
Perhaps if jcx were to reread my quote, he would realize that this experiment was performed originally in the mid 1970s, with purely analog signals.  He may not be aware that this was a pre digital audio era, at least with respect to commonly available hardware.  I also apologize if I have not made it clear that I have researched sampling theory as well as single and multirate digital signal processing all the way back to it's very beginnings at Ma Bell, prior to World War I.  Perhaps one of his intellect and digital audio product design background assumes dolts and hucksters such as myself would try to digitize 50KHz signals at 44.1 or 48KHz, even though I have designed several dozen ds and multibit da audio components over the last 38 years.
 
My suggestion would be for many of you to stop taking yourselves so seriously.  This is a hobby.  It is to be enjoyed.  I am amazed at the pugnacity I have seen in two sound science threads I have spent time digesting.  There are at least two ways to make others feel bad - one is to make wrong what another has said, and another is to explain to someone else exactly what he meant to say.  I see a lot of that on those threads.
 
It reminds me of a joke of a bear hunter who misses the shot on the bear, who is then captured by the bear who has his way sexually with the hunter before he releases him.  This then happens two more times.  The third time, the bear says to the hunter - "Hey buddy, are you sure you are in this for the hunt?"  Are all of you sound science guys really here for the hunt of good audio??
 
We will see if this post survives deletion.  I hope so.
 
Finally, in the spirit of the topic of this thread, my lifetime experience has taught me that a damned good audio experience is available from redbook recordings.  I believe that making audio components in very limited quantities for a very unlimited price is puffery.  I also know that I have sold schiitloads of DACs with absolutely no promise of anything other than specifications and the greatest majority of users are very happy.  I also know there are people who will never be happy, period.  If that describes you, then please do not buy a Schiit DAC.  As I write, I am listening to one of my $100 DACs and I am enjoying the music.


----------



## Joe Bloggs

baldr said:


> [COLOR=454434]It reminds me of a joke of a bear hunter who misses the shot on the bear, who is then captured by the bear who has his way sexually with the hunter before he releases him.  This then happens two more times.  The third time, the bear says to the hunter - "Hey buddy, are you sure you are in this for the hunt?"  Are all of you sound science guys really here for the hunt of good audio??[/COLOR]




I have no beef with the rest of your post but this sounds insulting. It's almost as though you're saying the sound science guys will keep taking it up their ass as long as they try to discuss with people what can make a real difference in audio and what cannot. :rolleyes:


----------



## RRod

Yes we are fully aware of how this hobby goes:
  
 .Person A spends umpteen simoleons on a bunch of tube stuff and DACs without filters to tame his HD800s and spends $30 on every new release of DSotM. Crowd reaction "Man, that guy really gets this hobby."
  
 .Person B spends a little as he can on audibly transparent equipment equipment, decent cans, and listens to albums that don't need any HD "fixes". Crowd reaction "You just don't get it, and you take this stuff way to seriously."
  
 Rolling my eyes on the floor.


----------



## limpidglitch

I can't really speak for anyone but myself, but I'm definitely not here for "the hunt of good audio". That stuff got old real fast.
  
 These days I spend my time hunting for the snark. I find that a far more fruitful pursuit.


----------



## Shembot

joe bloggs said:


> I have no beef with the rest of your post but this sounds insulting. It's almost as though you're saying the sound science guys will keep taking it up their ass as long as they try to discuss with people what can make a real difference in audio and what cannot.


 

 I took from this anecdote the idea that there's a difference between "discussing with people what can make a real difference in audio and what cannot," which is constructive and appreciated, versus using attacks on people's experiences and credibility to discount their contributions to the discussion. The former is a great way to further our understanding of audio, while the latter, well, isn't.
  
 There's also a point at which discussions can lose sight of seeking maximum enjoyment of music in favor of bickering about minutiae in a manner that just turns people off to the whole hobby. I find for myself that it's good to take a step back to gain some perspective and re-center on what really matters. 
  
 I can only speak for myself, of course, but I appreciate the candor of Mike's posts about these things -- they help me remember that this is a hobby that can be enjoyable for everyone while engendering spirited, but polite, debate that can improve all of our experiences.


----------



## RRod

shembot said:


> I took from this anecdote the idea that there's a difference between "discussing with people what can make a real difference in audio and what cannot," which is constructive and appreciated, versus using attacks on people's experiences and credibility to discount their contributions to the discussion. The former is a great way to further our understanding of audio, while the latter, well, isn't.
> 
> There's also a point at which discussions can lose sight of seeking maximum enjoyment of music in favor of bickering about minutiae in a manner that just turns people off to the whole hobby. I find for myself that it's good to take a step back to gain some perspective and re-center on what really matters.
> 
> I can only speak for myself, of course, but I appreciate the candor of Mike's posts about these things -- they help me remember that this is a hobby that can be enjoyable for everyone while engendering spirited, but polite, debate that can improve all of our experiences.


 
  
 That's all good and fine until you have to counter the argument of "2>1 therefore hi-res is better" for the trillionth time.


----------



## Joe Bloggs

baldr said:


> jcx said:
> 
> 
> > in fact Moffat's
> ...




I think the experiment wasn't adequately described for replication in that one paragraph (nor were you intending for it to be replicated with that brief blurb, I suppose). What were these "pulses" and what was done to verify that you were able to detect two pulses?

Here I illustrate two extremes of what you might have been doing (at 384000Hz sample rate):
1. Two noise bursts of square envelope, separated by 20 microseconds of silence
2. Two sine half-waves of 200 microseconds length, separated by 20 microseconds of silence

In the former case there's hardly any evidence of there being a break in the sound, even looking at the waveform, whereas in the latter case you have two obvious pulses... 220 microseconds apart, not 20 microseconds apart. Nevertheless, my cloth ears can't really perceive even the latter as two pulses. The lower two plots illustrate what happens if you downsample each waveform to 44.1kHz.


----------



## jcx

the "its all good fun excuse again?
  
 I seem to be getting repeated nontechnical put downs by a couple of guys who tell us they don't take themselves seriously
  
  
 you just walked in all innocent of the ongoing controversies - are unconsciously using "loaded phrases" that here invoke long standing arguments?
  
  
 maybe "._.another is to explain to someone else exactly what he meant to say."_
  
 is about how to have a technical dialog - restate the other's position in your terms, get feedback on what they still recognize, acknowledge that you got right, work on improving your mutual understanding of each other's positions and reasoning
  
 as mentioned, in my experience, between EEs "pugnacious" isn't about "making the other person feel bad" - some ideas, conceptualization, inferences are just wrong - no need for "social lube"
  
 we all have been, expect to be on both ends learning, working in complex subjects
  
 in fact the competitive aspect, "not wanting to sound stupid", is incentive to debugging your thinking - it actually engages other filters and scans internally when you think about how it will sound to someone else, especially if they are expected to pick apart your reasoning
  
  
 so what is right about the following, how is my point misreading it? - to me literal technical inaccuracy is involved
  


> > ...Not unusual until one considers that 20 microseconds corresponds to a square wave of 50KHz. And then, there is the time domain – home of spatial cues which audio measurement traditionalists ignore...


 


>


 the 50 kHz part sounds to me like a erroneous implication that will mislead many – in fact such examples have misled, get repeatedly presented as if implying 1/t ultrasonic human hearing, needed 2x, 4x sample rates, we can point to similar examples all over Sound Science and beyond
  
 my bolded 16/44 and positive abx results does address the 50 kHz hand wave - as I spelled out
  
  
 the unneeded swipe at "_time domain – home of spatial cues which audio measurement traditionalists ignore_" is insulting our intellectual depth too, considered baiting here in Sound Science - outside SS I think its damaging to Head-Fi too  
  
  
 I used free LTspice to generate the test files - the anti-alias filter could be better, uses cascaded filters I already had in sim, but is I think adequate for the example
  

  
  
  
 and my main point - debugging the "50 kHz" the implication that you would need higher sample rate - is that I can *positively* discriminate in abx at 44.1k - not the weak negative


----------



## castleofargh

shembot said:


> I took from this anecdote the idea that there's a difference between "discussing with people what can make a real difference in audio and what cannot," which is constructive and appreciated, versus using attacks on people's experiences and credibility to discount their contributions to the discussion. The former is a great way to further our understanding of audio, while the latter, well, isn't.
> 
> There's also a point at which discussions can lose sight of seeking maximum enjoyment of music in favor of bickering about minutiae in a manner that just turns people off to the whole hobby. I find for myself that it's good to take a step back to gain some perspective and re-center on what really matters.
> 
> I can only speak for myself, of course, but I appreciate the candor of Mike's posts about these things -- they help me remember that this is a hobby that can be enjoyable for everyone while engendering spirited, but polite, debate that can improve all of our experiences.


 
  
 when discussing how junk food isn't healthy, I can also come in and say "but it tastes so good, stop worrying", it's easy to get the crowed go my way with stuff like that. but you can't just go hakuna matata anytime someone asks a question. if we all had done that, there wouldn't be any digital audio to talk about and no internet.

  
 and if it's about joy, then all I need for music is bunch of buddies, something to drink and a lousy stereo playing some old offspring songs. that's always when I enjoyed music the best. what's highres doing for my enjoyment? I can't even hear a difference. to me highres is guy selling me a bigger bottle with the same content at higher price and pretending like it's a revolution. when I ask what's better, he would say "just listen", and I would answer, "but I don't hear an difference?", and he would say "yeah but just listen it's better, trust me I work at marketing, look at those stair step graphs that represent no reality whatsoever". 
 not only I gain nothing, but the guy is taking me for a fool. not exactly my idea of enjoying music.
   but I guess that's just me. people buying the bigger bottle thinking they get the ultimate stuff, they sure can enjoy thinking they have it all. at least until some copyright expires and the new ultimate stuff comes up explaining how the last stuff was crap. history repeating. file formats are the laundry powder of audio, each time the new one explains how the old one wasn't really making stuff white. but we should always only doubt the old one, this time for sure they're not lying!!!! they know they were the ones selling the old crap too...
  
  
 it's easy to use rhetoric instead of talking about facts. and I might just have convinced more people right now with my personal rant than I ever could talking about blind tests and nyquist. but nyquist was right and we do get 90db+ of goodness from a CD. and people do fail blind tests. so what's all this about? joy? I don't think so.


----------



## kstuart

rrod said:


> Yes we are fully aware of how this hobby goes:
> 
> .*Person A spends umpteen simoleons *on a bunch of tube stuff and DACs without filters to tame his HD800s and spends $30 on every new release of DSotM. Crowd reaction "Man, that guy really gets this hobby."
> 
> ...


 
 and


> people *buying the bigger bottle *thinking they get the ultimate stuff


 
  
  
 It's interesting how often anti-audiophile posts are about money.
  
 If it were scientific, then the price would not matter - it would only be about how the audio boutique equipment cannot be better than the mass produced equipment - regardless of whether they are the same price or not.
  
 Better or not better, can be determined scientific.   Oscilloscopes are unaware of price.
  
 So, the obvious conclusion is that opposition to audiophiles is emotional and the invocation of science is just a veneer.


----------



## RRod

kstuart said:


> and
> 
> 
> It's interesting how often anti-audiophile posts are about money.
> ...


 
  
 You seem to be leaving out the part where companies charge money for their product, and knowing how much a product cost affects our perception of its abilities. And we often do discuss whether audio boutique equipment has audible benefits over more reasonably-priced equipment or not.
  
 And part of the whole focus on here is determining better by doing scientific things like blind tests, so I really don't get your beef at all.


----------



## kstuart

rrod said:


> kstuart said:
> 
> 
> > and
> ...


 

 So you are agreeing with me - if " Audio Gold Corporation " charged $500 for their amplifier instead of $20,000, then you would not be bothering to do any testing at all against " Giant Asian Corporation" 's $500 amplifier.
  
 That's not science, it is human emotion about pricing.
  
 In contrast, if you have two different anti-cancer drugs, the price is nowhere in the scientific studies, and no one tests them solely because they are expensive or cheap.


----------



## castleofargh

kstuart said:


> rrod said:
> 
> 
> > Yes we are fully aware of how this hobby goes:
> ...


 

  what's that anti-audiophile thing you're talking about? does that mean RRod and I hate people who like music? 
	

	
	
		
		

		
			




 and sorry I'm not a rich scientist with unlimited funds, but one guy at home who actually cares about the cost of things and being lied to.
  
 care to explain what you're actually talking about?
  
  
 edit: oh so we hate highres because it's expensive? if that's your message, then it's true for me. having the same album in several available formats and sizes at the same price would be very fine for me(although I probably would still go for 16/44 flac for the size).


----------



## RRod

kstuart said:


> So you are agreeing with me - if " Audio Gold Corporation " charged $500 for their amplifier instead of $20,000, then you would not be bothering to do any testing at all against " Giant Asian Corporation" 's $500 amplifier.
> 
> That's not science, it is human emotion about pricing.
> 
> In contrast, if you have two different anti-cancer drugs, the price is nowhere in the scientific studies, and no one tests them solely because they are expensive or cheap.


 
  
 A test should happen when someone makes a claim that is testable. If someone claims that the ODAC is audibly indistinguishable from the Yggy, that should be tested, regardless of their price. The fact is that high price gear is harder to get tested because a) fewer people have it and b) those who have it don't want to find out it's not audibly better than something cheaper. But that has nothing to do with when a test is proper or not. How has anything we said smacked differently than that?


----------



## kstuart

I'm purely talking about the motivation.
  
 About 95% of the people who support "objective audio testing" over subjective impressions, eventually make some post that expresses anger over the existence of $20,000 products.
  
 If all of the amplifiers were a uniform $500, no one would care enough to spend hours doing specialized testing.


----------



## RRod

kstuart said:


> I'm purely talking about the motivation.
> 
> About 95% of the people who support "objective audio testing" over subjective impressions, eventually make some post that expresses anger over the existence of $20,000 products.
> 
> If all of the amplifiers were a uniform $500, no one would care enough to spend hours doing specialized testing.


 
  
 The anger I expressed was over people assuming that because you don't have the $20000, you can't know what "good sound" is. That's a bit of a different matter.
  
 If I woke up one day and all amps were the same price, I would assume all marketing departments had somehow disappeared overnight…\o/ Still, at some point someone would claim one was better than another and we'd be off, perhaps with a bit less haste, I'll grant you.


----------



## Joe Bloggs

kstuart said:


> I'm purely talking about the motivation.
> 
> About 95% of the people who support "objective audio testing" over subjective impressions, eventually make some post that expresses anger over the existence of $20,000 products.
> 
> If all of the amplifiers were a uniform $500, no one would care enough to spend hours doing specialized testing.




Well, science is motivated by human needs just as much as anything else.

If the expected result of a 4 degrees Celsius rise in global temperature were an even warming of all arctic locations to comfortable habitable temperatures, increased agricultural landmass, no desertification, no high power hurricanes, and somehow the arctic ice would still be all there and not cause sea level rise to submerge major cities, well, you can bet there wouldn't have been even a fraction of the research effort put into whether the projected temperature rise is man-made or nature-based. Or if there were, the focus of the research would have been how to pump even more greenhouse gases into the air, not reduce it. :rolleyes:

Or, going by your logic, you could say that the sum total of climate research is the result of scientists venting their frustration at not having enough money to each own a personal Learjet, a helicopter car with a 30 litre engine, and a 10,000 feet personal palace that's always climate controlled at a cool 70 degrees in the middle of Nevada (built entirely out of 1000 y.o. Amazon hardwood), and in general not being rich enough to be a significant contributor to global warming. :rolleyes: :rolleyes:


----------



## jcx

Lets see, how many times in connection with quoting


> Not unusual until one considers that 20 microseconds corresponds to a square wave of 50KHz


 
  
 in a thread titled: *Why 24 bit audio and anything over 48k is not only worthless, but bad for music.*
  
 did I mention that* I can hear the difference at 44.1* – so that* no 50 kHz hearing is needed*
  
 Quote: 





baldr said:


> jcx said:
> 
> 
> > in fact Moffat's
> ...


  
 
perhaps if Mike were to read closely in the 1st place?
 


> Perhaps if jcx were to reread my quote, he would realize...
> ...such as myself would try to digitize *50KHz signals *at 44.1 or 48KHz


 
 
 you seem to have made a logical leap – that just because the 20 us gap does have 50 kHz and above frequency components that hearing 50 kHz was involved in hearing the difference between the described waveforms
  
 it doesn't matter when the experiment was performed, or if your expectations were “primed” by Heil, the AMT's extended frequency response
 or a mindset that every wild claim about human hearing must be true if the claimants are sincere
  
 for the purpose of this thread my demonstration is that no 50 kHz hearing, encoding was needed
  
  
 the more complete:
  
 



> Perhaps if jcx were to reread my quote, he would realize that this experiment was performed originally in the mid 1970s, with purely analog signals.  He may not be aware that this was a pre digital audio era, at least with respect to commonly available hardware.  I also apologize if I have not made it clear that I have researched sampling theory as well as single and multirate digital signal processing all the way back to it's very beginnings at Ma Bell, prior to World War I.  Perhaps one of his intellect and digital audio product design background assumes dolts and hucksters such as myself would try to digitize 50KHz signals at 44.1 or 48KHz, even though I have designed several dozen ds and multibit da audio components over the last 38 years.
> 
> My suggestion would be for many of you to stop taking yourselves so seriously.  This is a hobby.  It is to be enjoyed.  I am amazed at the pugnacity I have seen in two sound science threads I have spent time digesting.  There are at least two ways to make others feel bad - one is to make wrong what another has said, and another is to explain to someone else exactly what he meant to say.  I see a lot of that on those threads.
> 
> ...


 
  
 shows you are using nasty rhetoric there - putting “fighting words” in my mouth that I never said, employing belittling tone, insulting “humor”
  
 when you appear to not even understand the point - which was that the described signals can be audibly discriminated when band limited to below20 kHz - sorry but that anecdote don't hunt
  
 makes the climb down a bit awkward – doesn't it?
  
  
 I don't doubt your or Jason's EE chops, even respect your unique depth of experience in audio
  
 but you both so far seem to be intent on sounding like the worst touchy “gurus” - that your word is final – none should dare question
  
 both of you seem intent on seeing reasonable tech, psychoacoustic questions, discussions as “attacks” – to me this appears to be playing to the guru/fanboy dynamic
  
  
 are there “pimply faced ankle biters”? - probably, some will adopt or be misled by fake “scientific” positions, get off on “debunking” with sophomoric logic, lack of knowledge
  
 is everyone with questions, different perspectives one?
  
 there are also decades experienced mixed signal engineers with strong long time hobby interest in audio, others with a variety of STEM backgrounds that can follow pretty detailed arguments
  
 are there “tin gods” in audio – I do think so, on both sides of most debates - try not sounding like them
  
 do you and Jason have to put on the Audio Guru act, demonize any but fanboys to sell Schiit? – I don't think so
  
  
 if you want it to be fun – if we are supposed to take prominent claims that you don't take yourselves too seriously then maybe you should show it
  
 which just might include looking with an open mind, avoiding stereotyping, responding to what is said without escalating the rhetoric


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## Baldr

It seems that I have been publicly called out on this forum for posts I've left not only here, but on other forums as well. I reacted with a post of my own, and have since had time to consider the entire situation. In my almost 68 years, I have made mistakes. Dr. Heil, inventor of the air-motion transducer not only told me of his pulse based audio tests, but gave me a recording of them back in the mid 1970s. He subsequently wrote and presented a paper to the AES (which I have a copy of somewhere). Since I believe he has now passed, the proper presentation of his technology is in the record.  It is definitely not my place to present his data.
  
 It was also a mistake to allow myself to be engaged by an ad hominem critical post by a forum member. My worst mistake was to descend to the same phallus-waving tech-narcissism established in the original post that publicly called me out. That does not serve any of the readers well. For the above two points, I apologize to all members of this forum, including the original poster. It is my experience that if one puts out slop, one gets it back. For that reason, I will not be further engaged on this matter.
  
 In the interest of positive discussion, I offer to the readers to the true experts on digital time domain optimization. Alan V. Oppenheim, Lawrence R. Rabiner, Ronald W. Shafer, and Ronald E. Crochiere. They have written much on the subject, from grad school level textbooks to various technical papers. Attacking me for their genius is futile.
  
 Until I began to study them 35 years ago, I did not realize that not only was I ignorant on the subject of digital time domain optimization, but I was ignorant of the fact that I was ignorant. This intellectual haboob did not clear until I comparatively modeled, implemented, measured and compared their various algorithms in various software based DSP processors. Detractors of their data are well advised to look before they leap.
  
 I do not imagine myself as any sort of audio genius. I've had the benefit of meeting many of them, however, and absorbed as much knowledge as I could. I am only a product of those I have met along the way. What I have been very skilled at is applying that knowledge to design and build dozens of analog and digital products over the last 38 years. These products have many thousands of users. I have no reason not to believe the great majority of them are happy for their value received. My skillset is making products happen. That experience is the genus of my opinions.
  
 Finally, a brief statement of my philosophy of audio comparison. I admit that I allow my ears and those of a few trusted listeners provide the lead, rather than strict abx tests to do so. If there is a consensus of sonic difference, it is either faith-based bias, or may bear further investigation. If I am convinced of a sonic difference, then there must be a way to verify so within the measurable physical universe as a condition of my declared superiority. I sell the specifications of my gear as opposed to the “sound”. I make no sonic claim for any of my products. I do question and will not be constrained by science. After all, today's science may well be only a subset of tomorrow's.
  
 I haven't completed compilation of evidence re my hypothesis of the apparent superiority of multi as opposed to ds d/a converter components. A more immediate focus is on reducing the cost of multibit converters. Experiments are underway which require the construction of time-intensive test equipment which has no recoupable development cost. I will get there.
  
 I am done fighting. It is a huge waste of time as well as a sorry spectacle.


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## Don Hills

baldr said:


> ...Finally, a brief statement of my philosophy of audio comparison. I admit that I allow my ears and those of a few trusted listeners provide the lead, rather than strict abx tests to do so. If there is a consensus of sonic difference, it is either faith-based bias, or may bear further investigation. If I am convinced of a sonic difference, then there must be a way to verify so within the measurable physical universe as a condition of my declared superiority. I sell the specifications of my gear as opposed to the “sound”. I make no sonic claim for any of my products. I do question and will not be constrained by science. After all, today's science may well be only a subset of tomorrow's....


 
  
 This bears repeating, especially the part that if a difference is audible, it must be measurable - somehow.


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## jcx

> It was also a mistake to allow myself to be engaged by an ad hominem critical post by a forum member


 
 I'm sorry that you still see it that way - how many times do I have to say I discuss Ideas - criticizing someone's statements, reasoning about a subject is not an attack on the person
  
  
 I too have "senior and graduate level" DSP course books - needed to learn polyphase time interpolating filter design in multiplexed sampled data systems requiring time aligned data output
  
 and I don't imagine myself an expert by those standards - many, probably most of the tens of thousands of EEs that have taken the actual courses, or done serious self study are ahead of me
  
  
 not knowing what you are doing in DSP specifically I haven't said there are no possible improvements
  
  
 I think I can still reasonably be skeptical of how important these unnamed improvements are for listening to digital audio music releases given so many other limitations in time, phase errors of transducers, inherent and added noise
  
  
 I am not at all a "meter reader" reductionist - Psychoacoustics is broad, complex and incomplete
  
  
 if you want a even more extreme example try inter aural time delay - clicks presented to opposite ears can be discerned to arrive at different times as small as 10, possibly 5 us, maybe 2 us? - the textbook number may still be 20 us - which is still intriguing
  
 and I do criticize many surrounding claims, Signal Theory fails of even Kunchur - the differences can be shown with 16/44 digital audio
  
 what I really try do debunk is the implication that 50, 100, even 200 kHz hearing is required to explain it - and after its all wrapped up Kunchur agrees, points to correlation - not 100 kHz hearing
  

  
  
 for my part the only "fight " was to get considered seriously, get you to pay attention to what was said - not being stereotyped, my argument ignored - get past your nonresponsive and non technical criticisms


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## JWolf

jcx said:


> shows you are using nasty rhetoric there - putting “fighting words” in my mouth that I never said, employing belittling tone, insulting “humor”
> 
> when you appear to not even understand the point - which was that the described signals can be audibly discriminated when band limited to below20 kHz - sorry but that anecdote don't hunt
> 
> makes the climb down a bit awkward – doesn't it?


 
  
 Why is it you are splitting your lines and making them double spaced? I would find it much easier to read if you didn't split your line and double space. Thanks.


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## Joe Bloggs

baldr said:


> It was also a mistake to allow myself to be engaged by an ad hominem critical post by a forum member. My worst mistake was to descend to the phallus-waving techs-narcissism established in the original post that publicly called me out. That does not serve any of the readers well. For the above two points, I apologize to all members of this forum, including the original poster. It is my experience that if one puts out slop, one gets it back. For that reason, I will not be further engaged on this matter.




While I have no idea what the original post was or whether you do indeed owe him an apology, calling the original-poster a "d**k-waving tech narcissist" and then apologizing to him for "descending to his level" doesn't really look like an apology to me.


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## coli

So... what's the conclusion? Someone write a tl;dr please?


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## KeithEmo

The problem with audibility tests is that they tend to be limited... and many of the ones that I see referenced over and over are _VERY_ limited (this often happens with other types of tests as well). For example, the Boston Audio Society test where they "introducing a Red Book CD record-playback loop into the signal chain" to test the audibility of the differences between high-res and "ordinary" digital audio files..... 
  
 The basic "logic" of that test is that:
  
 1) We all know that there are measurable differences between high-res and ordinary-res files (wider bandwidth; better S/N ratio).
 2) If we pass our high-res signals through a "CD quality loop" we will eliminate those differences.
 3) If we hear no difference when we do so, then we can reasonably conclude that the differences were inaudible (because we didn't hear a difference when we eliminated them). 
  
 Now, here are the "gaps" in their methodology:
  
 1) It was never confirmed that the differences _POSSIBLE_ between high-res files and regular ones were present in their samples. I don't recall seeing any test results showing that any of the sample content they chose actually included any signal content in the range of frequencies which high-res files can reproduce and ordinary files cannot.
  
 2) It was never confirmed that, if any of that "extra information" was there in their test samples, that their equipment could reproduce it. If we're trying to test whether anyone can hear the "ultrasonic content" present in a high resolution disc, we need to confirm both that it is present in our test samples, _AND_ that our test equipment (including the speakers) can reproduce it. (Unless we confirm all those unknowns, all we really know is that the high-res discs _MIGHT_ contain content that would be eliminated by passing it through the standard-res loop).
  
 In other words, what they did was logically equivalent to an audiologist handing you a headset, setting a dial on his equipment to 18 kHz, and declaring that "18 kHz was inaudible to you" when you didn't report hearing anything - without confirming whether his equipment was actually putting out a signal at 18 kHz, or whether the headphones he used could reproduce it. (In logical terms, if you heard the tone, that would confirm all three conditions; but your not hearing it only confirms that _EITHER_ you can't hear it, or the headphones can't produce it, or the output simply isn't working.) In their test, their null result proved that _EITHER_ there is nothing audible in a high-res signal that can't be reproduced perfectly by a standard-res signal, _OR_ that there was nothing present in the particular sample content they chose which couldn't be reproduced perfectly by a standard-res signal, _OR_ that, if there was something present in the particular sample content they chose which couldn't be reproduced perfectly by a standard-res signal, it couldn't be reproduced by their test setup anyway (perhaps because of some limitation of their disc player or speakers).
  
 To put it simply, in order to validate that their test procedure would actually test what it was intended to they should have:
  
 1) Tested their sample content to ensure that there was in fact some measurable difference between the high-res and the standard-res version.
 2) Confirmed that their electronic reproduction chain could accurately reproduce both signals and the differences between them.
 3) Used a measurement microphone to confirm that their speakers could accurately reproduce both signals, and the difference between them, _IN THE AIR_.
  
 At this point, they could state with validity that they were actually delivering the correct test signals, and so could expect accurate results.
  
 And, unfortunately, any "audibility testing" conducted by an individual is even more limited. You or I can only confirm whether a certain difference is audible with the content we currently have, and using the equipment we currently own (or have access to). The problem is that, by confirming that we hear no difference with our current music and equipment, we can't infer that there won't be some audible difference with the content and equipment we might own tomorrow or next year...  and, as I've noted before, we'd hate to spend money to put together a great collection of content, only to find out later that it has some audible shortcoming that's clearly audible with the new speakers we buy next year, or with some new recordings we may buy later. Therefore, the quest for "something that is audibly as good as possible" rather than simply "something that has no flaws that are audible at the moment" isn't really unreasonable.
  
 Incidentally, I agree with you 100% - that the biggest problem today is simply poor quality mastering (or even music that is deliberately mastered with minimal dynamic range and maximal loudness to appeal to a certain audience). But that is, as you say, a different battle.....  (However, I do think that, overall, it's "a good thing" that it is at least becoming obvious that customers do in fact care about audio quality... )
  
  
 Quote:


castleofargh said:


> for the interest of how things work, I always enjoy reading about jitter, digital and analog filters, LSB, oversampling, nyquist, and music's actual dynamic range. same for the ever growing information we have on psycho acoustic. but I fail to see why they should lead an argument about audibility? shouldn't audibility tests be the valid answer to audibility concerns?
> 
> and shouldn't we do our own audibility tests to come up with our own conclusions and choices? as far as I'm concerned, from time to time I notice a difference in the highres file, I convert it to 16/44 and abx it with the cd rip I have(if the differences aren't very obvious. with stuff like "random access memory", ABX really wasn't needed^_^). and sure enough the differences are still very much on the downsampled version. demonstrating that I'm listening to 2 different masters and not to highres vs CD. to me that's important so that we avoid fighting the wrong war.
> based on my tests, clearly my war is on mastering job, and never once was about highres vs cd where trying to ABX the highres file to a downsampled then upsampled file, resulted in guessing statistics for as long as I remember doing it. I am yet to find one album that is the same master but sounds different when played in highres. I've looked long and hard, and listened on a variety of gears throughout the years. nothing, nada, rien.
> ...


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## RRod

keithemo said:


> And, unfortunately, any "audibility testing" conducted by an individual is even more limited. You or I can only confirm whether a certain difference is audible with the content we currently have, and using the equipment we currently own (or have access to). The problem is that, by confirming that we hear no difference with our current music and equipment, we can't infer that there won't be some audible difference with the content and equipment we might own tomorrow or next year...  and, as I've noted before, we'd hate to spend money to put together a great collection of content, only to find out later that it has some audible shortcoming that's clearly audible with the new speakers we buy next year, or with some new recordings we may buy later. Therefore, the quest for "something that is audibly as good as possible" rather than simply "something that has no flaws that are audible at the moment" isn't really unreasonable.


 
  
 What year is this going to happen then, Keith? SACD came out in 1999, yet for some reason we aren't yet in a world where equipment is expected to be speced out beyond 20kHz or with 144dB of SNR; why might that be? And I agree with you that various published tests leave something to be desired, but I also think that you should probably also then go around the rest of the forum and ask all these people claiming "huge" differences with hi-res just how well their equipment is reproducing hi-res content. Personally, my E-MU delivers pretty flat performance up to 48kHz, and Sennheiser would have me believe my HD800s are only -3dB at 44.1kHz (which I can't really test myself, unfortunately). Intermodulation tests at 96ksps stay perfectly quiet even at high volumes (not so for 192ksps), so for up to that sample rate I'm at least somewhat sure my ABX results for hi-res testing aren't totally fallacious. I mean, what equipment would you have someone use that you would consider kosher?


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## Joe Bloggs

> 1) It was never confirmed that the differences POSSIBLE between high-res files and regular ones were present in their samples. I don't recall seeing any test results showing that any of the sample content they chose actually included any signal content in the range of frequencies which high-res files can reproduce and ordinary files cannot.
> 
> 2) It was never confirmed that, if any of that "extra information" was there in their test samples, that their equipment could reproduce it. If we're trying to test whether anyone can hear the "ultrasonic content" present in a high resolution disc, we need to confirm both that it is present in our test samples, AND that our test equipment (including the speakers) can reproduce it. (Unless we confirm all those unknowns, all we really know is that the high-res discs MIGHT contain content that would be eliminated by passing it through the standard-res loop).




We're getting somewhere here though. All those arguments made that hi-res e.g. fills in the gaps in those standard-res "stairsteps" even within 20kHz, arguments about "timing accuracy" of signals within 20kHz, 19kHz signals "fading in and out", etc. etc. At least those can be put behind if we agree that you need reproduced >22kHz signals for the redbook standard to be tested against, no?


----------



## KeithEmo

In order to do a valid test, you need to be able to "control for all the variables" - and, until you do so, then your test is _NOT_ 100% conclusive. I agree with you that a lot of the things people think they hear are probably just imagination, or placebo effect, and also that a lot of other things are so trivial or occur so rarely that they probably don't matter, but, as someone with a science background, I am forced to differentiate between scientific facts and generalizations that aren't based solidly on facts. 
  
 As for equipment, I do know that there are test microphones claimed to have response well past 50 kHz, so it should be possible to find an SACD, or some other form of high-res file, that has content which includes harmonics out to 30 kHz or 40 kHz, play it through an amplifier, and see if we can find a pair of headphones or set of speakers which our microphone shows are actually reproducing those harmonics. If we do find them, then we'll have a valid test setup; and, if not, then we'll be forced to conclude that we are simply unable, for technical reasons, to perform a conclusive test. (It happens all the time in "real scientific circles" - where someone has an interesting theory, but the limitations in current technology prevent them from being able to test it. In the case of audio, we might even conclude that, since we couldn't find any content that contained significant information at the frequencies in question, it really doesn't matter... however, even then, we couldn't conclude that,_ IF_ such content existed, the difference wouldn't be audible.)
  
 In other words, we need to be very specific in the way we phrase our questions.
  
 Here are several different questions. Think very carefully about the answer to each - based on the results of all the tests you've read about,:
  
 1) "Is it possible, under any conditions, using any recording equipment, and any content, for a human being to hear a difference between a high-res recording and a standard-res one?"
  
 2) "Has anyone proven conclusively, using current sources and current equipment, that audible differences between high-res and standard-res files exist?"
  
 3) "Has anyone proven conclusively, with every possible combination of current source material and current equipment, that audible differences between high-res and standard-res files do NOT exist?"
  
 4) "Has anyone proven conclusively that there is no difference between high-res and standard-res files that will be audible to any human being, with any possible combination of content and equipment that currently exists or might ever be developed?"
  
 I would summarize all of the information that I've read about so far as follows:
  
 Technically, high-res audio files have wider frequency response and a better S/N ratio than standard-resolution files. However, several tests of reasonable but limited scope have been conducted, and none of them has produced evidence to suggest that these differences are audible to a significant number of test subjects, under a variety of typical listening conditions. Another data point is provided by a large collection of anecdotal evidence from people who claim to hear a significant difference. However, at least so far, whenever these claims have been tested using scientific test methods, they have not been shown to have merit, so it seems reasonable to conclude that some or all of them are simply the result of expectation bias or the placebo effect.
  
 (To put it bluntly, many people out there believe quite strongly in things that, by available scientific information, seem less likely. And, every now and then, something widely believed by science to be true turns out _NOT_ to be true. I remember when we were all taught that "matter was made up of protons, neutrons, and electrons - which were the smallest indivisible units of matter"; as it turns out, this is a useful model, but we now know that it isn't in fact true. And the only "evidence" we have so far that "high-res files aren't audibly superior" is simply a lack of evidence that they are... which is sufficient justification for an assumption, but not a statement of fact.)
  
 To answer your final question directly, I would use a wider variety of equipment, in the hope of providing a good cross sample. Specifically, I would have added a variety of high-end DACs and at least a few pairs of electrostatic headphones to the test list - because I personally find them to do a good job of making tiny differences audible in general. I would also invite people, including both end users and equipment vendors, to submit equipment for consideration. And, if a difference was in fact audible with any of them, then I would be forced to conclude that "an audible difference exists"... but, if no difference showed up in the tests, I would be forced to state the list of equipment with which no difference was detected (and I would refrain from making generalizations that extended to equipment I hadn't tested). I would leave it to the person reading the test to draw their own conclusions as to whether the failure to demonstrate a difference with xxx pieces of equipment - as listed - was sufficient for them to decide "it probably didn't matter" or not.  
  
 (The fact is that, in all of the tests I've read about, a truly tiny percentage of all the audio equipment currently in existence was tested, which sure makes it a stretch to extend those conclusions to all of the equipment out there.)
  
 Quote:


rrod said:


> What year is this going to happen then, Keith? SACD came out in 1999, yet for some reason we aren't yet in a world where equipment is expected to be speced out beyond 20kHz or with 144dB of SNR; why might that be? And I agree with you that various published tests leave something to be desired, but I also think that you should probably also then go around the rest of the forum and ask all these people claiming "huge" differences with hi-res just how well their equipment is reproducing hi-res content. Personally, my E-MU delivers pretty flat performance up to 48kHz, and Sennheiser would have me believe my HD800s are only -3dB at 44.1kHz (which I can't really test myself, unfortunately). Intermodulation tests at 96ksps stay perfectly quiet even at high volumes (not so for 192ksps), so for up to that sample rate I'm at least somewhat sure my ABX results for hi-res testing aren't totally fallacious. I mean, what equipment would you have someone use that you would consider kosher?


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## KeithEmo

Absolutely....
  
 Most of the arguments that include pictures of jagged stairsteps, with explanations of how such an unruly looking signal can't possibly sound anything other than awful, are either gross exaggerations or deliberate attempts to scare people with bad science, and most of the remaining few are simply poor explanations put forth by folks who don't understand the science themselves. The 16/44k Red Book CD standard provides for excellent sound quality (although you may have to dig around to find CDs that are mastered well enough to prove the point).
  
 The audible differences between a high-res version of some particular content, and a Red Book version of it, which is produced from the same master, and with equal care, should be tiny at most. We're talking about (maybe) slight shifts in imaging which (might) be caused by microsecond shifts in phase relationships, and equally subtle alterations in how certain instruments (may) sound due to the loss or alteration of upper harmonics which many people classify as "ultrasonic" and "inaudible". The reason it's difficult to even do reliable comparisons is that the relatively simple process of converting from 192k to 44k, using the best commercial rate-conversion software available, produces differences of similar magnitude. (This makes it very difficult, if not impossible, to separate audible differences that might be due to the different sample rate from the tiny differences that are simply artifacts of the conversion process itself.)
  
 Where things get a little tricky is that not all information can be expressed as a frequency. For example, a square wave is a collection of several harmonics, in particular proportions, and in a particular phase relationship to each other. If the frequency response of a device or recording isn't correct, then square waves played through it will both look unusual and sound wrong. However, if the frequency response is correct, but the phase response is wrong, you can end up with a signal that contains all the right amounts of energy at each frequency, but has a waveform that looks nothing like the original. Since our hearing seems to work _mostly_ by analyzing energy like a spectrum analyzer, this difference is _mostly_ inaudible. However, some research suggests that we are in fact somewhat sensitive to some aspects of the shape of the waveform itself - like the precise arrival time of the leading edge of it. One theory is that, even if we include all the audible frequencies up to 20 kHz so that we don't hear anything missing, because the missing harmonics above 20 kHz contribute to the overall shape of the wave, some other aspect of our hearing (perhaps the mechanism that figures out spatial location from phase relationships) may detect that the wave shapes are now incorrect, which may result in a perceived shift in the location of that instrument in the sound stage. In other words, the basic claim is that, even though we don't "hear" sound above 20 kHz, some of that information above 20 kHz does in fact contribute to other things we perceive about the sound - like its location - or even some other as yet not fully defined detail. And so we somehow sense when that information is altered or discarded. There have been some tests that at least suggest that this may happen - but they are far from conclusive. There is also lots of anecdotal evidence that a lot of people claim to hear a subtle difference.
  
 However, to put this in "practical perspective", what we're talking about is a tiny bit of information that may be audible and, if it is audible, and if it manages to get picked up in the recording process, may contribute to a slight improvement in realism or fidelity. (Or, to put that perspective differently, I've heard one or two superbly recorded Red Book CDs that sound better than the vast majority of the high-res recordings I own... which proves just how close the capabilities of both formats are.)
  
 We are _NOT_ talking about some huge dramatic difference - we're talking about a difference far smaller than, for example, the difference between different speakers, or different phono cartridges.
  
 Quote:


joe bloggs said:


> We're getting somewhere here though. All those arguments made that hi-res e.g. fills in the gaps in those standard-res "stairsteps" even within 20kHz, arguments about "timing accuracy" of signals within 20kHz, 19kHz signals "fading in and out", etc. etc. At least those can be put behind if we agree that you need reproduced >22kHz signals for the redbook standard to be tested against, no?


----------



## RRod

keithemo said:


>


 
  
 Specifics, my good man. I'll agree with you that the final countdown, paradigm-shifting, über-proctored test of hi-res audibility done by whatever society should include a range of equipment randomized across the test subjects. But my question was more: "right now, if you had to make such a list, what would put on it, and would any of that stuff be within the reach of your mid-to-high-but-not-summit-fi member of this forum?"


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## old tech

Keith, in regard to one of you points on the methodology I do recall reading somewhere that the paper was criticized by a label on those grounds. The claim is that the SACDs and DVD-As used in the study were produced from 16/44 or 24/44 masters. If there is any truth behind the claim then more serious issues are raised. Have the labels misled consumers? Are they in effect saying that from 1999 to 2007, audiophiles were quite happy with 16/44 as long as they believe it to be hi Res? I think Moran responded to this and other criticisms over the years.


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## castleofargh

from my still limited understanding of a full sound system, I'm done with highres. it's as simple as that.  the only time I go get some, is when my friends keep telling me about some latest remaster and it's not available in 16/44.
 I never bought a SACD player because I find the tech idiotic and that's my consumer's way to vote, but then again there are a few albums I wish I could get.
 I would go back to concern myself with highres the day we get revolutionary improvements on speakers/headphones. when 0.1% THD stops being seen as amazing at the output of a headphone, then in my super basic logic, stuff below will matter again.  I know that's a narrow minded view of sound, not everything below will be masked by distortions, and that there is distortions and distortions. and maybe stuff happening in the time domain I know nothing about. but I just can't get myself to care about what's happening below -96db when I'm using gears that can't guaranty the first 60db to be clean.
  
  
 so I agree with you Keith that my own tests only talk about me and my gears at the moment. and I kind of believe that things could be different with better drivers and everything(also it might be wishful thinking), but that's the thing, I don't really expect any revolution in the transducer's domain. they make small progresses, but it's more in how cheaper stuff get closer to the good ones, than the good ones really putting the old good ones to shame. so I fail to have much expectations.
 my hopes are toward letting the sound engineers go back to doing their best instead of the loudest and fastest. we all agree on this, but in the world economy few studios will do that. 
	

	
	
		
		

		
		
	


	



 and also HRTF and DSPs becoming the usual thing.  but all I see are people who don't understand a thing, crying "stranger danger" anytime someone wants to touch their sound, and begging to be left alone in the 20th century.
  
 so I have hopes that sound can get better, but it doesn't involve highres audio and even then I'm not optimistic that it will come soon.


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## KeithEmo

Here goes (and you'll notice that most of it isn't expensive). Also, as an industry rep, I'm sort of precluded from "recommending" specific equipment. Again, and to be very clear, this is equipment that I am either both familiar with and consider to be pretty good at revealing differences in other equipment, or that has a reputation for being "neutral" and "revealing". This means that it lacks overwhelming colorations, and seems to me to be pretty close to neutral. (I'm not mentioning equipment that, even though I think it sounds good, doesn't seem to me to be very revealing.) Note that, since we're testing for audible differences, and not for personal opinions of "what sounds _good_", a lot of the equipment I would recommend would be characterized as "analytical" or "dry" to many audiophiles - which is just fine; after all, we are performing an analytical test.)
  
 1) I would use a separate DAC, so I would say that almost _ANY_ disc player would be fine. It should be tested to verify that it actually produces a bit perfect output, and doesn't upsample or otherwise alter the bitstream. I would probably pick the latest bottom Oppo model (at the moment that would be the 103), but anything that delivers bit-perfect response would be fine. Likewise, any computer should work fine as a source. Again, the only requirement would be that the playback software provides a bit-perfect output rather than re-sampling or doing anything else "interesting".
  
 2) I would use several middle-of-the-line DACs, and I would specifically avoid DACs with tube output stages, and models otherwise known for various euphonic colorations. Therefore, I would _EXCLUDE_: DACs with tube output stages, non-oversampling DACs, and DACs with "unusual" procesing built-in. An Emotiva DC-1, or a Schiit Gungnir, or a Wyred4Sound DAC2, or any Benchmark DAC should work fine. (Note that I am _NOT_ excluding Sabre DACs. Even though I believe that most of them introduce some euphonic coloration, I find it to be of the sort that emphasizes differences in other areas rather than covers them up.) If we're using a computer source, which would almost certainly suggest a USB connection, I would specify that the DAC have an _ASYNCHRONOUS_ USB input - to eliminate the major effects of jitter on the input signal. If the DAC includes the option of multiple user-selectable oversampling filters, I would let the test subject pick the one they like - but I would exclude any that specifically produce unusual frequency response (like "slow roll-off filters" which produce a significant drop at 20 kHz) - because they may mask differences that would otherwise be obvious. (It's probably easier just to avoid DACs with selectable filters and frequency responses other than very flat.)
  
 3) For speakers, I would strongly suggest models known for being revealing (note that this doesn't specify that they sound "good"). I've found our Emotiva Airmotiv line to be good at this, and I assume that most speakers with folded ribbon (or "true" ribbon) tweeters would be as well. I also assume that most electrostatic speakers would excel in this regard (although I'm not especially familiar with current models, and some are not). Again, I would specifically avoid speakers known for euphonic colorations (like high-efficiency horns, and speakers known to have rolled-off high ends). I would definitely avoid speakers with horn tweeters, vintage models with cone tweeters, and unusual designs with large heavy drivers equalized to produce high frequencies (like Bose 901's) - because I don't think most of them reproduce transients well.
  
 4) For amplifiers, I would suggest any "normal" solid state power amp (meaning a current model from.... Crown, Emotiva, Marantz, Parasound, Rotel, etc.). Again, I think most current middle of the road models would work well, but I would avoid tube amps and hybrids with tube stages, which are known to produce euphonic coloration. Since some Class D amps are known for being somewhat colored, and I'm not familiar enough with the various models to list which is which, I would stick with "standard Class A/B amps".  
  
 5) Alternately, to avoid the complexity and cost of selecting several good speakers, I would probably prefer headphones. I find electrostatics to be the most revealing: I think Koss ESP-950's would be an excellent choice (since they come with their own amplifier, they also eliminate that variable, and they cost less than $1000). Any of the Stax models I've heard would also work well. (Again, though, if using a separate headphone amplifier, I would avoid using a tube model, because many of them produce various euphonic colorations, which may include "smoothing over" differences in the program source.) Any of the higher-end dynamic models from Sennheiser, or AKG, or Beyerdynamic would probably also work well, although I find electrostatics to be more revealing. I would _AVOID_ planars because, while I think many of them sound quite good, I also find many to be somewhat colored - and in the direction of "smoothing the high end" - which seems to me to be likely to conceal differences in high-frequency and transient response.
  
 (You will also notice a preference on my part to choose components that are specifically good at reproducing transients and high-frequencies, and not to say much about low frequency response. This is mostly because, from personal experience, and from my technical theoretical knowledge, I don't expect high-res files to reproduce low frequencies any differently than standard-res files, but I _DO_ expect significant variations in that regard between various speakers and amplifiers. Therefore, I expect any differences that may be audible to be in the areas of high-frequency response or transient response, or related to them.)
  
 6) For test content, I personally happen to prefer rock and pop music, so I'm not all that familiar with many classical tracks - or how they should sound. Also, sadly, most rock and pop music isn't mastered especially well (IMHO). However, here are a few of the albums and tracks that I normally use to "show off high-res sound reproduction" and which I think would be good to include:
  
 - The Tempest (Reference Recordings - 176k)
 - The Eagles - Hotel California (24/192k HDTracks)
 - Grateful Dead - Studio Album Remasters - American Beauty (24/192 HDTracks)
 (slightly problematic for testing high-res because the new remasters have been "heavily restored", and so sound very different from the originals)
 - Kodo - Tataku (SACD)
 - Pink Floyd - Dark Side of the Moon (high-res version from the Immersion set)
 - Alison Krauss - New Favorite (SACD)
  
  
 Quote:


rrod said:


> Specifics, my good man. I'll agree with you that the final countdown, paradigm-shifting, über-proctored test of hi-res audibility done by whatever society should include a range of equipment randomized across the test subjects. But my question was more: "right now, if you had to make such a list, what would put on it, and would any of that stuff be within the reach of your mid-to-high-but-not-summit-fi member of this forum?"


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## interpolate

To be honest guys, you only get what you pay for and the rest is just a commercially induced placebo effect.
  
 24-bit is just a binary range of possible values right. In digital it's purely mathematical and has to be converted to analogue again. Where your limitation is the human hearing range.
  
 I suppose the best way to look at it is, if you have the studio masters in FLAC or WAV format for example then it's how the artist intended.


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## KeithEmo

At one level you're entirely wrong.... or may be. The whole point of this discussion is the question of whether the current digital technology has in fact reached the limit of what we can hear (that's a technical question). For example, if you or I can in fact hear some audible difference between two recordings, or two DACs, then either one of them fails to be better than the limits of our hearing, or both of them fail. (If they sound different, then, at most, _ONE_ of them can be right, and perhaps neither. If we can hear a difference, then both _CANNOT_ be audibly perfect.) You've only reached the level of "just placebo effect" once you've passed that point.
  
 However, at another level, you have a point. If the "master itself" isn't much like the "original" (whatever we mean by that) to begin with, then delivering actual fidelity to that master may be sort of moot. Now, when you're talking about physical instruments, then there is some sort of absolute standard... regardless of who's playing it, or how good or bad the composition he's playing is, a recording of a flute should in fact sound like a flute. However, with modern music, often there is no "actual original" - and, in fact, the "band" may have recorded separate tracks, in different studios, in which the "original" only exists in the mind of the mixing engineer, or the instruments themselves may only be "sounds" created in a computer.
  
 And, yes, if I were to "Photoshop a picture for a web page", I might not worry about artifacts and flaws that I know won't be visible when people look at that picture scaled to the size of a postage stamp, it's also possible that an artist who knows that his or her recording will be distributed in MP3 format may allow flaws and imperfections such that it really won't sound better when reproduced at 16/44 (or 24/192) - because they know that their audience will never hear it that way. (In fact, for years, some artists have deliberately added "record ticks" to certain tracks, and phony film scratches to videos, for "artistic reasons".) And, in that case, you might fairly argue that there's nothing to be gained by reproducing the poor quality original at better quality.
  
 You also get into a sort of philosophical argument in that some artists and engineers may actually create a master that has been "tweaked" specifically "to sound good on certain equipment". For example, a song that has been mixed specifically to sound good on a car radio, may have anomalies designed to "work well with the way a car radio sounds", and may actually only "sound as the engineer intended", when played on a car radio - in which case you might argue that "playing it on a better system will _NOT_ sound as he intended it" - and may actually sound worse. This argument usually comes up in the context of whether it's more accurate to listen to an album using "studio monitors" (so you can hear what the mixing engineer heard), or on home speakers (which is what you can assume the engineer expected you to listen on).
  
 (If you were planning to display a painting by Rembrandt, would it be more accurate to display it in "pure white light", or in the light of lamps like those that may have been used in Rembrandt's studio, or in light that matches the light in the room where it was intended to hang when it was originally commissioned? Reasonable arguments can be made for all of those possibilities.)
  
 However, to most audiophiles, the overriding idea is that we want to hear the recording exactly as the artist or engineer intended it, and, to be sure we do, we would want to have a system that can reproduce whatever is in that recording with an accuracy that exceeds our ability to notice the differences. (But, yes, the requirements of a system that can achieve that may vary depending on the source itself.)
  
  
 Quote:


interpolate said:


> To be honest guys, you only get what you pay for and the rest is just a commercially induced placebo effect.
> 
> 24-bit is just a binary range of possible values right. In digital it's purely mathematical and has to be converted to analogue again. Where your limitation is the human hearing range.
> 
> I suppose the best way to look at it is, if you have the studio masters in FLAC or WAV format for example then it's how the artist intended.


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## RRod

keithemo said:


>


 
  
 Thanks for the elaboration, and basically confirms that I've given hi-res its own fair chance as far as my own personal testing.


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## RRod

keithemo said:


> Where things get a little tricky is that not all information can be expressed as a frequency. For example, a square wave is a collection of several harmonics, in particular proportions, and in a particular phase relationship to each other. If the frequency response of a device or recording isn't correct, then square waves played through it will both look unusual and sound wrong. However, if the frequency response is correct, but the phase response is wrong, you can end up with a signal that contains all the right amounts of energy at each frequency, but has a waveform that looks nothing like the original. Since our hearing seems to work _mostly_ by analyzing energy like a spectrum analyzer, this difference is _mostly_ inaudible. However, some research suggests that we are in fact somewhat sensitive to some aspects of the shape of the waveform itself - like the precise arrival time of the leading edge of it. One theory is that, even if we include all the audible frequencies up to 20 kHz so that we don't hear anything missing, because the missing harmonics above 20 kHz contribute to the overall shape of the wave, some other aspect of our hearing (perhaps the mechanism that figures out spatial location from phase relationships) may detect that the wave shapes are now incorrect, which may result in a perceived shift in the location of that instrument in the sound stage. In other words, the basic claim is that, even though we don't "hear" sound above 20 kHz, some of that information above 20 kHz does in fact contribute to other things we perceive about the sound - like its location - or even some other as yet not fully defined detail. And so we somehow sense when that information is altered or discarded. There have been some tests that at least suggest that this may happen - but they are far from conclusive. There is also lots of anecdotal evidence that a lot of people claim to hear a subtle difference.


 
  
 So what would it take to convince someone on such a point. My feeling is that I've taken an ideal 1k square wave at a silly high rate (256*48000), decimated it down to a 96ksps file A, sinc filtered everything above 20kHz to make file B, ABXed A and B and haven't hear a whit of difference, and that's getting pretty much as many leading edges into my ears as I can. Any phase errors introduced by either the resampling or my DAC+amp would be in there too.
  
 The frustrating thing is that such a test is considered, on most of this site at least, nothing next to some guy "sensing" a difference in his transients in an sighted evaluation of two different masters of a recording made on tape.


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## JWolf

A 1k square sine wave is worthless as a test when talking about complex music.


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## RRod

jwolf said:


> A 1k square sine wave is worthless as a test when talking about complex music.


 
  
 Point proven.


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## interpolate

Don't get me wrong what high res recordings or sacd quality masters i own are subliminally different rather than distinctly audibly different. 

My own 24-bit files are less susceptible to aliasing when mixed with other waveforms.....whatever that means.


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## KeithEmo

I agree with you entirely - and I'm afraid I've been playing Devil's advocate here because I get very OCD when people make generalizations. In science, it is very difficult to prove absolute generalizations, especially when a subject as variable as human beings is involved. And, while I'm inclined to take the side that the majority of weird things that audiophiles believe really are the result of placebo effect or some other sort of expectation bias, I've also seen a few that weren't - which makes me unwilling to make blanket statements without lots of corroborating evidence. (And I can name you several instances of blanket statements about audio that have in fact turned out to be wrong, or to have easily demonstrable exceptions.)
  
 So, to answer your question.... Because the differences themselves are easily measured, and all we're talking about is the limits of human perception, which I don't believe have been fully explored, it would take a lot of evidence to convince me of the generalization - and I don't honestly believe anyone is going to expend the effort and expense necessary to explore the subject that thoroughly.
  
 Therefore, I would much rather simply rephrase it as a practical question rather than an absolute generalization.....
  
 - Have I ever heard a difference between a high-res file and the equivalent standard-res file where I could say with certainty that the difference was audible BECAUSE the file was a high-res file and wasn't due to mastering differences or conversion artifacts? No.
  
 - Am I aware of any test that has shown that a statistically significant number of people could hear such differences? No.
  
 - Would I suggest that someone should buy the high-res version of a file even if they can't hear a difference? Probably not.
 (But I wouldn't discourage someone from buying a high-res version of an album that sounded better - even if there was a distinct possibility that the real difference was due to different mastering or something else.)
    


rrod said:


> So what would it take to convince someone on such a point. My feeling is that I've taken an ideal 1k square wave at a silly high rate (256*48000), decimated it down to a 96ksps file A, sinc filtered everything above 20kHz to make file B, ABXed A and B and haven't hear a whit of difference, and that's getting pretty much as many leading edges into my ears as I can. Any phase errors introduced by either the resampling or my DAC+amp would be in there too.
> 
> The frustrating thing is that such a test is considered, on most of this site at least, nothing next to some guy "sensing" a difference in his transients in an sighted evaluation of two different masters of a recording made on tape.


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## JWolf

The more bits there are the more accurate the sampling.  If you convert an analogue signal to digital and back again you introduce quantisation errors, in both processes, leading to distortion.  With 24 bits the size of the errors are about one thousandth of those occurring when 16 bits are used.  This is what is relevant, not the dynamic range (which, as mentioned, is adequately covered by 16 bits).


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## RRod

jwolf said:


> The more bits there are the more accurate the sampling.  If you convert an analogue signal to digital and back again you introduce quantisation errors, in both processes, leading to distortion.  With 24 bits the size of the errors are about one thousandth of those occurring when 16 bits are used.  This is what is relevant, not the dynamic range (which, as mentioned, is adequately covered by 16 bits).


 
  
 You have assumed that, because the quantization errors are smaller, they are less audible, which is of course the exact thing that you need to prove.


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## Arunabha Lahiri

Yes I agree. But the quantization error is hardly noticeable in 16 bit audio in normal audio playback volume. And nobody listen to music that loud, it will cause permanent deafness. So I guess that's another argument against 24bit audio.


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## Arunabha Lahiri

rrod said:


> You have assumed that, because the quantization errors are smaller, they are less audible, which is of course the exact thing that you need to prove.




Actually he is right as quantization errors are smaller, they are less audible. Higher bit rates do result less quantization noise. And higher bit rate do mean higher dynamic range. There is more difference in the loudest part and the softest part. A 16 bit recording has a dynamic range of near 120 dB. A 24 bit recording has max dynamic range 144dB.a human ear has dynamic range of 140 dB. So 16 dB recording is more than enough to completely cover up the quantization noise. In fact to hear the quantization noise there should be a music volume of 96dB higher than the quantization noise. And that's absurd. So 16 bit is more than enough. 24 bit is kinda unnecessary. Though human mind works in a strange way. And belief is the greatest religion in the universe.


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## RRod

arunabha lahiri said:


> Actually he is right as quantization errors are smaller, they are less audible. Higher bit rates do result less quantization noise. And higher bit rate do mean higher dynamic range. There is more difference in the loudest part and the softest part. A 16 bit recording has a dynamic range of near 120 dB. A 24 bit recording has max dynamic range 144dB.a human ear has dynamic range of 140 dB. So 16 dB recording is more than enough to completely cover up the quantization noise. In fact to hear the quantization noise there should be a music volume of 96dB higher than the quantization noise. And that's absurd. So 16 bit is more than enough. 24 bit is kinda unnecessary. Though human mind works in a strange way. And belief is the greatest religion in the universe.


 
  
 If the quantization on 16 bits is already inaudible under normal listening conditions, then it isn't any less audible in 24-bits. So we're in agreement but for semantics.


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## Arunabha Lahiri

rrod said:


> If the quantization on 16 bits is already inaudible under normal listening conditions, then it isn't any less audible in 24-bits. So we're in agreement.




Yes we are.. Technically we both are stating the obvious.


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## coli

jwolf said:


> The more bits there are the more accurate the sampling.  If you convert an analogue signal to digital and back again you introduce quantisation errors, in both processes, leading to distortion.  With 24 bits the size of the errors are about one thousandth of those occurring when 16 bits are used.  This is what is relevant, not the dynamic range (which, as mentioned, is adequately covered by 16 bits).


 
 Actually, I have never heard any difference between 24/16bits. Sample rates yes, but not bit rates. And I have a very accurate setup nowadays.
  
 Edit: I'll take that back. I have heard some really great sounds out of movie mixes but I'm not going to make them 16bit and compare...


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## 441879

Great thread folks. I didn't read every post, but definitely enough to convice me that the limitations on my system have nothing to do with bit rates. CD quality recordings, properly mastered, are good enough for my 50 year old ears and if I want better sound I should spend more on my headphones.


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## KeithEmo

Absolutely. I've heard enough regular CDs to demonstrate that they can sound remarkably good. The sad part is how many of them fail to _live_ up to that level of sound quality.
  
 Quote:


will f said:


> Great thread folks. I didn't read every post, but definitely enough to convice me that the limitations on my system have nothing to do with bit rates. CD quality recordings, properly mastered, are good enough for my 50 year old ears and if I want better sound I should spend more on my headphones.


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## aphex27

http://www.realhd-audio.com/?p=74
  
 I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.
  
 Exactly what my ears heard.


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## JWolf

aphex27 said:


> http://www.realhd-audio.com/?p=74
> 
> I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.
> 
> Exactly what my ears heard.


 
  
 The article linked from the article you linked isn't there.


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## aphex27

jwolf said:


> The article linked from the article you linked isn't there.


 

 Can you clarify? I just linked 1 article..it doesn't work?


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## JWolf

aphex27 said:


> Can you clarify? I just linked 1 article..it doesn't work?


 
  
 There a link about DSD in the article you linked that doesn't work.


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## aphex27

jwolf said:


> There a link about DSD in the article you linked that doesn't work.


 

 Could you take a screenshot? I don't think there are any links in that interview..


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## RRod

aphex27 said:


> http://www.realhd-audio.com/?p=74
> 
> I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.
> 
> Exactly what my ears heard.


 
  
 Your ears have either heard it blind or they haven't. If they haven't, your arguments won't get much traction around here. If they have, then we'd love to know which track you can differentiate.


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## aphex27

rrod said:


> Your ears have either heard it blind or they haven't. If they haven't, your arguments won't get much traction around here. If they have, then we'd love to know which track you can differentiate.


 

 And we'd love to have you dispute this interview with arguments. (and by we I mean I)


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## RRod

aphex27 said:


> And we'd love to have you dispute this interview with arguments. (and by we I mean I)


 
  
 Well, we can start with the horse's mouth, if indeed this is the same Mr. Waldrep:
 http://www.youtube.com/watch?v=Z5S_DI99wd8&t=3m38s


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## aphex27

rrod said:


> Well, we can start with the horse's mouth, if indeed this is the same Mr. Waldrep:
> http://www.youtube.com/watch?v=Z5S_DI99wd8&t=3m38s


 

 I'm sorry, I don't think you understood. Mr. Waldrep is the interviewer. John Siau is the interviewee. I am talking about Mr. Siau's presentation of facts.
  
 (Btw Waldrep's hate of pono (and rightly so) is well documented, I don't understand what the video you linked has to do with this discussion)


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## aphex27

Maybe this will help clear up my first post
  
 "
John Siau 
MAY 24, 2013 AT 1:16 PM

 Where does Benchmark stand on DSD vs. PCM, and why?
 1. Benchmark recognizes that DSD (64x and higher) has significant advantages over 44.1/16 PCM.
 2. Benchmark recognizes that high-resolution PCM (96/24 and higher) has significant advantages over 44.1/16 PCM.
 3. Benchmark’s measurements and calculations show that the performance of 64x DSD is almost identical to the performance of 20-bit 96 kHz PCM (the in-band SNR of 64x DSD is about 120 dB). 64x DSD does not have any time-domain, frequency domain, or linearity advantage over 96 kHz PCM. DSD marketing materials have been very misleading.
 4. 24-bit 96 kHz PCM has a 24 dB noise advantage over 64x DSD (144 dB vs. 120 dB), but this 24 dB noise difference is completely masked by the noise produced by other components in our recording and playback systems, and by the noise limitations of our recording and listening spaces.
 5. Benchmark recognizes that 64x DSD and 96/24 PCM formats outperform most of the recording and playback chain. Bandwidth of either digital transmission system meets or exceeds the bandwidth of our microphones, amplifiers, and speakers. Likewise, the SNR of either digital transmission system meets or exceeds the noise performance of microphones, microphone preamplifiers, and power amplifiers. In addition, these digital transmission systems both exceed the performance of most A/D and D/A converters. 64x DSD and 96/24 PCM are not the factors limiting the performance of our audio systems. Focusing on DSD vs. PCM will distract us from much bigger issues in the recording and playback chain. Any sonic advantage of one digital system over the other will be very small when compared to improvements that can be made in other parts of the signal chain.
 6. 64x DSD and 96/24 PCM both offer excellent sonic performance as distribution formats. PCM is more compact, but DSD provides better copy protection (a frustration to those of us who use music servers, but an important consideration for copyright holders).
 7. Every A/D and D/A converter that Benchmark has produced uses Sigma-Delta conversion with equally-weighted 1-bit conversion elements. Benchmark never used multi-level conversion because of the THD issues caused by the linearity errors that are common to all multi-level systems. Benchmark has always placed high priorities on THD and linearity, at the expense of SNR. Sigma-delta 1-bit DACs tend to produce more noise than multi-level systems (such as ladder DACs), but the 1-bit systems achieve near-perfect linearity, which in our opinion is much more important than SNR. Benchmark has overcome the noise limitations of 1-bit conversion through the use of parallel 1-bit conversion systems. Our DAC2 sums the outputs of four balanced converters. Each of these four converters has sixteen equally-weighted balanced 1-bit converters (for a total of 64) that are summed together to improve the SNR of the system. These 64 1-bit converters can be driven from a 32-bit PCM signal, or from a 1-bit DSD signal. Either way, the performance is nearly identical, and none of the multi-bit THD issues exist. In this sigma-delta configuration there is almost no difference between the in-band performance of PCM vs. DSD. The only measurable difference at the output of the DAC2 is that 64x DSD signals produce about 8 dB more noise in-band than 96/24 PCM (due to the SNR limitations of DSD). Ultrasonic noise is not an issue at the output of the DAC2 because we are careful to remove the ultrasonic noise produced by DSD noise shaping. These same filters also remove the ultrasonic images that are always produced by D/A conversion (DSD or PCM).
 8. The ultrasonic noise produced by DSD noise shaping must be removed after D/A conversion. It cannot be removed from the DSD signal before D/A conversion. This noise is due to the 6-dB SNR of the 1-bit DSD transmission system. Aggressive noise-shaping must be used in the DSD A/D, and at least once more in the mastering process. This noise-shaping is used to achieve an excellent SNR in the audible band by moving most of the 1-bit quantization noise to ultrasonic frequencies. Each time this process is applied, the quality of the DSD audio degrades (noise and distortion both increase). For this reason, the quality of DSD degrades very quickly in the mixing and mastering process. DSD has produced impressive results when the mixing and mastering processes have been omitted from the signal chain. To date, most of the DSD vs. PCM listening tests have omitted these processing steps. Unfortunately very few recordings can be produced without some mixing, editing, and mastering. Cascaded DSD noise-shaping processes should be avoided. For this reason, Benchmark does not recommend recording and mixing in DSD.
 9. The 24-dB noise advantage that 24/96 PCM has over 64x DSD begins to become significant in the mixing and mastering processes. In terms of in-band noise, each DSD noise-shaping process is equivalent to at least 16 cascaded 24-bit dither processes. In terms of distortion, there is no comparison; the DSD noise-shaping process adds distortion while the PCM dithering process is distortion-free.
 10. If the ultrasonic noise of DSD is not removed after D/A conversion, it will usually cause distortion in the playback system. The slew-rate limitations of most power amplifiers will fold the ultrasonic noise into the audible band causing distortion that is not harmonically related to the music. If the power amplifier has sufficient slew rates to pass the ultrasonic frequencies, similar problems will occur in the speakers. For these reasons, the ultrasonic noise must be removed from a DSD source after D/A conversion or before amplification.
 11. Benchmark introduced 64X DSD on the new Benchmark DAC2 converter family. This gives our customers the ability to play DSD recordings in native format. Existing DSD recordings should not need to be converted to PCM to be enjoyed on a Benchmark converter.
 12. Currently there is no practical way to play SACD disks through a high-quality outboard converter. SACD copy protection holds most existing DSD recordings captive to the limited quality of the low-cost conversion systems built into SACD players. It is our hope that many of the fine recordings that exist on SACD disks will be released for purchase as DSD downloads.


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## RRod

So the counter-argument is that you:
 a) Don't have any music that even pushes 16-bits in terms of audibility of quantization distortion
 b) Won't hear anything from a hi-res file that is high-passed over 20kHz
  
 Feel free to show how either of these statements is false. 24bits has advantages for recording that I don't think anyone argues with, but for playback there is no need for the extra bits. What high frequencies are useful for I haven't a clue. Perhaps you could elaborate on Mr. Siau's statement:
 "Benchmark recognizes that high-resolution PCM (96/24 and higher) has significant advantages over 44.1/16 PCM."
  
 since he doesn't explicitly list the advantages. The only thing he really seems to harp on is THD and linearity, and again feel free to show you can hear the advantages in these outcomes in hi-res/DSD recordings in blind test conditions.
  
 Also, once again, we're talking about someone with skin in the hi-res game.


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## aphex27

That's the thing..they're talking about a very different thing (DSD vs PCM), not trying to prove anything with regards to 16/44.1 vs 24/96..
 Benchmark has no horse in this race..they do both DSD and PCM..the guy is explaining his honest opinion based on scientific facts (not that i understand them , I'm not an electrical engineer)
  
 "
*MW:* Wasn’t archiving their whole reason for coming up with it in the first place? It was going to be used to take their analog masters in their vault and putting in a format that they thought would preserve the most fidelity, right?
*JS:* Yeah. And conceptually it looked like a simple approach. And, DSD significantly outperformed the 16-bit PCM systems that were common at the time. As a distribution format, DSD is definitely a big step above 44/16 CDs, and we want to give people the best possible playback of the wonderful DSD recordings that already exist.
*MW:* And they tried to put in the successor to the CD and that’s where we got a format war.


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## RRod

aphex27 said:


> That's the thing..they're talking about a very different thing (DSD vs PCM), not trying to prove anything with regards to 16/44.1 vs 24/96..
> Benchmark has no horse in this race..they do both DSD and PCM..the guy is explaining his honest opinion based on scientific facts (not that i understand them , I'm not an electrical engineer)
> 
> "
> ...


 
  
 Well he started off your list above with the comment about hi-res and DSD both being superior to Redbook. And making both hi-res PCM and DSD, he would seem to have TWO horses in the race.
  
 As far as the quote there, he says "outperformed", so where are the results that DSD *audibly* outperformed Redbook PCM in *blind* test conditions? I'm all fine with companies arching in hi-res "just in case," but at the end of the day what matters is what we, the listeners, hear. To that end, hi-res PCM and DSD are useful for making recordings that can optimally mapped to 16/44.1, but for delivery good old Redbook is just fine. So again, what actual blind listening tests are you calling upon for your claim:


aphex27 said:


> I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.


----------



## aphex27

I'm calling upon the authority of a master electrical engineer. But I see where where you're coming from. How could I set up an unambiguous test? I think I have pretty good ears..my only condition is that I can choose the music (not some lady gaga crap)


----------



## castleofargh

aphex27 said:


> http://www.realhd-audio.com/?p=74
> 
> I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.
> 
> Exactly what my ears heard.


 
 funny how reading the same article, I come to totally different conclusions. 
	

	
	
		
		

		
			




 the benchmark guy spends pretty much the all time saying in as kind a way as he can muster, how DSD is a nonsense format and makes everything more complicated for everybody. for the master engineer, for the DACs, for the end user. all that for an effective resolution no better than highres PCM. I have a hard time reading praises for DSD in this article.
  
 he talks about the possible resolutions of the different formats, not about audibility. only you did that.
 nobody here will challenge the fact that DSD can measure better in the 20hz-20khz than 16/44. it's not a mystery, not news, and not a debate.
 it's when you say you hear the difference, that we start to have more doubts. even more so when you don't care to explain how you made sure that you really heard a difference.


----------



## RRod

aphex27 said:


> I'm calling upon the authority of a master electrical engineer. But I see where where you're coming from. How could I set up an unambiguous test? I think I have pretty good ears..my only condition is that I can choose the music (not some lady gaga crap)


 
  
 The thing is other master electrical engineers don't think people need to be paying any more $$ for hi-res, so really at this point people just need to decide for themselves. It doesn't seem like anyone has the time/resources/interest to do a new large-scale test of this stuff.
  
 The first thing to setting up a test is to verify that your system isn't doing things like producing audible IMD with hi-res files. See the intermodulation tests here. Once you've got that then the typical recommendation is to use the ABX utility of foobar. Being on Linux and hating black-boxes, I resample in SoX, first taking the track down to Redbook and then taking it back up to hi-res (to avoid biases from how your system handles switching sampling rates). We also have to agree on how to handle problems, e.g. the resampling causing audible clipping.


----------



## aphex27

rrod said:


> The thing is other master electrical engineers don't think people need to be paying any more $$ for hi-res, so really at this point people just need to decide for themselves. It doesn't seem like anyone has the time/resources/interest to do a new large-scale test of this stuff.
> 
> The first thing to setting up a test is to verify that your system isn't doing things like producing audible IMD with hi-res files. See the intermodulation tests here. Once you've got that then the typical recommendation is to use the ABX utility of foobar. Being on Linux and hating black-boxes, I resample in SoX, first taking the track down to Redbook and then taking it back up to hi-res (to avoid biases from how your system handles switching sampling rates). We also have to agree on how to handle problems, e.g. the resampling causing audible clipping.


 
  
  


castleofargh said:


> funny how reading the same article, I come to totally different conclusions.
> 
> 
> 
> ...


 

 Good stuff dudes..i wish I could continue but don't have the time..cheers


----------



## RRod

aphex27 said:


> Good stuff dudes..i wish I could continue but don't have the time..cheers


 
 Ta.


----------



## XenHeadFi

aphex27 said:


> http://www.realhd-audio.com/?p=74
> 
> I have been convinced. No subjectivity about it. 16/44.1<<<<<<<<<<<DSD(kinda=)96/24.
> 
> Exactly what my ears heard.


 
  
 Did you read what you posted? Siau clearly lists several reasons why DSD is WORSE than 96/24. See points #4 (PCM has better SNR), #8 (DSD requires "Aggressive noise shaping" to remove ultrasonic noise introduced by the DSD format itself. "[T]he quality of DSD degrades very quickly" when processed, like from mixing and mastering.), #9 (PCM's 24 DB SNR advantage is a boon to mixing and mastering, not including the degradation of processing DSD.)  The only benefit over 96/24 is #6 (DSD is able to implement DRM).
  
 No where does he say that 96/24 is better for playback. He makes it clear that 96/24 exceeds all current ADC and DAC.


----------



## aphex27

xenheadfi said:


> *Did you read what you posted?* Siau clearly lists several reasons why DSD is WORSE than 96/24. See points #4 (PCM has better SNR), #8 (DSD requires "Aggressive noise shaping" to remove ultrasonic noise introduced by the DSD format itself. "[T]he quality of DSD degrades very quickly" when processed, like from mixing and mastering.), #9 (PCM's 24 DB SNR advantage is a boon to mixing and mastering, not including the degradation of processing DSD.)  The only benefit over 96/24 is #6 (DSD is able to implement DRM).
> 
> *No where does he say that 96/24 is better for playback*. He makes it clear that 96/24 exceeds all current ADC and DAC.


 
 lol
  
*JS:* Yeah. And conceptually it looked like a simple approach. And, DSD significantly outperformed the 16-bit PCM systems that were common at the time. As a distribution format, DSD is definitely a big step above 44/16 CDs, and we want to give people the best possible playback of the wonderful DSD recordings that already exist.
*MW:* And they tried to put in the successor to the CD and that’s where we got a format war.
*JS:* Yep. Moving forward, we should focus on 24/96, and 24/192 downloads as these formats offer the best quality available.


----------



## aphex27

castleofargh said:


> funny how reading the same article, I come to totally different conclusions.
> 
> 
> 
> ...


 

 Ok, thanks, I understand your point. To be honest what I understood from the article is that hi res PCM is better than CD quality in terms of reproduction (and obviously production) and I assumed he meant within the human hearing range. Not that I understand any of the science behind it


----------



## castleofargh

aphex27 said:


> castleofargh said:
> 
> 
> > funny how reading the same article, I come to totally different conclusions.
> ...


 

 well because of noise shaping and low pass filtering, SACD resolution is a lot less than what the data number would suggest. but it's still very good, there is no debate about this either. just like highres has the potential to be better than CD. the question being what music we record on it? and what is our own hearing threshold for those resolutions? 
 when you take an overly compressed justin bieber, increasing the file resolution is only adding zeroes everywhere while really doing nothing for the signal. and in practice, we seem to fail to hear anything past 20khz(and actually below that for the average adult), so CD can do it. and we seem to fail to hear something 80db below music, so is there really a need for more than the dynamic range of a CD?
 most logic and most tests agree that it's going from mighty hard to impossible to tell CD from higher resolutions, that's why people like me get suspicious when I read about hearing superiority of a format. because too often what people call the audible superiority of a format is expectation bias, or listening on different devices so one can of course sound better than the other without the format having anything to do with it, some don't notice that they are listening to 2 different masters... that's why we ask for as much controls as possible during a listening test, to make sure all those possibilities are not mistaken for the actual format differences. 
 but that doesn't mean I reject the measurable superiority of the format. highres can do better than CD and measurably so. that's a fact. when we argue about audibility we never argue about measurement, those are separate matters and only a few lost lambs can't see how they don't always have to be one and the same.
 some like me think "I fail to hear better then CD, so CD is good enough", others think "why settle with good enough when you can have higher resolution?". both are legit choices IMO. there isn't really a wrong answer to making our own choices for our own reasons ^_^.


----------



## aphex27

castleofargh said:


> well because of noise shaping and low pass filtering, SACD resolution is a lot less than what the data number would suggest. but it's still very good, there is no debate about this either. just like highres has the potential to be better than CD. the question being what music we record on it? and what is our own hearing threshold for those resolutions?
> when you take an overly compressed justin bieber, increasing the file resolution is only adding zeroes everywhere while really doing nothing for the signal. and in practice, we seem to fail to hear anything past 20khz(and actually below that for the average adult), so CD can do it. and we seem to fail to hear something 80db below music, so is there really a need for more than the dynamic range of a CD?
> most logic and most tests agree that it's going from mighty hard to impossible to tell CD from higher resolutions, that's why people like me get suspicious when I read about hearing superiority of a format. because too often what people call the audible superiority of a format is expectation bias, or listening on different devices so one can of course sound better than the other without the format having anything to do with it, some don't notice that they are listening to 2 different masters... that's why we ask for as much controls as possible during a listening test, to make sure all those possibilities are not mistaken for the actual format differences.
> but that doesn't mean I reject the measurable superiority of the format. highres can do better than CD and measurably so. that's a fact. when we argue about audibility we never argue about measurement, those are separate matters and only a few lost lambs can't see how they don't always have to be one and the same.
> some like me think "I fail to hear better then CD, so CD is good enough", others think "why settle with good enough when you can have higher resolution?". both are legit choices IMO. there isn't really a wrong answer to making our own choices for our own reasons ^_^.


 

 Agree completely..the recording/mastering is by far the most important part of the chain. So maybe people who make records in hi-res (I believe most of them do in fact want to give us a better listening experience) have better recording standards and we confuse the better recording for the format. On a somewhat related sidenote, I've been listening to the Beatles albums that came out on Tidal yesterday. These are are the 2009 remasters which I'd never heard before. I did a little research and found out that these remasters were a 5 year project by some very competent engineers. I know every Beatles song inside/out and can easily say give me these remasters in 192kbps over any previous version in DSD128 ..


----------



## XenHeadFi

aphex27 said:


> lol
> 
> *JS:* Yeah. And conceptually it looked like a simple approach. And, DSD significantly outperformed the 16-bit PCM systems that were common at the time. As a distribution format, DSD is definitely a big step above 44/16 CDs, and we want to give people the best possible playback of the wonderful DSD recordings that already exist.
> *MW:* And they tried to put in the successor to the CD and that’s where we got a format war.
> *JS:* Yep. Moving forward, we should focus on 24/96, and 24/192 downloads as these formats offer the best quality available.


 
  
 I apologize as I just read the bulleted list that you posted. Now, I've read the entire interview and the part you quoted comes at the very end without any explanation or reasons why it is better. Siau does say it, so I am wrong in my post.
  
 The vast majority of the article was spent pretty much saying DSD is a horrible format but that Benchmark makes the best DSD DAC to handle its playback. I didn't know that DSD pretty much filters out everything from 47kHz and up or how limiting DSD is for playback usability. As a user, a DSD file pretty much has to be converted to PCM before crossfades/fadeouts/fadeins/gapless playback can be implemented from our music players, but they can't if that DSD file has DRM, like for SA-CD. Even if you could, "consumer" DSD is 1-bit so any manipulations introduces artifacts. Another horrible attempt by Sony to lock-in users into a format that is extremely (edit: NOT) user-friendly and only offers stronger DRM as a benefit (Memory Sticks, UMD, SecureROM even on audio CDs!, etc).
  


aphex27 said:


> Agree completely..the recording/mastering is by far the most important part of the chain. So maybe people who make records in hi-res (I believe most of them do in fact want to give us a better listening experience) have better recording standards and we confuse the better recording for the format. On a somewhat related sidenote, I've been listening to the Beatles albums that came out on Tidal yesterday. These are are the 2009 remasters which I'd never heard before. I did a little research and found out that these remasters were a 5 year project by some very competent engineers. I know every Beatles song inside/out and can easily say give me these remasters in 192kbps over any previous version in DSD128 ..


 
  
 Totally agree with you here. Releasing better mastered music is right now the most important thing. The sample rate and bit rate wars are just a distraction.
  
 Edit: LOL, I originally wrote that DSD was user-friendly...


----------



## sonitus mirus

Initially, I was worried that the powerful and acquisitive music industry would attempt to push higher bit/sample rate files in an effort to get people to repurchase their audio libraries  Unfortunately for this lawyer-controlled industry, many people just don't care about expensive, higher quality music, and when any original music was simply transcoded directly to a new format, people started catching on and there has been some backlash from consumers.   I believe we are always going to have different quality levels of music with regards to the format.  This works out best for the music conglomerates as there is no need to remaster and upgrade their entire, vast catalog; and they can select which music can be upgraded to target a specific market for maximum profitability.  
  
 What concerns me is that any well-mastered, higher quality release will only be made available in an expensive HD file, with a lower quality format somehow being inferior either due to a lack of care when creating them or intentionally to artificially create value with the technically higher quality formats.  I can not pass an ABX test with any of the HD tracks I have purchased and my own MP3 created from these tracks, so I don't feel there is any benefit to me from having the more expensive format, provided that the lower bit/sample rate version was competently created.
  
 I want good music, and if HD files are the direction needed to get the masters up to snuff, so be it.  The industry will either have to drastically lower the prices, or continue to make cheaper versions available to the masses.  If the streaming files and lower bitrate files are audibly transparent to me, I'm laughing all the way to the bank as I will save a lot of money and can stream the smaller files practically everywhere I go.  If these lower bitrate files are somehow being gimped, I'd probably begrudgingly spend more money on some select music I love to ensure I was hearing the best version.  I'm closely monitoring how things turn out, and it is a major reason for my interest and participation on music/audio sites such as this one.


----------



## coli

From my experience, a lot of the HD recording actually sound worse than CD release. Looking closer, it seem a large part of the music industry equipment/software is 44.1 and then they just upsample it and the suckers buys them for a higher price... Movie industry workflow seems to be different though (and tends to sound better when it comes to hi-res)
  
 Edit: Major labels, movies, broadcasters are the only ones doing hi-res right, everyone else are a crap shot, and does them wrong.


----------



## castleofargh

coli said:


> From my experience, a lot of the HD recording actually sound worse than CD release. Looking closer, it seem a large part of the music industry equipment/software is 44.1 and then they just upsample it and the suckers buys them for a higher price... Movie industry workflow seems to be different though (and tends to sound better when it comes to hi-res)
> 
> Edit: Major labels, movies, broadcasters are the only ones doing hi-res right, everyone else are a crap shot, and does them wrong.


 

 old stuff shouldn't come as a surprise, and even modern stuff where people mix whatever sample they like wherever they got it, some 44.1 for sure. even mp3 might end up in the mix. is it still highres music? when only 1 track in 20used to make the song isn't highres, is the song highres? at this point we see all the ridicule of this resolution based system as it becomes more philosophy than sound quality.
 and remaster when I grew up,they had been linked to "best of" in my mind. a way to resell the same old stuff in poor quality with a cheap fast remaster that too often sounded worst than the original.
 to the point that subjectively, I still to this day have a negative bias when I read "remastered" on a cover.  it should make me dream of all the improved stuff modern tech brought out, but it actually scares me away most of the time.


----------



## s0ny

I bought several hi-res 24-bit albums, 96 kHz, 192 kHz and DSD. Were they wasted money? I paid $25 for each online. My local retailer sells brand new CD's for $12, which seems a better deal (physical disc, booklet, case, just more retail value-for-money). I read a lot of threads and websites such as these:
  
http://people.xiph.org/~xiphmont/demo/neil-young.html
  
 and it seems like an adult can't hear anything above CD-quality. Then, there's the statement that it matters in quality experience even if you can't hear those frequencies/resolution. If anything above CD-quality is pointless, then I should stop buying hi-res music? I honestly have a huge preference for retail, because of the physical product, but I'm willing to go for hi-res if the quality is audible.


----------



## JWolf

s0ny said:


> I bought several hi-res 24-bit albums, 96 kHz, 192 kHz and DSD. Were they wasted money? I paid $25 for each online. My local retailer sells brand new CD's for $12, which seems a better deal (physical disc, booklet, case, just more retail value-for-money). I read a lot of threads and websites such as these:
> 
> http://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> and it seems like an adult can't hear anything above CD-quality. Then, there's the statement that it matters in quality experience even if you can't hear those frequencies/resolution. If anything above CD-quality is pointless, then I should stop buying hi-res music? I honestly have a huge preference for retail, because of the physical product, but I'm willing to go for hi-res if the quality is audible.


 
  
 It's not just about frequencies above what you can get at 16/44.It's about making the frequencies you can hear the best they can be. yes I know you'll get a lot of people saying the reason Hi-Res sounds better is because of a better master used. But it's about what's available. If you can, get a CD of any of your Hi-Res music and compare. Listen for yourself what you hear. Don't listen to what anyone else says. Do your own comparing and make up your own mind and then come back hre and post your concitions.


----------



## castleofargh

jwolf said:


> s0ny said:
> 
> 
> > I bought several hi-res 24-bit albums, 96 kHz, 192 kHz and DSD. Were they wasted money? I paid $25 for each online. My local retailer sells brand new CD's for $12, which seems a better deal (physical disc, booklet, case, just more retail value-for-money). I read a lot of threads and websites such as these:
> ...


 

 I totally disagree with what you just said about highres differences, so I totally agree with your conclusion. ^_^ don't listen to what anyone else says
	

	
	
		
		

		
		
	


	



  
@s0ny the end result is how happy you are listening to your music or owning your music. music is about pleasure, real or imagined, what really matters is how we feel about it not how it really is. to me a CD means I'll have to rip it. for 1 CD it's nothing, for hundreds of CDs, how much of my life did I waste doing that?
 now of course just downloading some file, it's not like having a physical object. I understand that that too. I guess you could go for DVD audio or SACD to get a physical object and highres, but then trying to get the music on a computer or a DAP a be a nightmare.  but in the end what matters is you. if one option annoys you, get rid of it. nothing justifies being bothered by something that is supposed to bring us joy.
 about highres, can you hear a difference? do you want a given master that wasn't pressed on CD?  those are mainly the questions you must ask yourself. for the rest, CD resolution is already amazing. we come from k7 tapes and vinyls, compared to that CDs are high fidelity already.


----------



## s0ny

I agree with the arguments. I also use a CD player occasionally, which makes having a CD more convenient. When I purchase a CD, I rip it to my DAP immediately. I was just wondering if it's biologically impossible to hear audio beyond 16-bit 44.1 kHz. If it is theoretically impossible, then I would have no reason to buy hi-res music anymore. If audio beyond CD-specs provide noticeable improvement of sound quality, then I'm willing to prefer hi-res purchases over CD's.


----------



## XenHeadFi

s0ny said:


> I agree with the arguments. I also use a CD player occasionally, which makes having a CD more convenient. When I purchase a CD, I rip it to my DAP immediately. I was just wondering if it's biologically impossible to hear audio beyond 16-bit 44.1 kHz. If it is theoretically impossible, then I would have no reason to buy hi-res music anymore. If audio beyond CD-specs provide noticeable improvement of sound quality, then I'm willing to prefer hi-res purchases over CD's.


 
 The universe will be around for a long time so impossible is a pretty strong gate to pass through. Listening to music is both physical (objective) and metaphysical (subjective). I lean heavily towards the objective sides for many things but I completely understand that the emotion of enjoying music is not purely objective.
  
 CastleofAaaaaaaaargh gives out good advice. There are reasons why I would buy hi-res music and it has nothing to do with capabilities of the format. It has everything to do with the mastering of that piece of music. If hi-res is the only way to get a great master, then I will spend the money on it. $20 spent on a good master is better than $10 completely wasted on a CD that is compressed to be flat...
  
 I now really treasure my late 1980's to mid 1990 CDs. They have dynamics! I just found out that Pretty Hate Machine (1989) has more dynamic range than any of Adele's albums.


----------



## upstateguy

castleofargh said:


> jwolf said:
> 
> 
> > s0ny said:
> ...


 
  
 A couple of things pop out at me.
  
*"...it seems like an adult can't hear anything above CD-quality."    *This begs the question of whether there is any extra audible material to hear in HiRez?
  
*"If you can, get a CD of any of your Hi-Res music and compare."  *This is completely ridiculous.  The only valid way to do this is to reprocess your HighRez material to CD quality and compare.
  
*"I totally disagree with what you just said about highres differences, so I totally agree with your conclusion...."   *I totally agree with what you totally disagree with and I am in complete agreement with your conclusion.


----------



## JWolf

upstateguy said:


> A couple of things pop out at me.
> 
> *"...it seems like an adult can't hear anything above CD-quality."    *This begs the question of whether there is any extra audible material to hear in HiRez?
> 
> ...


 
  
 Doesn't matter if we can hear past 20Khz or not. What matters is the quality of the music we can hear.
  
 Your valid way is invalid. They way to do it is to listen to what's available. if all you can get is the Hi-Res and the CD, then you listen to both and decide what you like. This is not a compare with a fictitious file that doesn't exist. You compare with what's available and then you decide if Hi-Res is worth it. I do wish a lot of the people onhere would stop trying to muddy things up with invalid comparisons.
  
 That last statement is so out of touch with reality. Reality is when we do our own comparisons and make our own conclusions instead of letting others dictate what we should or should not be able to hear. And stop quoting old obsolete articles on what we should or should not be able to hear,
  
 s0ny, just do your own comparisons and come up with your own conclusions Forget that article you linked. It's obsolete. Forget what others tell you you should or should not be able to hear. Just do your own listening and make up your own mind.


----------



## Arunabha Lahiri

At the end music is all about entertainment and mental as well as spiritual satisfaction. And at the end all that matters is the music. If someone is happy with high resolution, it's his choice and he has complete liberty to do so. I believe the purpose of this thread is to make people aware of the advertising policy of different audio company and I would say in that case the thread is successful. But if the good masters are only reserved for high res in the future, then that's the way to go. Even apple is rumored to give high res support in iPhone 7. So we can't ignore that either. Our brain is a mysterious thing. Even now we cannot decipher it's functions and all we have are some statistics and hypothesis. We can say that anything above cd quality is pointless. But at the end of the day everything depends on the user's choice and satisfaction.


----------



## Roly1650

s0ny said:


> I bought several hi-res 24-bit albums, 96 kHz, 192 kHz and DSD. Were they wasted money? I paid $25 for each online. My local retailer sells brand new CD's for $12, which seems a better deal (physical disc, booklet, case, just more retail value-for-money). I read a lot of threads and websites such as these:
> 
> http://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> and it seems like an adult can't hear anything above CD-quality. Then, there's the statement that it matters in quality experience even if you can't hear those frequencies/resolution. If anything above CD-quality is pointless, then I should stop buying hi-res music? I honestly have a huge preference for retail, because of the physical product, but I'm willing to go for hi-res if the quality is audible.




That link is an excellent place to start and as relevant today as the day it was published, there have been zero developments in the meantime which would have rendered it obselete. Anybody saying otherwise is either offering bad advice or has no clue what they are talking about.

The biggest problem with online downloads is you have no way of knowing what the files actually are until after you've downloaded them. The website preview option is usually at lower resolution than the downloaded files, the website maintaining bare-bones bandwith for cost reasons. So there's many an audiophile banging on about how wonderful his 24 bit files are, blissfully unaware that he's actually got 16/44 files in a 24/96/192 container, (ie: upsampled). It really is a crapshoot as to what you actually get, the download websites themselves don't necessarily have 100% the right information on the files, for whatever reason.

With the price of cd's being where they are and the quality possible, (obvious if audiophiles can get tricked into believing cd quality is actually hi-res, see above) then it would still seem like the way to go. Sure you may have to rip the cd, but that's a routine excercise.


----------



## s0ny

For some reason, ripping CD's is enjoyable to me. I just like seeing the progress.

If hi-res music is taken directly from the original master source, providing more real data in the same songs, then I can imagine it to sound different, even if you can't hear the highest frequencies. Being a version that sounds the way it was originally recorded makes it interesting for me. It doesn't necessarily sound better though.

I have some music that has been remastered multiple times over the past decades and the remastered CD versions sound more impactful and impressive than the DSD that I bought. I guess I prefer forward mids and compressed sound. The DSD had raving reviews from audiophiles, but to my ears, it sounded too neutral and weak. The remastered CD-versions were obviously compressed, amplified and edited, but they do sound more impressive because of it. I opened one of the best-sounding 24-bit 192 kHz hi-res purchases in Audacity. It looked totally brickwalled but sounds great. The reviews of that album said the producer did major effort to improve the master for that hi-res version. Even though that album looked like a fake hi-res in Audacity, it sounded way better to me than the perfect DSD.

Like everyone mentioned above, it's probably the source used and how/if it's edited to determine how good it sounds. I almost bought some hi-res albums yesterday, but I had a new years deal of $8 per CD (brand new, sealed). I received them today.


----------



## JWolf

roly1650 said:


> That link is an excellent place to start and as relevant today as the day it was published, there have been zero developments in the meantime which would have rendered it obselete. Anybody saying otherwise is either offering bad advice or has no clue what they are talking about.
> 
> The biggest problem with online downloads is you have no way of knowing what the files actually are until after you've downloaded them. The website preview option is usually at lower resolution than the downloaded files, the website maintaining bare-bones bandwith for cost reasons. So there's many an audiophile banging on about how wonderful his 24 bit files are, blissfully unaware that he's actually got 16/44 files in a 24/96/192 container, (ie: upsampled). It really is a crapshoot as to what you actually get, the download websites themselves don't necessarily have 100% the right information on the files, for whatever reason.
> 
> With the price of cd's being where they are and the quality possible, (obvious if audiophiles can get tricked into believing cd quality is actually hi-res, see above) then it would still seem like the way to go. Sure you may have to rip the cd, but that's a routine excercise.


 
  
 I do stick by my statement that the article is obsolete. When we listen to music, we do not listen to specific frequency waveforms. Music is not like that. Muysic is a lot more complex and it is a lot more difficult to fill in any missing information. A waveform is easy to fll in the gaps so it is indistinguishable from the original. But music is not like that. That's why we have had the CD vs. LP argument for a long time. The thing is, Hi-Res music isn't just about extending the high end past what you get from 44.1Khz. It's about capturing as much of the audio as you can in all the frequencies of the master recording.
  
 Some say Hi-Res is better because it's been remastered better than the CD. Even if that's true, then if you buy the CD you are buying a lesser quality recording.  A lot are saying to downsample/convert the Hi-Res recording and compare that. That's rubbish. You have to compare the Hi-Res with the best currently available on CD because you aren't going to be buying a CD downsampled from the Hi-Res and this, that makes the comparison not really valid.
  
 So compare the Hi-Res with the best on Cd and judge for yourself. Don't listen to obsolete articles on waveforms and how 16/44.1 is just fine for waveforms. Music is more complex and thus do your own thinking.


----------



## JWolf

s0ny said:


> For some reason, ripping CD's is enjoyable to me. I just like seeing the progress.
> 
> If hi-res music is taken directly from the original master source, providing more real data in the same songs, then I can imagine it to sound different, even if you can't hear the highest frequencies. Being a version that sounds the way it was originally recorded makes it interesting for me. It doesn't necessarily sound better though.
> 
> ...


 
  
 I've head some highly compressed Hi-Res that are not worth being in Hi-Res. It really is a shame that good music can be allowed to be ruined like that.


----------



## nick_charles

jwolf said:


> ... specific frequency waveforms. Music is not like that. Muysic is a lot more complex and it is a lot more difficult to fill in any missing information.


 
  
  
 This is at least the second time you have said this. The first time I ignored it, but you are spreading misinformation. It does not matter how complex a waveform is as long as none of the component frequencies are above fs/2 then it can be reconstructed perfectly well with a DAC operating at a given fs. the argument that a complex wave is harder to render is incorrect. Please visit here Jim leSurf on Waves 
  


> _where B is the signal bandwidth. Given this information we can, therefore, reconstruct the actual shape of the original continuous signal at any instant ‘in between’ the sampled instants. It should also be clear that this reconstruction is not a guess but a true reconstruction._


----------



## JWolf

nick_charles said:


> This is at least the second time you have said this. The first time I ignored it, but you are spreading misinformation. It does not matter how complex a waveform is as long as none of the component frequencies are above fs/2 then it can be reconstructed perfectly well with a DAC operating at a given fs. the argument that a complex wave is harder to render is incorrect. Please visit here Jim leSurf on Waves


 
  
 Have you compared Hi-Res to CD-Res and if so, what was your conclusion based on your listening?


----------



## RRod

jwolf said:


> I do stick by my statement that the article is obsolete. When we listen to music, we do not listen to specific frequency waveforms. Music is not like that. Muysic is a lot more complex and it is a lot more difficult to fill in any missing information. A waveform is easy to fll in the gaps so it is indistinguishable from the original. But music is not like that. That's why we have had the CD vs. LP argument for a long time. The thing is, Hi-Res music isn't just about extending the high end past what you get from 44.1Khz. It's about capturing as much of the audio as you can in all the frequencies of the master recording.
> 
> Some say Hi-Res is better because it's been remastered better than the CD. Even if that's true, then if you buy the CD you are buying a lesser quality recording.  A lot are saying to downsample/convert the Hi-Res recording and compare that. That's rubbish. You have to compare the Hi-Res with the best currently available on CD because you aren't going to be buying a CD downsampled from the Hi-Res and this, that makes the comparison not really valid.
> 
> So compare the Hi-Res with the best on Cd and judge for yourself. Don't listen to obsolete articles on waveforms and how 16/44.1 is just fine for waveforms. Music is more complex and thus do your own thinking.


 
  
 The ideal reconstruction of a PCM signal is perfectly compatible with a frequency-view; that we can't attain the ideal doesn't mean suddenly the view is thrown out the window.
  
 I'll agree with you that it's missing the point a bit to talk about what a resample of a hi-res track sounds like if you can't get the master on anything but hi-res. That's the results of the sad cycle of loudness, where companies started to make things loud, people started to accept it and then expect it, and now good sound is considered a "niche." Still, that you can take a hi-res master and take it down to Redbook with no audible issues just shows how unnecessary the intertwining of hi-res downloads and good mastering really is.
  
 To @s0ny: Find a master you like and get it. If having it in hi-res is a pain for your hardware/software setup, it is quite easy to take it down to CD and have it sound just as good. Do searches for your albums on sites like Steve Hoffman's forum and the loudness database to help you find masters you might want.


----------



## L8MDL

jwolf said:


> I've head some highly compressed Hi-Res that are not worth being in Hi-Res. It really is a shame that good music can be allowed to be ruined like that.




Examples?


----------



## Cerastes

jwolf said:


> When we listen to music, we do not listen to specific frequency waveforms.


 
 Indeed, however you have to understand that those specific frequency waveforms dictate WHAT we can hear... one does not simply listen to something that is not there.
  


jwolf said:


> Muysic is a lot more complex and it is a lot more difficult to fill in any missing information.


 
 Not true per se... sure music can be somewhat complex, however at the end of the day it's pure math.
  


jwolf said:


> That's why we have had the CD vs. LP argument for a long time.


 
 The only reason why we are still having this argument is simply because being a hipster is a thing, and people are having nostalgia boners and those people as a whole have no idea what they are talking about and cannot accept the fact that the CD or digital in general is the superior format overall.


----------



## RRod

l8mdl said:


> Examples?


 
  
 A version of _Countdown to Extinction_ available from a certain hi-res retailer sounds a total mess compared to either the original release or the MoFi version. Bloated bass, sibilant voices, bleh.


----------



## JWolf

l8mdl said:


> Examples?


 
  
 R.E.M - Fables of Reconstruction
  
 That one is well brick-walled at DR8. Tracks are either DR8 or DR9.


----------



## s0ny

rrod said:


> The ideal reconstruction of a PCM signal is perfectly compatible with a frequency-view; that we can't attain the ideal doesn't mean suddenly the view is thrown out the window.
> 
> I'll agree with you that it's missing the point a bit to talk about what a resample of a hi-res track sounds like if you can't get the master on anything but hi-res. That's the results of the sad cycle of loudness, where companies started to make things loud, people started to accept it and then expect it, and now good sound is considered a "niche." Still, that you can take a hi-res master and take it down to Redbook with no audible issues just shows how unnecessary the intertwining of hi-res downloads and good mastering really is.
> 
> ...


I've been reading Steve Hoffman's forum for two days. I also check a lot of reviews before I buy.


----------



## KeithEmo

The DRM used in SACDs is much more extreme than just blocking certain things. An SACD player, by license, is not allowed to give you an "in the open" digital audio stream that is derived from that SACD content at full quality. This means that an SACD player can convert the DSD to PCM and give you a PCM output _ONLY OVER HDMI_ - because the HDMI connection is encrypted. An SACD player is not allowed to give you a two-channel PCM output, for example, over Coax or Toslink, that you can connect to a two-channel PCM DAC. So, for example, if you play an SACD on an Oppo, it will give you an output that has been converted to PCM, or a DSD output, _BUT ONLY VIA HDMI._ (Any pre/pro will be able to play the PCM version, and any pre/pro that supports DSD, like the Emotiva XMC-1, will be able to play the DSD version; but there are only one or two DACs that will accept an HDMI input at all.)
   In other words, if you get a DAC that supports DSD playback, it can play DSD files that you download, played from a computer or something like an Oppo, and some illegal bootlegs, but there's no way you can use it as an external DAC to play your SACDs. (You can use an external DAC with your CD player, but, because of the DRM, you _CAN'T_ use one with an SACD player. And you can use an external DSD DAC to play files from your computer, if both support DOP, but you can't play SACDs on your computer or RIP them with it.)
  
 The general gist of Siao's article seems pretty clear..... Ignoring whether you can actually hear the difference or not, SACD is slightly better than 16/44, and 24/96 is better than 16/44, but DSD is NOT better than 24/96 in any meaningful way. (Or, to put it another way, when the choice was between CD and SACD, SACD was arguably better, but there is no logical reason to claim that SACD is better than 24/96 digital audio.)
  
  
 Quote:


xenheadfi said:


> I apologize as I just read the bulleted list that you posted. Now, I've read the entire interview and the part you quoted comes at the very end without any explanation or reasons why it is better. Siau does say it, so I am wrong in my post.
> 
> The vast majority of the article was spent pretty much saying DSD is a horrible format but that Benchmark makes the best DSD DAC to handle its playback. I didn't know that DSD pretty much filters out everything from 47kHz and up or how limiting DSD is for playback usability. As a user, a DSD file pretty much has to be converted to PCM before crossfades/fadeouts/fadeins/gapless playback can be implemented from our music players, but they can't if that DSD file has DRM, like for SA-CD. Even if you could, "consumer" DSD is 1-bit so any manipulations introduces artifacts. Another horrible attempt by Sony to lock-in users into a format that is extremely (edit: NOT) user-friendly and only offers stronger DRM as a benefit (Memory Sticks, UMD, SecureROM even on audio CDs!, etc).
> 
> ...


----------



## L8MDL

rrod said:


> A version of _Countdown to Extinction_ available from a certain hi-res retailer sounds a total mess compared to either the original release or the MoFi version. Bloated bass, sibilant voices, bleh.




The poor reditions appear to be copies of the 2012 25th anniversary CD which also had those bad dr nimbers. The sites can only sell what the record companies give them.


----------



## RRod

l8mdl said:


> The poor reditions appear to be copies of the 2012 25th anniversary CD which also had those bad dr nimbers. The sites can only sell what the record companies give them.




Since the material is 24/96 and has content above 20k that isn't just noise or filter cruft, it can't just be a CD copy. Still sounds bad, though.


----------



## old tech

aphex27 said:


> And we'd love to have you dispute this interview with arguments. (and by we I mean I)


 
 Well here is another interview, this time with Roger Sanders, and Audio Engineer and musician who designs and builds what is arguably the best electrostatic speakers/amp combination you can buy.  Not only does he dismiss that 16/44 <<< 24/96 for playback, he also argues that PCM >>> DSD.  What do think are the flaws in his arguments?
  
http://www.monoandstereo.com/2013/11/interview-with-roger-sanders.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+monoandstereo%2FHOym+%28MONO+AND+STEREO+Ultra+High+End+Audio+Magazine%29


----------



## spruce music

old tech said:


> Well here is another interview, this time with Roger Sanders, and Audio Engineer and musician who designs and builds what is arguably the best electrostatic speakers/amp combination you can buy.  Not only does he dismiss that 16/44 <<< 24/96 for playback, he also argues that PCM >>> DSD.  What do think are the flaws in his arguments?
> 
> http://www.monoandstereo.com/2013/11/interview-with-roger-sanders.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+monoandstereo%2FHOym+%28MONO+AND+STEREO+Ultra+High+End+Audio+Magazine%29


 
 Don't think I see any flaws myself.


----------



## castleofargh

old tech said:


> Well here is another interview, this time with Roger Sanders, and Audio Engineer and musician who designs and builds what is arguably the best electrostatic speakers/amp combination you can buy.  Not only does he dismiss that 16/44 <<< 24/96 for playback, he also argues that PCM >>> DSD.  What do think are the flaws in his arguments?
> 
> http://www.monoandstereo.com/2013/11/interview-with-roger-sanders.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+monoandstereo%2FHOym+%28MONO+AND+STEREO+Ultra+High+End+Audio+Magazine%29


 
 he might just have summed up 90% of the sound science sub forum in one long interview.
  
 I sure have zero love for DSD. not for the tech(hard to copy onto other supports, 1bit nonsense), not for the choice to make some masters exclusively available on DSD when most of the time they were mastered in PCM anyway.


----------



## aphex27

old tech said:


> Well here is another interview, this time with Roger Sanders, and Audio Engineer and musician who designs and builds what is arguably the best electrostatic speakers/amp combination you can buy.  Not only does he dismiss that 16/44 <<< 24/96 for playback, he also argues that PCM >>> DSD.  What do think are the flaws in his arguments?
> 
> http://www.monoandstereo.com/2013/11/interview-with-roger-sanders.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+monoandstereo%2FHOym+%28MONO+AND+STEREO+Ultra+High+End+Audio+Magazine%29


 

 I don't see any..in fact the past few days I've been listening to the albums I downloaded in DSD, on Tidal at 16/44.1..I don't think I can tell a difference..
 I will freely admit that the amazing recording quality had me fooled that it was the medium rather than the recording..
 I urge anyone who is into classical to try some recordings from Pentatone records on Tidal..pretty amazing recording quality


----------



## 441879

old tech said:


> Well here is another interview, this time with Roger Sanders, and Audio Engineer and musician who designs and builds what is arguably the best electrostatic speakers/amp combination you can buy.  Not only does he dismiss that 16/44 <<< 24/96 for playback, he also argues that PCM >>> DSD.  What do think are the flaws in his arguments?
> 
> http://www.monoandstereo.com/2013/11/interview-with-roger-sanders.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+monoandstereo%2FHOym+%28MONO+AND+STEREO+Ultra+High+End+Audio+Magazine%29




He sounds exactly like an engineer (which I am as well). The one thing that really bothers me about high end audio is how much pure BS there is and it's nice to see someone call it as such.


----------



## icebear

Very interesting interview. He obviously doesn't care about being polarizing. He has solid arguments for his positions although a lot of audiophiles will not like his stance against analog (tape and vinyl). Sure thing he is spot on about the recording being the most important part of the entire audio chain and not the data format. And also that high rez for consumer replay is a complete waste vs red book when all the recording and processing has been done with enough headroom the final consumer format can be 16/44.1 without any loss when done properly.
 I am not sure if he has heard any First Watt amps or is familiar with Nelson's "home-brew" but for me 500+ watts into 8 ohms are definitely beyond overkill


----------



## blade007

_Brad Meyer and David Moran_ from the _Audio Engineering Society_ did such a study. Subjects sat at a chair and listened to a SACD/DVD-A sound source directly vs piping through a 16bit/44.1kHz A/D/A device. Subject were asked which source was superior.

 Out of 554 trials, 276 picked the pure SACD/DVD-A source. That is 49.82%, and is pretty much 50/50 chance.

 The study concluded,
  


> Our test results indicate that all of these recordings
> could be released on conventional CDs with no audible
> difference.


  
 Source: http://www.aes.org/e-lib/browse.cfm?elib=14195
 http://drewdaniels.com/audible.pdf
  
 When you're looking at sample rates above 44.1kHz, there won't be any audible difference. Anyone who says otherwise is either lying, or deluded.
 HOWEVER, it's worth noting that sample rates above 44.1kHz are still important when it comes to recording/production/engineering, simply due to potential aliasing issues when editing/bouncing/rendering. But this is the only reason that they are important, and they hold no value in the consumer market.
  
 Another test: http://archimago.blogspot.com.au/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html?m=1
  
  
 So, it's all placebo effect. Stuff like AK380 is marketing bs. I mean you could spend all that money on a high end headphone/amp etc. 16 bit/44.1 is all you need unless you want your dogs/cat or bats to listen. It's funny because I was going to buy Ak380. Not anymore.


----------



## JWolf

blade007 said:


> _Brad Meyer and David Moran_ from the _Audio Engineering Society_ did such a study. Subjects sat at a chair and listened to a SACD/DVD-A sound source directly vs piping through a 16bit/44.1kHz A/D/A device. Subject were asked which source was superior.
> 
> Out of 554 trials, 276 picked the pure SACD/DVD-A source. That is 49.82%, and is pretty much 50/50 chance.
> 
> ...


 
  
 Ignore those articles that say that 16/44.1 is all you need. What you should do is listen for yourself. Personally,I do hear a difference in favor of Hi-Res. But you need to make that choice/distinction for yourself. A lot of those articles/blog posts say how well 16/44.1 reproduce sine waves. I don't disagree there. But music is more complex.


----------



## Roly1650

nick_charles said:


> This is at least the second time you have said this. The first time I ignored it, but you are spreading misinformation. It does not matter how complex a waveform is as long as none of the component frequencies are above fs/2 then it can be reconstructed perfectly well with a DAC operating at a given fs. the argument that a complex wave is harder to render is incorrect. Please visit here Jim leSurf on Waves







jwolf said:


> A lot of those articles/blog posts say how well 16/44.1 reproduce sine waves. I don't disagree there. But music is more complex.



@nick_charles gave a perfect explanation of why you are incorrect. He even linked to an excellent article on the why's and wherefores, which would have enlightened you, if you'd chosen to read it. The sampling theorem is as true for higher sample rates as it is for 16/44. You repeating this pile of crap won't magically make it right and at this stage you aren't doing anymore than trolling the thread with misinformation again.


----------



## castleofargh

jwolf said:


> Ignore those articles that say that 16/44.1 is all you need. What you should do is listen for yourself. Personally,I do hear a difference in favor of Hi-Res. But you need to make that choice/distinction for yourself. A lot of those articles/blog posts say how well 16/44.1 reproduce sine waves. I don't disagree there. But music is more complex.


 
  
 yeah you told that to us almost word for word last page too. but you don't actually have any evidence of what you're saying. not for your opinion that highres sounds different(let me guess, sighted evaluation?), and not about music being any more difficult for 16/44 than redoing 1 single 20khz sine wave.
 it could look counter intuitive to some, but that's how it is. the difficulty is not how many waves are recorded at once, because waves add up at one physical point into 1 single value of pressure in the air, and one single value of voltage in the analog path at a single instant T.  so the hard part is only ever to be able to move from one value to the next fast enough to redo the content before the direction changes(up or down). and what is the fastest changing content of music in the 20hz-20khz audible range? the 20khz sine wave!  all the music below is factually changing slower than one single 20khz sine.
 so doing 20khz right is in fact evidence that we can do everything that is slower and your argument is false.
 and this could be demonstrated(contrary to your statement).  take super complex music from 20hz to 15khz, you can redo it with let's say 35khz sample rate(some margin for the low pass filter). but you will fail to redo the 20khz sine wave correctly with that sample rate because you won't have 2 points per period and the low pass filter can't save that.
  
 no DAC is drawing a signal by adding points one at a time and leave it at that. until you understand the purpose of band limiting, you won't get what the low pass filter really does to the analog signal and you will stay with your false instinctive concept of digital audio. it's ok, I was like you, everybody here was like you and thought like you at some point in life. you can't know what you don't know. but you sure could consider that maybe you don't know as well as you think, after you see so many people and read so many papers contradicting your idea of digital audio.
 if you find that boring to read, or didn't do much math or physics at school, it's very ok not to bother. I can drive a car and don't understand 90% of how it works
	

	
	
		
		

		
		
	


	




. there is no need to be a rocket scientist to enjoy music. and there is nobody telling you to stop buying highres tracks if you want to.
 but if you come here telling others how things are, you have to know your stuff !


----------



## RRod

White noise is pretty much as complex as you can get from a time perspective: no correlation between time intervals, unexpectedly large swings in signal value, etc. I doubt musical signals, inherently a bit more "sine-y" than noise, get more complicated. Yet I don't see people complaining about how 44.1ksps noise sounds different than 192ksps noise. Could it be because a) a periodic reconstruction makes perfect sense for a set of samples, b) a periodic reconstruction can be made in terms of cosine/sine sums, and c) most people can't hear sine waves above 20kHz? Yet instead of admitting any limits on human perception, people chose to make vague assaults against well-established and rigorous theory. Sigh.


----------



## speedracer1

blade007 said:


> _Brad Meyer and David Moran_ from the _Audio Engineering Society_ did such a study. Subjects sat at a chair and listened to a SACD/DVD-A sound source directly vs piping through a 16bit/44.1kHz A/D/A device. Subject were asked which source was superior.
> 
> Out of 554 trials, 276 picked the pure SACD/DVD-A source. That is 49.82%, and is pretty much 50/50 chance.
> 
> ...


 
 Hello blade007,
  
 I'm not questioning the results of the study with respect to the 50/50 split  But I would  like to know a bit about the subjects and their background?
  
 Listening to music is more about training the mind to listen then the ear to hear sound.  For example were the subjects audiophiles or musicians?  Many people simply don't have a frame of reference to actually know what actual instruments sound like let alone mixing it with the detailed process of sound reproduction.  Therefore it makes it difficult to make an objective decision and good choices.  I'm an avid audiophile for many years and a musician and I've had my fair share of audio systems including the AK 380 and just because you read about a study doesn't make it gospel. A great recording is a great recording regardless of the media (16/44.1, 48, 96 SACD) it is placed on.  
  
 Sincerely,
 -Speed


----------



## JWolf

castleofargh said:


> yeah you told that to us almost word for word last page too. but you don't actually have any evidence of what you're saying. not for your opinion that highres sounds different(let me guess, sighted evaluation?), and not about music being any more difficult for 16/44 than redoing 1 single 20khz sine wave.
> it could look counter intuitive to some, but that's how it is. the difficulty is not how many waves are recorded at once, because waves add up at one physical point into 1 single value of pressure in the air, and one single value of voltage in the analog path at a single instant T.  so the hard part is only ever to be able to move from one value to the next fast enough to redo the content before the direction changes(up or down). and what is the fastest changing content of music in the 20hz-20khz audible range? the 20khz sine wave!  all the music below is factually changing slower than one single 20khz sine.
> so doing 20khz right is in fact evidence that we can do everything that is slower and your argument is false.
> and this could be demonstrated(contrary to your statement).  take super complex music from 20hz to 15khz, you can redo it with let's say 35khz sample rate(some margin for the low pass filter). but you will fail to redo the 20khz sine wave correctly with that sample rate because you won't have 2 points per period and the low pass filter can't save that.
> ...


 
  
 I don't need scientific results for YOU or anyone to make you own opinion. It's up to you what you hear/feel/think. It's nobody's job to tell you what you like to listen to be it CD or Hi-Res of some kind. Linking articles/blog posts/etc. is worthless. It's up to each of us to decide what it is we prefer. If we go making up our minds based on what someone else says, then we could lose out because we might not be choosing what we feel is best. This is subjective. You can quote numbers and show sine waves and so forth, but it's what we hear that matters most of all.


----------



## castleofargh

jwolf said:


> I don't need scientific results for YOU or anyone to make you own opinion. It's up to you what you hear/feel/think. It's nobody's job to tell you what you like to listen to be it CD or Hi-Res of some kind. Linking articles/blog posts/etc. is worthless. It's up to each of us to decide what it is we prefer. If we go making up our minds based on what someone else says, then we could lose out because we might not be choosing what we feel is best. This is subjective. You can quote numbers and show sine waves and so forth, but it's what we hear that matters most of all.


 
 sure you make your own opinion. and if it's to stay in the subjective realm, then please do that. because it's everybody's problem when you justify your opinion with a false technical statement.


----------



## JWolf

castleofargh said:


> sure you make your own opinion. and if it's to stay in the subjective realm, then please do that. because it's everybody's problem when you justify your opinion with a false technical statement.


 
  
 I'm not giving any technical information because I have no idea if these technical links are correct or not. If I hear that Hi-Res is better than CD, then to me, those links are all wrong. So I'm not posting a link to something that may not be correct. The only links that I know are correct are the ones showing how 16/44.1 can handle a sine wave.


----------



## nick_charles

jwolf said:


> The only links that I know are correct are the ones showing how 16/44.1 can handle a sine wave.


 
  
 Would you be so kind as to remind me of which (some at least) of these links you are referring to, it will help me understand your position better, thanks. There are many highly misleading "how it works" type links replete with stairstep representations of reconstructed  waves(*) that are willfully used to befuddle , and lets face it , con unwary punters!
  
  
  
 * - excluding NOS DACs which can exhibit such stairsteps such as this NOS DAC


----------



## JWolf

nick_charles said:


> Would you be so kind as to remind me of which (some at least) of these links you are referring to, it will help me understand your position better, thanks. There are many highly misleading "how it works" type links replete with stairstep representations of reconstructed  waves(*) that are willfully used to befuddle , and lets face it , con unwary punters!
> 
> 
> 
> * - excluding NOS DACs which can exhibit such stairsteps such as this NOS DAC


 
  
 The link that comes to mind as rubbish is the link to xiph.org from the first message in this thread.


----------



## Cerastes

jwolf said:


> The link that comes to mind as rubbish is the link to xiph.org from the first message in this thread.


 
 Not sure if joking or... care to elaborate what scientific articles on that site you find rubbish?


----------



## JWolf

cerastes said:


> Not sure if joking or... care to elaborate what scientific articles on that site you find rubbish?


 
  
 The one linked in the first post of this thread. It's all about sine waves and has very little to do with music.


----------



## castleofargh

let me see if I get it right, you admit not to understand the technicalities, yet you feel like you're in a good position to make a judgment and call rubbish one of the most accessible article you can hope to find on the web that isn't pure snake oil or trying to sell you something.
 nothing making you a little uncomfortable with that? like the talking without knowing part? or maybe the calling rubbish someone's benevolent work that you don't understand?
  
 we have in sound science a poor reputation of bashing the innocent guy that doesn't know much. and I hope to get that reputation to go away with time and efforts, but when I read your last few posts, it's really hard not to get angry at you for talking completely out of place.


----------



## old tech

speedracer1 said:


> Hello blade007,
> 
> I'm not questioning the results of the study with respect to the 50/50 split  But I would  like to know a bit about the subjects and their background?
> 
> ...



 


Read the link in that post to the drewdaniels site. It provides the test's methodology including information on the test subjects. They included musicians, audiophiles and recording producers - none of whom scored better than the subjects without a "frame of reference". Says it all really.


----------



## old tech

jwolf said:


> The one linked in the first post of this thread. It's all about sine waves and has very little to do with music.



 

Wow, so you dismiss all the science and factual technical details behind digital signals, along with hundreds of years of accumulated human knowledge about sound and human hearing because what you think you hear proves it wrong? Not only is that absurd, it reeks of narcissism - rather than look at why you may be hearing a difference in the face of all the evidence, saying instead I am right, everyone else (even the people who invented it) are is wrong. Do you believe digital audio is some sort of magic bestowed on us from some higher power and therefore beyond the understanding of mere mortals?


----------



## Joe Bloggs

There's two issues here that both sides are conflating:

1. Whether it's _possible_ to make 16/44.1 CD material to sound the same as high-res material

This is a technical question, and the answer is "yes". Human hearing limits in terms of frequency and S/N, etc.

2. Whether it then follows that the CD-quality material you can buy on the market all sound the same as high-res material you can buy on the market.

This is a marketing question, and for a significant part the answer is "no"--different masters.

JWolf takes the fact of (2) and tries to discredit the answer to (1), when there's little to no relationship between the two.

On the other hand, back when JWolf was presenting the fact of (2), the objectivists also tried to discredit it with (1), which was also not right.


----------



## spruce music

jwolf said:


> The one linked in the first post of this thread. It's all about sine waves and has very little to do with music.


 

 What would convince you that you are mistaken about this?


----------



## old tech

joe bloggs said:


> There's two issues here that both sides are conflating:
> 
> 
> 
> ...



 

I don't believe anyone was discrediting (2), even within CD's there are different sounding versions of the same album eg remasters.


----------



## asymcon

I'm a big fan of 48/20, yes, that's 20bit at 48kHz.
 16bit needs to be noise-shaped to actually reach ATH, which reduces dynamic range on high frequencies and makes audio post-processing more difficult and lossy.
 24bit on the other hand is overkill and waste of space - no track or sample I have ever heard utilized full 24bit dynamic range.
  
 20bit is perfect spot as the original signal is retained in its lossless form without need for noise-shaping and it's spot-on for dynamic range of most of those "hi-res" recordings.
 When compressed through FLAC or WavPack 48/20 is usually 33% smaller than 48/24.


----------



## JWolf

Have any of you actually just listened for yourself? That's all that's need to be done. You don't have to read this article or that post. You don't need to know the science behind digital audio. You just need to listen and make your own decision as to what you prefer. HAVE YOU ACTUALLY LISTENED? All you seem to be doing is to try to discredit Hi-Res audio with meaningless articles/blog posts. Just listen.
  
 Post yor results of listening. And no ABX please.IT's OK that you know what the source is you are listening to. So just listen. That's all.


----------



## Joe Bloggs

jwolf said:


> Have any of you actually just listened for yourself? That's all that's need to be done. You don't have to read this article or that post. You don't need to know the science behind digital audio. You just need to listen and make your own decision as to what you prefer. HAVE YOU ACTUALLY LISTENED? All you seem to be doing is to try to discredit Hi-Res audio with meaningless articles/blog posts. Just listen.
> 
> Post yor results of listening. And no ABX please.*IT's OK that you know what the source is you are listening to.* So just listen. That's all.




Why is sighted "testing" ok?

And why is ABX not ok?

Because only results from the flawed methodology of the former conforms with your expectations?

Is it not enough for us to concede that hi-res albums are worth buying in real life (for the mastering quality), even if you may safely downsample them to 16/44.1 for actual usage?


----------



## Joe Bloggs

In other news, any 44.1kHz material (including hi-res resampled to 44.1kHz) sounds miles *better* than any hi-res material to me on all my music systems (X7 in stock form excepted), because of real-time DSP correction processing for my loudspeakers and headphones that work only at 44.1kHz:

http://www.head-fi.org/t/782131/why-high-res-audio-is-bad-for-music-take-2

(I could set it all up to work at 48kHz as well. 192kHz, not so much. DSD? Forget it.)


----------



## JWolf

joe bloggs said:


> Why is sighted "testing" ok?
> 
> And why is ABX not ok?
> 
> ...


 
  
 ABX puts too much pressure into the listening. Listening should just be stress free. Even though you can tell which recording is which, you can still make up your own mind.
  
 If the Hi-Res sounds better than the CD version, converting to 16/44.1 will also sound better than the CD version. So go for it if that's what you want to do. The thing I don't get though is if we can make CD quality music sound better, why is it not better to start with?


----------



## Joe Bloggs

jwolf said:


> ABX puts too much pressure into the listening. Listening should just be stress free. Even though you can tell which recording is which, you can still make up your own mind.
> 
> If the Hi-Res sounds better than the CD version, converting to 16/44.1 will also sound better than the CD version. So go for it if that's what you want to do. The thing I don't get though is if we can make CD quality music sound better, why is it not better to start with?




There is no rule saying ABX must consist of 15 second clips switched between quickly. Go listen to a whole song, whole album, any way of listening you like. Reason fast switching is usually used is because discriminatory powers are found to be better that way, in contrast to the popular claim that one must settle down with a system for days, have a drink with it, sleep with it, etc. to tell the difference. If that's the way you roll, you can roll that way in ABX as well--just be prepared to find what little discriminatory power you actually had to start with disappear in thin air. Also, a statistically significant number of trials should be performed no matter how long each trial takes.

Reason why "it's not better to start with" is because I've made it sound better *for my system*. It is system-specific processing that must be performed by each listener individually for his / her system, or at best for the same model of headphones.


----------



## asymcon

Hi-res audio could make sense when you prefer specific downsampling algorithms for bit/sample rate reduction for final mixdown.
 For example flat vs. noise-shaped 16bit. Music of the Spheres by Mike Oldfield for example has tons of dithering noise and some people might not like it, so they dither the 24bit master using TPDF.


----------



## Cerastes

jwolf said:


> IT's OK that you know what the source is you are listening to.


 
 Depends. If you are doing a comparison then it's definitely NOT OK, since we have this thing called confirmation bias.
  
  


jwolf said:


> ABX puts too much pressure into the listening. Listening should just be stress free.


 
 That's really up to the listener and what kind of mentality / personality they have. If they cannot just listen to the music on any equipment  / situation that is not their own choosing, then maybe they shouldn't be talking about differences or stuff like this in the first place due to the inherit bias they have.
  
  


jwolf said:


> If the Hi-Res sounds better than the CD version, converting to 16/44.1 will also sound better than the CD version.


 
 The only reason why it would sound better is that it has been mixed & mastered differently than it's CD counterpart.
  
  


jwolf said:


> The thing I don't get though is if we can make CD quality music sound better, why is it not better to start with?


 
 I'm pretty sure we all are up for better music quality, however upping the settings that are beyond human hearing to begin with doesn't magically do that; that's only snake oil ... we can only achieve better quality music when audio / mastering engineers start doing their job properly.


----------



## asymcon

> We can only achieve better quality music when audio / mastering engineers start doing their job properly.


 
 +1
  
 That's why as a composer, I mix/master my track on my own.


----------



## JWolf

cerastes said:


> I'm pretty sure we all are up for better music quality, however upping the settings that are beyond human hearing to begin with doesn't magically do that; that's only snake oil ... we can only achieve better quality music when audio / mastering engineers start doing their job properly.


 
  
 But, are you sure we are not getting any benefits in the audio range?


----------



## RRod

asymcon said:


> Hi-res audio could make sense when you prefer specific downsampling algorithms for bit/sample rate reduction for final mixdown.
> For example flat vs. noise-shaped 16bit. Music of the Spheres by Mike Oldfield for example has tons of dithering noise and some people might not like it, so they dither the 24bit master using TPDF.


 
  
 People must listen to music much louder than I do and have much better tone thresholds to account for the torment shaped dither gives them.


----------



## asymcon

@JWolf - Does it matter? If it put smile on your face then that's what matter.
 I couldn't tell difference between 48k and 96k...
 BUT
 I was able to detect sinewave tones in 38kHz range produced by CCFL 1.2kV drivers on multiple occasions. When measured with directional microphone, there were no other harmonics in the audible range, only that pulsating sine going from 36 to 40kHz.
 Goes both ways I guess...


----------



## asymcon

@RRod - I personally don't mind those high noise levels, but it's possible in certain situations, especially with powerful noise-shaping in quiet portions, the high-frequency noise is disturbing.
 Yes, most audiophiles listen close to 85dBA.


----------



## Cerastes

jwolf said:


> But, are you sure we are not getting any benefits in the audio range?


 

 Depends how old you are and what your hearing capabilities are.
  
 Cheapest way to test your hearing if it's "High-Res ready" is to buy a dog whistle... or just visit your local audiologist.
  
 Personally as a 29yo male I can only hear up to 17 kHz now, so anything beyond CD specifications are wasted on me.


----------



## RRod

asymcon said:


> @RRod - I personally don't mind those high noise levels, but it's possible in certain situations, especially with powerful noise-shaping in quiet portions, the high-frequency noise is disturbing.
> Yes, most audiophiles listen close to 85dBA.


 
  
 And RMS level of shaped dither should be in the -70dbFS range, so we're at 15dBA. So if we're willing to consider people listening in NR20ish rooms, then ok I guess.


----------



## asymcon

rrod said:


> And RMS level of shaped dither should be in the -70dbFS range, so we're at 15dBA. So if we're willing to consider people listening in NR20ish rooms, then ok I guess.


 

 There's huge variety of noise-shaping filters, SoX have bunch built-in, could be interesting to actually test NS audibility.
 Anyways, it does present practicality issue - say I want to EQ notch 16.23kHz on both channels. After such processing I have to apply the same noise-shaping algorithm as the one used in original recording to keep original DR, but since there's such huge variety of filters, it's just easier to work with 20bit.


----------



## RRod

asymcon said:


> There's huge variety of noise-shaping filters, SoX have bunch built-in, could be interesting to actually test NS audibility.
> Anyways, it does present practicality issue - say I want to EQ notch 16.23kHz on both channels. After such processing I have to apply the same noise-shaping algorithm as the one used in original recording to keep original DR, but since there's such huge variety of filters, it's just easier to work with 20bit.


 
  
 Well no one argues to work in higher integer or float formats, it's about effects on the end product when you do the final conversion via SoX or whatever. Do you often notch at 16kHz?


----------



## asymcon

@RRod Lots of movie soundtracks have this little bugger around 16k:

  
 It's super painful to my ears, and I usually wonder if those sound engineers left it there on purpose.
  
 I meant matching NS with already noise-shaped 16bit recording. But I shouldn't probably care and just use TPDF dither 
	

	
	
		
		

		
		
	


	




 
 EDIT: It's 50.3dBFS in right channel and 56.3 in left one


----------



## 441879

jwolf said:


> The one linked in the first post of this thread. It's all about sine waves and has very little to do with music.




Fourier proved a long time ago that all wave forms (including music) can be described as a series of sine waves summed together. He used it to explain harmonics and other real world observable phenomena. It's pretty undisputed that music is sine waves.

Its my understanding that it's how your brain hears as well- a bunch of hairs inside your ear that sinusoidally vibrate at different frequencies.


----------



## castleofargh

jwolf said:


> Have any of you actually just listened for yourself? That's all that's need to be done. You don't have to read this article or that post. You don't need to know the science behind digital audio. You just need to listen and make your own decision as to what you prefer. HAVE YOU ACTUALLY LISTENED? All you seem to be doing is to try to discredit Hi-Res audio with meaningless articles/blog posts. Just listen.
> 
> Post yor results of listening. And no ABX please.IT's OK that you know what the source is you are listening to. So just listen. That's all.


 
 yeah just listen without controls, because as we know that works perfectly and we never get fooled by any kind of biases. much better to deny that 10 problems exists with the test that says I'm right, than acknowledge ABX because "it stresses me".  that's your logic, not logic.
  
 external elements, like reading the damn resolution of the file when pretending to test if we recognize it... or not checking if it's the same master, or the same loudness, or if it's not your gear that makes the 2 resolutions to sound different... all those stuff could cause you to experience a difference(real or not) but how do you know if that difference has anything to do with the resolution of the signal? well you don't. you heard or you think you heard a difference and jump to the conclusion that highres is better and dismiss all the other possibilities. that's called confirmation bias https://en.wikipedia.org/wiki/Confirmation_bias
  your "way" of testing is like playing cluedo when you have decided from the start that colonel mustard is the killer, the game starts, there is a murder, one of the guys in the room is colonel mustard and you stop the game right there and go shouting "see I knew it was mustard, just look!".  of course on a statistical level you will be tend to be wrong, but as you won't go any further, you will never know if you were wrong.
  
 but if what you were looking for is the truth instead of showing you agree with yourself, then you would also test for all the other possibilities before coming with your unjustified certainties. we try to do just that with some ABX. we take a highres file down convert then convert back up to 24/96, so that the music content is only what a 16/44 can do, but in the highres container so that we make sure it's not our DAC or playier or windows mixer... that creates an audible difference between 16/44 and 24/96. also we don't know what music is playing, so no confirmation bias. as the file used to make both samples is the same highres file, we also avoid being fooled by 2 different masters or 2 different loudness levles. so yes some guy may panic a little the first 2 or 3 times he tries to pass an abx, so what, just train a little until you feel more relaxed. that's one little problem to solve all the above mentioned ones. 
  
 and just in case you believe you don't get fooled like anybody else and your senses are always giving you the real thing:
 how our brain mixes other senses with sound and gives priority to vision when it gets 2 conflicting cues (like reading that 2 files are different but hearing the same sound...)

  
  
 how some things are not always what they seem:

 2 different tones right? nope, up and down tiles have the same grey and you can't see it unless you hide the middle of the pic.
  
  
 or how thinking you know something makes you more likely to perceive it even when it's only in your head:

 and if you think making 2 cakes still has the potential for the 2 cakes to really taste different (not the same place in the oven or whatever), then try this one:

  
  
 if after that you don't see the need of blind testing to remove what you think you know about the tracks, I can do nothing for you.


----------



## JWolf

cerastes said:


> Depends how old you are and what your hearing capabilities are.
> 
> Cheapest way to test your hearing if it's "High-Res ready" is to buy a dog whistle... or just visit your local audiologist.
> 
> Personally as a 29yo male I can only hear up to 17 kHz now, so anything beyond CD specifications are wasted on me.


 
  
 That's not at all what I said. I asked are we getting any benefits with Hi-Res in the audible range?


----------



## JWolf

castleofargh said:


> or not checking if it's the same master


 [/quote]

  
 When you buy, you buy what's available. If the Hi-Res copy and the CD copy are different masters, then that's what you have to work with. Now in some cases, you can get he same master from some of the shops that sell Hi-Res downloads. But inmost cases, you can't so you go with what's available. If it turns out that Hi-Res is better because better care has been taken with the mastering of the Hi-Res vs. the CD, then so be it. As has been said before, if you feel the Hi-Res is better only because of the better mastering, then you can always convert down to 16/44.1 and use that for your listening.


----------



## RRod

jwolf said:


> That's not at all what I said. I asked are we getting any benefits with Hi-Res in the audible range?


 
  
 You dismiss the importance of out ability to hear tones and then ask about phenomena in the audible range, which is defined based on our ability to hear tones...


----------



## Rearwing

castleofargh said:


> yeah just listen without controls, because as we know that works perfectly and we never get fooled by any kind of biases. much better to deny that 10 problems exists with the test that says I'm right, than acknowledge ABX because "it stresses me".  that's your logic, not logic.
> 
> external elements, like reading the damn resolution of the file when pretending to test if we recognize it... or not checking if it's the same master, or the same loudness, or if it's not your gear that makes the 2 resolutions to sound different... all those stuff could cause you to experience a difference(real or not) but how do you know if that difference has anything to do with the resolution of the signal? well you don't. you heard or you think you heard a difference and jump to the conclusion that highres is better and dismiss all the other possibilities. that's called confirmation bias https://en.wikipedia.org/wiki/Confirmation_bias
> your "way" of testing is like playing cluedo when you have decided from the start that colonel mustard is the killer, the game starts, there is a murder, one of the guys in the room is colonel mustard and you stop the game right there and go shouting "see I knew it was mustard, just look!".  of course on a statistical level you will be tend to be wrong, but as you won't go any further, you will never know if you were wrong.
> ...





 Right on, I'm plus 1 with this. After years of eye witness testimony being proven to be plain wrong, I am certain that blind tests are the only way forward, we can be biased by so many issues it is almost laughable.


----------



## castleofargh

jwolf said:


> castleofargh said:
> 
> 
> > or not checking if it's the same master


 
  
 When you buy, you buy what's available. If the Hi-Res copy and the CD copy are different masters, then that's what you have to work with. Now in some cases, you can get he same master from some of the shops that sell Hi-Res downloads. But inmost cases, you can't so you go with what's available. If it turns out that Hi-Res is better because better care has been taken with the mastering of the Hi-Res vs. the CD, then so be it. As has been said before, if you feel the Hi-Res is better only because of the better mastering, then you can always convert down to 16/44.1 and use that for your listening.
 [/quote]


 but then we're talking about how well the master was done, not resolution. if you had come here saying you like a certain list of masters better in the available highres version (SACD or PCM) compared to the master available on CD, nobody would have had anything to say. we would have discussed taste and mastering talent and I might have just bought a few albums of your list because I'm not against new good music. but you didn't do that, you came saying that highres was audibly different and superior to CD, used some sudo science to argument while pissing all over a very ok article.
 your post right now makes a lot of sens and I agree, the stuff you said before, it was real bad on many levels. if we could continue with the new direction, I would be very happy.


----------



## JWolf

rrod said:


> You dismiss the importance of out ability to hear tones and then ask about phenomena in the audible range, which is defined based on our ability to hear tones...


 
  
 Some are saying they cannot hear above 17Khz and others are saying similar things, So, what I am asking is if Hi-Res makes a difference in the audible audio range.


----------



## JWolf

castleofargh said:


> but then we're talking about how well the master was done, not resolution. if you had come here saying you like a certain list of masters better in the available highres version (SACD or PCM) compared to the master available on CD, nobody would have had anything to say. we would have discussed taste and mastering talent and I might have just bought a few albums of your list because I'm not against new good music. but you didn't do that, you came saying that highres was audibly different and superior to CD, used some sudo science to argument while pissing all over a very ok article.
> your post right now makes a lot of sens and I agree, the stuff you said before, it was real bad on many levels. if we could continue with the new direction, I would be very happy.


 
  
 Well, I can say the Paul McCartney Hi-Res are very good. The SACD version of _Jeff Wayne's Musical War of the Worlds_ is also very good.


----------



## RRod

jwolf said:


> Some are saying they cannot hear above 17Khz and others are saying similar things, So, what I am asking is if Hi-Res makes a difference in the audible audio range.


 
  
 And I'm saying that it's hard to know what to say if you don't accept a time/frequency duality. For instance, hi-res will have better rise times, but those can be seen as due entirely to the presence of high frequency content. In this case there is no way to drop the frequencies and keep the time-based phenomenon. I don't know of a time-based feature that shows up in hi-res that we can actually hear; maybe someone else does.


----------



## old tech

jwolf said:


> But, are you sure we are not getting any benefits in the audio range?


 
 Why would there be?  The analog reconstruction in the 20-20khz audio range is exactly the same with 44.1 as it is with 96 or 192.  The noise floor and dynamic range at 16 bits already exceeds the capacity of most music and analog transducers.  So where is this benefit in the audio range - assuming of course your equipment is ok?
  
 As for your other question of just listening, yes I can hear the difference between many hi res album and its CD versions which is also apparent on double blind listens.  The difference always is the better mastering of the hi res album.  I have also heard several hi res albums which are inferior in sound to their CD version (eg SACD of DSOTM vs the 83 Sony mastered black triangle CD), again it is the mastering - though if I wanted to be cheeky I could use this as an example to mount a case that CDs sound better than hi res...
  
 However, if I take a hi res recording and downsample it to 16/44, I can hear no difference even when "just listening".  The lack of hearing a difference of course could just be expectation bias too - ie I don't expect to hear a difference so I don't - which is why I do double blind tests to confirm differences I am hearing, or not hearing.


----------



## Joe Bloggs

joe bloggs said:


> There's two issues here that both sides are conflating:
> 
> 1. Whether it's _possible_ to make 16/44.1 CD material to sound the same as high-res material
> 
> ...




Worth repeating as we inch toward that elusive consensus, which may continue to elude us because of the above. I think JWolf gets it now--his only rebuttal to castle's otherwise excellent post is the fact of different hi-res vs CD masters available and entirely valid.


----------



## coli

blade007 said:


> _Brad Meyer and David Moran_ from the _Audio Engineering Society_ did such a study. Subjects sat at a chair and listened to a SACD/DVD-A sound source directly vs piping through a 16bit/44.1kHz A/D/A device. Subject were asked which source was superior.
> 
> Out of 554 trials, 276 picked the pure SACD/DVD-A source. That is 49.82%, and is pretty much 50/50 chance.
> 
> ...


 
 Nah, it depends on your equipment and how good/bad it is. Eg: 48khz is a much easier clock frequency than 44.1khz for a lot of equipment out there, even some hightly expensive snake/ I mean audiophile ones....


----------



## dprimary

coli said:


> Nah, it depends on your equipment and how good/bad it is. Eg: 48khz is a much easier clock frequency than 44.1khz for a lot of equipment out there, even some hightly expensive snake/ I mean audiophile ones....


 

 How is it any easier? Most DAC will clock to any frequency you lock them to, as long as you don't push them beyond their maximum.


----------



## coli

dprimary said:


> How is it any easier? Most DAC will clock to any frequency you lock them to, as long as you don't push them beyond their maximum.


 
 You can literally hear the difference. Eg: PS Snakeoil DAC.


----------



## limpidglitch

coli said:


> You can literally hear the difference. Eg: PS Snakeoil DAC.


 
  
 How do you know it's the clock, and not, say, the AA-filter?


----------



## dprimary

coli said:


> You can literally hear the difference. Eg: PS Snakeoil DAC.


 

 I really have no idea what you trying to say. If I play a 44.1 recording at 48 yes I will hear a big difference.


----------



## JWolf

dprimary said:


> I really have no idea what you trying to say. If I play a 44.1 recording at 48 yes I will hear a big difference.


 
  
 Just how will you hear a big difference when you play a 44.1 recording at 48? That's the biggest load of rubbish I've heard in a long time.


----------



## Joe Bloggs

coli said:


> You can literally hear the difference. Eg: PS *Snakeoil* DAC.




No comment except, what a name :rolleyes:



jwolf said:


> dprimary said:
> 
> 
> > I really have no idea what you trying to say. If I play a 44.1 recording at 48 yes I will hear a big difference.
> ...




I guess he's saying playing a 44.1kHz recording at 48kHz speed, like a sped up record... :rolleyes:


----------



## spruce music

coli said:


> Nah, it depends on your equipment and how good/bad it is. Eg: 48khz is a much easier clock frequency than 44.1khz for a lot of equipment out there, even some hightly expensive snake/ I mean audiophile ones....


 

 There are computer sound cards and such which often are set to work only at 48 khz clock (or their used to be even my 4 year old laptop isn't that way now).  All other sample rates get converted to 48 khz.  Often the conversion wasn't great quality.
  
 Pretty much anything not of bottom of the barrel quality will have clocks derived from 2 crystals.  One crystal tuned to be even multiples of 48, 96,192 or even 384 khz.   One crystal tuned for use at 44, 88, 176, and 352 khz.   Neither more difficult or easier than the other.


----------



## mmerrill99

spruce music said:


> coli said:
> 
> 
> > Nah, it depends on your equipment and how good/bad it is. Eg: 48khz is a much easier clock frequency than 44.1khz for a lot of equipment out there, even some hightly expensive snake/ I mean audiophile ones....
> ...


How many ES9018 DACs use only one 100MHz clock? Bottom of the barrel?


----------



## Joe Bloggs

mmerrill99 said:


> How many ES9018 DACs use only one 100MHz clock? Bottom of the barrel?




Well, a 112896000Hz clock (112.896MHz) is just so freaking high a clock rate that it's an integer multiple of both 44.1kHz and 48kHz. Or 384kHz and 352.8kHz. Or 512x DSD. So there


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > How many ES9018 DACs use only one 100MHz clock? Bottom of the barrel?
> ...




Wasn't talking about 112.896MHz - was talking about any ESS DAC + 100MHz with PLL - so right back at ya :tongue_smile:


----------



## KeithEmo

mmerrill99 said:


> How many ES9018 DACs use only one 100MHz clock? Bottom of the barrel?


 
  
 There are a whole lot of different facts "going on" here.... so here's a bit of clarification....
  
 1) In the "old days", back when PLLs were used a lot, a DAC did simply "lock onto" the incoming signal - and so would work with most frequencies within its operating range. However, most modern DACs use a crystal-based oscillator to regenerate or lock onto the signal, and those have a very narrow range - so most modern DACs will only lock onto frequencies that are quite close to one of the standard sample rates. However, this isn't a problem because virtually all modern signal sources also use a crystal oscillator, so you're very unlikely to see a source operating at a non-standard frequency anyway.
  
 2) Many modern DACs do in fact use one crystal to generate all the frequency multiples of 44k (44k, 88k, 176k) and another to generate multiples of 48k (48k, 96k, 192k). However, there are other ways of doing it that use a single oscillator frequency (like a programmable clock generator). It is also true that many DACs from a few years ago did in fact have trouble with certain sample rates - usually due to limitations in specific integrated circuits used in them. You need to avoid overgeneralizing.... certain chips do perform much better if you use them with two separate crystals for the two sets of sample rates, but you shouldn't assume that this is true for other chips, and other DAC topologies.
  
 Our original Emotiva XDA-2 DAC wouldn't allow 176k via USB - because of a limitation in the USB interface chip we used; even though the DAC itself supports all standard sample rates via its other inputs, and all sample rates except 176k via USB; the newer Gen2 model used a newer USB interface chip which didn't have that limitation. Many DACs and USB-to-S/PDIF converters from a few years ago had this limitation, and wouldn't support 88k, or both 88k and 176k, via USB, while still supporting the higher 192k sample rate. However, it was a limitation of certain specific (and popular) chip sets. (So it would be fair to say that certain chips "have trouble with 88k or 176k"; but not to generalize that statement to everyone.) 
  
 2a) Back when virtually all DACs used either one or two fixed-rate crystals, units that lacked separate 44k and 48k clocks often didn't perform as well as those with separate clocks, and so it became a "sort of fact" that better DACs usually used separate crystals. (Again, that isn't true for DACs that use other methods altogether.)
  
 3) The Sabre DACs do the equivalent of performing a sample rate conversion on the incoming signal to reduce or eliminate jitter. They run on anywhere between one and three clocks (depending on how they're connected), and not all of those clocks are especially critical to the overall performance of the DAC.
  
 The whole point of a well designed DAC is for the designer to know which clocks (and power supplies) are critical, and which ones aren't, and to expend their effort (and parts budget) efficiently to improve the ones that actually matter to the sound quality. Using a poor quality clock in an important location will degrade sound quality; but using a high quality clock in a non-critical location simply raises costs while not improving sound quality at all. And, while using all high end parts may be nice if you have an unlimited budget, if your budget is limited, using expensive or "elegant" parts where you don't need them reduces the budget you have to spend where it matters, and so can actually hurt your performance in the end.


----------



## dprimary

jwolf said:


> Just how will you hear a big difference when you play a 44.1 recording at 48? That's the biggest load of rubbish I've heard in a long time.


 

 the pitch will change the speed will change pretty hard to miss


----------



## spruce music

mmerrill99 said:


> How many ES9018 DACs use only one 100MHz clock? Bottom of the barrel?


 

 "Pretty much anything" has different meaning than if I had said "everything" which is why I didn't say everything.  In general at least in most gear I have run into the better stuff uses two crystal clocks in recent years.  Yes ESS Dacs are not bottom of the barrel, but they are one reason I didn't say everything.  They are resampling and using one clock.  Benchmark did (and as far as I know still do) resample all rates to an in between rate for reasons of jitter reduction.  There are some examples of doing it a few other ways too. 
  
 So hopefully you can discern the difference in meaning now.


----------



## cjl

dprimary said:


> the pitch will change the speed will change pretty hard to miss


 

 No it won't. We aren't talking about just taking the same samples and playing them back at 48ks/s, we're talking about resampling to 48k, which won't change the pitch or speed at all.


----------



## KeithEmo

It sort of depends on the context....
  
 In "the old days" there were such things as tape recorders with "pitch" controls that had a very wide range, and there were a few digital devices that offered a similar option (mostly intended to correct recordings that were "off pitch" or to adjust the length of audio recordings to match the video they were supposed to go with). There were even some dictation machines that would let you speed up playback if you could type faster (modern versions of this digitally shift the pitch back to where it belongs after changing the base speed). And today there are a few special effects plugins that still allow you to deliberately shift the speed - with our without correcting the pitch - and do all sorts of other weird stuff.
  
 However, today, if you "pick a different sample rate" on a playback device, virtually all modern devices and programs will re-sample the audio and convert it to the new sample rate. (If this is done properly, there should be little or no audible difference.) The problem is that some devices and programs will do a much better job than others, and some are pretty bad. This is a common occurrence when you play digital audio on a computer; by default, both Windows and Apple OS/X are set for a single default sample rate, and convert anything they play to that sample rate; this means that anything you play that starts out at any other sample rate is being resampled (you need to select specific player programs and options to override this behavior). Likewise, there are common sample rates that are used by certain things by default - for example, 44.1k is the (only) allowed sample rate for Red Book CD audio, and 48k is commonly used by the audio that accompanies digital video (so, if you RIP the audio from a movie DVD, and don't convert it, you'll probably end up with 48k audio; the higher end formats support higher sample rates as well, but 48k is still the "bottom sample rate" for most of them).
  
 Quote:


cjl said:


> No it won't. We aren't talking about just taking the same samples and playing them back at 48ks/s, we're talking about resampling to 48k, which won't change the pitch or speed at all.


----------



## dprimary

cjl said:


> No it won't. We aren't talking about just taking the same samples and playing them back at 48ks/s, we're talking about resampling to 48k, which won't change the pitch or speed at all.


 
 I specifically said "If I play a 44.1 recording at 48 yes I will hear a big difference." resampling was not talked about.


----------



## dprimary

> In "the old days" there were such things as tape recorders with "pitch" controls that had a very wide range, and there were a few digital devices that offered a similar option (mostly intended to correct recordings that were "off pitch" or to adjust the length of audio recordings to match the video they were supposed to go with). There were even some dictation machines that would let you speed up playback if you could type faster (modern versions of this digitally shift the pitch back to where it belongs after changing the base speed). And today there are a few special effects plugins that still allow you to deliberately shift the speed - with our without correcting the pitch - and do all sorts of other weird stuff.


 
 We still can and still have to. Most master clocks have fixed, pull and pull down for video and film which are slightly off the standard 44.1,48, 96 ....
 Many also have the ability to set any sample rate you want or need. How they are able to vary the rate is not exactly clear to me, most master clocks have very high precision clocks, and many have options to lock to a 10MHz atomic clock. If you have a complex with many control rooms it would make sense to have an atomic master distributed to the master clocks in each control room.


----------



## gregorio

dprimary said:


> Most master clocks have fixed, pull and pull down for video and film which are slightly off the standard 44.1,48, 96 ....
> Many also have the ability to set any sample rate you want or need. How they are able to vary the rate is not exactly clear to me, most master clocks have very high precision clocks, and many have options to lock to a 10MHz atomic clock. If you have a complex with many control rooms it would make sense to have an atomic master distributed to the master clocks in each control room.


 
  
 In film, we commonly used pull-up or pull-down rates in the digital intermediate, post-prod phase, so that we could edit at video speed and then convert back to (35mm) film speed. That workflow is less common today, although there is still a common issue of the 0.1% difference between video speed and film speed. Generally the masterclocks we use are not extremely high precision, they are very decent crystal oscillators running at many tens of megahertz but there's absolutely no need for an atomic clock (which run well into the GHz range), that's more like one of those ridiculous audiophile excesses! There are different schemes employed, some of which are fairly complex or fairly simple PLLs or some combination but locking clock signals together with the accuracy needed for film audio doesn't benefit from atomic clocks.
  
 G


----------



## Hitec

People keep comparing bad resolution to another bad level of resolution.  Too me the test should be about what can be done to get to an analog sound.  Are we saying here that no one can hear the difference between an analog two-track tape, and a CD recording.  At the end of the day we want the effortless  sound of analog that we have heard for centuries, but with the perfection of digital.  How do we do that?  During the shift to digital, I noticed that all of the audio equipment being sold at audio stores started to sound like the music was super congested, with the imaging the size of a dot.  The systems use to sound big, spacious, and not hard.  Also, CD's hurt your ears at a loud volume.  It is like the music images are fighting one another.  Music has become a robot.  The soul of he music is half dead.  So why are we comparing CD to MP3 to whatever else.  All of them are bad.  We should be comparing them to which one sounds like analog.  It can't be better than analog because whether you can hear it or not, and depending what infinite level you observe, CD's will always have missing samples.  
 I think the goal would be to recreate analog to the atom.


----------



## WindowsX

I disagree that we don't need atomic clock. I use masterclock in my digital transport/dac. Even with 0.1/0.02ppm precision won't come close to 0.05ppb (0.00005ppm). It made digital playback sounds pretty close to analog vinyl records. When I demonstrate the effect of masterclock and atomic clock in my system, all audiophiles who listened to my system can clearly tell the difference apart.
  
 Regards,
 Windows X


----------



## dprimary

gregorio said:


> In film, we commonly used pull-up or pull-down rates in the digital intermediate, post-prod phase, so that we could edit at video speed and then convert back to (35mm) film speed. That workflow is less common today, although there is still a common issue of the 0.1% difference between video speed and film speed. Generally the masterclocks we use are not extremely high precision, they are very decent crystal oscillators running at many tens of megahertz but there's absolutely no need for an atomic clock (which run well into the GHz range), that's more like one of those ridiculous audiophile excesses! There are different schemes employed, some of which are fairly complex or fairly simple PLLs or some combination but locking clock signals together with the accuracy needed for film audio doesn't benefit from atomic clocks.
> 
> G


 

 The output of an atomic clock is only 1-to 20 MHz, with 10MHz being the most common. Internally they can go into the GHz range.
  
 I am not even suggesting  that an audiophile needs an atomic, a master clock or any external clock for that matter. Master clocks are only needed for production, and it is as you point out getting less necessary to even do pull-ups and pull-down very more rare to need to change the speed to something completely odd. You still run into the need to do though often it is from something that was done wrong it some other time. 
  
 Clocks create as many problems as they solve. For people involved in production they are a everyday part of life and most tend to notice when something is going wrong. Unfortunately many don't, the most common is the devices in the signal chain have to be locked to the same clock but are not causing small ticks and pops. Another issue is you cannot run word clock signals very far before you are worse off then where you started. 
  
 House sync and distributed masterclocks are fairly difficult to implement in a large complex of studios and editing suites. Even a small studio has to deal with much longer distances then people do in their homes.
  
 This is all tangent anyway. My original point is a DAC does not have a preference of a clock rate and should not sound any different  at different rates (as long it is the same rate the recording was done at). If it does it is broken - get it fixed.
  
 When people claim to hear " night and day differences" on things that you should not. The answer is simple - it is broken - get it fixed.


----------



## OddE

windowsx said:


> I disagree that we don't need atomic clock. I use masterclock in my digital transport/dac. Even with 0.1/0.02ppm precision won't come close to 0.05ppb (0.00005ppm).
> 
> It made digital playback sounds pretty close to analog vinyl records. When I demonstrate the effect of masterclock and atomic clock in my system, all audiophiles who listened to my system can clearly tell the difference apart.


 
  
 -And this test was done blind, of course. You didn't first let them listen to a stock system, then explain that you were now about to connect an expensive, atomic clock reference to the system, which would lift the veil and make the system sound much more analog-like and whatnot - before putting on another track and asking them whether they could tell the difference?


----------



## WindowsX

odde said:


> -And this test was done blind, of course. You didn't first let them listen to a stock system, then explain that you were now about to connect an expensive, atomic clock reference to the system, which would lift the veil and make the system sound much more analog-like and whatnot - before putting on another track and asking them whether they could tell the difference?


 
  
 No. I secretly removed atomic clock connection while changing cables and they asked me what happened because the sound got worse. I've done this few times and everyone can notice it. The reason why I didn't let them listen to inferior performance first because getting used to better sound before dropping down is easier to notice.
  
 Regards,
 Windows X


----------



## jnorris

Turntables and tape players all spin at rates that are internally generated and monitored, with no common reference.  Turntables and tape decks suffer from belt wear, varying values of internal components (RC networks that define clock speeds, specifically), warpage, drag, and a host of other afflictions that affect overall playback speed.  It would seem to me that all your test did is serve to verify that the increased speed accuracy of digital music sounded more "analog" (that is to say better) than a digital source whose lack of an absolute clock reference more closely emulated an actual analog source.


----------



## castleofargh

hitec said:


> People keep comparing bad resolution to another bad level of resolution.  Too me the test should be about what can be done to get to an analog sound.  Are we saying here that no one can hear the difference between an analog two-track tape, and a CD recording.  At the end of the day we want the effortless  sound of analog that we have heard for centuries, but with the perfection of digital.  How do we do that?  During the shift to digital, I noticed that all of the audio equipment being sold at audio stores started to sound like the music was super congested, with the imaging the size of a dot.  The systems use to sound big, spacious, and not hard.  Also, CD's hurt your ears at a loud volume.  It is like the music images are fighting one another.  Music has become a robot.  The soul of he music is half dead.  So why are we comparing CD to MP3 to whatever else.  All of them are bad.  We should be comparing them to which one sounds like analog.  It can't be better than analog because whether you can hear it or not, and depending what infinite level you observe, CD's will always have missing samples.
> I think the goal would be to recreate analog to the atom.


 
 the term analog for an audio media means loosely that the music played and the music recorded are following the same time stamps and that the process is continuous. if there was 3 seconds between this sound and that sound, then it will take 3 seconds to read it on the analog support. that's the general idea and sure enough an entire song can be buffered into the computer in almost no time at all, and then sent in packets to the DAC, which really isn't an analog method.
  
 now how many masters are still on tape to this day? 
	

	
	
		
		

		
			





 as it happens, for years the recordings have been done digitally on tape(tapes that save ones and zeros), most analog tapes done before have since been transferred to digital tapes, or simply stored on hard discs. so tapes or vinyls nowadays are made from digital for the very vast majority!!!!!!!!
  
 and the reason is pretty obvious, if you take the analog out when playing a CD, the timing will be more accurate(so closer to analog master recording at time=T) than a vinyl or a consumer tape player.
 analog is related to a time stamp, it has never and will never be a proof of signal fidelity. digital audio improved signal fidelity, what the mastering engineers did of it has nothing to do with the support(loudness war), and the fact is that violin alone in real life tends to sound harsh to my ears. so when it sounds harsh on the CD I don't go blaming the CD for ruining the sound. we look for fidelity or we don't, keeping up the analog argument while pretending to look for fidelity is misguided as digital audio outperforms all analog supports in signal fidelity.
  
 IMO you just don't like how some things sound and you need a coating of distortions, noise, and possibly some treble roll off, and super low crosstalk gluing the music together, to feel like the sound is nice and "good" like it was before. and that's ok, taste is taste and I also dislike harsh sounds for the sake of fidelity or strong trebles. but don't mistake what you enjoy and fidelity.
  


windowsx said:


> I disagree that we don't need atomic clock. I use masterclock in my digital transport/dac. Even with 0.1/0.02ppm precision won't come close to 0.05ppb (0.00005ppm). It made digital playback sounds pretty close to analog vinyl records. When I demonstrate the effect of masterclock and atomic clock in my system, all audiophiles who listened to my system can clearly tell the difference apart.
> 
> Regards,
> Windows X


 
  jitter measurements will account for the clock and for anything else that could go wrong in the timing of the signal.  as far as I know, the papers on the subject tend to agree that ok modern gear shouldn't have us concerned with jitter, because it stays below what seems to be audible anyway.
 so while an atomic clock sounds super cool and impressive, and can probably improve the signal in at least a measurable way, it should still sound very much the same. if your experiment gave the results you talk about, I bet my left arm that at least one of the situation had a problem. be it really unusually high jitter you never meet on digital consumer gears, or something else actually worse.
  
  
  
  
  
 am I alone finding that those 2 posts side by side advocating the total opposite from one another are kind of ironic? one asking to go back to trusting mechanical precision for time, the other one saying that we need no less than atomic precision for good sound. funny accident ^_^.


----------



## WindowsX

While I agree that using external clock can bring problems as it can solve because I can't find audiophile BNC cable that works properly in market and ended up having to make ones myself. However, the precision and improvements of timing accuracy is worth the investment for me.
  
 Have you tried implementing high quality external clocks like Esoteric G-0rb or PRS10/Antelope 10M clock with masterclock that supports 10M clock yourself? This is the optimal solution I found for my digital audio setup.
  
 Regards,
 Windowws X


----------



## icebear

castleofargh said:


> ....
> am I alone finding that those 2 posts side by side advocating the total opposite from one another are kind of ironic? one asking to go back to trusting mechanical precision for time, the other one saying that we need no less than atomic precision for good sound. funny accident ^_^.


 
 You're not alone 
	

	
	
		
		

		
		
	


	




.
 Doing cell phone recordings and arguing about software improvements for bit perfect pc based music reproduction and atomic clock precision is pretty funny, too
	

	
	
		
		

		
		
	


	




.


----------



## cjl

windowsx said:


> While I agree that using external clock can bring problems as it can solve because I can't find audiophile BNC cable that works properly in market and ended up having to make ones myself. However, the precision and improvements of timing accuracy is worth the investment for me.


 
 A cable is a cable (as long as it isn't a poorly designed cable that doesn't meet spec). You don't need an audiophile BNC cable.


----------



## WindowsX

icebear said:


> You're not alone
> 
> 
> 
> ...


 
  
 Fallacy spotted! That's not very scientific statement. All I did say here is sharing the best I find from my experience. There's no need to use such foul speech like that. Looks like voicing the different ideas isn't welcomed here. Bye.
  
 Quote: 





cjl said:


> A cable is a cable (as long as it isn't a poorly designed cable that doesn't meet spec). You don't need an audiophile BNC cable.


 
  
 I tried making 3 different sleeving from the same cable. Somehow, as BNC cable, they all sound different. But I tested with something like $20-30k equipment level so it may not show such significant different on 3-4 digits products range. But I agree that cables don't need to be expensive and always encourage my friends to use proper designed cables instead of highend cables.
  
 Regards,
 Windows X


----------



## bfreedma

windowsx said:


> icebear said:
> 
> 
> > You're not alone
> ...




At least two of the cables must be defective.


----------



## cjl

windowsx said:


> I tried making 3 different sleeving from the same cable. Somehow, as BNC cable, they all sound different. But I tested with something like $20-30k equipment level so it may not show such significant different on 3-4 digits products range. But I agree that cables don't need to be expensive and always encourage my friends to use proper designed cables instead of highend cables.
> 
> Regards,
> Windows X


 

 I use a $50k oscilloscope with BNC connectors all the time at work for all kinds of interesting (and extremely high frequency/low amplitude) measurements. You don't need an audiophile BNC cable.


----------



## sonitus mirus

cjl said:


> I use a $50k oscilloscope with BNC connectors all the time at work for all kinds of interesting (and extremely high frequency/low amplitude) measurements. You don't need an audiophile BNC cable.


 
  
 Yes, and I'm confident that any engineer designing $30K audio equipment is similarly using basic BNC connectors with their test equipment.


----------



## WindowsX

I'm also using basic BNC cables. The problem is most audiophile cables don't follow the standards or specifications of applications properly. What I'm trying to convey is more revealing equipment shows the effects of used sleeving on cable. I didn't notice this change when I was using $1,xxx equipment though.
  
 Regards,
 Windows X


----------



## cjl

sonitus mirus said:


> Yes, and I'm confident that any engineer designing $30K audio equipment is similarly using basic BNC connectors with their test equipment.


 

 Almost definitely, though it's astonishing how expensive some scope probes can get (even just passive probes). Of course, those have good reasons for being expensive usually.


----------



## dprimary

hitec said:


> People keep comparing bad resolution to another bad level of resolution.  Too me the test should be about what can be done to get to an analog sound.  Are we saying here that no one can hear the difference between an analog two-track tape, and a CD recording.  At the end of the day we want the effortless  sound of analog that we have heard for centuries, but with the perfection of digital.  How do we do that?  During the shift to digital, I noticed that all of the audio equipment being sold at audio stores started to sound like the music was super congested, with the imaging the size of a dot.  The systems use to sound big, spacious, and not hard.  Also, CD's hurt your ears at a loud volume.  It is like the music images are fighting one another.  Music has become a robot.  The soul of he music is half dead.  So why are we comparing CD to MP3 to whatever else.  All of them are bad.  We should be comparing them to which one sounds like analog.  It can't be better than analog because whether you can hear it or not, and depending what infinite level you observe, CD's will always have missing samples.
> I think the goal would be to recreate analog to the atom.


 

 Can I hear the difference when transferring analog tape to 44.1/16 on good equipment? No. Can I hear the difference transferring 44.1/16 digital to analog two track? yes, every time. Making a recording sound like and analog recording format has never been my goal. Every analog recording format is lossy, adds distortions, and compression (which is just another distortion).
  
 For about a decade it was common the record basic tracks to analog 2" then transfer it into digital for additional tracks and mixing. For the people that learned to love all the distortions that analog tape gave them it worked perfectly.
  
 My goal has always been for it to sound exactly like it did from the microphone through the preamp to an active monitor . Analog recording formats don't even come close, it never plays back what was put into it.


----------



## gregorio

hitec said:


> It can't be better than analog because whether you can hear it or not, and depending what infinite level you observe, CD's will always have missing samples.
> I think the goal would be to recreate analog to the atom.


 
  
 1. There are no missing samples in CD. Except at frequencies beyond human hearing, CD already has more samples than necessary!
  
 2. While there are some who try to recreate analog, that is generally not the goal. A common goal is to recreate the original acoustic energy, not a (relatively) highly imperfect analog representation of it!
  


windowsx said:


> I disagree that we don't need atomic clock. I use masterclock in my digital transport/dac. Even with 0.1/0.02ppm precision won't come close to 0.05ppb (0.00005ppm). It made digital playback sounds pretty close to analog vinyl records. When I demonstrate the effect of masterclock and atomic clock in my system, all audiophiles who listened to my system can clearly tell the difference apart.


 
  
 Unless you have a defective or a terribly designed DAC, an external masterclock will degrade system performance. Although extremely unlikely, it might even degrade it to the level of vinyl.


dprimary said:


> For people involved in production they are a everyday part of life and most tend to notice when something is going wrong.


 
  
 Even in music production, an external masterclock is usually unnecessary these days. External clocking is always inferior and most music production today occurs "in the box", so there is no need to lock different bits of physical kit together and therefore no benefit to a masterclock. In TV/film this is often not the case though, we usually have to at least lock the picture equipment to the audio equipment and therefore an external masterclock can be advantageous.
  
 G


----------



## WindowsX

gregorio said:


> Unless you have a defective or a terribly designed DAC, an external masterclock will degrade system performance. Although extremely unlikely, it might even degrade it to the level of vinyl.


 
  
 I don't think Emm Labs/Esoteric/dCS/Wadia have defective design. Have you tried those gears in your system?
  
 Regards,
 Windows X


----------



## dprimary

gregorio said:


> Even in music production, an external masterclock is usually unnecessary these days. External clocking is always inferior and most music production today occurs "in the box", so there is no need to lock different bits of physical kit together and therefore no benefit to a masterclock. In TV/film this is often not the case though, we usually have to at least lock the picture equipment to the audio equipment and therefore an external masterclock can be advantageous.
> 
> G


 
 Agreed, for most small studio master clocks will cause more problems then it solves, today it is likely everything is happening in one box so it would be silly to connect it to a master clock. I often have to deal with large complexes of studios, along with broadcast. Some systems have live venues, recording and broadcast so sync and master clock become very important.
  
 Most production people are only going to spend money on master clock when they really need one. They would much rather spend money on more microphones, preamps, and software.


----------



## WindowsX

"most small studio master clocks will cause more problems then it solves"
  
 I agree. Master clocks aren't things for small scale implementation. Most studios and stereo systems implementing master clocks should spend well on others first. If you still have budget and want to improve the production quality, that's when master clock comes into the play. I wouldn't invest on master clock before speakers, amp, and sources after all. 
  
 Regards,
 Windows X


----------



## spruce music

https://www.soundonsound.com/sos/jun10/articles/masterclocks.htm
  
 Might wish to read this shootout/test of external clocks from a few years back. 
  
 The quick takeaway is even with expensive precision clocks using an external clock increased jitter in each case. If you don't need it to sync multiple AD or DA devices don't use one.


----------



## WindowsX

I need masterclock in my stereo system for single reclock that feed the word clock to both transport/dac. Your suggested link is one article from pro audio magazine. It's like reading audiophile magazine article to suggest equipment in some ways.
  
 But truthfully speaking, I hardly see any decent implementation of master clock systems too. A lot of my audiophile friends who used to implement masterclock now stopped using it since he couldn't get the timing and fine-tuning right. So yeah, getting masterclock to work right is hard to do. It really can cause problems as much as it solves.
  
 Regards,
 Windows X


----------



## watchnerd

jwolf said:


> Post yor results of listening. And no ABX please.*IT's OK that you know what the source is you are listening to*. So just listen. That's all.


 
  
 Wow, so much lack of knowledge in this statement.


----------



## watchnerd

jwolf said:


> if you feel the Hi-Res is better only because of the better mastering, then you can always convert down to 16/44.1 and use that for your listening.


 
  
 I do this all the time, both actively via SoX, but also passively when I stream something high res and it gets down-converted.


----------



## Hitec

Nice: live to 2-track to CD's here.  The Wildfire line.  http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3.   Why isn't music processed today to give us this type of sound...  If the high resolution samples don't matter, then what is broken in today's process?  Of course this sample is suppose to be filled with ALL that analog noise.  I think I like that noise 
  
 http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3


----------



## watchnerd

hitec said:


> Nice: live to 2-track to CD's here.  The Wildfire line.  http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3.   Why isn't music processed today to give us this type of sound...  If the high resolution samples don't matter, then what is broken in today's process?  Of course this sample is suppose to be filled with ALL that analog noise.  I think I like that noise
> 
> http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3


 
  
 If you like tape artifacts, there are a number of DAW plugins that can add that to a recording during the mixing processing.


----------



## JWolf

hitec said:


> Nice: live to 2-track to CD's here.  The Wildfire line.  http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3.   Why isn't music processed today to give us this type of sound...  If the high resolution samples don't matter, then what is broken in today's process?  Of course this sample is suppose to be filled with ALL that analog noise.  I think I like that noise
> 
> http://mapleshaderecords.com/cds/mp3s/bigjoemaher1.mp3


 

 This is a thread about Hi-Res vs.CD. Don't link us crappy mp3.


----------



## Hitec

I'm talking about good production, mastering, and fidelity.  You can hear that in the MP3 that I referenced.  So, imagine what you would hear from the CD version.  I think mp3 formats are crappy, but they can be made to sound better than a crappy CD, that did not have fidelity.


----------



## castleofargh

making crap sound like crap is fidelity well applied. ^_^


----------



## Hitec




----------



## watchnerd

hitec said:


> I'm talking about good production, mastering, and fidelity.  You can hear that in the MP3 that I referenced.  So, imagine what you would hear from the CD version.  I think mp3 formats are crappy, but they can be made to sound better than a crappy CD, that did not have fidelity.


 
  
 Actually, I'm going to disagree with that: 
  
 The drum kit sounds like crap, like a combination of a cardboard box and a garbage can lid.
  
 Bad mic placement and/or mic choice on the drum.


----------



## Hitec

Thanks for your honest reply watchnerd - All the music is real, with real people playing.   Is that what you use - a drum kit.  The musicians that recorded there disagree with you - they treasured their relaxed, natural, real life setting and production experience.
  
 Hey, check these below refs. out.  The CD versions are waay better - doesn't have that wavering sound.  None of my David Sanborn CD's and other CD's sound as good as this stuff.  When I play these, heads turn, and I have to tell them that it is  an old school live to 2-track with no filters, compression or anything.  Note, I'm into listening and buying, not making and promoting.  People tell me that these references sound closer to real.  Should I reply,that they are wrong, and that my David Sanborn CD's  sound better (using Sanborn as an example). ..and that they just can't appreciate the newer technology that is built into latter CD's?
  
 24bit, 48bit, 192khz, whatever, can you tech guys just help us make the music sound like the performers are there with us in the house - you know like they show on Star Trek when they go to the simulation deck   The stuff today, does not sound close.  Also, everyone acts like there is NO room for improvement.  In time, you will see that there was PLENTY of room for improvement - you just don't know about it yet.  Maybe there is yet another unknown periodic element waiting to be discovered.
  
  
 link note:
 highlight, cntrl-c (copy)
 cntrl-v (paste)
  
 free samples
  
 1. http://mapleshaderecords.com/cds/mp3s/tonywilliamson1.mp3
 2. http://mapleshaderecords.com/cds/mp3s/johncocuzzi1.mp3
 3. http://www.mapleshaderecords.com/cds/mp3s/norristurney1.mp3
 4. http://mapleshaderecords.com/cds/mp3s/tonywilliamson3.mp3
 5. http://www.mapleshaderecords.com/cds/mp3s/michaelcarvin7.mp3
  
  
 48k and beyond may be noted here as worthless on the technical side, and it may also be worthless to the non-tech saavy that just want something that is developed with more fidelity that they can hear.  It does not matter what was technically proven if a layman does not understand the proof.  So, I have no idea why people enjoy the samples that I referenced, but most I encouter like what they hear in comparison to what has been pushed to them. Granted these all overall have the same room sound, since that's all you have when you don't use anything but live to 2-track.
  
 So what does 24bit or 48khz or anything greater improvements have to do with what we hear.  I think what we hear should be the ultimate goal.  Why are we building a perfect bridge that no one uses.
  
  
 Thanks,


----------



## dprimary

watchnerd said:


> Actually, I'm going to disagree with that:
> 
> The drum kit sounds like crap, like a combination of a cardboard box and a garbage can lid.
> 
> Bad mic placement and/or mic choice on the drum.


 

 It is recorded with PZM's . It takes all the acoustical character out of the room. I like it better then common over damped lifeless "modern jazz" that I often hear. I hate acoustic recordings that I can't hear the room, I have no sense of the size of the room or where the walls are.


----------



## dprimary

hitec said:


> Thanks for your honest reply watchnerd - All the music is real, with real people playing.   Is that what you use - a drum kit.  The musicians that recorded there disagree with you - they treasured their relaxed, natural, real life setting and production experience.
> 
> Hey, check these below refs. out.  The CD versions are waay better - doesn't have that wavering sound.  None of my David Sanborn CD's and other CD's sound as good as this stuff.  When I play these, heads turn, and I have to tell them that it is  an old school live to 2-track with no filters, compression or anything.  Note, I'm into listening and buying, not making and promoting.  People tell me that these references sound closer to real.  Should I reply,that they are wrong, and that my David Sanborn CD's  sound better (using Sanborn as an example). ..and that they just can't appreciate the newer technology that is built into latter CD's?
> 
> 24bit, 48bit, 192khz, whatever, can you tech guys just help us make the music sound like the performers are there with us in the house - you know like they show on Star Trek when they go to the simulation deck   The stuff today, does not sound close.  Also, everyone acts like there is NO room for improvement.  In time, you will see that there was PLENTY of room for improvement - you just don't know about it yet.  Maybe there is yet another unknown periodic element waiting to be discovered.


 
 I appreciate what the guy is trying to do. Many of my favorite recordings was done with a pair of microphone. Even for rock my drums start with the room mic's, most of the sound is that pair of mic's. For classical I have never used more then a pair of microphones.  
  
 Musicians tend to like a very close mic'ed sound since they are often very intimate with their instrument. They think theirs sounds dead on but everyone else's in the band doesn't sound right. Close mic'in does not sound right to anyone listening to the band.
  
 A few problems I hear in the recordings is the room is too small for the recording method.  PZM's have a unique sound which is not always very accurate. They solve a certain problem, that I would prefer to solve in correcting the room acoustics. The low frequency response of of a PZM is dependent on the area of the boundary. The engineer has decided that PZM's are the only solution, eliminating all other microphones. There is a reason large studios have large collections of microphones, what works in one case almost never works in the next.


----------



## dprimary

hitec said:


> So what does 24bit or 48khz or anything greater improvements have to do with what we hear.  I think what we hear should be the ultimate goal.  Why are we building a perfect bridge that no one uses.
> 
> 
> Thanks,


 
 Exactly, audio recording and reproduction has never been better. It is far beyond what anyone can hear. Hardly anyone is trying to get all they can out of it.


----------



## Hitec

Thanks dprimary for that information!  Yes, the room is small.  That gives me a more unbias thought process towards understanding what is happening.  Seems this recording style is difficult, like driving a wild mustang that doesn't have all the BMW help-me-drive features built-in   Now, I'm in search of something that is considered, a pretty good recording - live and canned mix tracks.


----------



## watchnerd

hitec said:


> Thanks for your honest reply watchnerd - All the music is real, with real people playing.   Is that what you use - a drum kit.  The musicians that recorded there disagree with you - they treasured their relaxed, natural, real life setting and production experience.


 
 A "drum kit" refers to the entire set up of the drummer - snare, kick drum, hi-hat, etc.  
	

	
	
		
		

		
			




  
 I make live recordings on a weekly basis, both live small venue jazz and live symphonic recordings.
  
 Capturing the acoustics is great, but the choice of mic and mic placement in that recording is not very good.  
  
 A simple pair of stand-mounted cardioids to get the instruments and the venue, plus a mic for the vocalist, could have done much better.
  
 It's nice that the musicians feel good about their relaxed recording, but the audio engineer needs to work on his skills.
  
 (if it was home-made by the musicians themselves, well, okay...but it's no better than a demo cut)


----------



## watchnerd

dprimary said:


> I appreciate what the guy is trying to do. Many of my favorite recordings was done with a pair of microphone. Even for rock my drums start with the room mic's, most of the sound is that pair of mic's. For classical I have never used more then a pair of microphones.
> 
> Musicians tend to like a very close mic'ed sound since they are often very intimate with their instrument. They think theirs sounds dead on but everyone else's in the band doesn't sound right. Close mic'in does not sound right to anyone listening to the band.
> 
> A few problems I hear in the recordings is the room is too small for the recording method.  PZM's have a unique sound which is not always very accurate. They solve a certain problem, that I would prefer to solve in correcting the room acoustics. The low frequency response of of a PZM is dependent on the area of the boundary. The engineer has decided that PZM's are the only solution, eliminating all other microphones. There is a reason large studios have large collections of microphones, what works in one case almost never works in the next.


 
  
 I would have tried my standard starting point for jazz quartets...
  
 1. 2 x AKG C414 stand-mounted in the far-field
 2. Mic for the vocalist using whatever they're used to
 3. If the stand-mounts aren't picking up enough bass, I'd add a Schoeps CMC6 on a boom
  
 This is my standard travel kit for jazz (sans piano).
  
 FWIW, I've made better live recordings than the sample in the far-field using a Shure MV88 over a Thunderbolt connection to an iPhone.


----------



## Joe Bloggs

If you're touting certain CDs as exemplary in recording / mastering and sound quality, and giving out samples of these CDs in the form of mp3s, surely you can agree that the distribution medium (be it CDs, 24/96 or DSD) is not the limiting factor, but rather the recording / mastering? That's what we were arguing for too!


----------



## OddE

windowsx said:


> I'm also using basic BNC cables. The problem is most audiophile cables don't follow the standards or specifications of applications properly.


 
  
 -The solution to this problem is both simple and cheap. Do not, repeat DO NOT buy 'audiophile' grade cables. (Or, simpler still - just don't buy any 'audiophile' kit. )
  
 Heck, any Belden coax assembly will be up to spec, durable and, when compared to 'audiophile' stuff - laughingly affordable.


----------



## JWolf

watchnerd said:


> Actually, I'm going to disagree with that:
> 
> The drum kit sounds like crap, like a combination of a cardboard box and a garbage can lid.
> 
> Bad mic placement and/or mic choice on the drum.


 
  
 Or it could be due to being made MP3.


----------



## JWolf

> Originally Posted by *Hitec* /img/forum/go_quote.gif
> 
> free samples


 
  
 If you are going to give us free samples, stop with the freaking MP3 and give us CD quality of better. Stop with the crappy audio files.


----------



## limpidglitch

jwolf said:


> If you are going to give us free samples, stop with the freaking MP3 and give us CD quality of better. Stop with the crappy audio files.


 
  
 I know they're just humble 128k mp3s, but I thought they sounded quite alright.


----------



## JWolf

limpidglitch said:


> I know they're just humble 128k mp3s, but I thought they sounded quite alright.


 
  
 But given that they can impact the sound of a drum kit and someone has mentioned the drums don't sound quite right, I would say it's the MP3 causing it. Also, MP3 is not what's being discussed here. It's 16/44.1 vs. Hi-Res. Please take down your MP3 links and put up at least CD resolution links. Sounds alright? Nope. It's time you moved to an iPod and iTunes and give up CDs.


----------



## watchnerd

limpidglitch said:


> I know they're just humble 128k mp3s, but I thought they sounded quite alright.


 
  
 Jeez, really?
  
 If you thought those sounded okay, then you definitely shouldn't waste any money on high-resolution stuff.


----------



## cjl

jwolf said:


> Or it could be due to being made MP3.


 

 Really? You've never heard a drum kit sound better than that on MP3? I know I have - that drum kit sounded pretty awful. It definitely wasn't just low bitrate distortion either.


----------



## watchnerd

cjl said:


> Really? You've never heard a drum kit sound better than that on MP3? I know I have - that drum kit sounded pretty awful. It definitely wasn't just low bitrate distortion either.


 
  
 What I heard wasn't due to digital artifacts (those sound different).  It was due to bad audio engineering.
  
 And, no, I'm not a fan of Kenny Sanborn, either....but I would suggest the promote go listen to some of the Reference Recordings that were made on analog tape to learn what can really be done with a tape deck and better engineering.


----------



## jnorris

I don't think any of you got the point of the recording.  It's a simple 2-mike recording of instruments in an untreated room, which, in this case, gives a pretty accurate rendering of how drums actually sound in those conditions (dynamic range notwithstanding).  News flash - unless you're the drummer (or you've got 10 or more ears that can be placed within inches of each drum), real drums do not sound like they do in multi-miked recordings, which are also compressed and equalized.   The drum sound was not caused by the MP3 encoding, especially at 128K, and to even imply that indicates a lack of understanding.  I would like to hear the recordings at 320K or lossless, however, as they would probably be very interesting under the headphones.


----------



## watchnerd

jnorris said:


> I don't think any of you got the point of the recording.  It's a simple 2-mike recording of instruments in an untreated room, which, in this case, gives a pretty accurate rendering of how drums actually sound in those conditions (dynamic range notwithstanding).  News flash - unless you're the drummer (or you've got 10 or more ears that can be placed within inches of each drum), real drums do not sound like they do in multi-miked recordings, which are also compressed and equalized.   The drum sound was not caused by the MP3 encoding, especially at 128K, and to even imply that indicates a lack of understanding.  I would like to hear the recordings at 320K or lossless, however, as they would probably be very interesting under the headphones.


 
  
 You're not getting the point of my critique:
  
 Even in a simple 2 mic recording, that's crappy.
  
 I listen to live jazz in a small venue every single week with no spot mic or reinforcement on the drums.  I've recorded this multiple times for my own collection and had it sound way better than this recording.  The results on this track are predictable if you don't have good mic placement and/or aren't using mics well-suited for the task.
  
 I agree that in real life drums don't sound like they do in studio-made albums, but they don't sound like this hash, either.


----------



## jnorris

I think the recordist was trying to get a fair amount of room ambiance and a sense of depth and air in the recording, so he moved the mic further out and way from the performers.  If he moved it in closer it would have resulted in a totally different sound - possibly one more to your liking.  You are right: placement and mic quality play a huge role.
  
 BTW, I am a drummer and have made dozens of recordings of my band with a simple single-point stereo mic.  Some excellent, some...not so much...


----------



## Hitec

Those were posted by Pierre Sprey on his site as demos.  I would have to ask him if I could demo sections of CD's in FLAC or something..
 I use to demo some of the CD's at International Auto Sound Challenge Association (IASCA) events, when IASCA use to be mostly audiophile competitions and not focused on bass-offs.
 IASCA even featured some of his recordings on media that was used to grade cars.  Great sound in a car is close to impossible, but the best out there is incredible for a car.


----------



## Hitec

..did just that, and I agree, I heard some very focused, clear, real room, live recordings that sounded very good to me at the reference sites. .. So, those kind of recordings can be produced, then that's what we want on the CD's today, unless resolutions greater than 24 bit, etc., has something to do with what we want to hear.  Why are we not putting the best on all the CD's that are being made.  I bet it is a business thing, or has something to do with profit.


----------



## Hitec

I was using a Sennheiser HD700 to listen.


----------



## KeithEmo

There are definitely a few CDs around that sound really good, and a lot more that don't.... and I can't imagine it's because it costs significantly more to make them sound good. (The cost to press a disc is exactly the same regardless of what's on it, so the only difference would be the recording and mastering part of the process.)
  
 The scary thing is that I suspect a lot of it has to do with the simple fact that most people can't hear the difference - and many may actually prefer bad sound. (Let me be more specific there.... Highly compressed music sounds better on low-end car systems, and on cheap ear buds, and on other crappy systems. And, even beyond that, many people seem to like music with lots of processing, and flashy special effects, like excessive reverb, and a zillion tracks of phony voice doubling. And, as long as people like those things, and few of them are complaining about the loss of sound quality that accompanies them, then the music industry will continue to provide what sells to the majority of their customers... and they aren't going to expend any effort to do a better job unless they believe it's costing them sales.)
  
 For that reason I believe that the current high-res music fad has value. Regardless of whether high-res is actually necessary, or whether high-res files inherently sound better than a Red Book CD, at least the existence of the high-res music market should serve to convince music producers that at least some of their customers actually do care about sound quality, and ARE willing to pay more for music that sounds better. And this realization should help drive a movement towards better quality overall. (Of course, the downside is that it encourages them to split the market - and to continue to produce lousy CDs, and separate high-quality high-res versions, which they can sell at a higher price.)
  
  
 Quote:


hitec said:


> ..did just that, and I agree, I heard some very focused, clear, real room, live recordings that sounded very good to me at the reference sites. .. So, those kind of recordings can be produced, then that's what we want on the CD's today, unless resolutions greater than 24 bit, etc., has something to do with what we want to hear.  Why are we not putting the best on all the CD's that are being made.  I bet it is a business thing, or has something to do with profit.


----------



## watchnerd

keithemo said:


>





> The scary thing is that I suspect a lot of it has to do with the simple fact that most people can't hear the difference - and many may actually prefer bad sound. (Let me be more specific there.... Highly compressed music sounds better on low-end car systems, and on cheap ear buds, and on other crappy systems. And, even beyond that, many people seem to like music with lots of processing, and flashy special effects, like excessive reverb, and a zillion tracks of phony voice doubling


 
  
 The mere existence of auto-tune would seem to validate this theory.
  
 But, also, as you said, many people are listening in mobile / crappy systems.


----------



## limpidglitch

watchnerd said:


> Jeez, really?
> 
> If you thought those sounded okay, then you definitely shouldn't waste any money on high-resolution stuff.


 
  
 That's a bit full on, isn't it?
  
 Of course it sounds like an above average demo recording, but I wouldn't blame that on the mp3 encoding.
 (notice I wrote that as a reply to JWolf claiming the opposite?)
  
  


jwolf said:


> But given that they can impact the sound of a drum kit and someone has mentioned the drums don't sound quite right, I would say it's the MP3 causing it. Also, MP3 is not what's being discussed here. It's 16/44.1 vs. Hi-Res. Please take down your MP3 links and put up at least CD resolution links. Sounds alright? Nope. It's time you moved to an iPod and iTunes and give up CDs.


 
  
 I'm perfectly happy with iTunes and an iPod.


----------



## JWolf

limpidglitch said:


> That's a bit full on, isn't it?
> 
> Of course it sounds like an above average demo recording, but I wouldn't blame that on the mp3 encoding.
> (notice I wrote that as a reply to JWolf claiming the opposite?)
> ...


 
  
 No wonder you think iTunes and an iPod sound good. 128k MP3 is not good. Pleaselink us with CD quality FLAC and we can forget your grievous error in linking us crappy MP3.


----------



## limpidglitch

jwolf said:


> No wonder you think iTunes and an iPod sound good. 128k MP3 is not good. Pleaselink us with CD quality FLAC and we can forget your grievous error in linking us crappy MP3.


 
  
 I haven't linked to any mp3 file.


----------



## cjl

jwolf said:


> No wonder you think iTunes and an iPod sound good. 128k MP3 is not good. Pleaselink us with CD quality FLAC and we can forget your grievous error in linking us crappy MP3.


 
 128k is audibly flawed, but it's a lot better than many audiophiles would have you believe. It can be surprisingly difficult with some files to distinguish 128k from FLAC, and 320 is nearly always indistinguishable. MP3 encoders have come a long way in the past decade.


----------



## JWolf

cjl said:


> 128k is audibly flawed, but it's a lot better than many audiophiles would have you believe. It can be surprisingly difficult with some files to distinguish 128k from FLAC, and 320 is nearly always indistinguishable. MP3 encoders have come a long way in the past decade.


 
  
 The best lossy format is Ogg Vorbis. MP3 encoders have gotten better, but they still can have easily noticeable differences from CD.


----------



## RRod

jwolf said:


> The best lossy format is Ogg Vorbis. MP3 encoders have gotten better, but they still can have easily noticeable differences from CD.


 
  
 Are you still talking 128kbps? if so then I don't think anyone will disagree re mp3, though Opus might have a bit of an edge on Vorbis. At 320kbps, though, I think many people would find it hard to detect the differences between mp3 and the original CD content.


----------



## cjl

jwolf said:


> The best lossy format is Ogg Vorbis. MP3 encoders have gotten better, but they still can have easily noticeable differences from CD.


 
 You're right that MP3 is not the best lossy format, but there are no easily noticeable differences from CD at 320kbps. There are occasionally some extremely subtle (but audible) differences, but in the great majority of cases, 320 will be audibly indistinguishable from CD.


----------



## watchnerd

jwolf said:


> The best lossy format is Ogg Vorbis. MP3 encoders have gotten better, but they still can have easily noticeable differences from CD.


 
  
 Ogg is great.  And Ogg is dead.
  
 RIP, Ogg, you were too late to market.


----------



## sonitus mirus

watchnerd said:


> Ogg is great.  And Ogg is dead.
> 
> RIP, Ogg, you were too late to market.


 
  
 It's not dead yet.  You can't put Ogg in the cart, it's against regulations.
  
 Spotify has 20 million paying subscribers,


----------



## JWolf

watchnerd said:


> Ogg is great.  And Ogg is dead.
> 
> RIP, Ogg, you were too late to market.


 
  
 Ogg Vorbis sounds better than MP3 and AAC plus is produces smaller files.


----------



## watchnerd

jwolf said:


> Ogg Vorbis sounds better than MP3 and AAC plus is produces smaller files.


 
  
 Not disagreement from me on that.
  
 But it's market penetration (does streaming actually count?) is practically nil, except for a few die hard open source leaning fans.
  
 The biggest private torrent tracker won't even let you post Ogg files to share anymore.  When even the pirates kick you to the curb you know you're marginal.


----------



## old tech

gregorio said:


> In film, we commonly used pull-up or pull-down rates in the digital intermediate, post-prod phase, so that we could edit at video speed and then convert back to (35mm) film speed. That workflow is less common today, although there is still a common issue of the 0.1% difference between video speed and film speed. Generally the masterclocks we use are not extremely high precision, they are very decent crystal oscillators running at many tens of megahertz but there's absolutely no need for an atomic clock (which run well into the GHz range), that's more like one of those ridiculous audiophile excesses! There are different schemes employed, some of which are fairly complex or fairly simple PLLs or some combination but locking clock signals together with the accuracy needed for film audio doesn't benefit from atomic clocks.
> 
> G


 

 Hey, welcome back!
  
 Your thread on the 24 bit myth exploded is what got me interested in sound science in the first place.


----------



## Hitec

Ok, I can't post any of my CD examples of the MP3's, but I will say from a listener's perspective, you should pickup the following CD, and let us know what you think.  It is one of the ones that I used at the car competitions as a demo to showoff .  ..for competition there was only one official CD of tests references used from car to car - you couldn't bring your own.  As I said before, if I can get this type of produced recording, I don't see why I would need 24 bits or greater, etc. if I can't hear what I'm getting.  I can hear a difference between the mp3's above and the CD versions.
  
 "Sweetman" -   http://shop.mapleshadestore.com/bSweetman-with-his-South-Side-Groove-Kings_b-Austin-Backalley-Blue/productinfo/02752/


----------



## Hitec

hitec said:


> Ok, I can't post any of my CD examples of the MP3's, but I will say from a listener's perspective, you should pickup the following CD, and let us know what you think.  It is one of the ones that I used at the car competitions as a demo to showoff .  ..for competition there was only one official CD of tests references used from car to car - you couldn't bring your own.  As I said before, if I can get this type of produced recording, I don't see why I would need 24 bits or greater, etc. if I can't hear what I'm getting.  I can hear a difference between the mp3's above and the CD versions.
> 
> "Sweetman" -   http://shop.mapleshadestore.com/bSweetman-with-his-South-Side-Groove-Kings_b-Austin-Backalley-Blue/productinfo/02752/


 
 Oh, and someone said they were a drummer:  Here's the CD that was produced starring Michael Carvin as the drummer - http://shop.mapleshadestore.com/bMichael-Carvin_b-Drum-Concerto-At-Dawn/productinfo/03732/.  It is a pure drummer's solo CD.  "Drum Concerto at Dawn".


----------



## watchnerd

hitec said:


> Ok, I can't post any of my CD examples of the MP3's, but I will say from a listener's perspective, you should pickup the following CD, and let us know what you think.  It is one of the ones that I used at the car competitions as a demo to showoff .  ..for competition there was only one official CD of tests references used from car to car - you couldn't bring your own.  As I said before, if I can get this type of produced recording, I don't see why I would need 24 bits or greater, etc. if I can't hear what I'm getting.  I can hear a difference between the mp3's above and the CD versions.
> 
> "Sweetman" -   http://shop.mapleshadestore.com/bSweetman-with-his-South-Side-Groove-Kings_b-Austin-Backalley-Blue/productinfo/02752/


 
  
 Are you just a Mapleshade sales bot?


----------



## Hitec

watchnerd said:


> Are you just a Mapleshade sales bot?


 
 Ha, I knew that was coming  I'm sure others in the background are shaking their heads and saying, "yea is he pro Maple or something".  No, but that's what it looks like doesn't it   Unfortunately, I'm wet behind the ears, and do not have any knowledge about any other recording companies, and so, Maple is the only one that I can speak of in order to help me understand their production techniques and how they would relate to 24 bit improvements, and higher frequencies improvements.  I do think they say that they sample recordings a 100 times a standard CD to be used as their master.  I'm am learning plenty from the comments here.  The one big thing is I am beginning to work on completing a personal theater in my basement, or music room, and so I want to set up the environment and the music properly.


----------



## watchnerd

hitec said:


> Ha, I knew that was coming  I'm sure others in the background are shaking their heads and saying, "yea is he pro Maple or something".  No, but that's what it looks like doesn't it   Unfortunately, I'm wet behind the ears, and do not have any knowledge about any other recording companies, and so, Maple is the only one that I can speak of in order to help me understand their production techniques and how they would relate to 24 bit improvements, and higher frequencies improvements.  I do think they say that they sample recordings a 100 times a standard CD to be used as their master.  I'm am learning plenty from the comments here.  The one big thing is I am beginning to work on completing a personal theater in my basement, or music room, and so I want to set up the environment and the music properly.


 
  
 Well, first off, you're barking up the wrong trees.
  
 1. Good recording has practically nothing to do with "24 bit improvements".  All modern ADCs are 24bit.  Crappy stuff is made in 24bit and amazing stuff is made in 24bit.
  
 2. Mapleshade has no widespread professional reputation to speak of.  Go look at Reference Recordings, Linn, Chesky, Nordic 2L for labels that consistently make well-produced recordings.
  
 3. Reference recordings are good for evaluating speakers, but setting up a room is more about understanding acoustics, especially in the bass area. Investing in a test microphone and learning how to measure is going to do more good for you when setting up a room than any single recording.


----------



## nick_charles

watchnerd said:


> 2. Mapleshade has no widespread professional reputation to speak of.


 
  
  
 They also make/promote some (shall we say) questionable audio enhancement items


----------



## watchnerd

nick_charles said:


> They also make/promote some (shall we say) questionable audio enhancement items


 
  
 Oh, you mean like $495 wood cutting boards on feet?
  

  
 I think I could make one of those for <$50.
  
 And then you have dumb statements like this:
  
 ""I hate mixing boards," says Sprey, who records everything live to two-track on analog tape. "With a mixing board, if there was a problem with the way [the soloist] sounded, they would adjust it with the mixing board. Here we adjust it with the microphone itself in front of the musician."
  
 Dude, seriously?
  
 If you adjust it at the microphone, you affect *everything*, not just the amplitude.  The timbre, the soundfield captured, how much of the other nearby instruments, the sibilance, etc.  In other words, you're undoing all the work you put into mic placement to get all those other attributes exactly right.
  
 The whole point of the mixing board is to adjust amplitude in isolation from everything else.  The fact that he doesn't get this (and nobody else does it like he does) speaks volumes.


----------



## Hitec

watchnerd said:


> Well, first off, you're barking up the wrong trees.
> 
> 1. Good recording has practically nothing to do with "24 bit improvements".  All modern ADCs are 24bit.  Crappy stuff is made in 24bit and amazing stuff is made in 24bit.
> 
> ...


 
 1. ..not sure I understand.. I thought the subject was about, "if 24 bit or greater would be bad for music".  My focus was on if there was an ability to hear a difference with the higher resolutions, etc.   If this forum can get to the bottom of a question and produce an answer for other users, then I think this is the correct tree for barking..  ..this tree could then advise on the next road that is needed to get to the next tree.
  
 2. Mapleshade has an outstanding reputation.  They started in 1986, 2 years before Chesky.  They are just not as popular as Chesky, since it is more of a personal passion for the music and artists - not profit.  I don't think that reputation makes music.  Passion makes music, and it does not have to be by popular demand - it's personal. Also, well produced may be in the eyes of the beholder.  This is a hobby at the heart for Mapleshade  ..don't know if you have met the folks out at Mapleshade, but they are about the nicest, passion-for-music people you can meet that are into making and engineering music first as a hobby - not ripping people off for a profit.  Also, the one with the unpopular reputation may be the one with the missing link that is needed for further progression.  They could be commenting on this forum at this moment - but they are dismissed from the thoughts of followers because they do not have a popular reputation that followers are seeking... btw, Mapleshade, Telarc, and Chesky were used on the same International Auto Sound Challenge Association (IASCA) competition CD media for judging and tuning cars.  The reason being, they all had something to offer.  Chesky's stuff was outstanding at measures all around, while Mapleshade selections were the best for their patented vivid imaging, and tonal quality.  This is Pierre Sprey's professional reputation:  https://www.google.com/?gfe_rd=ssl&ei=yzypVs-wJIGd-AXAoYFw#q=Pierre+Sprey.  Music is the man's passion - not profit.
  
 3. I already own two RTA's from Audio Control - SA3052, and another  mobile sized one.  I use them to help my ears with localizing and aiming speakers, imaging, stage width, depth, tonality, linearity, frequency smoothing and balancing, phase, crossover points, etc. My practice started with competing, and judging in IASCA SQ competitions.  I need to diver further into figuring out how to setup a room.  In the car, it was full of Dynamat, padding on the dash, etc.  Very fun sound challenge that is just about impossible to accomplish..


----------



## watchnerd

hitec said:


> 1. ..not sure I understand.. I thought the subject was about, "if 24 bit or greater would be bad for music".  My focus was on if there was an ability to hear a difference with the higher resolutions, etc.   If this forum can get to the bottom of a question and produce an answer for other users, then I think this is the correct tree for barking..  ..this tree could then advise on the next road that is needed to get to the next tree.


 
  
 This has already been answered countless times:
  
 1. 24bit is useful for recording / production when compared to 16bit (some argue 20bit is enough)
  
 2. 24bit gains nothing except fatter files for playback
  
 I guess you didn't like that answer?
  
  
  


hitec said:


> 2. Mapleshade has an outstanding reputation.  They started in 1986, 2 years before Chesky.  They are just not as popular as Chesky, since it is more of a personal passion for the music and artists - not profit.  I don't think that reputation makes music.  Passion makes music, and it does not have to be by popular demand - it's personal. Also, well produced may be in the eyes of the beholder.  This is a hobby at the heart for Mapleshade  ..don't know if you have met the folks out at Mapleshade, but they are about the nicest, passion-for-music people you can meet that are into making and engineering music first as a hobby - not ripping people off for a profit.  Also, the one with the unpopular reputation may be the one with the missing link that is needed for further progression.  They could be commenting on this forum at this moment - but they are dismissed from the thoughts of followers because they do not have a popular reputation that followers are seeking... btw, Mapleshade, Telarc, and Chesky were used on the same International Auto Sound Challenge Association (IASCA) competition CD media for judging and tuning cars.  The reason being, they all had something to offer.  Chesky's stuff was outstanding at measures all around, while Mapleshade selections were the best for their patented vivid imaging, and tonal quality.  This is Pierre Sprey's professional reputation:  https://www.google.com/?gfe_rd=ssl&ei=yzypVs-wJIGd-AXAoYFw#q=Pierre+Sprey.  Music is the man's passion - not profit.


 
  
 All of the links on the first page of Google point to a military / defense expert....
	

	
	
		
		

		
			




  
 As for the other guy, how many Grammy's has he won (unlike Chesky, Nordic 2L)?  How many AES papers has he published?  Oh, right 0 in both cases.
  
 I don't care how nice the folks think he is. That doesn't matter.


> 3. I already own two RTA's from Audio Control - SA3052, and another  mobile sized one.  I use them to help my ears with localizing and aiming speakers, imaging, stage width, depth, tonality, linearity, frequency smoothing and balancing, phase, crossover points, etc. My practice started with competing, and judging in IASCA SQ competitions.  I need to diver further into figuring out how to setup a room.  In the car, it was full of Dynamat, padding on the dash, etc.  Very fun sound challenge that is just about impossible to accomplish..


 
  
 Sounds like you're in a good spot to learn acoustics.


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## Hitec

watchnerd said:


> Oh, you mean like $495 wood cutting boards on feet?
> 
> 
> 
> ...


 
 The mixing board may consist of adjusting more than amplitude - correct, how about frequency, stereo position, dynamics and effects, etc.   In this case, they would be screwing up all kinds of things from the original.


----------



## Hitec

watchnerd said:


> This has already been answered countless times:
> 
> 1. 24bit is useful for recording / production when compared to 16bit (some argue 20bit is enough)
> 
> ...


 
 Uhh, either you are joking, or actually around 18 years old.  I am not downing Chesky or any other music and engineering professional.  I think that ALL have something to offer no matter how POPULAR one becomes.  You, on the other hand, are being disrespectful about a fellow music professional that has not said anything negative about you - don't even know who you are.  Also, you've probably never meet Chesky or Pierre.  Your comments are taking things well beyond the lead title.  This forum was created to post and comment about the following:  "Why 24 bit audio and anything over 48k is not only worthless, but bad for music.."  Some people disagree with what you said about 24bits.  Based on what I have researched, from REPUTABLE sources, I agree with you.  ..but how reputable are YOU to answer this technical question.  The paper that was posted for this title was very good to me.  Do you have a paper that you can post to backup your statements on 24 bit?  I know I don't.  Please answer that instead of making  comments that are demeaning to others.


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## watchnerd

hitec said:


> Uhh, either you are joking, or actually around 18 years old.  I am not downing Chesky or any other music and engineering professional.  I think that ALL have something to offer no matter how POPULAR one becomes.  You, on the other hand, are being disrespectful about a fellow music professional that has not said anything negative about you - don't even know who you are.  Also, you've probably never meet Chesky or Pierre.  Your comments are taking things well beyond the lead title.  This forum was created to post and comment about the following:  "Why 24 bit audio and anything over 48k is not only worthless, but bad for music.."  Some people disagree with what you said about 24bits.  Based on what I have researched, from REPUTABLE sources, I agree with you.  ..but how reputable are YOU to answer this technical question.  The paper that was posted for this title was very good to me.  Do you have a paper that you can post to backup your statements on 24 bit?  I know I don't.  Please answer that instead of making  comments that are demeaning to others.




You're the one who keeps posting Mapleshade stuff into a thread that has nothing to do with the topic.

Given that this is the Sound Science forum, you shouldn't expect a warm reception for a guy who sells high priced wooden products with pretty astounding claims for what amount to thick cutting boards.

As for 24bit, there are so many articles on this, both at Hydrogen Audio and at the AES that you only need to search.

But let's flip this around:

What special benefits at playback time do you think 24bit gives you?


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## watchnerd

hitec said:


> The mixing board may consist of adjusting more than amplitude - correct, how about frequency, stereo position, dynamics and effects, etc.   In this case, they would be screwing up all kinds of things from the original.




You're confusing recording with post production...


----------



## dprimary

hitec said:


> 3. I already own two RTA's from Audio Control - SA3052, and another  mobile sized one.  I use them to help my ears with localizing and aiming speakers, imaging, stage width, depth, tonality, linearity, frequency smoothing and balancing, phase, crossover points, etc. My practice started with competing, and judging in IASCA SQ competitions.  I need to diver further into figuring out how to setup a room.  In the car, it was full of Dynamat, padding on the dash, etc.  Very fun sound challenge that is just about impossible to accomplish..


 
 To set up a room you will need more then a RTA, there is plenty of fairly low cost tools. A RTA cannot measure room acoustics.
  
 If you don't already have it you should pick up a copy of the Master Handbook of Acoustics by Everest I think it in the 6th edition now. It is fairly easy to read for a book on acoustics and hits on most of the principals of room acoustics. I still reread it from time to time, I think have I four editions of it. 
  
 Home Recording Studio: Build It Like the Pros by Rod Gervais is a good book to help understand construction of acoustical spaces.
  
Acoustic Absorbers and Diffusers: Theory, Design and Application by Trevor J. Cox is an excellent text for people wanting to have a deeper understanding of acoustics. 
  
If you really want to get deep into it look for texts from, Olson, Kuttruff,  and Beranek.
  
I don't think texts from Newell are that helpful. Take anything from a self claimed "Expert" with a grain of salt. There is as much or more snake oil in acoustics as there is in audio. Most is from people trying to sell you something (often questionable), or they really don't understand it but rather make things up instead of learning the physics.
 

  
 The challenge in room acoustics is getting the right balance of materials to control the room acoustics, and every room is different. You can build two rooms with exactly the same dimensions and have them sound completely different, how the room is constructed can drastically change the acoustics even though to the eye they look the same.


----------



## spruce music

watchnerd said:
			
		

> All of the links on the first page of Google point to a military / defense expert....
> 
> 
> 
> ...


 

 There is no other guy.  The guy who worked in military defence (seems he had a major say in F16 design and A10 Warthog) is the same guy at Mapleshade.  He does make nice recordings or the few I have heard are very nice.  Some of his other ideas I don't believe, but the recordings are good.  Don't need no stinkin' grammy's to tell me it is good.


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## watchnerd

spruce music said:


> There is no other guy.  The guy who worked in military defence (seems he had a major say in F16 design and A10 Warthog) is the same guy at Mapleshade.  He does make nice recordings or the few I have heard are very nice.  Some of his other ideas I don't believe, but the recordings are good.  Don't need no stinkin' grammy's to tell me it is good.


 
  
 Subjectively, it's obviously a matter of taste.  
  
 But technically his recordings are, well...rudimentary and compromised.
  
 As for being good demo material, it's really only testing imaging and soundstaging. Everything else is easy-peasy stuff for any competent speaker due to limited dynamic and spectral range.


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## spruce music

watchnerd said:


> Subjectively, it's obviously a matter of taste.
> 
> But technically his recordings are, well...rudimentary and compromised.
> 
> As for being good demo material, it's really only testing imaging and soundstaging. Everything else is easy-peasy stuff for any competent speaker due to limited dynamic and spectral range.


 

 Yes well this and worse can be said in criticism of about 99% of all recordings.  The technique of recording is similar to a jecklin disk and two omnis.  There are far worse things to do with a recording.  As you say subjectively it is a matter of taste.  The recording quality is generally higher than average even with some of the far out ideas he has about various aspects involved.  I don't find using two mics and moving people around to be somehow misguided or improper.  It is a different method.  One he can sometimes get to work well.  I guess I somehow missed where good recordings had to test the limits of a system to be good.  If they sound nice, portray the music well and are enjoyed by those who like the genre I think it is a successful recording.
  
 You might be interested to know, despite some of his loonier ideas, he thinks more than CD/Redbook for distribution is pretty much of no benefit.  Which is another place where it fits in this thread.


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## watchnerd

I don't know if this paper has already been posted to the thread or not, apologies if it has but I don't recall seeing any discussion of it::
  
Evaluation of Sound Quality of High Resolution Audio
  
  
 Challenge to conclude much with only 27 subjects and stats mostly below the significance line.


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## sonitus mirus

watchnerd said:


> I don't know if this paper has already been posted to the thread or not, apologies if it has but I don't recall seeing any discussion of it::
> 
> Evaluation of Sound Quality of High Resolution Audio
> 
> ...


 
  
 The results seem to be all over the place.  It was easier for the subjects to identify samples between 24 and 16 bit than it was to determine between HRA and a 128kbps MP3.  It's like rock, paper, scissors.
  
 Edit: From section 3.1, I'm still trying to calculate how they got a 60.3% correct rate when the subjects answered the same or not the same in 112 evaluations.  For 95% confidence that the subjects were not guessing, 65 of 112 correct answers would be required.  60.3% is about 67.5 correct answers.  It looks like the test just barely is over 95% confidence, and then the test is concluded.  If an additional session was performed, it may have made the results statistically insignificant.  We don't know if the test was always supposed to be 112 evaluations, or they chose to stop after this amount because the results coincided with what the supporters of this test were hoping to find.  Also, not sure if the same/not the same evaluations included all formats, so that the more easily identifiable lower quality mp3 file was helping to make the overall results statistically significant.  If only CD and HRA were used, would the results be more than guessing?


----------



## watchnerd

sonitus mirus said:


> The results seem to be all over the place.  It was easier for the subjects to identify samples between 24 and 16 bit than it was to determine between HRA and a 128kbps MP3.  It's like rock, paper, scissors.
> 
> Edit: From section 3.1, I'm still trying to calculate how they got a 60.3% correct rate when the subjects answered the same or not the same in 112 evaluations.  For 95% confidence that the subjects were not guessing, 65 of 112 correct answers would be required.  60.3% is about 67.5 correct answers.  It looks like the test just barely is over 95% confidence, and then the test is concluded.  If an additional session was performed, it may have made the results statistically insignificant.  We don't know if the test was always supposed to be 112 evaluations, or they chose to stop after this amount because the results coincided with what the supporters of this test were hoping to find.  Also, not sure if the same/not the same evaluations included all formats, so that the more easily identifiable lower quality mp3 file was helping to make the overall results statistically significant.  If only CD and HRA were used, would the results be more than guessing?


 
  
 Even weirder is that they were more easily able to discriminate between high and low bit-rate MP3 than between HRA/CD and low-bit rate MP3.
  
 Maybe the 30 second rest period between samples was too long and audio memory faded?


----------



## nick_charles

watchnerd said:


> Even weirder is that they were more easily able to discriminate between high and low bit-rate MP3 than between HRA/CD and low-bit rate MP3.
> 
> Maybe the 30 second rest period between samples was too long and audio memory faded?


 
  
  
 Part of my day job is peer-reviewing academic papers. The paper as presented is short on a lot of details that would help it to be evaluated. For starters the training session might be interpreted as the experimenters pointing out specific artifacts to be listened for which would be loading the dice when it comes to MP3. Why did they have to apply a second low pass and resample the CD to 48K - it should not matter but whey did they have to do it. Was the order of presentation randomized - I hope so. I agree that 30 seconds seems too long a gap. They take Oohashi seriously which is worrying and use a super tweeter , we know that the super tweeter in Oohashi's experiment was considered by some as a problem due to the IMD it created.
  
 The numbers literally do not add up. If there are 27 subjects then each subject that prefers A over B represents 3.7 % - the possible additions of 3.7% are
  
 3.7, 7.4, 11.1, 14.8, 18.5, 22.2, 25.9, 29.6, 33.3, 37, 40.7, 44.4, 48.1, 51.8, 55.5, 59.2, 62.9, 66.6, 70.3, 74, 77.7, 81.4, 85.1, 88.8, 92.5, 96.2, 99.9 (exc rounding error) 
  
 57.4
 61.1
  
 are impossible to get unless you have half a subject !  So a subject half preferred A over B which makes no sense unless they preferred A in 1 test and B in another which suggests random guessing not preference
  
  


> For example, in the pair of CD with MP3(L), 61.1 % of the subjects preferred CD to MP3(L).


 
  
 As above just not possible unless you count half correct tests!
  
 What I think they did was collect the 54 trials for each pair and tot up how often A was preferred over B regardless of whether the subject stated the same preference both times, this approach seems flawed


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## spruce music

This is a two alternative forced choice test for preference.  So the statistics are different than your typical ABX.  75% would represent 95% confidence the results weren't random.  One of the differences in 2AF preference testing is 75% is the amount for 95% confidence regardless of the number of samples. The only one getting close was the high to low bitrate MP3 comparison.  The results for the others fail to indicate a preference was perceived at a level of confidence at 95%. 
  
 30 seconds between samples is long enough to exceed everyone's echoic memory and was a poor choice for testing accuracy.  Averaging results from disparate stimuli is also an improper way to analyze the results.


----------



## watchnerd

So is the consensus of this forum that the paper in question is too poorly conducted to conclude anything?


----------



## spruce music

watchnerd said:


> So is the consensus of this forum that the paper in question is too poorly conducted to conclude anything?


 

 The results do not support the idea that the various formats were audible.   However, there were a few issues with how the test was conducted that could be better.  As it stands the results don't provide evidence of an audible difference.  It comes very close for high vs low rate MP3 and other testing shows that is perceptible.  So this reinforces the idea this could have been better.  Mainly imo by not having the 30 seconds between selections.  I would expect if that were changed 128 k vs 320 k mp3 should result in very high scores for a difference perceived.


----------



## Maconi

watchnerd said:


> sonitus mirus said:
> 
> 
> > The results seem to be all over the place.  It was easier for the subjects to identify samples between 24 and 16 bit than it was to determine between HRA and a 128kbps MP3.  It's like rock, paper, scissors.
> ...


 
  
 I'm still new to the science of audio but don't we only have a few seconds of real retention and after that it's just whatever our brain felt like storing in our memory?
  
 https://en.wikipedia.org/wiki/Echoic_memory
  
 http://www.harbeth.co.uk/usergroup/showthread.php?878-Human-hearing-hearing-loss-and-audio-memory
  
 That could possibly be why they had a hard time I suppose.


----------



## watchnerd

maconi said:


> I'm still new to the science of audio but don't we only have a few seconds of real retention and after that it's just whatever our brain felt like storing in our memory?
> 
> https://en.wikipedia.org/wiki/Echoic_memory
> 
> ...


 
  
 Yes, this phenomenon is what I was referring to.


----------



## charleski

sonitus mirus said:


> The results seem to be all over the place.  It was easier for the subjects to identify samples between 24 and 16 bit than it was to determine between HRA and a 128kbps MP3.  It's like rock, paper, scissors.
> 
> Edit: From section 3.1, I'm still trying to calculate how they got a 60.3% correct rate when the subjects answered the same or not the same in 112 evaluations.  For 95% confidence that the subjects were not guessing, 65 of 112 correct answers would be required.  60.3% is about 67.5 correct answers.  It looks like the test just barely is over 95% confidence, and then the test is concluded.  If an additional session was performed, it may have made the results statistically insignificant.  We don't know if the test was always supposed to be 112 evaluations, or they chose to stop after this amount because the results coincided with what the supporters of this test were hoping to find.  Also, not sure if the same/not the same evaluations included all formats, so that the more easily identifiable lower quality mp3 file was helping to make the overall results statistically significant.  If only CD and HRA were used, would the results be more than guessing?


 

 Wow, that paper would barely pass muster as an undergrad project. The experimental design is atrocious. It looks like they're trying to test whether people can spot the difference between 'high-res' audio and a standard 16/48 stream, so why include all the mp3 stuff? The 60.3% 'correct rate' appears out of thin air and should be ignored, they just seem to have made this up.
  
 57.4% expressed a preference for HRA over CD, which is the question they were seemingly attempting to ask. The methodology for the sessions is unclear, possibly because of linguistic difficulties, but let's assume that each of the 28 subjects heard the HRA--CD pairing 4 times (twice in each session for two sessions), giving 112 total presentations. The problem with this is that 57.4% of 112 is 64.288 ... why isn't that an integer? They provide absolutely no data on whether subjects were consistent in their preference (did those who preferred HRA the first time also prefer it on the other 3 occasions?), which is a major failing and slightly suspicious. But OK, they end up with a 1x2 category table: one category (listened to the two pieces) x two outcomes (preferred HRA or CD). With the null hypothesis that each each outcome should be 50% we get a chi-sq value [ Σ((E-O)^2/E) ] of 2.453 with one degree of freedom. This gives a p-value of 0.117, not remotely significant.
  
 Let's start with a different null hypothesis, that automotive engineers are able to design a proper psychological test. From the data in this paper I think _that_ null hypothesis can be firmly rejected.


----------



## limpidglitch

And what's up with the reference list? It's really thin and all in Japanese, giving us very little chance to second-guess their interpretation of the literature.
  
 This: Subjective Evaluation of High Resolution Audio under In-Car Listening Environments, appears to be a follow-up of sorts. 
 Haven't read it, but the claims made in the abstract seems just as dubious. I smell industry funding.


----------



## castleofargh

it looks to me that you're all in league with degrasse tyson, trying to fool honest citizens into thinking that CD is enough and that the earth isn't flat.


----------



## nick_charles

charleski said:


> Wow, that paper would barely pass muster as an undergrad project. The experimental design is atrocious. It looks like they're trying to test whether people can spot the difference between 'high-res' audio and a standard 16/48 stream, so why include all the mp3 stuff? The 60.3% 'correct rate' appears out of thin air and should be ignored, they just seem to have made this up.
> 
> 57.4% expressed a preference for HRA over CD, which is the question they were seemingly attempting to ask. The methodology for the sessions is unclear, possibly because of linguistic difficulties, but let's assume that each of the 28 subjects heard the HRA--CD pairing 4 times (twice in each session for two sessions), giving 112 total presentations. The problem with this is that 57.4% of 112 is 64.288 ... why isn't that an integer? They provide absolutely no data on whether subjects were consistent in their preference (did those who preferred HRA the first time also prefer it on the other 3 occasions?), which is a major failing and slightly suspicious. But OK, they end up with a 1x2 category table: one category (listened to the two pieces) x two outcomes (preferred HRA or CD). With the null hypothesis that each each outcome should be 50% we get a chi-sq value [ Σ((E-O)^2/E) ] of 2.453 with one degree of freedom. This gives a p-value of 0.117, not remotely significant.
> 
> Let's start with a different null hypothesis, that automotive engineers are able to design a proper psychological test. From the data in this paper I think _that_ null hypothesis can be firmly rejected.


 
  
  
 The first test had only 27 subjects who did each A/B test twice for a total of 54 trials for each pair.
  
 In the 2nd experiment instead of resampling the CD source to 48K they resampled the HRA source down to 48K with a 20K low pass and 24 or 16 bit versions - so not quite the same thing. The 28 subjects included 24 from the first test. So 4 new subjects, so we cannot directly compare expt 1 and 2..The 57.4% preference refers to the first expt 27 subjects. But the numbers for the 2nd are impossible unless some results were excluded...or there were in fact 224 trials i.e 4 x two pairs (8 * 28) which works numerically.


----------



## watchnerd

castleofargh said:


> it looks to me that you're all in league with degrasse tyson, trying to fool honest citizens into thinking that CD is enough and that the earth isn't flat.


 
  
 CDs? The physical things that deteriorate over time and can be broken?
  
 No, CDs are horrible and should stay dead.
  
 Now I've you're talking about standard-definition lossless audio, I'll agree.
  
 That, plus tubes.


----------



## sonitus mirus

nick_charles said:


> The first test had only 27 subjects who did each A/B test twice for a total of 54 trials for each pair.
> 
> In the 2nd experiment instead of resampling the CD source to 48K they resampled the HRA source down to 48K with a 20K low pass and 24 or 16 bit versions - so not quite the same thing. The 28 subjects included 24 from the first test. So 4 new subjects, so we cannot directly compare expt 1 and 2..The 57.4% preference refers to the first expt 27 subjects. But the numbers for the 2nd are impossible unless some results were excluded...or there were in fact 224 trials i.e 4 x two pairs (8 * 28) which works numerically.




Ah yes, 135 of 224 nets the 60.3% in the second experiment.


----------



## nick_charles

sonitus mirus said:


> Ah yes, 135 of 224 nets the 60.3% in the second experiment.


 
  
  
 However, this was the overall rate for all tests which they presented as 60.3% of subjects successfully detecting the difference. So in 135 trials out of 228 a correct answer was given. But we do not know how many times the pairs were balanced - i.e were the sequences correctly randomized was it always 2 A and 2B and did subjects get feedback (hopefully not) after the trials. Were they told that there would be a fixed no of A and B - in fact any details wild be welcome.
  
 But an overall 60.3% rate does not mean that 60.3% of subjects were able to correctly discriminate - that would be 16.884 subjects which again does not add up.


----------



## charleski

nick_charles said:


> The first test had only 27 subjects who did each A/B test twice for a total of 54 trials for each pair.
> 
> In the 2nd experiment instead of resampling the CD source to 48K they resampled the HRA source down to 48K with a 20K low pass and 24 or 16 bit versions - so not quite the same thing. The 28 subjects included 24 from the first test. So 4 new subjects, so we cannot directly compare expt 1 and 2..The 57.4% preference refers to the first expt 27 subjects. But the numbers for the 2nd are impossible unless some results were excluded...or there were in fact 224 trials i.e 4 x two pairs (8 * 28) which works numerically.


 

 Ugh, yes, they had two cohorts. That'll teach me for trying to decipher these things late on a Friday night. 27 subjects in the first group does give an integer result for the 57.4%, but if they only did 54 presentations then it's even further from significance.
  
 They specifically talk about '112 evaluations' for the 2nd experiment, though as you mention that doesn't provide an integral result for the 60.3%. There might even have been 448 presentations of each pair: 'Each session consisted of four paired-stimuli patterns', 'Each subject joined four sessions'. Anyway, even at 112 presentations the 60.3% figure reaches significance. The problem then is that they failed to show an ability to discriminate between 24/192 and 16/48 signals (despite their attempt to paint the 57.4% number as a positive result in the conclusion), but are claiming their subjects _could _discriminate between 24/48 and 16/48 signals, implying that the presence of high-frequency material _decreases _our ability to discern bit-depth. This, obviously, doesn't make any sense so they tried to ignore it.
  
 Apart from mentioning 'patterns with permutation' they don't give any details of how the patterns were arranged within each session. Was the ordering independently randomised, was it interleaved (i.e. A-B, B-A, A-B, B-A), or was there some other arrangement? They failed to provide any sort of control, and the confusing results from the mp3 tests (in which more people preferred 128kbps than 320kbps mp3 when each was compared to HRA) suggest that we're simply looking at different levels of extraneous bias.


----------



## JWolf

With 16-bit sampling of an analogue signal, the input maximum amplitude range is divided into 65536 (2 to the 16th power) levels, and the value of the input signal at each sample point is approximated by the nearest level. With 24-bit sampling, you're dividing the input range into 16777216 (2 to the 24th power) steps, which clearly gives a much more accurate representation of the sampled signal.
  
 This process is known as quantisation, and the differences between the 'real' analogue input values and their quantised approximations ('quantisation error') effectively adds some noise, which is known as 'quantisation noise'. If we have smaller steps, we have, on average, less quantisation error per sample, and hence less quantisation noise.


----------



## varzyl

I personally just tried to discern 24bit\192khz FLAC and Opus 128 VBR (16bit\48khz) and miserably failed (on Foobar+ABX, using Norah Jones "What am I to you"). I managed to pass the Golden Ear from Philips, but I remember the Silver level 128kbps mp3 as one of the most difficult test.
  
 I guess I'll keep ripping my 16\44.1 CDs to MP3 -v0 for the peace of mind and compabity though.


----------



## L8MDL

So the "quantisation noise" is why my 192/24 files sound "warmer", right? Just like tape noise, seems to me...


----------



## Pootis

No real difference between 16/44.1 and 24/96. Lossy vs. lossless is very doable in a blind test however.


----------



## cjl

jwolf said:


> With 16-bit sampling of an analogue signal, the input maximum amplitude range is divided into 65536 (2 to the 16th power) levels, and the value of the input signal at each sample point is approximated by the nearest level. With 24-bit sampling, you're dividing the input range into 16777216 (2 to the 24th power) steps, which clearly gives a much more accurate representation of the sampled signal.
> 
> This process is known as quantisation, and the differences between the 'real' analogue input values and their quantised approximations ('quantisation error') effectively adds some noise, which is known as 'quantisation noise'. If we have smaller steps, we have, on average, less quantisation error per sample, and hence less quantisation noise.


 

 While technically true, this quantization noise is at below -90dBFS, so it isn't really relevant to normal listening.


----------



## sonitus mirus

jwolf said:


> With 16-bit sampling of an analogue signal, the input maximum amplitude range is divided into 65536 (2 to the 16th power) levels, and the value of the input signal at each sample point is approximated by the nearest level. With 24-bit sampling, you're dividing the input range into 16777216 (2 to the 24th power) steps, which clearly gives a much more accurate representation of the sampled signal.
> 
> This process is known as quantisation, and the differences between the 'real' analogue input values and their quantised approximations ('quantisation error') effectively adds some noise, which is known as 'quantisation noise'. If we have smaller steps, we have, on average, less quantisation error per sample, and hence less quantisation noise.


 
  
 According to the Nyquist-Shannon Sampling Theorem, I don't see how you can get more accurate than a perfectly reconstructed analog signal.  Perfect is perfect.  Sure, the dynamic range is greater with 24-bit, but the accuracy of the signal is not any more precise.


----------



## JWolf

But you have to admit though if there is error with figuring out what goes between one sample and the next in 16-bits will be lessened or removed with 240bits.


----------



## spruce music

sonitus mirus said:


> According to the Nyquist-Shannon Sampling Theorem, I don't see how you can get more accurate than a perfectly reconstructed analog signal.  Perfect is perfect.  Sure, the dynamic range is greater with 24-bit, but the accuracy of the signal is not any more precise.


 

 Now this is not quite right.  There is less than total perfect reconstruction with limited sample depth.  Quantization noise is a result of that inaccuracy.  Quantization noise is lower for 24 bit than with 16 bit.  Now real world thermal noise in circuits will keep you from true 24 bit noise levels, but 20 bit or so is possible in practice.  Dither lets you record signals into the noise somewhat.  So 16 bit can effectively reach near 120 db or 20 bit equivalent.  So effectively with dither there is little actual difference in accuracy though there is a potential difference.  There are a small number of electronics that manage 21 or 22 bit worth.  There is likely no audible difference in these if done well.


----------



## castleofargh

jwolf said:


> But you have to admit though if there is error with figuring out what goes between one sample and the next in 16-bits will be lessened or removed with 240bits.


 

  yes that one value per sample can be more accurately defined in 24bit, but that value being a guitar+ another+ a voice+.... +quantization noise. that's all inside that 1 value at instant t, thanks to the "magic" of sound waves. when changing the bit depth to 24bit, the one and only thing being improved is the quantization noise and all possible sounds recorded below -96db(good luck with recoding that clearly). if it was real close to the recorded instruments, then it would matter to lower it, but when it's 16bit down already, it starts to look like seeking improvement just because we can. not because it has a purpose(I'm talking about us listeners, the recording people do have a use for that improved dynamic range).


----------



## watchnerd

castleofargh said:


> yes that one value per sample can be more accurately defined in 24bit, but that value being a guitar+ another+ a voice+.... +quantization noise. that's all inside that 1 value at instant t, thanks to the "magic" of sound waves. when changing the bit depth to 24bit, the one and only thing being improved is the quantization noise and all possible sounds recorded below -96db(good luck with recoding that clearly). if it was real close to the recorded instruments, then it would matter to lower it, but when it's 16bit down already, i*t starts to look like seeking improvement just because we can. not because it has a purpose(I'm talking about us listeners, the recording people do have a use for that improved dynamic range).*


 
  
 Seriously, this thread has become pointless because:
  
 1. 24bit for recording is the norm, for multiple reasons (that's how ADCs come these days, extra overhead for editing).
  
 2. 24bit for playback is stupid overkill waste of space because the extra dynamic range isn't needed and people can't ABX the same mastering  at 24bit vs 16bit
  
 Everything else is magical thinking.


----------



## gregorio

jwolf said:


> But you have to admit though if there is error with figuring out what goes between one sample and the next in 16-bits will be lessened or removed with 240bits.


 
  
 It's not so much an interpolation error as it is an error in the sample values themselves. You're always going to get quantisation error, regardless of the bit depth you employ, it's simply a matter of whether the resultant quantisation noise has any impact (which it doesn't with 16bit or higher) or how you deal with the resultant noise with lower bit depths.
  


spruce music said:


> Now real world thermal noise in circuits will keep you from true 24 bit noise levels, but 20 bit or so is possible in practice.


 
  
 I would personally phrase this a bit differently. I would say that 20bit or so is possible in theory but in practice we're probably talking of no more than around 12bit. The limiting factor "in practice" is not the bit depth of the recording format but what we are recording and what we are recording with.
  
 Quote:


l8mdl said:


> So the "quantisation noise" is why my 192/24 files sound "warmer", right? Just like tape noise, seems to me...


 
  
 No, the quantisation noise is inaudible (unlike tape noise). If you are hearing a difference there are only 4 possibilities for what's causing the "warmer" sound at 192/24:
  
 1. You are imagining a difference where none actually exists.
 2. The 44.1/16 version you are using for comparison has been deliberately butchered in some way to sound different (less warm) than your 192/24 version.
 3. The 192/24 files are causing IMD on your system, which for some reason you are interpreting as a "warmer" sound.
 4. Your DAC has a fault (or design flaw) when playing back files other than 192/24.
  
 #1 or #2 are the most likely explanations.
  


watchnerd said:


> 1. 24bit for recording is the norm, for multiple reasons (that's how ADCs come these days, extra overhead for editing).


 
  
 I certainly agree with your other points and "that's how ADCs come these days" but it's not because of extra overhead for editing. No extra overhead is required for editing. It's purely to give extra overhead (headroom) when recording, no other reason.
  
 G


----------



## charleski

gregorio said:


> I certainly agree with your other points and "that's how ADCs come these days" but it's not because of extra overhead for editing. No extra overhead is required for editing. It's purely to give extra overhead (headroom) when recording, no other reason.
> G


 
 Quantisation noise is additive. Let's say you need to mix together two full-scale 16bit signals, A and B. Each has -6.02*16-1.76 = -98.08dB of quantisation noise. You're using a 16bit digital mixer and you know you need to decrease the gain on these signals, otherwise the output will be clipped. That's easy enough - you apply -3dB of gain to both of them. But they're still 16-bit signals, and still have the same level of quantisation noise, which can never go below -98.08dB. So you add the two together and you have a full-scale signal containing A+B, but this now has -95.08dB of quantisation noise.
  
 Quantisation noise is essentially rounding error. Any operation you perform on the signal will introduce additional rounding error. So when you're doing a lot of this you want to operate in a higher bit depth to minimise the amount of extra noise you're adding. Mixing and editing at 24 bits gives you more than enough overhead to be sure that your final signal will have minimal noise when output at 16bits.


----------



## limpidglitch

gregorio said:


> I would personally phrase this a bit differently. I would say that 20bit or so is possible in theory but in practice we're probably talking of no more than around 12bit. The limiting factor "in practice" is not the bit depth of the recording format but what we are recording and what we are recording with.
> 
> G


 
  
 You're assuming we are only recording, and not synthesizing.

 On that note, quantization error isn't strictly speaking inevitable either, but that's really just an academic point.


----------



## gregorio

charleski said:


> Let's say you need to mix together two full-scale 16bit signals, A and B. Each has -6.02*16-1.76 = -98.08dB of quantisation noise. You're using a 16bit digital mixer ...


 
  
 I'll stop the quote there because the rest of your post is based on using a 16bit digital mixer, when in practice, there are no professional 16bit mixers. I'm not even sure if there ever were any 16bit mixers! There are not even any 24bit mixers I'm aware of! I personally use a 56bit mixer which has been a commercial standard for more than a decade. That and 64bit (float) are the most common professional mixing bit depth standards today. Yes, there are still rounding errors but they are at around -300dB!
  


limpidglitch said:


> You're assuming we are only recording, and not synthesizing.


 
  
 Yes but there are extremely few recordings of purely synthesized music. Most synths today do not purely synthesize sound, they are effectively sophisticated sample playback units which apply some synthesis techniques, IE., they are based on recordings.
  
 G


----------



## charleski

gregorio said:


> I'll stop the quote there because the rest of your post is based on using a 16bit digital mixer, when in practice, there are no professional 16bit mixers. I'm not even sure if there ever were any 16bit mixers! There are not even any 24bit mixers I'm aware of! I personally use a 56bit mixer which has been a commercial standard for more than a decade. That and 64bit (float) are the most common professional mixing bit depth standards today. Yes, there are still rounding errors but they are at around -300dB!


 

 Early digital mixers like the Soundstream deck used to master _Tusk_ were indeed 16bit. By the mid-80s 20bit digital mixers were available, which offered a huge improvement.
 Most modern editors use double- or quad-precision bit depths for intensive internal calculations within a module but then place a 24bit signal back on the bus.
  
 I note that you're agreeing with me and watchnerd that you need more than 16 bits for editing ...


----------



## RRod

charleski said:


> Early digital mixers like the Soundstream deck used to master _Tusk_ were indeed 16bit. By the mid-80s 20bit digital mixers were available, which offered a huge improvement.
> Most modern editors use double- or quad-precision bit depths for intensive internal calculations within a module but then place a 24bit signal back on the bus.
> 
> I note that you're agreeing with me and watchnerd that you need more than 16 bits for editing ...


 
  
 Seems like there are many early digital recordings that still hold up well today, though. I guess they didn't have a whole lot of editing done by today's standards?


----------



## gregorio

charleski said:


> Most modern editors use double- or quad-precision bit depths for intensive internal calculations within a module but then place a 24bit signal back on the bus.


 
  
 The older ProTools TDM systems had 24bit paths for parts of the signal chain but those 24 bits were maintained, even at very low gain settings, by using a much larger bit depth bus. With the newer ProTools HDX systems the signal remains at 64bit float throughout, until the mix is printed/output. Most other DAWs maintain signal paths of 32 or 64bit float.
  
 Quote:


charleski said:


> I note that you're agreeing with me and watchnerd that you need more than 16 bits for editing ...


 
  
 No, for editing you don't need any more bits than your recording file format, say 16bit. Editing does not incur any rounding errors, etc. For mixing, yes, 56bit or 64bit float is useful to (effectively) eliminate rounding errors. As a file format, 24bit is only useful for providing extra headroom during recording.
  
 G


----------



## KeithEmo

There's a factor here that everyone seems to neglect - and that is the way in which humans seem to store ALL memories (and it explains how we can sometimes pick out seemingly minor differences - even with very short term "acoustic memory"). In most situations, we simply don't actually remember what we see or hear (or taste) at all. Instead, when we're talking about familiar things, what we tend to have in our heads is more like a "standard version" and a "modifier"... although this may vary by the subject. So, for example, when you remember the spicy mustard you had last night, you're probably really remembering a generalized memory of "spicy mustard" - and your "memory" from last night is more like the "+1" that people use on bulletin boards to signify "again". And, if you had ketchup last night that was especially sweet, rather than remember in detail what it tasted like, you tend to remember a general flavor for "ketchup" plus a modifier "sweet".
  
 The reason this is relevant for audio is that it also means that you tend to remember DIFFERENCES more than you remember individual details. So, for example, you don't have a detailed "acoustic memory" of the last time you heard your favorite song, but you do have a relatively detailed composite "sound image" of the last fifty times you heard it - sort of averaged together. And, if you hear it again today, on a different pair of speakers, in your mind you'll be comparing today's version to that composite version, and noticing and remembering the differences. And so, just like you remember the stone that you tripped over, even though you don't remember every stone in your sidewalk, you tend to notice how each version you listen to is different from the "standard version" you have in your memory. (And, so, even though you may not remember every note of the song, you'll still notice the one that's missing, or the one that you never noticed before - because it seems to be "something new" compared to the "standard memory", and both will stand out. This allows us, even though we don't remember every song in detail, to still notice how a given speaker, or a given DAC, may be different. It also means that most of us are a lot more sensitive to slight differences in the way something we are very familiar with sounds than we would be to slight differences between different versions of things we aren't familiar with.
  
 For another example, just think of trying to pick someone out of a police lineup who you only saw for a few seconds, as compared to recognizing an old friend in those same few seconds, or noticing that your old friend had a new hairstyle... Since you already have a "composite memory" of your old friend, with a lot of detail, even trivial differences may often be very obvious. (And this also leads to the phenomenon where you may "fill in" details with information from that composite memory image - and confusing them with new information... so, for example, you may 'remember" that your friend was carrying a briefcase last night, even though they actually weren't, "because they always carry a briefcase".)
  
 In the audio world, this mechanism is why your current speakers come to seem "normal" after a few days or weeks, and you become very aware of how other things are "different" from that "normal", and why you tend to notice differences in speakers that are "bright" or "dull" or "smooth" compared to your current "normal" reference model.     
  
 Because of this sort of thing, it's not at all a simple matter of "you can't compare how two things sound because you can't remember what things sound like from minute to minute". (And it's also why the MP3 version of a song may sound just fine; until you hear a better version, which contains information that wasn't there before.)
  
 The upshot of all this is that, rather than "remember what it sounded like last time", it's more like you become trained to know "what you think it should sound like", and tend to compare the current version you're listening to to that "standard version".
  
  
 Quote:


maconi said:


> I'm still new to the science of audio but don't we only have a few seconds of real retention and after that it's just whatever our brain felt like storing in our memory?
> 
> https://en.wikipedia.org/wiki/Echoic_memory
> 
> ...


----------



## watchnerd

keithemo said:


>


 
  
 Yes, cognition and memory have an notable and testable effect on auditory perception.
  
 But what does any of that have to do with 24bit?
  
 Unless you're saying 24bit files play to the the placebo effect, in which case, sure, but so do a lot of other things.


----------



## cjl

sonitus mirus said:


> According to the Nyquist-Shannon Sampling Theorem, I don't see how you can get more accurate than a perfectly reconstructed analog signal.  Perfect is perfect.  Sure, the dynamic range is greater with 24-bit, but the accuracy of the signal is not any more precise.


 
 You have to be a little careful here - Shannon-Nyquist says that you can perfectly reconstruct a sampled, bandwidth-limited signal (so long as the bandwidth limitation is below 1/2 of the sample frequency), but as part of the theorem, it assumes perfect sampling (no error in the sampled points). Quantization error due to limited bit depth does indeed prevent this perfect reconstruction, since your samples are actually wrong. Now, the magnitude of this error is very small (< -90dBFS for 16 bit), but it does exist, and can be a problem during recording and mixing. The solution is to record, process, and mix at 24+ bit, and then drop it down to 16 bit for playback (since -90dB is inaudible anyways).


----------



## charleski

rrod said:


> Seems like there are many early digital recordings that still hold up well today, though. I guess they didn't have a whole lot of editing done by today's standards?


 
 Some of the mixing on _Tusk_ was done on analog 24-track tape, with the Soundstream used only for the final mixdown and mastering. At least that's what Bruce Rothaar seems to be reported saying here. If you zoom in on the tails of the tracks they're pretty noisy, but at over 80dB down you can't really hear it, and as the engineer said, "we wanted to get a more raw sound".
  


gregorio said:


> No, for editing you don't need any more bits than your recording file format, say 16bit. Editing does not incur any rounding errors, etc. For mixing, yes, 56bit or 64bit float is useful to (effectively) eliminate rounding errors. As a file format, 24bit is only useful for providing extra headroom during recording.
> 
> G


 
 Editing is any manipulation of the source material and includes gain changes, mixing, filtering, etc. No-one does simple cut-and-paste outside the most purist classical recordings, and I'm not interested in irrelevant semantics.
  
 ProTools HDX uses a 32-bit floating-point process depth, which is a 23-bit mantissa with 8 bits for the exponent. It's basically a more flexible form of 24bit that allows the audio level to be scaled over a far wider range, but 32bit float and 24bit fixed have roughly the same resolution (which is why it makes no sense to apply dither on a 32->24bit conversion). The mixer plugin doubles that resolution for summing, but the result heads out at 32bits, the same is true for most plugins these days. Keeping the signal at 64bit throughout would be tremendously wasteful.


----------



## gregorio

charleski said:


> Editing is any manipulation of the source material and includes gain changes, mixing, filtering, etc. No-one does simple cut-and-paste outside the most purist classical recordings ...


 
   
 Everyone does simple copy and paste in every music genre and, classical music is probably the genre where it occurs the least! Editing isn't just cut and paste though, it's also copying, moving, removing and replacing. Gain changes are mixing, not editing, as is filtering, EQ and any other processing.
  
 Quote:


charleski said:


> I'm not interested in irrelevant semantics.


 
  
 Are you interested in avoiding confusion or making incorrect statements due to the incorrect use of terminology? Are you interested in not making yourself appear foolish when using pro audio terms because you don't know what they mean?
  
 Quote:


charleski said:


> ProTools HDX uses a 32-bit floating-point process depth, ... The mixer plugin doubles that resolution for summing, but the result heads out at 32bits, the same is true for most plugins these days. Keeping the signal at 64bit throughout would be tremendously wasteful.


 
  
 No, ProTools HDX is 64bit float, not just individual plugins but the entire internal signal path. It is not wasteful because the HDX cards have been specifically designed for a 64bit signal path and nothing else. And BTW, the result "heads out" at 24bit.
  
 G


----------



## L8MDL

gregorio said:


> Originally Posted by *L8MDL*
> 
> 
> 
> ...


 
  
 I was only half-serious when I posted that statement. I am curious, however, about whether or not quantisation noise is "inaudible". You seem to indicate it is, unless the player is not working correctly. Others say 192/24 can cause audible distortion and therefore 96/24 is better.


----------



## charleski

gregorio said:


> No, ProTools HDX is 64bit float, not just individual plugins but the entire internal signal path. It is not wasteful because the HDX cards have been specifically designed for a 64bit signal path and nothing else. And BTW, the result "heads out" at 24bit.
> 
> G


 
 If some salesman spun you a tale of ProTools having a full 64bit throughput then you were getting sold a bill of goods.The depth used for the processing bus (which is 32bit float in HDX, not 24) is the same whether you run ProTools natively or with the DSP. The plugin depth also has the same double precision in both versions.
  
 Do you have anything to say which isn't completely irrelevant? Please don't pretend to have some special command of 'pro audio terms' 
	

	
	
		
		

		
			





 .


----------



## RRod

l8mdl said:


> I was only half-serious when I posted that statement. I am curious, however, about whether or not quantisation noise is "inaudible". You seem to indicate it is, unless the player is not working correctly. Others say 192/24 can cause audible distortion and therefore 96/24 is better.


 
  
 At 16 bits the quantization error should have peaks near -90dBFS (so necessarily lower RMS). In my listening room there is no way I hear such a signal without setting the volume at abnormal listening levels.


----------



## limpidglitch

gregorio said:


> Yes but there are extremely few recordings of purely synthesized music. Most synths today do not purely synthesize sound, they are effectively sophisticated sample playback units which apply some synthesis techniques, IE., they are based on recordings.
> 
> G


 
  
 I have no problem finding examples of music that's generated purely from mathematical algorithms. The demo scene is full of it, just to give an obvious example, but it's also a much wider phenomenon.
  
 But even if you think that's a bit niche, popular music is full of examples that has been created wholly within a DAW. And as common home computers has become more powerful (since the early 00's, or so), sample based synthesis has been steadily giving way for realtime digital PWM-, AM-, FM- etc. synthesis. Hell, these days even your phone is powerful enough to do some pretty sophisticated synthesizing. I know Mouse on Mars, for example, has been using an iPhone as a separate musical instrument in some of their recordings.


----------



## dprimary

charleski said:


> Early digital mixers like the Soundstream deck used to master _Tusk_ were indeed 16bit. By the mid-80s 20bit digital mixers were available, which offered a huge improvement.
> Most modern editors use double- or quad-precision bit depths for intensive internal calculations within a module but then place a 24bit signal back on the bus.
> 
> I note that you're agreeing with me and watchnerd that you need more than 16 bits for editing ...


 

 Soundstream was not a mixer but a recorder, Soundstream could do editing possibly even crossfades of edits, but that only happened back at Soundstream. I can't think of any digital mixers till the early 90's except the old yamaha keyboard mixers. Neve had the DSP 1 in the early 80's which was 24 bit bus and 32 bit mixing but only a few where built and I don't know if any were used in music production.


----------



## gregorio

Quote:


charleski said:


> Please don't pretend to have some special command of 'pro audio terms'
> 
> 
> 
> ...


  
 The difference between "editing" and "mixing" doesn't require a "special command" of pro audio terms, a novice level of understanding is sufficient. And, I don't claim or pretend to have any better a command of pro audio terms than any other audio pro who's been in the business for over two decades.
  
 Quote:


charleski said:


> Do you have anything to say which isn't completely irrelevant?


 
  
 I'm responding directly to your statements, are you saying that your statements are irrelevant?
  
 Quote:


charleski said:


> If some salesman spun you a tale of ProTools having a full 64bit throughput then you were getting sold a bill of goods.The depth used for the processing bus (which is 32bit float in HDX, not 24) is the same whether you run ProTools natively or with the DSP.


 
   
 It's got nothing to do with anything a salesman has spun, try instantiating any 32bit AAX DSP plugin in PT HDX 11 or higher and see how far you get, it's 64bit only! The data paths within the HDX cards are also 64bit. That's not to say that some calculations within 64bit plugins may not occur at 32bit but both bit depths and sample rates are virtually impossible to track these days without being privy to information only generally known to the individual software developers. Audio data which "heads out" of HDX can only do so through a HDX ADC/DAC, which are 24bit not 32bit float.
  
 Regardless of whatever happens to bit depth "under the hood" in the mixing environment, my original post stands: The only benefit of recording 24bit is the additional headroom and no greater bit depth than the recording format is required for editing.
  
 Quote:


limpidglitch said:


> I have no problem finding examples of music that's generated purely from mathematical algorithms. ... But even if you think that's a bit niche, popular music is full of examples that has been created wholly within a DAW.


 
  
 I didn't say there are no examples of music which is generated purely from mathematical algorithms, just that it's extremely rare. But this has nothing to do with whether or not it's been created wholly within a DAW. As I mentioned, most soft synths (used wholly within a DAW) are not generating sound purely from mathematical algorithms, most are based on sample playback plus some synthesis techniques, envelope manipulation being a standard example. Those underlying samples are generally samples of recordings and are therefore subject to the same limitations as other recordings.
  
 G


----------



## charleski

dprimary said:


> Soundstream was not a mixer but a recorder, Soundstream could do editing possibly even crossfades of edits, but that only happened back at Soundstream. I can't think of any digital mixers till the early 90's except the old yamaha keyboard mixers. Neve had the DSP 1 in the early 80's which was 24 bit bus and 32 bit mixing but only a few where built and I don't know if any were used in music production.


 
  
 Yes, but from what I can find out _Tusk _was indeed mastered at the Soundstream base in Utah. See the link I provided earlier (it's quite long, but the section about _Tusk _is around a third of the way down). This produced addtional problems as 'Fleetwood Mac flew to Salt Lake City for the editing session. There was pressure to get the editing completed by four a.m. so they could be in their Lear Jet, over the Grand Canyon, on drugs, at dawn.’ (
	

	
	
		
		

		
		
	


	




). I strongly supect that EQ and dynamics, etc were done in analogue before the inputs were digitised, but the 16bit/50kHz digital signals could be mixed on a PDP11/60. Unfortunately there's no record I can find on whether they bounced tracks down to digital and then folded them back in, but the tape recorder that was part of the system could only handle 4 channels. It does seem that this was a very different situation to something like Springsteen's _The River_ from 1980, which was completely mixed in analogue on an API console and the Sony PCM1600 that featured on the album artwork only carried the final master. It's worth noting that Fleetwood Mac dumped digital completely for their next album, saying it 'was a complete waste of time' and that they preferred the 'softening' provided by tape. This was probably a reference to the limiting from tape's built-in high-frequency saturation, but it's tempting to speculate on how much accumulated quantisation noise played a role here.
  
 You're right, the Neve DSP1 was a very interesting animal, though only 8 were actually put into use. For those who are interested there's a long article on it here. This was only actually delivered in 1985, though a year earlier Neve had built a much smaller transfer deck for Tape One called the Attila which had EQ and dynamics and seems to have used the same technology. The DSP1 was very advanced for its era and used 24bits for the EQ and dynamics modules and 32bits for summing. All AD/DA was 16bit at first, but some later deliveries were upgraded to 18bit.
  
 For those who didn't have £300,000 for a DSP1 the first actually attainable digital console was probably Yamaha's DMP7 from 1987, a 16/44.1 device that processed effects with DSP chips running at 23bits. As noted in that last link, you could chain up to 6 of these 8-track mixers together, giving a final SNR of -60dB (ick).
  
  


gregorio said:


> ...


 
 Please take this somewhere else. This isn't a ProTools forum, but I'm sure you can find one where people will walk you through it.


----------



## spruce music

I don't really have a dog in this fight about Pro Tools bit depth.  Here is the basic info claimed from Pro Tools themselves.
  
 http://www.avid.com/US/products/Pro-Tools-HDX/features
  
 32 bit float for processing and 64 bit float for mixer depth.  Either is plenty it seems to me that processing errors are a non issue.


----------



## watchnerd

I"m not even sure why we're debating the bit-level of DAWs, floating point, etc.  It's completely off topic.
  
 Post #1 of this thread is a quote from Pono, which is all about distribution of playback formats.


----------



## charleski

watchnerd said:


> I"m not even sure why we're debating the bit-level of DAWs, floating point, etc.  It's completely off topic.
> 
> Post #1 of this thread is a quote from Pono, which is all about distribution of playback formats.


 
 Well, if the question is 'do we need 24bit for playback at home' then it's really boring, because the answer is 'No'.


----------



## RRod

charleski said:


> Well, if the question is 'do we need 24bit for playback at home' then it's really boring, because the answer is 'No'.


 
  
 Heck I wouldn't even want a system capable of it. Toddler gets a hold of the remote one time and we all get flash-banged.


----------



## gregorio

charleski said:


> Yes, but from what I can find out _Tusk _was indeed mastered at the Soundstream base in Utah.  ... "Fleetwood Mac flew to Salt Lake City for the editing session."


 
  
 Huh? Why are you persisting in what is now a deliberate confusion of terms? Fleetwood Mac went to Soundstream for the editing, as your quote states, not for the mixing or mastering! For the last time; mixing, mastering and editing are different tasks with different names to reflect that fact. BTW, Tusk was mastered by Capitol Records, presumably at their mastering studio in Hollywood.
  


charleski said:


> This isn't a ProTools forum, but I'm sure you can find one where people will walk you through it.


 
  
 I'm sure I can. Indeed, being a certified ProTools instructor myself, I'm one of those people who walk others through it!
  
 Quote:


watchnerd said:


> I"m not even sure why we're debating the bit-level of DAWs, floating point, etc.  It's completely off topic.


 
  
 charleski brought it up when he wandered off talking about theoretical 16bit mixers which don't exist and actually, you started it by saying you needed more bits for editing. Ergo, it's all your fault! 
	

	
	
		
		

		
		
	


	



  
 G


----------



## watchnerd

charleski said:


> Well, if the question is 'do we need 24bit for playback at home' then it's really boring, because the answer is 'No'.


 
  
 Exactly.
  
 /closethread


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## JWolf

Yes you do need 24-bit playback at home because anything that doesn't handle 24-bit is not going to sound good enough.


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## watchnerd

jwolf said:


> Yes you do need 24-bit playback at home because anything that doesn't handle 24-bit is not going to sound good enough.


 
  
 Yay, a believer in the religion!
  
 Care to elaborate on why you think this is true?


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## cjl

To be fair, for computer audio, 24 bit can be nice if you're using digital volume control, especially at the low range of the digital volume control. Of course, this just requires that the computer/DAC be 24 bit capable - the audio files themselves can still be 16 bit without any issues.


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## Pootis

jwolf said:


> Yes you do need 24-bit playback at home because anything that doesn't handle 24-bit is not going to sound good enough.


 
  
 96 dB is certainly good enough.


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## spruce music

24 bit is also potentially useful if you do room/speaker correction digitally.  You can still feed a 16 bit source into the 24 bit hardware of course without issue.


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## spruce music

gregorio said:


> charleski brought it up when he wandered off talking about theoretical 16bit mixers which don't exist and actually, you started it by saying you needed more bits for editing. Ergo, it's all your fault!
> 
> 
> 
> ...


 

 One account of Tusk was they took 24 track analog tapes to Soundstream who then mixed/edited that into the final stereo version. They wanted to avoid mixing those 24 tracks in the analog realm. 
  
_The album was mixed digitally on Soundstream's new digital master recorder under the supervision of Soundstream engineer Rich Feldman. The decision to utilize the digital mixdown system came after considering a number of alternative mixing systems and a demonstration of the equipment. "when you A-B it," explains Mick, "you can definitely tell the difference. I know we were originally afraid it would introduce some artificial quality in the final sound but it~didn't." Dashut adds: "we didn't want to add any more surface tape noise to the noise that was already on the 24-track tape when we mixed it down, and with the digital mixdown, that doesn't happen. Digital just reproduces what you put into it, and nothing more."

 Editing of the mixed tapes had to be completed in Salt Lake City, Soundstream's headquarters, because the equipment needed to accomplish the intercuts and fades is located there. Fleetwood explains: "Any intercuts you might have just done with a razor blade on a conventional machine, you have to do on a special machine. That's the only major inconvenience, since we're always chopping and changing and splicing tapes."_
  
 http://www.fleetwoodmac-uk.com/articles/presskits/pk_tusk.html


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## JWolf

pootis said:


> 96 dB is certainly good enough.


 
  
 It's not about what's good enough. It's about DACs that only handle 16-bit are going to be old and not all that good They will not sound as good as a more updated DAC that can handle 24-bit. Cowon J3 only handles 16-bit. Fiio X3II handles 24-bit. X3II sound much better than the J3.


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## ubs28

24 bit or over is not worthless. Good luck mixing and mastering with 16 bit source files. The final mix will not be good.


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## Pootis

jwolf said:


> It's not about what's good enough. It's about DACs that only handle 16-bit are going to be old and not all that good They will not sound as good as a more updated DAC that can handle 24-bit. Cowon J3 only handles 16-bit. Fiio X3II handles 24-bit. X3II sound much better than the J3.


 
  
 No, not necessarily. A 16 bit DAC with 16 bit performance is good enough to be transparent.


----------



## castleofargh

jwolf said:


> Yes you do need 24-bit playback at home because anything that doesn't handle 24-bit is not going to sound good enough.


 

 144db resolutions. it sounds better!!!!! 
 but don't forget your 144db resolution DAC, 144db resolution amp, your usual 144db resolution headphone. and of course try to listen at 144db SPL for better results.
	

	
	
		
		

		
			





 maybe more because you also would want to have all your music above the ambient noise of your room for better "micro details". 
	

	
	
		
		

		
		
	


	



  
 you keep insisting on something irrational just because you refuse to look at the big picture. you're stuck with you half understood "24bit has better resolution than 16bit"(which is true on a way, but also not as relevant as you think, because we're dealing with sound waves), but you disregard what those extra points to define the data are really used for(to express values quieter than16bit). you forget to look at what your sound system can do, even with a great sound system the final output into your ears will have a hard time being as good as 96db of actual resolution. then you disregard that a human can at best notice 0.1db variation in ideal conditions(in practice 0.1db difference still is below what people need to pass a bind test). or all the evidence suggesting that we have a real hard time noticing more than 60db of dynamic at once(maybe because it's really hard to imagine a listening situation where you'll get much more than 70db of clean dynamic in a room?).  then you disregard the ambient noise that was in the recording room and bound to mask at least some of the quietest sounds from instruments. the resolution of the microphones, the recording that probably wasn't done at 150db SPL(duh). the bits lost while adjusting the different tracks in the mixing process so that at least some of the sounds don't have the maximum dynamic anymore.
  
 yet you keep coming with nothing but confidence and tell us all about how it obviously sounds better to have the song in 24bit... you don't see like half a dozen reasons why you must be wrong even if you have amazing hearing and amazing records played on amazing sound system?
  if your sound system was such that 16/44 really didn't sound like 24/96 or higher(and I'm still waiting for you to demonstrate that you are more than a dreamer on that matter), that would be specific to your sound system, and still most likely related to sample rate and the filters, not bit depth.
  
 it's ok to be wrong, but please try not to drag others with you. use highres because you want to, or because you believe marketing, or simply because it has more data(relevant data, not sure, but more data still). but please pretty please, if you're going to lie, lie to yourself. stop trying to drag other people with you.


----------



## castleofargh

jwolf said:


> pootis said:
> 
> 
> > 96 dB is certainly good enough.
> ...


 

 oh yeah another great example of cherry picked evidence. I eat raw potato, then I eat raw tomato. the tomato tasted better, therefore red tastes better than whatever color was the potato. point proven. not.
 this is sound science, the evidence makes the conclusion. not the other way around.
 between 2 completely different DAPs made many years apart, by 2 different brands, the one reason you pick for sound difference is how one doesn't play 24bit? seriously? why not go with: one is Chinese one is Korean so Chinese sounds better? this is ludicrous.
  
 oh and AFAIK the J3 had a 24bit wolfson DAC chip...
  
 please stop.


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## gregorio

ubs28 said:


> Good luck mixing and mastering with 16 bit source files. The final mix will not be good.


 
  
 Why not? Why would mixing and mastering with 24bit files be any better than with 16bit files?
  
 G


----------



## JWolf

castleofargh said:


> oh yeah another great example of cherry picked evidence. I eat raw potato, then I eat raw tomato. the tomato tasted better, therefore red tastes better than whatever color was the potato. point proven. not.
> this is sound science, the evidence makes the conclusion. not the other way around.
> between 2 completely different DAPs made many years apart, by 2 different brands, the one reason you pick for sound difference is how one doesn't play 24bit? seriously? why not go with: one is Chinese one is Korean so Chinese sounds better? this is ludicrous.
> 
> ...


 
  
 But the J3 does not allow 24-bit. Just 16-bit/48 is the best you get. The J3's output is rolled off and it sounds it.
  
 What I am trying to say is that DACs that only do 16-bit won't sound as good as a good DAC that does 24-bit. It has nothing to do with 24 v 16. It has to do with the DAC chip being better and the electronics being better. So with a CD, the DAC that does 24-bit should sound better than the DAC that only does 16-bit.


----------



## Pootis

jwolf said:


> But the J3 does not allow 24-bit. Just 16-bit/48 is the best you get. The J3's output is rolled off and it sounds it.
> 
> What I am trying to say is that DACs that only do 16-bit won't sound as good as a good DAC that does 24-bit. It has nothing to do with 24 v 16. It has to do with the DAC chip being better and the electronics being better. So with a CD, the DAC that does 24-bit should sound better than the DAC that only does 16-bit.


 
  
 No, not necessarily.


----------



## dprimary

ubs28 said:


> 24 bit or over is not worthless. Good luck mixing and mastering with 16 bit source files. The final mix will not be good.


 

 We mixed and mastered 16bit files for decades. Most of it sounds better then what is currently being released.


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## watchnerd

jwolf said:


> But the J3 does not allow 24-bit. Just 16-bit/48 is the best you get. The J3's output is rolled off and it sounds it.
> 
> What I am trying to say is that DACs that only do 16-bit won't sound as good as a good DAC that does 24-bit. It has nothing to do with 24 v 16. It has to do with the DAC chip being better and the electronics being better. So with a CD, the DAC that does 24-bit should sound better than the DAC that only does 16-bit.


 
  
 Whether that's true or not, that's a separate issue from the files.
  
 All my DACs are 24bit because they use relatively recent chips. That doesn't mean I need to feed them higher bitrate files.


----------



## JWolf

watchnerd said:


> Whether that's true or not, that's a separate issue from the files.
> 
> All my DACs are 24bit because they use relatively recent chips. That doesn't mean I need to feed them higher bitrate files.


 
  
 I never said you had to feed a DAC capable of 24-bit 24-bit files. I said a DAC capable of 24-bit will most likely sound better than a DAC only capable of 16-bit.


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## JWolf

dprimary said:


> We mixed and mastered 16bit files for decades. Most of it sounds better then what is currently being released.


 
  
 That's probably because of better recordings and/or better mastering.


----------



## limpidglitch

jwolf said:


> I never said you had to feed a DAC capable of 24-bit 24-bit files. I said a DAC capable of 24-bit will most likely sound better than a DAC only capable of 16-bit.


 
  
 While that might be true, what is the causal relationship?

 Any given 16-bit DAC has a high likelihood of either being in a portable device, be really really old, or be a part of some esoteric audiophile design.


----------



## JWolf

limpidglitch said:


> While that might be true, what is the causal relationship?
> 
> Any given 16-bit DAC has a high likelihood of either being in a portable device, be really really old, or be a part of some esoteric audiophile design.


 
  
 No way would a 16-bit only DAC be classed as audiophile. Any DAC that only does 16-bit would be classed as obsolete.


----------



## limpidglitch

jwolf said:


> No way would a 16-bit only DAC be classed as audiophile. Any DAC that only does 16-bit would be classed as obsolete.


 
  
 A bunch of NOS DACs do. And a lot of legacy units are still highly regarded.


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## icebear

Just as a reminder:
 There is always the ingnore button for peace of mind


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## ubs28

dprimary said:


> We mixed and mastered 16bit files for decades. Most of it sounds better then what is currently being released.




If you don't use alot of compression and limiting, then you can get away with it based on my experience (everything ITB except for a few hardware synthesizers)

Maybe this problem also doesn't happen when mixing with analog gear?


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## sterling1

I've gotta tell ya, the 16 bit D/A and A/D converters in my Sony PCM-7010 DAT Recorders, which I purchased back in 1994, are still as musical as anything I've since used in the studio for recording. In fact, back in 1982 when CD's appeared Sony had a slogan for the concept, "perfect sound, forever perfect". They told the truth. Despite, bit and bites out there today, it seems 16/44.1 has not been surpassed for musically. And, although there were certainly a few glitches apparent  in digital to analog conversion at it's introduction, by the end of the 1980's, these obstacles to perfect sound were resolved.


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## watchnerd

jwolf said:


> I never said you had to feed a DAC capable of 24-bit 24-bit files. I said a DAC capable of 24-bit will most likely sound better than a DAC only capable of 16-bit.


 
  
 Ah, sorry for misinterpreting....but the thread is about formats (when it's on topic...), so a diversion into chips was unexpected.


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## charleski

dprimary said:


> We mixed and mastered 16bit files for decades. Most of it sounds better then what is currently being released.


 
  
 As I mentioned above, digital mixing desks from the mid-80s on would use 16bit converters on the analogue in/outputs, but upconvert this to a higher bitdepth for internal efffects and summing modules (23bits for the DMP7, 24/32bits for the DSP1). You would only really get bitten by noise accumulation if you bounced partial mixes down to a 16bit track and then dropped that back in for further mixing, as shown by the way the SNR plummeted when you chained multiple DMP7s together. It's probably only the really early efforts at digital mixing that ran into these problems. Modern DAWs are doing their bit for the bitdepth wars by boasting much larger depths to their accumulators, but only _really _need these when running recursive effects like convolution or IIR filters.


----------



## gregorio

charleski said:


> As I mentioned above, digital mixing desks from the mid-80s on would use 16bit converters on the analogue in/outputs, but upconvert this to a higher bitdepth for internal efffects and summing modules (23bits for the DMP7, 24/32bits for the DSP1). You would only really get bitten by noise accumulation if you bounced partial mixes down to a 16bit track and then dropped that back in for further mixing, as shown by the way the SNR plummeted when you chained multiple DMP7s together. It's probably only the really early efforts at digital mixing that ran into these problems.


 
  
 If we're going off topic down digital mixing memory lane, then we have to consider that digital mixing didn't really start becoming mainstream studio practice until the early '90s, with products like Yamaha's DMR8, DMC1000 and Neve's Capricorn leading the way. Up until then there wasn't really any purely digital mixing. Commonly digital was used as the recording medium or recording + editing medium but mixed on analogue desks. An obvious exception was in the classical music world but even then it wasn't really mixed digitally in the usual sense, it was mainly mixed acoustically (EG. By the conductor and with mic placement), with little or no processing.
  


charleski said:


> Modern DAWs are doing their bit for the bitdepth wars by boasting much larger depths to their accumulators, but only _really _need these when running recursive effects like convolution or IIR filters.


 
  
 That's not entirely true. Very high bit-depths are useful in the film sound world, where there are commonly hundreds and sometimes up to around a thousand or so channels of complexly routed audio to contend with. It's certainly useful having such a huge window to play with and not having to be overly meticulous all the time about gain staging, introducing noise or loosing resolution. Not to mention the hundreds of plugins, including maybe a dozen or so reverbs.
  
 G


----------



## frodeni

This topic has not evolved much, despite the ton of posts.
  
 It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.
  
 I use an Oppo HA-1, and even for 16/44.1, the buffer must be set for max. Increasing to 32bit, decreases the SQ, for a 16/44.1 source. I have nothing to prove, as I know my hearing, thank you. I also know the drawbacks of A-B testing. Its flaws.
  
 By theory, buffer increase should not increase SQ. But it does.
  
 By theory, increasing bit dept to 32bit should not degrade the sound, as it is a pure bit shift operation. But it does.
  
 The second makes sense, if the firs one is true, but it is bloody hard to understand why. It really does not matter what I understand, as the 16/44.1 clearly sound the best, with the buffer at max.
  
 A controlled test, well, to my knowledge there is no such thing. It really does not even matter: Music is a subjective experience anyway. If I am able to hear harmonies and articulation with one setup, and not with another, well, that is what I do. No matter how I hard I try. People being stupid, denying that, well, that is on them. I know what I hear. My hearing is honed and well trained, and I am far more critical than you guys will ever be, if you get stuck where you are at.
  
 You guys need to move on. You actually ruin this forum, if you attack like you guys do. Time to grow.


----------



## watchnerd

frodeni said:


> This topic has not evolved much, despite the ton of posts.
> 
> It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.
> 
> ...


 
  
 How do you know it makes the sound better?
  
 If it's sighted listening, that's suspect.


----------



## bfreedma

frodeni said:


> This topic has not evolved much, despite the ton of posts.
> 
> It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.
> 
> ...





Sure. All of us "stupid people" who know that sighted evaluations have long been proven to be unreliable and are still waiting for you "smart people" to be able to support your claims via blind testing or measurements indicating an actual audible difference.

Product marketing or actual scientific evidence, it's so hard to know which is more accurate.....


----------



## watchnerd

frodeni said:


> I use an Oppo HA-1, and even for 16/44.1, the buffer must be set for max. Increasing to 32bit, decreases the SQ, for a 16/44.1 source. I have nothing to prove, as I know my hearing, thank you. I also know the drawbacks of A-B testing. Its flaws.
> 
> By theory, buffer increase should not increase SQ. But it does.
> 
> By theory, increasing bit dept to 32bit should not degrade the sound, as it is a pure bit shift operation. But it does.


 
  
 In any case, whatever is happening with the settings in your Oppo is a completely different topic.  This thread is about high resolution audio file formats.


----------



## charleski

frodeni said:


> This topic has not evolved much, despite the ton of posts.
> 
> It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.


  
 If you have an alternate methodology to suggest, please do so. The* most annoying part* of these 'subjectivist' claims is their complete inability to address the issue in anything resembling an honest manner.
  


> I use an Oppo HA-1, and even for 16/44.1, the buffer must be set for max. Increasing to 32bit, decreases the SQ, for a 16/44.1 source. I have nothing to prove, as I know my hearing, thank you. I also know the drawbacks of A-B testing. Its flaws.


  
 No, you don't.
  


> By theory, buffer increase should not increase SQ. But it does.
> 
> By theory, increasing bit dept to 32bit should not degrade the sound, as it is a pure bit shift operation. But it does.
> 
> The second makes sense, if the firs one is true, but it is bloody hard to understand why. It really does not matter what I understand, as the 16/44.1 clearly sound the best, with the buffer at max.


  


> A controlled test, well, to my knowledge there is no such thing. It really does not even matter: Music is a subjective experience anyway. If I am able to hear harmonies and articulation with one setup, and not with another, well, that is what I do. No matter how I hard I try. People being stupid, denying that, well, that is on them. I know what I hear. My hearing is honed and well trained, and I am far more critical than you guys will ever be, if you get stuck where you are at.


  
 People run controlled tests _all the time_. It's a well-defined and clearly-understood regimen. You know that science thing? That science thing that brought you modern medicine and all the shiny toys we like to play with? It's founded on a strict method, and that method _works_. There's no point throwing a tantrum when it produces results you don't like.
  


> You guys need to move on. You actually ruin this forum, if you attack like you guys do. Time to grow.


 
  
 Please, feel free to move on. This is the Sound _Science _fourm, you can find plenty of Sound Mythology forums elsewhere.


----------



## frodeni

watchnerd said:


> How do you know it makes the sound better? ...


 
  
 I have answered that. Read the post. Take it in.
  
  


bfreedma said:


> Sure. All of us "stupid people" who know that sighted evaluations have long been proven to be unreliable and are still waiting for you "smart people" to be able to support your claims via blind testing or measurements indicating an actual audible difference. ....


 
  
 I actually addressed that. And have flawed this many times by now. "Sighted evaluation" goes both ways, and you need to make room to flaw pure denial as well. As stated long, long, time ago. Move on.
  
  


watchnerd said:


> In any case, whatever is happening with the settings in your Oppo is a completely different topic.  This thread is about high resolution audio file formats.


 
  
 No. It is not. It is about if differences is audible, and why. All the replies answered here, are either flat denial, or about methodology. Gear is a crucial part of that methodology, if you know what that even means.
  


charleski said:


> Please, feel free to move on. This is the Sound _Science _fourm, you can find plenty of Sound Mythology forums elsewhere.


 
  
 Denial is one form of putting it. Science is not denial of what is not understood, quite the contrary. Science is a pretty wide field of practice, and what I just wrote falls well within its bounds.
  
 Rigging tests to prove false results, is not science. As I pointed out ages ago. You guys do not even take into account when gear corrupts your results. If you are this blind to test rigging, none of you will be able to prove any credible scientific results. If any work passed, it needs a critical review. This is pretty basic.
  
 This is clearly going no where. If any of you starts to speak the language of real science, I might follow up on that. More mocking and stupidity, no, I will not follow up on that.
  
 There are well established norms in science for data gathering. Before throwing out globally accepted and basic scientific methodology, do your homework guys.


----------



## charleski

frodeni said:


> Denial is one form of putting it. Science is not denial of what is not understood, quite the contrary. Science is a pretty wide field of practice, and what I just wrote falls well within its bounds.
> 
> Rigging tests to prove false results, is not science. As I pointed out ages ago. You guys do not even take into account when gear corrupts your results. If you are this blind to test rigging, none of you will be able to prove any credible scientific results. If any work passed, it needs a critical review. This is pretty basic.
> 
> ...


 
  
 No, you're going to have to demonstrate some actual understanding of scientific method if you want anyone to take you seriously.
  
 If you don't want to be mocked, try speaking 'the language of real science' yourself instead of making completely unsubstantiated claims. Your vague accusations of 'rigged tests' are simple intellectual dishonesty.
  
 There may well be problems with the current regimens that are used. The solution to that is to explore new methodology or analysis that _retains _the protection against bias that is currently in place (specifically double-blind testing) while introducing more sophisticated elements. The solution is NOT to throw everything out and pretend you can spot a difference because you 'know your hearing'. Because guess what? You don't.


----------



## bfreedma

frodeni said:


> watchnerd said:
> 
> 
> > How do you know it makes the sound better? ...
> ...




This is going nowhere because you are in the Sound Science forum and somehow believe your personal experience acquired via a process utterly absent of any valid controls trumps known science.

BTW, insulting everyone here is not a substitute for supplying support for your methodology. If you have any actual peer reviewed data to contribute or explanation of your claims of rigged tests, now would be an excellent time to provide it. Your hand waving isn't a substitute for the above.


----------



## watchnerd

frodeni said:


> I have answered that. Read the post. Take it in.
> 
> 
> 
> I actually addressed that. And have flawed this many times by now. "Sighted evaluation" goes both ways, and you need to make room to flaw pure denial as well. As stated long, long, time ago. Move on.


 
  
 No, you didn't address it.
  
 And by your answer it's clear you don't understand the difference between "knowing" and "believing".
  
 From everything you've said so far, your belief that it sounds better could be placebo effect and you don't have any evidence that it isn't.
  


frodeni said:


> No. It is not. It is about if differences is audible, and why. All the replies answered here, are either flat denial, or about methodology. Gear is a crucial part of that methodology, if you know what that even means.


 
  
 You were talking about the effects of your gear on 16bit/44.1khz files.  This thread is about 24bit files.   If you want to debate the merits of the settings in your Oppo on 16bit files, that's a different topic.


----------



## frodeni

watchnerd said:


> And by your answer it's clear you don't understand the difference between "knowing" and "believing".


 
  
 Really? Maybe you care to tell us then, the difference between "knowing" and "believing"?
  
 Don't be shy now. Hit us with what you got, and your sane reasoning for me being mad. Go on.


----------



## watchnerd

frodeni said:


> Really? Maybe you care to tell us then, the difference between "knowing" and "believing"?
> 
> Don't be shy now. Hit us with what you got, and your sane reasoning for me being mad. Go on.


 
  
 If you understand the placebo effect or confirmation bias, it's self-explanatory.


----------



## watchnerd

frodeni said:


> Really? Maybe you care to tell us then, the difference between "knowing" and "believing"?
> 
> Don't be shy now. Hit us with what you got, and your sane reasoning for me being mad. Go on.


 
  
 Also, if you prefer video as your means of education, here is a video of an AES panel discussion on how cognition effects hearing (among other things).


----------



## frodeni

bfreedma said:


> This is going nowhere because you are in the Sound Science forum and somehow believe your personal experience acquired via a process utterly absent of any valid controls trumps known science.


 
  
 Again, No. There are many forms of science, what you probably argue would be called "objectivism". That was practiced in the 1800s, and have its flaws. There are plenty of other philosophies within the sciences. A good staring point for the novice, could be this one:
  
https://en.wikipedia.org/wiki/Philosophy_of_science
  
 It is just the tip of a great iceberg.
  
 Another one is this one, in particular false negative, is of greatest interest for this very thread:
  
https://en.wikipedia.org/wiki/False_positives_and_false_negatives#False_negative_error
  
 I got basic training in both physics, social science, and computer science on the basic philosophies applied. What people can sense, is also firmly within psychology. I have even read some of that.
  
 Go on. Read up. Then come back. We do not live in the 1800s anymore.


----------



## frodeni

watchnerd said:


> Also, if you prefer video as your means of education, here is a video of an AES panel discussion on how cognition effects hearing (among other things).


 
  
 How do _know_ that video is real? How do you _know_ that you exist? Insane as it sounds, that is actually vital for that accusation you throw at me.


watchnerd said:


> If you understand the placebo effect or confirmation bias, it's self-explanatory.


 
  
 How on earth are you going to explain the difference of knowing and believing with to these two hypothesis? How do you _know_ that these are true? How is this self-explanatory? This gives the definition and the difference of these terms how exactly?
  
 First is the hypothesis a belief or a knowledge?
  
 Go on. You are way up above your head as it is. Swimming to the surface, would be a wise move.
  
 And for the record, I have just watched ten minutes of the video yet, and all being said thus far, is known to me. I actually have written quite a few post on this site, covering just that.
  
 In particular, I loved the statement about false negatives, and how you needed to take that into consideration, to get any valid results. Particularly, since you of all people sent me there.
  
 It sort of just proves the point being told in the part I have seen so far, just in this case, it is your bias getting a beating.


----------



## bfreedma

frodeni said:


> bfreedma said:
> 
> 
> > This is going nowhere because you are in the Sound Science forum and somehow believe your personal experience acquired via a process utterly absent of any valid controls trumps known science.
> ...




More generalized distraction and hand waving in the absence of your ability to support your subjective opinions and resultant claims.


----------



## watchnerd

>


 
   
 Quote:


frodeni said:


> How do _know_ that video is real? How do you _know_ that you exist? Insane as it sounds, that is actually vital for that accusation you throw at me.
> 
> How on earth are you going to explain the difference of knowing and believing with to these two hypothesis? How do you _know_ that these are true? How is this self-explanatory? This gives the definition and the difference of these terms how exactly?
> 
> First is the hypothesis a belief or a knowledge?


 
  
 First of all:
  
 Re-hashing basic metaphysical philosophy and everything Decartes, Hume, Hegel, Wittgenstein, etc wrote over last several centuries isn't necessary to discuss modern, experiment based research in psychoacoustics.
  
 Second of all:
  
 You're dodging the issue that you don't have any empirical evidence to support your claim.
  


frodeni said:


> And for the record, I have just watched ten minutes of the video yet, and all being said thus far, is known to me. I actually have written quite a few post on this site, covering just that.


 
  
 If it's known to you, and you actually understand it, then you should be able to apply it to yourself and your experiences.
  
 Or do you think you are somehow a unique human being who is immune to confirmation bias and placebo effects?


----------



## frodeni

watchnerd said:


> If it's known to you, and you actually understand it, then you should be able to apply it to yourself and your experiences.
> 
> Or do you think you are somehow a unique human being who is immune to confirmation bias and placebo effects?


 
  
 Well, I have now looked through the entire video of yours, and as a recording artists, at which I still am a newbe, it was quite interesting. In particular, I liked his methodology, in which he uses both knowledge and his senses to try to get an accurate understanding as possible. In particular, I liked to be noticed that DAW EQ is a non destructive filter. I am not arguing the math.
  
 But he on and on again, ask people to listen, but not what to listen for. Most people cannot hear lossy compression effects, but when taught how to listen for them, can then pinpoint them. Just like the Zepplin tune.
  
 As I have stated many times now, if you do not know specifically what to listen for, and are perfectly able to hear the difference before the ab-test, you will not either by how ab-test normally is run. If you can hear them, can pinpoint them, then ab-test simply do not matter. The infamous Tidal web test is one example, as the so called hifi on that test, I cannot hear what I do from the Tidal app. I cannot tell the difference using the web test.
  
 As for your "placebo effect" it will always be there. No matter what you do. You can try to counter it, but as with this effect in all walks of life, it will effect your perception of hearing anyway. Having said that, I can tell my son and daughter voices apart. My hearing is actually made for doing so.
  
 As for false negatives, I ran into the issue with USB cabling. I really did not believe in it, but gave it a try. Since I clearly heard the difference, I was rather shocked. That is somewhere on these forums to.
  
 As for false positives, compressed lossy music sometimes have better separation of instruments. Why is that? Probably because a USM as in photography similar effect. I have even written about that before.
  
 Now, how is that explanation of knowing and believing of yours going? For such a bold and completely unfounded statement, do you have anything to show at all? You cannot even describe this in your own words, can you? But insult using it, that you can.
  
 Have you read up on false negatives now? Did you also get the contradictions in the video you linked to? First they claim that you cannot overcome bias, then they apply their own bias in a number of examples?
  
 Also, giving a hypothesis, that is non valid as of my listening experience is not very impressive. The differences I speak of, is indifferent as to where my head is placed, using speakers. And certainly so when using closed cans. That hypothesis is easy to flaw, but I simply have better things to do. He did not prove that placement hypothesis in any way, he just threw it out there as usual. No real test that could get a true negative seems to be carried out at all. That is not science.
  
 You do not understand the old masters of philosophy. You do not get that this "bias" of yours, is nothing new. It has been debated for thousand of years, by people who deared to discuss them. You do not. You are not even close to master this, if your writing is anything to go by. You just make a lot of seriously false claims about me. Why?
  


watchnerd said:


> Re-hashing basic metaphysical philosophy and everything Decartes, Hume, Hegel, Wittgenstein, etc wrote over last several centuries isn't necessary to discuss modern, experiment based research in psychoacoustics.


 
  
 Sure they are. Some of the claims in that video of yours, violates basic scientific philosophy. How irritating that might be, it is still the case. There is an awful lot of bias in that video of yours, and sometimes it slips a bit too much. There are quite a few false interpretations of experiment results, and yes, generally speaking, many of argumentative flaws are more than discussed by say Decartes.
  
 The next time you pick on somebody, like me, you better know your basics first. I have yet to see any ABX proof of anything in here, only results indicating something, but no proof. It is not proof just because something proofs you believes right. Most "tests" in here, are openly described as rigged, as people do not know basic testing methodology. They do not know the philosophy science is founded on.
  
 So my answer is simply: Ditto. I got personal experience, as do you guys. You guys have bias too, but you sure do not act like you do.


----------



## castleofargh

frodeni said:


> This topic has not evolved much, despite the ton of posts.
> 
> It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.
> 
> ...


 
  you self proclaimed "I have nothing to prove, as I know my hearing". would you accept this from anybody else? of course not, you don't even accept the results of blind tests from other people when they don't fit your world view. so what you're some VIP with an all pass on reasoning and facts? you know how tests are flawed(or so you say), you know how people make mistakes because of overly simplified data. and your way to address all those potential flaws, is to make empty claims while saying that you don't need to prove them because you're you. and that's apparently something special that we should all accept as valid evidence of the truth. 
	

	
	
		
		

		
			




  
 whatever the subject, you have pieces of evidence that differences are audible, or you don't. it doesn't even matter if you're right or not. you're not so much better than everybody else that you and you alone can make claims and be exempted from proving them.


----------



## RRod

frodeni said:


> But he on and on again, ask people to listen, but not what to listen for. Most people cannot hear lossy compression effects, but when taught how to listen for them, can then pinpoint them. Just like the Zepplin tune.


 
  
 Yes and there comes a bit-rate where these artifacts cannot be detected in a blind test. Yet some will say "but but, I can still hear them if I know which file is lossy." So what would this be due to other than the bias of knowing the answer when you take the test?
  
  


frodeni said:


> Having said that, I can tell my son and daughter voices apart. My hearing is actually made for doing so.


 
  
 Straw man. We can make examples all day of sounds that are easy to tell apart; what's the point?
  
  


frodeni said:


> As for false negatives, I ran into the issue with USB cabling. I really did not believe in it, but gave it a try. Since I clearly heard the difference, I was rather shocked. That is somewhere on these forums to.
> 
> As for false positives, compressed lossy music sometimes have better separation of instruments. Why is that? Probably because a USM as in photography similar effect. I have even written about that before.


 
  
 False negatives and positives have a proper statistical definition; these examples do not meet them.
  


frodeni said:


> So my answer is simply: Ditto. I got personal experience, as do you guys. You guys have bias too, but you sure do not act like you do.


 
  
 Actually we do act like we have bias, which is exactly why we try do tests that are as controlled as possible to make determinations. Taking a test with the answers in front of you shows no modicum of control.


----------



## cjl

frodeni said:


> The infamous Tidal web test is one example, as the so called hifi on that test, I cannot hear what I do from the Tidal app. I cannot tell the difference using the web test.


 
 The interesting thing about this is that the Tidal test actually had some very audible flaws, so the fact that you couldn't tell the difference is surprising.


----------



## frodeni

castleofargh said:


> you self proclaimed "I have nothing to prove, as I know my hearing". would you accept this from anybody else? of course not, you don't even accept the results of blind tests from other people when they don't fit your world view. ...


 
  
 So, you are still here? I accept that people experience what they do, and in this case, I accept that. The two of us are quite different, as I find people that can tell things appart by their senses, when I cannot, quite intriguing. The things that defies my belief, is rather interesting to me. Those are the ones that typically gives me the most growth. Staying in bed in safe land won't.
  
 As for this "blind tests have proven", is false. Even when carried out by the book, they only prove that the test candidates were not able to tell the difference, given the circumstances. Some even argue that is not proving anything, it is just a logical implicit deduction. Not an equivalence, which is true. This is the ABC of science, if you cannot take the heat, get out.
  
  


rrod said:


> ...
> 
> Actually we do act like we have bias, which is exactly why we try do tests that are as controlled as possible to make determinations. Taking a test with the answers in front of you shows no modicum of control.


 
  
 Hi there RRod. It has been a while. You know that the test conditions are rigged to only reveal false positives. It was on the table back then. It was simply not set up for false negatives.
  
 You also dismiss anything other than this AB testing, that you even rigg. And as pointed out by me, you do not really know what you are testing, using a lot of current gear. As in most high end gear using an USB interface.
  
 I do not have time for this, as said, I have nothing to prove. If that pisses you off, so be it. That is your problem. It does not change what I am hearing.
  
 Sure, if you could only behave, I could go far into details of my experience. You sort of get a pretty grey answer then, as I by no means think I know this stuff. I only know what I hear, and constantly need to change my perception, as my senses and reasoning meet. You guys are simply unable to carry out such an argument or sharing, as you do not have the respect of others, that it commands.
  
 I have had enough for now. You guys are not going to grow any, and will remain at where you are now, if I check back in a year or five. I will only join in, if someone reasonable joins in. We probably end up having a conversation with 10-15 post in between any one of ours, telling us how stupid we are. That we know no science, even if that is exactly what we are discussing. That is how bad this sub-forum has become.
  
 Bye now.


----------



## RRod




----------



## watchnerd

> We probably end up having a conversation with 10-15 post in between any one of ours, telling us how stupid we are. That we know no science, even if that is exactly what we are discussing.


 
  
 I'll agree with you on that part.
  
 People who aren't familiar with either the concepts and the specific published studies on psychoacoustics continue to question the entire premise of blind listening tests, while also offering no viable alternatives or data themselves.  They confuse philosophical arguments with data, and practice sophistry rather than analysis. That hasn't changed in at least 30-40 years.
  
 Nobody changes sides.  Those who prefer magical thinking will continue doing so.


----------



## watchnerd

frodeni said:


> [redacted wasted space]


 
  
 Are you going to actually share some data that provides evidence of your claims?
  
 Because you're doing a lot of writing, but it doesn't bolster your claims.


----------



## frodeni

cjl said:


> The interesting thing about this is that the Tidal test actually had some very audible flaws, so the fact that you couldn't tell the difference is surprising.


 
  
 No not really. But if you told me the flaws, then maybe I could. There seemed to be some difference, but I could not put my finger on it specificly.
  
 I actually listened for the specific strengths of the sound from the Tidal app on my PC: They were not there. It takes quite a lot of training, as it is for tuning a guitar, to pinpoint specifics. Once you know what to listen for, it is hard to belive you could not hear it before. By training, the listening experience grow, at least for me. Music become more and more fun.
  
 My mm400 has a problem with a tiny metal piece, that makes a noise, given loud enough music, at a specific frequency. Sine I was told of the issue, I knew what it was, when I heard it. It was a hard thing to crack down on, since the answer is not easily given.
  
 If you could pinpoint a flaw, please share. The sonic traits I usually listen for, were simply not distinctly different for that test.


----------



## sonitus mirus

frodeni said:


> No not really. But if you told me the flaws, then maybe I could. There seemed to be some difference, but I could not put my finger on it specificly.
> 
> I actually listened for the specific strengths of the sound from the Tidal app on my PC: They were not there. It takes quite a lot of training, as it is for tuning a guitar, to pinpoint specifics. Once you know what to listen for, it is hard to belive you could not hear it before. By training, the listening experience grow, at least for me. Music become more and more fun.
> 
> ...


 
  
 http://www.head-fi.org/t/770352/how-well-can-you-hear-audio-quality#post_11662420


----------



## nick_charles

watchnerd said:


> Nobody changes sides.


 
 I did  and others here have had similar epiphanies 
  
 For years I was a true believer that everything was different and everything made a difference I went through all sorts of contortions from spiked speaker stands and isolation tables and hand-braided solid core mains cable for speaker wire - I even bought several expensive RCA cables certain that there must be a difference as so many people expressed that opinion, ***** I even bought two turntables twenty years after I went digital (the grotesque noise from vinyl drove me nuts) . It took me many years and some backsliding (Just bought a new SMSL M8 - well it was cheap on Massdrop and it has good measurements) to arrive at a more or less rational attitude - a switch box an external ADC and FooBar's ABX plugin slowly brought me to realize that there is a lot of magical thinking out there and that I had been a victim of it. the stupid thing is that I have two degrees in Psychology and that in no way stopped me from being hoodwinked, funny really !


----------



## castleofargh

frodeni said:


> castleofargh said:
> 
> 
> > you self proclaimed "I have nothing to prove, as I know my hearing". would you accept this from anybody else? of course not, you don't even accept the results of blind tests from other people when they don't fit your world view. ...
> ...


 
 this is sound science, self confidence doesn't determine the reality of things. theories, tests, and measurements try to. you pretend to be the rational side of the argument, while doing everything to prove that you don't get any of it.
 you show exaggerated self confidence, as if human senses couldn't be fooled by countless biases.
 you decide that things are right or wrong depending entirely on how they agree with your opinion.
 you present zero evidence but sure love to make empty claims.
  
 in short, you're one guy with nothing but opinions trying to force them onto others through sheer rhetoric as if they were facts. I see no rational and no science in this. and I (and a few others as you can see)would like you to stop.
  
 now for the second part, oh the irony!  making fun of me and talk about logical implicit deduction for something I didn't write that you made up just to win your own argument. priceless!
 when exactly did a straw-man argument become logical and the ABC of science? as you're doing the same with the answer to RRod, I would also appreciate it if you would stop.
  
 this is sound science, if you have an opinion, offer it as an opinion. if you have a fact, offer it with the evidence that can lead to accept them as facts. if you have no better than trickery rhetoric and "I know what I hear", please leave this sub section. you can find plenty of "impression threads" in other sections of the forum where your posts will be very fine.


----------



## watchnerd

nick_charles said:


> I did  and others here have had similar epiphanies
> 
> For years I was a true believer that everything was different and everything made a difference I went through all sorts of contortions from spiked speaker stands and isolation tables and hand-braided solid core mains cable for speaker wire - I even bought several expensive RCA cables certain that there must be a difference as so many people expressed that opinion, ***** I even bought two turntables twenty years after I went digital (the grotesque noise from vinyl drove me nuts) . It took me many years and some backsliding (Just bought a new SMSL M8 - well it was cheap on Massdrop and it has good measurements) to arrive at a more or less rational attitude - a switch box an external ADC and FooBar's ABX plugin slowly brought me to realize that there is a lot of magical thinking out there and that I had been a victim of it. the stupid thing is that I have two degrees in Psychology and that in no way stopped me from being hoodwinked, funny really !


 
  
 Well, "nobody" should have been "it's rare".
  
 I have to spot-check myself often, too.  Not much with digital, but with my turntable - heck yes.  Because vinyl is so messy and flawed, and prone to tweakery actually making a difference (cartridge alignment being an example), it's easy to get into a delusional mind set.


----------



## nick_charles

watchnerd said:


> Well, "nobody" should have been "it's rare".
> 
> I have to spot-check myself often, too.  Not much with digital, but with my turntable - heck yes.  Because vinyl is so messy and flawed, and prone to tweakery actually making a difference (cartridge alignment being an example), it's easy to get into a delusional mind set.


 
  
 Jim Johnston's box that just went click clack is a priceless example !


----------



## old tech

frodeni said:


> ...As for your "placebo effect" it will always be there. No matter what you do. You can try to counter it, but as with this effect in all walks of life, it will effect your perception of hearing anyway. Having said that, I can tell my son and daughter voices apart. My hearing is actually made for doing so.
> 
> As for false negatives, I ran into the issue with USB cabling. I really did not believe in it, but gave it a try. Since I clearly heard the difference, I was rather shocked. That is somewhere on these forums to.
> 
> ...


 
  
 Your arguments are strikingly familiar to those used by psychics and practitioners of homeopathy, astrology and other paranormal subjects.


----------



## singboo1000

I've spent some time reading this as a lurker. The thing is that it doesn't really matter which argument is right. Logic dictates that with what we know of physics and the human ear then the 16/44 encoding would be more than adequate. Yet, some people prefer different. Surely that's ok? Even if it's placebo effect. 
  
 All I know is that there is a mountain of difference between available encodings of different albums and that is what matters to me. I have bought higher bit rate encodes before as I preferred the released version of others available. Not really a decision based upon the bit rate itself. 
  
 Same goes for the equipment. I have a nice DAP. Like a fool, I bought it knowing that there is little way to justify the cost. However, having tried over a dozen of the things with various headphone combinations, I settled on the one I liked the sound of best. No science just chose the one from the plate that tasted best to me.
  
 That's okay is it not? Don't see why people are so heated in their views here or feeling the need to defend their preference with pseudo science.


----------



## RRod

nick_charles said:


> Jim Johnston's box that just went click clack is a priceless example !


 
  
 And it would seem there are two ways to take the revelation:
 a) "So I was just making it all up in my head? Guess I learned that lesson; please put the box away now."
 b) "Well the box *did*make things sound better in my mind, so I guess I need to use it all the time from now on."
  
 Many people on this site would call b) a reasonable response, because all that matters in the end is how subjectively good you feel about your music experience. To some of us, a) is the best way, because we get the same actual sound coming into our ears but without being a slave to the box.


----------



## spruce music

nick_charles said:


> I did  and others here have had similar epiphanies
> 
> For years I was a true believer that everything was different and everything made a difference I went through all sorts of contortions from spiked speaker stands and isolation tables and hand-braided solid core mains cable for speaker wire - I even bought several expensive RCA cables certain that there must be a difference as so many people expressed that opinion, ***** I even bought two turntables twenty years after I went digital (the grotesque noise from vinyl drove me nuts) . It took me many years and some backsliding (Just bought a new SMSL M8 - well it was cheap on Massdrop and it has good measurements) to arrive at a more or less rational attitude - a switch box an external ADC and FooBar's ABX plugin slowly brought me to realize that there is a lot of magical thinking out there and that I had been a victim of it. the stupid thing is that I have two degrees in Psychology and that in no way stopped me from being hoodwinked, funny really !


 

 Nice post.  I share a similar experience.  A technical background and should have known better you would think.  I spent years on what I eventually realized was an irrational approach based upon ideas that were in no way reflective of reality.


----------



## frodeni

I really should not, but this is the rudest and most uneducated post I have read by a moderator, for quite some time now.
  
 Quote:


> Originally Posted by *castleofargh* /img/forum/go_quote.gif
> 
> ...
> 
> ...


 
  
 That was the teaching every student at the University of Oslo had to take in the 1990s. Most still do. Not something I ever made up, but what was considered the bare minimum a scientist of any kind had to know to be able to conduct any science. This particular point was devoted many weeks of teaching.
  
 I once had a full two hours lecture on why a man who had a blue tongue, not necessarily had eaten blue berries. We were also taught basic logic, highlighting what could be proven by promoting a hypothesis and what that really "proved". Science is not really "knowledge" but rather a field of most probably assumptions. Usually referred to as hypothesis.
  
 My asking for a definition of knowlegde and beliefs, is well placed to. Science really do not know anything, it is all up to the individual to choose his own beliefs. At some point, something is proven beyond doubt, and is considered "knowledge". It can still be wrong. Just like the laws of Newton was in the old days. Laws even mathematically proven, or so they thought. Most scientific knowledge is nothing but assumptions people believe in: As is most knowledge in life.
  
 This is at the heart of science, and not something I made up. It is the very fabric of any science, and well recognized.
  
 The main argument against objectivism, is that any such knowledge, must be subjectively experienced. Straight back to the senses. Arguing that everything is experienced through the senses, thus nothing can be purely objective, is an age old argument. One I am keenly aware of. Most science on hearing, is purely empirical, and based on subjective claims. Even worse, most stem from research using simple sine waves or poorly conducted tests. I see no reason to let people tell me what to sense by hearing, based on this. None. Dictating people what to sense, is a dangerous route to take.
  
 Rigging a test to mock people, like the one in that video of late, is simply unethical. As is much of the pushing in here, as it has a serious part of mocking built into it. Also, the test are biased and rigged, favoring false positives, and not the three others. In particular, false negatives are almost impossible to detect. This is not anecdotal, it is classic science in which you need to consider all four possible outcomes, if testing one parameter. It is the most common flaw, in any science, not to consider all four.
  
 Those who do not belive in any difference between 16 and 24 bit, getting a blind test with one 16 and one 24 bit sample, how can they produce a false negative? You guys also actively agitate that no such difference exists, mocking anyone who claim otherwise. That in itself would flunk any work of science, as this is active manipulation of the test subjects. People have been thrown out of universities for far less.
  
 Selection is critical to. Random only works to find random skill. Only one person is needed to prove that something is humanly possible. The tests do not seem to recognize this, as far as I have seen. If just one person makes it way past slump, then the negation is falsified.
  
 Any hypothesis needs only to be falsified once, to be proven false. It does not matter how many times it has been supported. Also pretty basic. Just not in this subforum.
  
 This is not a true science forum. It never was, it probably never will be.
  
 People with shared belief in objectivism have found each other, and that is all fine. Most of you speak of objectivism, which is fine. But all science it is not. This subforum should be renamed Objectivists forum. A new forum should be made and named science forum, with current forum as a sub forum. That way, the rest of could discuss the science of things by a loose definition of science, and objectivists could discuss things purely objectivistic in their sub forum.
  
 I might set up a test in some far future, and just be the one to kill some claims. If I came in with a near perfect score on a 16 vs 24 bit test, would anyone of you believe me? What does that really take? What is in such a test for me? A **** load of conspiracy theories, and a ton of ridicule and mocking for sure. What else? Why should I even care? Why would I feel any need to please you, acting like you do? Give it some thought. I have nothing to prove.
  
 As always, I will follow up on sensible answers. Hopefully I will be able to find those, in all the noise: I will give it a fair try.


----------



## Hitec

rrod said:


> And it would seem there are two ways to take the revelation:
> a) "So I was just making it all up in my head? Guess I learned that lesson; please put the box away now."
> b) "Well the box *did*make things sound better in my mind, so I guess I need to use it all the time from now on."
> 
> Many people on this site would call b) a reasonable response, because all that matters in the end is how subjectively good you feel about your music experience. To some of us, a) is the best way, because we get the same actual sound coming into our ears but without being a slave to the box.


 
 Here' a rant to add......  We are designed to prove only in the material world.  We don't believe in anything that cannot be proved.  ..this may be the biggest problem that is limiting our concept of life.
  
 I think that their is more to music than we think.  Those movie scores really affect you.  Music moves you mentally - calculate that...  If you are sitting still and all of a sudden you hear boommooom - you get scared.  When you hear high sounds they seem friendlier.  As a result, music can be used to tantalize the soul.  It speaks to us in music language that relate to us physically.  There are things that we still don't know about ourselves, and the world, yet we cross it off as being impossible but not possible.  Where do you go if you keep straight into outer space.  Some people will say no where - well prove it.  Some people will say you will go on and on and on - prove it.  Some say, you will go in a giant circle and return - prove it.   ..believe in God ... glass half full, or half empty.  This stuff goes on and on.  What I'm getting to is, hey, music may be doing something to us beyond just hearing.  ..why does a song come on and make you think of a happy or sad time.  At that moment, you were not even thinking of the bass, mids, hi's, etc.   Maybe we should judge 24bit music by how it makes you feel, rather than by if you can hear the leaf fall.
  
 ..ever notice how you can just quietly stand over someone while they are sleeping and a lot of the times, they will feel your presence and open their eyes and wonder why you are staring at them.  How was that possible without hearing or seeing.  Science should not be in charge of ruling everything out.  If so, then science should have ruled that we are not alive, especially since it can't prove or disapprove God. If a person say's A sounds better than B, then it could be that there is something in the music that is actually moving them and allowing them to really enjoy what they are hearing - can't be physically explained.  It is probably something that can't be measured or proven, since we are basically in the caveman stages of our mentality.  Can you imagine arguing with a caveman.  In one thousand years, a lot of what we believe or didn't know today, will make us look like cavemen.  Open your mind to the possibilities".


----------



## frodeni

sonitus mirus said:


> http://www.head-fi.org/t/770352/how-well-can-you-hear-audio-quality#post_11662420


 
  
 If I got this right, base and treble is lifted in the uncompressed file by 1db?
  
 Even knowing that, I am still not able to distinctly separate the samples in the Tidal test. Used Chorme, Win10, Oppo HA-1, and HD800. Still sound about the same to me, and I cannot put my finger on any specific sonic trait to separate the samples.
  
 What in particular should I listen for?


----------



## cjl

Closer to 2dB, based on this spectral analysis: http://cdn.head-fi.org/2/23/900x900px-LL-2384b094_Tidal.png
  
 As for what I would listen for, it depends on how good your high frequency extension is. If you have fairly decent high frequency hearing, I'd guess that the high frequency attenuation might be a bit more audible, but if not, you should listen for a bit more fullness/warmth to the bass in one compared to the other (which isn't surprising, since one has the bass attenuated by 2dB below 100Hz or so). I can hear both the differences in the highs and in the lows pretty clearly when doing a fast switch ABX, but it's much more subtle with a slow switch comparison (for me at least).


----------



## frodeni

cjl said:


> Closer to 2dB, based on this spectral analysis: http://cdn.head-fi.org/2/23/900x900px-LL-2384b094_Tidal.png
> 
> As for what I would listen for, it depends on how good your high frequency extension is. If you have fairly decent high frequency hearing, I'd guess that the high frequency attenuation might be a bit more audible, but if not, you should listen for a bit more fullness/warmth to the bass in one compared to the other (which isn't surprising, since one has the bass attenuated by 2dB below 100Hz or so). I can hear both the differences in the highs and in the lows pretty clearly when doing a fast switch ABX, but it's much more subtle with a slow switch comparison (for me at least).


 
  
 I am past 40, and my hearing is not what it used to be. Just a simple fact of life. The HD800 extend far enough, my hearing do not.
  
 If you hear the difference using a browser, then you probably do. I just do not.
  
 Using the browser, there is no such thing as a quick ABX switch. I need to listen to whole sample before switching. My whole point is that the sound is so degraded being played by the browser, that any on the strengths I usually experience are not there.
  
 Obviously, I am not the master of minor amplitude changes, and others outclass me in that aspect. Not what I am used to, but this time, is what it seems like.
  
 That would also give a hint, why people might be focused on this loudness, doing ABX. I simply do not get why. The things I listen for, is only affected by a minor degree by change in volume. Soundstage is pretty stable. As is imaging. Articulation is only lesser affected. Separation is also fairly stable. As are harmonies. None of the things I need to enjoy the music is in any of the samples, when played by the browser: I cannot find the differences I normally listen for, as the reproduction is somehow degraded too far.
  
 I cannot tell a 2db drop? Well, that was a surprise. Even how poor and unknown the sound was.


----------



## castleofargh

> Quote:Originally Posted by *frodeni* /img/forum/go_quote.gif


 
  
   I'm being rude... the only reason why I haven't removed you entirely is because I'm always scared of abusing my power. and with each new post you make me regret not having done so the first time I wanted to.
 your 2 tier logic where everything is questionable and can't be totally trusted, except your empty claims, is the rude element of this topic. answering a post by making up your own straw man argument instead of dealing with what is really discussed, that's being rude to the poster. trying to fool people into thinking you're right with rhetoric instead of clean arguments, that's being rude to every reader of the topic.  this is below, the post I started to answer to because it was a little rude, had nothing to do in sound science, is off topic, and came out of nowhere for no reason. 


frodeni said:


> This topic has not evolved much, despite the ton of posts.
> 
> It all comes down to a limited set of people, not taking in the critics of others. Methodology is simply flawed, and reasoning is made by overly simplified empirical data.
> 
> ...


 
 and after that kind of silly post full of kanye west (or trump?) impersonation, you want to give us lessons on science, logic, respect, and human nature. talk about the pot calling the kettle black.
  Quote:


frodeni said:


> sonitus mirus said:
> 
> 
> > http://www.head-fi.org/t/770352/how-well-can-you-hear-audio-quality#post_11662420
> ...


 
 are you really talking about this ? http://test.tidalhifi.com/ 
 someone should be in prison and the website taken down. just listen for a general feel of the music being louder and you should get 5 out of 5. if it's difficult for one song, focus on low or high end sounds, but still loudness is all you need. apparently I didn't know but when I boost my EQ I increase resolution. at least that's what this fraud of a test is saying.
  
  
  
  
  


hitec said:


> I think that their is more to music than we think.


 
 good luck with that. you realize that you're listening to music made and processed based on what we know of music right? so we can record the mysterious element, convert it into a fluctuating voltage, then into some digital or analog support, then back to voltage, then back to air pressure thanks to a headphone or speaker made with shape and volumes expecting a certain acoustic reaction and we get it just like the simulation said we would. and with all that under all those different forms, the mysterious element would travel unsuspected? what do you figure are the odds for something like this to be possible? for us to carry and reproduce something we haven't figured out, yet get it pretty right at the end? 
	

	
	
		
		

		
			




  
  any emotional charge on music or other things should be discussed with a psychologist. you're the one pushing your emotional baggage onto a music, a piece of art, or a cake, and finding that it's more than what reality says it is. a song you like the first time you hear it, I may dislike it, so how is that element supposed to be attributed to the music?
 if I like a song and listen to it 5000 in 2 weeks, I bet I will start to hate it. but it's still the same song. your all argument is discussing the human mind and emotions, not music and certainly not the audible fidelity of an audio signal.
  
 and it's not that we don't believe in anything that cannot be proved, it's just that we don't like people spamming us with "it exists" when they have zero evidence of it. the matter is purely about not making claims until we can prove them.
 but you believe whatever you want, that's being an individual, we have no obligation to think like the guy next to us(that wouldn't be much fun anyway). but when talking to other people in our shared reality, trying to force our own vision as if it was factual truth, while having zero evidence and saying you don't need to present any. that's the kind of stuff a dictator or a bully would try. a normal respectful human would mind the chance of claiming something that may not be true, and thus being a vector of disinformation to the community. so he would refrain from making empty claim for that reason, and would limit his claims to what he can demonstrate.
 and as this is the sound science section, science methods are what we expect, and when it comes to human sense, there is nothing more effective than a blind test for the science guys. that's why blind tests are used in many industries, the drugs are tested that way and they're a much more serious matter than testing for minute audible differences in a file format. but somehow the testing methods that decide upon the life and death of people, aren't good enough for some audiophiles to test if they really hear something or if they get tricked by money and marketing. hard not to be amazed.
  
 on a side note: a person standing over you, you will most likely hear him(move, breath), or smell him(perfume, hormones, bad breath...)***. you just went and concluded something inexplicable because you didn't think about testing the explicable. not because it really can't be explained. when sleeping our hearing and smell are ***still fully active.
  
***edit: @Don Hills(thanks) just mentioned that I was wrong about smell, and this seems to agree with him http://www.sciencedaily.com/releases/2004/05/040518075747.htm
 so science just proved me wrong (once again)^_^. hearing is right though.


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## RRod

hitec said:


> What I'm getting to is, hey, music may be doing something to us beyond just hearing.  ..why does a song come on and make you think of a happy or sad time.  At that moment, you were not even thinking of the bass, mids, hi's, etc.   Maybe we should judge 24bit music by how it makes you feel, rather than by if you can hear the leaf fall.


 
  
 24-bit is a technical consideration and is thus open to analysis in that way. Many people get shivers listening to Caruso's cylinder recordings; that doesn't mean we can't discuss their technical aspects and decide they aren't the way to go for music dissemination.


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## frodeni

castleofargh said:


> ... are you really talking about this ? http://test.tidalhifi.com/
> someone should be in prison and the website taken down. just listen for a general feel of the music being louder and you should get 5 out of 5. if it's difficult for one song, focus on low or high end sounds, but still loudness is all you need. apparently I didn't know but when I boost my EQ I increase resolution. at least that's what this fraud of a test is saying.
> ...


 
  
 Yes.
  
 Funny how you suddenly hear things I cannot.
  
 Here we are, you telling me to listen for a "general feel of the music being louder". There is such an irony to that. That probably is the best description you can give, which is fine with me.
  
 If you say you can hear a difference, then you probably can.
  
 I still cannot pinpoint any difference. Switching between 320kbps and lossless within the Tidal application, well that is another story. In that case I can pinpoint a lot of distinct sonic traits, with fairly high accuracy.
  
 In a bisarr twist of fate, you suddenly made my argument. I will leave you at that, as my point is utterly made by now.


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## watchnerd

hitec said:


> As a result, music can be used to tantalize the soul.  It speaks to us in music language that relate to us physically.  There are things that we still don't know about ourselves, and the world, yet we cross it off as being impossible but not possible....What I'm getting to is, hey, music may be doing something to us beyond just hearing.  ..why does a song come on and make you think of a happy or sad time.  At that moment, you were not even thinking of the bass, mids, hi's, etc.   Maybe we should judge 24bit music by how it makes you feel, rather than by if you can hear the leaf fall.


 
 [removed stuff about outer space]
  
 Double-blind testing doesn't, a prior, negate emotional or even spiritual reactions unless said reactions are also tied into a sighted experience.  I find Beethoven's 9th Symphony to be a spiritually meaningful piece of music, but I don't need to see the objects in question to know whether version A vs B moves me emotionally...or if it's the same.


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## watchnerd

frodeni said:


> Yes.
> 
> Funny how you suddenly hear things I cannot.
> 
> ...


 
  
 How did he prove any of your argument(s)?
  
 The Tidal situation is not placebo...the two files have a measurable difference well within the range of audibility.
  
 As a reminder:
  
 Perceptual coaching can help a subject statistically pass an ABX test if the difference actually exists and is audible.
  
 Perceptual coaching cannot help a subject statistically pass an ABX test if there is no difference or isn't audible.


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## sonitus mirus

watchnerd said:


> How did he prove any of your argument(s)?
> 
> The Tidal situation is not placebo...the two files have a measurable difference well within the range of audibility.
> 
> ...


 
  
 To add a bit about coaching, Tidal offers a tip for anyone taking their online test that specifically calls out the exact differences identified between the two files through signal analysis.
  
 http://test.tidalhifi.com/intro 





> Compressed music weakens for instance the pressure in the bass, details in cymbals and gives the sound less headroom.


 
  
 Tidal was even telling everyone that they should be hearing a slightly lower volume with bass frequencies and cymbals precisely where we find the -2dB EQ variations with the lossy files.  
  
pressed music weakens for instance the pressure in the bass, details in cymbals and gives the sound less headroom.​


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## cjl

Yep - I generally find cymbals are one of the easiest instruments to pinpoint a treble rolloff.


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## mmerrill99

I believe what Frodeni is referring to is Castleofargh suggestion for doing an ABX test along the lines of "general feel of the music being louder"

I have never seen this advice being termed "perceptual coaching". The term generally refers to the training given in the conscious identification of audible differences or impairments as this is the usual approach to ABX testing - the specific & conscious identification of differences - it's one of the great overlooked problems of ABX testing, incorrect or insufficient training in specific audible impairments. It strikes me that this instruction to use the "general feel of the music being louder" shows a lack of understanding of the psychoacoustic issues of ABX testing - the fact that without a very specifically identifiable audible difference, ABX testing is guaranteed to return a null result. 

But, if this is a new approach to ABX testing, can anyone give examples of ABX tests that have been performed with this instruction "how one feels about the music"?


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## jcx

*"guaranteed*" is an unscientific overstatement - focus and training helps, can sometimes change results from "cannot reject the null hypothesis" to "statistically significant" - but you don't always have to have a "name" for the difference
  
 the "forced choice" feature of controlled listening tests has proven valuable at times - people experienced with DBT, ABX have shown statistically significant discrimination when they still feel at a loss to describe the difference, don't even consciously "feel" the difference, thought they were guessing randomly
  
 Quote:


> psychoacoustic issues of ABX testing - the fact that without a very specifically identifiable audible difference, ABX testing is *guaranteed* to return a null result.


 
  
  
 but overall yes it does help a lot when the sound difference can be given a suggestive name/description connected with the rest of our experience
  
  
 even better is the up/down threshold test when the difference has a metric that correlates with audibility at higher levels, the test starts at those higher levels, people getting some number of trials correct then get the stimulus level reduced for the next series, miss too many and it steps back up - so the training is built in
  
 when the test reverses direction a few times its taken as evidence that the audible threshold for that subject,  those test conditions has been reached
  
  
 as many have pointed out your "devils advocate" attack on everything isn't itself particularly intellectually sound - some things have lots more evidence going in their favor than others


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## mmerrill99

jcx said:


> "guaranteed" is an unscientific overstatement


I used it for the psychological impact it would have on the reader 





> - focus and training helps, can sometimes change results from "cannot reject the null hypothesis" to "statistically significant" - but you don't always have to have a "name" for the difference


How many of these "feel good" ABX results can you point to that stand up to scrutiny?



> the "forced choice" feature of controlled listening tests has proven valuable at times - people experienced with DBT, ABX have shown statistically significant discrimination when they still feel at a loss to describe the difference, don't even consciously "feel" the difference, thought they were guessing randomly


Indeed, I'm sure this can happen but does this make it any more convincing? I often see a single statistically positive ABX result being rejected on this basis - so can you show a large scale, statistically significant ABX result which has used this "feel good" method? 


> > psychoacoustic issues of ABX testing - the fact that without a very specifically identifiable audible difference, ABX testing is *guaranteed* to return a null result.
> 
> 
> 
> ...


Correct & the whole impetus in ABX tests is towards this method of conscious identification of audible differences but as I said if this is a new approach, I'm eager to see the results.



> even better is the up/down threshold test when the difference has a metric that correlates with audibility at higher levels, the test starts at those higher levels, people getting some number of trials correct then get the stimulus level reduced for the next series, miss too many and it steps back up - so the training is built in
> 
> when the test reverses direction a few times its taken as evidence that the audible threshold for that subject,  those test conditions has been reached


Would these be up/down differences in the "goodness of feeling" or some other up/down audible aspect being used?



> as many have pointed out your "devils advocate" attack on everything isn't itself particularly intellectually sound - some things have lots more evidence going in their favor than others


_"devils advocate" attack on everything_ is a very unscientific overstatement - guess you are using it for psychological effect on the reader? I'm asking for your "lots more evidence" if you would care to provide it.

As soundandmotion once stated - I'm skeptical of the skeptics & you obviously don't like being asked the skeptical questions & presume to put yourself forth as the spokesman for others who feel the same? 

BTW, science also uses "devils advocate" techniques quite extensively - it's a well known approach to finding logical issues with any particular hypothesis - go figure


----------



## charleski

frodeni said:


> That was the teaching every student at the University of Oslo had to take in the 1990s. Most still do. Not something I ever made up, but what was considered the bare minimum a scientist of any kind had to know to be able to conduct any science. This particular point was devoted many weeks of teaching.
> 
> I once had a full two hours lecture on why a man who had a blue tongue, not necessarily had eaten blue berries. We were also taught basic logic, highlighting what could be proven by promoting a hypothesis and what that really "proved". Science is not really "knowledge" but rather a field of most probably assumptions. Usually referred to as hypothesis.


  
 Right, so you start off with a hypothesis, that files containing 24bit samples sound better than files with 16bit ones. You then do an experiment to test this. Guess what, people have done this, *repeatedly*. In almost every case the result was that the subjects couldn't detect a difference, and those that did claim to show a difference were found to have serious methodological flaws.
  
 Now, let me let you into a secret. It's very hush-hush, so don't let anyone know I told you.
 In science, when the evidence decisively points to a particular conclusion, scientists end up thinking that's what they have to accept, even when it destroys their most fondly-held theories. The history of sicence is littered with theories that were found to be bad and were rejected.
 It doesn't always work, it's not perfect. Sometimes theories that have been thrown out are later found to provide a better explanation of the facts. But that happens when new evidence comes to light that can't be explained. It's not a matter of debate, it's a matter of finding the new evidence.
  
 Right now there's a perfectly good explanation for why you think your music sounds better in 24bit: you're fooling yourself. People are very _very_ good at fooling themselves, and that's why scientific method has evolved such a robust set of measures to prevent this from interfering with the evidence.
  


> My asking for a definition of knowlegde and beliefs, is well placed to. Science really do not know anything, it is all up to the individual to choose his own beliefs. At some point, something is proven beyond doubt, and is considered "knowledge". It can still be wrong. Just like the laws of Newton was in the old days. Laws even mathematically proven, or so they thought. Most scientific knowledge is nothing but assumptions people believe in: As is most knowledge in life.
> 
> This is at the heart of science, and not something I made up. It is the very fabric of any science, and well recognized.


  
 Utter BS. Science is not about beliefs. Period.
 Newton's laws, by the way, are still perfectly valid within their limits and are still in active use across the entire world. You don't integrate a tensor to find out how fast your car will go, F=ma works just fine.
 I will accept that you didn't make this up, though. You're repeating the same garbage you'll find on any Creationist website.
  


> As always, I will follow up on sensible answers.


  
 Please don't.

  
  
  


mmerrill99 said:


> I believe what Frodeni is referring to is Castleofargh suggestion for doing an ABX test along the lines of "general feel of the music being louder"
> 
> I have never seen this advice being termed "perceptual coaching". The term generally refers to the training given in the conscious identification of audible differences or impairments as this is the usual approach to ABX testing - the specific & conscious identification of differences - it's one of the great overlooked problems of ABX testing, incorrect or insufficient training in specific audible impairments. It strikes me that this instruction to use the "general feel of the music being louder" shows a lack of understanding of the psychoacoustic issues of ABX testing - the fact that without a very specifically identifiable audible difference, ABX testing is guaranteed to return a null result.
> 
> But, if this is a new approach to ABX testing, can anyone give examples of ABX tests that have been performed with this instruction "how one feels about the music"?


 
  
 I would actually agree that the ABX test's reliance on conscious discrimination represents a weakness. But it is possible to test subconscious responses to stimuli that have no direct conscious correlate, and this has been done extensively in visual perception. These same effects are certainly present in the auditory system, in fact lossy mp3 and aac compression is based on the same phenomenon of perceptual masking found in vision, which is a subsconscious process.
  
 But if you want to probe subsconscious modulations like this you have to come up with a coherently-testable parameter. These tests work by showing how a conscious response is modulated by stimuli that are not consciously preceivable, a process called masked priming. So there has to be some conscious decision-making that is being measured: Is the instrument an oboe or a bassoon? Is the sound near or far? etc.
  
 Measuring mood is a lot trickier, largely because there are so many things that affect your mood, and listening to music is a relatively minor consideration in it. You can try measuring mood change, assessing it before and after listening to the music, but that still doesn't guard against random effects: even the most beautiful music in the world isn't going to raise your mood if it reminds you of that disastrous date you had last month.
  
 And lastly, the sad fact is that this is a commercial process. As we have seen, all the evidence in the world isn't going to stop people frothing at the mouth and claiming the tests were rigged if they produce results counter to their expectations. Manufacturers want tests that show you should buy their more expensive products, and consumers want the same so they can justify their purchase. While the ABX test's reliance on conscious discrimination is a weakness in general scientific terms, it's justifiable in that it was designed to test precisely that sort of claim: that fancy expensive gadgets could be consciously discriminated from nasty cheap ones.
  
 I certainly think it would be worthwhile to develop some form of masked priming test for musical perception. I very much doubt that it would show any difference between 96/24 and 44.1/16, but if it did I'd certainly have to consider it very carefully. If such results could be reliably repeated I'd have to change my opinion. But I can tell you right now that no matter how sophisticated the test, if it didn't show any effect you would still find audio mythologists rejecting the negative results out of hand and clinging to their faith.


----------



## Guidostrunk

What a great post! 



hitec said:


> Here' a rant to add......  We are designed to prove only in the material world.  We don't believe in anything that cannot be proved.  ..this may be the biggest problem that is limiting our concept of life.
> 
> I think that their is more to music than we think.  Those movie scores really affect you.  Music moves you mentally - calculate that...  If you are sitting still and all of a sudden you hear boommooom - you get scared.  When you hear high sounds they seem friendlier.  As a result, music can be used to tantalize the soul.  It speaks to us in music language that relate to us physically.  There are things that we still don't know about ourselves, and the world, yet we cross it off as being impossible but not possible.  Where do you go if you keep straight into outer space.  Some people will say no where - well prove it.  Some people will say you will go on and on and on - prove it.  Some say, you will go in a giant circle and return - prove it.   ..believe in God ... glass half full, or half empty.  This stuff goes on and on.  What I'm getting to is, hey, music may be doing something to us beyond just hearing.  ..why does a song come on and make you think of a happy or sad time.  At that moment, you were not even thinking of the bass, mids, hi's, etc.   Maybe we should judge 24bit music by how it makes you feel, rather than by if you can hear the leaf fall.
> 
> ..ever notice how you can just quietly stand over someone while they are sleeping and a lot of the times, they will feel your presence and open their eyes and wonder why you are staring at them.  How was that possible without hearing or seeing.  Science should not be in charge of ruling everything out.  If so, then science should have ruled that we are not alive, especially since it can't prove or disapprove God. If a person say's A sounds better than B, then it could be that there is something in the music that is actually moving them and allowing them to really enjoy what they are hearing - can't be physically explained.  It is probably something that can't be measured or proven, since we are basically in the caveman stages of our mentality.  Can you imagine arguing with a caveman.  In one thousand years, a lot of what we believe or didn't know today, will make us look like cavemen.  Open your mind to the possibilities".


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## castleofargh

mmerrill99 said:


> I believe what Frodeni is referring to is Castleofargh suggestion for doing an ABX test along the lines of "general feel of the music being louder"
> 
> I have never seen this advice being termed "perceptual coaching". The term generally refers to the training given in the conscious identification of audible differences or impairments as this is the usual approach to ABX testing - the specific & conscious identification of differences - it's one of the great overlooked problems of ABX testing, incorrect or insufficient training in specific audible impairments. It strikes me that this instruction to use the "general feel of the music being louder" shows a lack of understanding of the psychoacoustic issues of ABX testing - the fact that without a very specifically identifiable audible difference, ABX testing is guaranteed to return a null result.
> 
> But, if this is a new approach to ABX testing, can anyone give examples of ABX tests that have been performed with this instruction "how one feels about the music"?


 

 it's not about abx, he was asking for cues as to what to look for. as this stupid test is really about hearing EQed bass and trebles when pretending to make us test highres vs lower res, I just explained how I passed this farce of a test. and general feeling of loudness was good enough for 4 out of five songs for me. the last one I had to focus on the trebles a lot to notice which one I should select.  for the other 4, if it felt louder I clicked highres and got it right...





 anyway 5 trials means nothing, this test is marketing pretending to play science by wearing a white coat. they ask "do you prefer rolled off bass and trebles? no? so pay for our stuff while thinking the difference had anything to do with resolution" I don't know how this is even legal.


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## watchnerd

guidostrunk said:


> What a great post!


 
  
 can't tell if serious


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## watchnerd

castleofargh said:


> I don't know how this is even legal.


 
  
 It might not be. At least in the US, you can report companies for violations of truth in advertising.


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## mmerrill99

charleski said:


> I would actually agree that the ABX test's reliance on conscious discrimination represents a weakness. But it is possible to test subconscious responses to stimuli that have no direct conscious correlate, and this has been done extensively in visual perception. These same effects are certainly present in the auditory system, in fact lossy mp3 and aac compression is based on the same phenomenon of perceptual masking found in vision, which is a subsconscious process.
> 
> But if you want to probe subsconscious modulations like this you have to come up with a coherently-testable parameter. These tests work by showing how a conscious response is modulated by stimuli that are not consciously preceivable, a process called masked priming. So there has to be some conscious decision-making that is being measured: Is the instrument an oboe or a bassoon? Is the sound near or far? etc.


Thank you for the links & a most interesting demonstration. I feel that I can tell the same from the different masked word but I have no way of verifying this - the switch to the word "WINDOW" appears different in quality in some instances - a more abrupt jarring of the visual change from "######"

Your points are good & it does raise one issue that I have been making in all my posts here - what goes on subconsciously is a bigger determinant of our perception of music than the conscious stuff we are aware of & this is the great truism that is at the heart of blind testing. We can often be affected by some unrecognisable or unquantifiable aspect of the music & we often hear the phrases - "I was fatigued after a short while listening to this" or it's opposite "I found the music more intelligible, more easy to follow the musical interplay"

Mostly these descriptions get denigrated & blind testing demanded as "proof" 



> Measuring mood is a lot trickier, largely because there are so many things that affect your mood, and listening to music is a relatively minor consideration in it. You can try measuring mood change, assessing it before and after listening to the music, but that still doesn't guard against random effects: even the most beautiful music in the world isn't going to raise your mood if it reminds you of that disastrous date you had last month.


Although I don't know the specific details of how these "mood" issues are being dealt with in the EEG, MEG & other neurological testing, I have a strong belief that good scientific procedures & investigative techniques are being used to minimise the possible influence of mood on the results. 



> And lastly, the sad fact is that this is a commercial process. As we have seen, all the evidence in the world isn't going to stop people frothing at the mouth and claiming the tests were rigged if they produce results counter to their expectations. Manufacturers want tests that show you should buy their more expensive products, and consumers want the same so they can justify their purchase. While the ABX test's reliance on conscious discrimination is a weakness in general scientific terms, it's justifiable in that it was designed to test precisely that sort of claim: that fancy expensive gadgets could be consciously discriminated from nasty cheap ones.


I would agree that good, well administered blind tests are worthwhile but I won't accept home-run blind tests as having any higher validity than normal listening (in fact I would rate it lower in validity). I certainly can't accept those who claim an "epiphany" from such tests that showed them the error of the ways!

But people want a good story to buy into a particular belief system - some stories appeal to some sides while other stories appeal to other sides. The fact of the matter is that we have subconsciously already decided - the story is just a rationalisation of this decision! 



> I certainly think it would be worthwhile to develop some form of masked priming test for musical perception. I very much doubt that it would show any difference between 96/24 and 44.1/16, but if it did I'd certainly have to consider it very carefully. If such results could be reliably repeated I'd have to change my opinion. But I can tell you right now that no matter how sophisticated the test, if it didn't show any effect you would still find audio mythologists rejecting the negative results out of hand and clinging to their faith.


I would like to see the same tests too & I agree with your prediction but I would also say that there would be a number of audio luddites who would also reject positive results - it's the people on the fence that would be swung


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## watchnerd

mmerrill99 said:


> Your points are good & it does raise one issue that I have been making in all my posts here - what goes on subconsciously is a bigger determinant of our perception of music than the conscious stuff we are aware of & this is the great truism that is at the heart of blind testing. We can often be affected by some unrecognisable or unquantifiable aspect of the music & we often hear the phrases - "I was fatigued after a short while listening to this" or it's opposite "I found the music more intelligible, more easy to follow the musical interplay"
> 
> Mostly these descriptions get denigrated & blind testing demanded as "proof"


 
  
 I don't see how these phenomena are in any way contradictory to blind testing.
  
 If it's unquantifiable and hard to describe, but reliably repeatable, it should pass a blind test just fine.


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## KeithEmo

gregorio said:


> No, the quantisation noise is inaudible (unlike tape noise). If you are hearing a difference there are only 4 possibilities for what's causing the "warmer" sound at 192/24:
> 
> 1. You are imagining a difference where none actually exists.
> 2. The 44.1/16 version you are using for comparison has been deliberately butchered in some way to sound different (less warm) than your 192/24 version.
> ...


 
  
 There is actually a fifth, and probably more common, reason.....
  
 Whenever people talk about "comparing two different sample rates", what they're usually really comparing is an original file, at some native sample rate, to a file created by performing a sample rate conversion. The problem there is that, because of both the necessary digital filtering that's part of the process, and probably simple flaws in the algorithms chosen, many converters introduce audible differences during the process - and it is then impossible to separate differences due to the different sample rates (if there are any) from differences caused by the conversion itself. And, even if you were to start with the same original, and then convert it to two different - lower - sample rates using the same software, there still might be audible differences caused by differences in how the program works at different output sample rates. (And, if you don't believe that different converters sound different - there are whole websites dedicated to comparing sample rate conversion programs - with measurements and audio samples - and at least some of them do in fact sound different.
  
 Therefore, it's not at all unlikely that the "less warm" sound being heard at the lower sample rate is different because it was changed as a side effect of the conversion process. And, likewise, it's distinctly possible that many of the "differences between PCM and DSD" that many people hear are really there, but are simply artifacts of the conversion process... regardless of which version is the original, and which way the conversion was done.


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## watchnerd

keithemo said:


> There is actually a fifth, and probably more common, reason.....
> 
> Whenever people talk about "comparing two different sample rates", what they're usually really comparing is an original file, at some native sample rate, to a file created by performing a sample rate conversion. The problem there is that, because of both the necessary digital filtering that's part of the process, and probably simple flaws in the algorithms chosen, many converters introduce audible differences during the process - and it is then impossible to separate differences due to the different sample rates (if there are any) from differences caused by the conversion itself. And, even if you were to start with the same original, and then convert it to two different - lower - sample rates using the same software, there still might be audible differences caused by differences in how the program works at different output sample rates. (And, if you don't believe that different converters sound different - there are whole websites dedicated to comparing sample rate conversion programs - with measurements and audio samples - and at least some of them do in fact sound different.
> 
> Therefore, it's not at all unlikely that the "less warm" sound being heard at the lower sample rate is different because it was changed as a side effect of the conversion process. And, likewise, it's distinctly possible that many of the "differences between PCM and DSD" that many people hear are really there, but are simply artifacts of the conversion process... regardless of which version is the original, and which way the conversion was done.


 
  
 In theory, maybe....
  
 But in practice when something like 320 kps MP3 is indistinguishable from lossless to most listeners, this would have to be a pretty egregious playback / transcode error to mess up lossless vs lossless to such an extent that it's easier to distinguish than high bitrate MP3.


----------



## KeithEmo

rrod said:


> And it would seem there are two ways to take the revelation:
> a) "So I was just making it all up in my head? Guess I learned that lesson; please put the box away now."
> b) "Well the box *did*make things sound better in my mind, so I guess I need to use it all the time from now on."
> 
> Many people on this site would call b) a reasonable response, because all that matters in the end is how subjectively good you feel about your music experience. To some of us, a) is the best way, because we get the same actual sound coming into our ears but without being a slave to the box.


 
  
 Now _THERE_'s a real philosophy question.... as opposed to a question of fact...
  
 What if I were to show you two little black boxes, and tell you that Box A was empty, while Box B contained the latest new patented circuit for making your audio sound wondrously better?
 And you were to listen to them and be thoroughly impressed at how much better your audio sounded after passing through Box B?
 And I were to then show you that both boxes were in fact empty?
 (We're taking the fact that Box B is really empty as a "given" here.)
  
 1) Would you then be willing to accept that the difference was strictly placebo effect?
 2) Would you actually feel deprived because you could no longer enjoy the improvement caused by Box B (since you knew it wasn't real)?
 3) Would you simply continue to enjoy the perceived benefits of Box B and insist that they were caused by some real but intangible "something"?
 4) Would you still be satisfied if you'd PAID me $1000 for the empty box?
 5) Would you be happier if nobody had told you the truth about Box B?
  
 This is like the old question of whether, if you were dying, and only had six months to live, you would prefer to be told the truth or not.....
 After all, you would probably be a lot happier for those last six months if you didn't know you were dying, right?
  
 It is my understanding that everyone participating in this thread is claiming that they want to know the factual truth of the situation.
 (As someone once said: "Reality is what's there - whether you believe it or not".)


----------



## KeithEmo

I'm inclined to believe the opposite - which is that most people don't hear the difference between 320k MP3 and lossless because they don't expect to (often because they simply don't care). The placebo effect can in fact work in both directions; you can imagine a difference that isn't there; or you can imagine that a subtle difference that's really there is not. I suspect that many people often don't hear a difference because they don't expect to hear one (and many people here seem to be ignoring _THAT_ possibility).
  
_NOTE_ that there is actually a quite simple and fair way to test this - which is scientifically and statistically valid. The trick here is that blind testing works so well at testing whether we can determine things because it doesn't rely on our own opinion on the subject. (In fact, a well thought out test doesn't even rely on our knowing that we perceived a difference.)
  
 Simply, if you want to see if someone can really discern a difference; then, before the test, tell them that there is in fact a difference which you expect to be audible to them. Then have them do their very best to discern that difference. If, after the test, you find that their answers correlate to random guessing within statistical limits, then you will have proven that they are really unable to discern a difference. And, if their performance suggests statistically that they did in fact discern a difference, then you will have proven that a difference exists.
  
 By providing an expectation bias in favor of hearing a difference you will have eliminated any bias in favor of _NOT_ hearing a difference. Therefore, you will greatly reduce the possibility of a "null placebo effect". At the same time, since the only way they can produce a result beyond random chance is if there is some real difference, if you get a non-random result, you will have shown that _SOME_ sort of difference exists (conscious or not).
  
 Quote:


watchnerd said:


> In theory, maybe....
> 
> But in practice when something like 320 kps MP3 is indistinguishable from lossless to most listeners, this would have to be a pretty egregious playback / transcode error to mess up lossless vs lossless to such an extent that it's easier to distinguish than high bitrate MP3.


----------



## watchnerd

> I'm inclined to believe the opposite - which is that most people don't hear the difference between 320k MP3 and lossless because they don't expect to (often because they simply don't care). The placebo effect can in fact work in both directions; you can imagine a difference that isn't there; or you can imagine that a subtle difference that's really there is not. I suspect that many people often don't hear a difference because they don't expect to hear one (and many people here seem to be ignoring _THAT_ possibility).
> 
> _NOTE_ that there is actually a quite simple and fair way to test this - which is scientifically and statistically valid. The trick here is that blind testing works so well at testing whether we can determine things because it doesn't rely on our own opinion on the subject. (In fact, a well thought out test doesn't even rely on our knowing that we perceived a difference.)
> 
> ...


 
  
  
 While possible, is there actually a large scale, peer-reviewed study of MP3 vs lossless audibility that people think was rigged?
  
 And, if such a thing does exist, does throwing out that data change the consensus findings, or are there enough other corroborating studies that the consensus remains?
  
 We can all make interesting hypothetical conjectures, but unless there is actual data it's just like shootin' the breeze at the bar about sports or who is hotter, Wonder Woman or Cat Woman.


----------



## nick_charles

mmerrill99 said:


> I would agree that good, well administered blind tests are worthwhile but I won't accept home-run blind tests as having any higher validity than normal listening (in fact I would rate it lower in validity). I certainly can't accept those who claim an "epiphany" from such tests that showed them the error of the ways!


 
  
 If by normal listening you mean the kind of narrative we see here such as


> The Source presents a flatter, up-front perspective with lots of image width, whereas the Yggy offers lots of depth and proportionate width. Yggy layers the performers front to back and separates voices and instruments in a way the Source does not. I prefer the Yggy here."


 
  
 This has greater validity than someone who compares an MP3 file created from a WAV file and converted back to a WAV file to the source WAV file and scores 15/15 using Foobar ABX plugin comparing two samples of the exact same length , perfectly time-aligned and with the same output format and concludes that for that sample the MP3 converted back is audibly different - really ?


----------



## KeithEmo

Darned if I know..... and, to be honest, I don't think it really matters....
  
 When you're discussing the difference between MP3 and lossless, I don't think the question (as we might phrase it here) really matters.
  
 I'd be quite willing to concede that the vast majority of people don't _NOTICE_ any difference. Bear in mind that this is a marketing question and not a technical one. The fact that the vast majority of files that are uploaded, downloaded, sold, bought, and played on portable devices all over the planet are in either AAC or MP3 format (both lossy) "proves" that, at least to most people, there is no _significant_ audible difference. (Note that I didn't say that those people can't hear a difference; the important issue here is that most of them probably haven't tried especially hard. All that matters is that they're perfectly happy with their MP3 files, and will continue to buy them. This may mean that they really can't hear a difference, or that they've never compared them, or even that they do hear an obvious difference, but don't care enough about it to pay the extra $5 for the CD.) And, since this is the sort of information, and the context, that matters to the sales department, they're going to run surveys and studies that lead to this sort of results.... the ones they can use.
  
 However, on the other side of the coin, with a given encoder (back when I cared), I was always able to find at least one or two audio files where I found the difference to be "glaringly obvious" - and I have little doubt that, if I wanted to, I could design a file with a test tone that would deliberately cause any given encoder to screw up in a way that would be audible. (Just as, with JPeG images, even though many look perfect, once you know how the technology works, it's trivial to craft a test image which cannot be encoded to look even passably good.) And, if someone wants to provide me with a specific version of a specific encoder, and finance about a week's work at my standard consulting rate, I'm pretty sure I can deliver you a specifically crafted test file that will sound glaringly obviously bad when encoded with that encoder. (In my experience, any "perceptual encoder" can be tricked into producing a bad output if you really set your mind to it - and I used to design test procedures for a living.)
  
 Now, the question of how many people notice the difference, and whether, perhaps, they would be willing to pay $1 extra (even though they won't pat $5 extra) for the CD-quality version, is of possible importance to the marketing department (and I'll bet the iTunes boys can tell you what percentage of their market notices a difference between 128k iTunes files and 256k ones).
  
 (However, all rhetoric aside, the absolute goal of perceptual encoding is to trick our human minds into failing to see the differences. Therefore, even if it were true that nobody could tell the difference between an MP3 and a lossless file, that _STILL_ wouldn't have serious bearing on whether there is an audible difference between different sample rates. All it would prove would be that "MP3 encoding really works". Perceptual encoding is essentially "slight of hand"; and a really good magician can fool most if not all of us - even though he can't fool a decent TV camera - and most of us realize we're being tricked.)
  
  
 Quote:


watchnerd said:


> While possible, is there actually a large scale, peer-reviewed study of MP3 vs lossless audibility that people think was rigged?
> 
> And, if such a thing does exist, does throwing out that data change the consensus findings, or are there enough other corroborating studies that the consensus remains?
> 
> We can all make interesting hypothetical conjectures, but unless there is actual data it's just like shootin' the breeze at the bar about sports or who is hotter, Wonder Woman or Cat Woman.


----------



## RRod

keithemo said:


> Now _THERE_'s a real philosophy question.... as opposed to a question of fact...
> 
> What if I were to show you two little black boxes, and tell you that Box A was empty, while Box B contained the latest new patented circuit for making your audio sound wondrously better?
> And you were to listen to them and be thoroughly impressed at how much better your audio sounded after passing through Box B?
> ...


 
  
 1) Sure, if we're defining the effect as a perceived improvement (rather than, say, actually curing the cancer)
 2) Perhaps initially, but over time I think I'd be just fine. Same thing that happened when I gave up expensive DACs and amps.
 3) Nah
 4) I'd probably choke you with box B ^_^
 5) "Happier" can only be judged relatively. Sadly I can't both know the truth about box B and not know the truth.
  
 The dying thing is a bad example, as it's a hard endpoint. I can take a few months to get over my box B encounter and have it just be a good anecdote for my remaining years; a bit different than knowing about terminal cancer.


----------



## nick_charles

keithemo said:


> Simply, if you want to see if someone can really discern a difference; then, before the test, tell them that there is in fact a difference which you expect to be audible to them. Then have them do their very best to discern that difference. If, after the test, you find that their answers correlate to random guessing within statistical limits, then you will have proven that they are really unable to discern a difference. And, if their performance suggests statistically that they did in fact discern a difference, then you will have proven that a difference exists.


 
  
 In the late 80s Masters and Clark amp test  60% of the listeners (15/25) were convinced that they could hear differences between amps (they called them believers)  and were able to describe those differences during the sighted portion, in fact even those skeptical beforehand reported differences during the sighted portion.
  
 In the DBT part of the test neither the believers nor skeptics (their words not mine) were able to discriminate reliably across amps but one group of skeptics actually managed one statistically significant positive test between a Pioneer and a Futterman amp (0.046)


----------



## frodeni

charleski said:


> ".. Right now there's a perfectly good explanation for why you think your music sounds better in 24bit: you're fooling yourself."


 
  
 By "you", you cannot speak about me. I never claimed that I have ever heard any improvement going 24bit. Never.
  
 There were no limits to the laws of Newton, prior to E=mc². Please prove otherwise, before going "your arguments are BS" on me.
  
 Seems like you deny basic logic, as the lack of equivalence in hypothesis testing. Maybe you rather care to prove this equivalence of yours, using sound logic?
  
 Once you realize the flaw in your argument, arguing "knowledge" is not that simple anymore. The same goes for this so called "fact" that the world can be described "objectively". Let alone this claim that "objectivity" is the only truth. You might believe that to be the case, but you cannot prove it. What typically happens, is that people prove their belief with their belief, as in circular argumentation.
  
 If you have some real science to point to, as an academic paper, that would help a lot. You obviously throw claims about scientific ABX tests that people fail, so it should be a easy thing then, to give the reference to them. Also, the scientific testing being flawed, should be easy to cite. That will give me a fair chance to backtrack your claim, as I do for all my work. I see no such references.


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > I would agree that good, well administered blind tests are worthwhile but I won't accept home-run blind tests as having any higher validity than normal listening (in fact I would rate it lower in validity). I certainly can't accept those who claim an "epiphany" from such tests that showed them the error of the ways!
> ...



So tell me, was your "epiphany" as a result of doing home-run blind tests? Is this what resulted in your "revelations"?


----------



## watchnerd

keithemo said:


>





> I'd be quite willing to concede that the vast majority of people don't _NOTICE_ any difference. Bear in mind that this is a marketing question and not a technical one. The fact that the vast majority of files that are uploaded, downloaded, sold, bought, and played on portable devices all over the planet are in either AAC or MP3 format (both lossy) "proves" that, at least to most people, there is no _significant_ audible difference.


 
  
 Actually, it doesn't prove anything about what they can audibly discriminate. It just demonstrates where their preferred marginal utility lies on a price curve.


----------



## nick_charles

mmerrill99 said:


> So tell me, was your "epiphany" as a result of doing home-run blind tests? Is this what resulted in your "revelations"?


 
  
 It was a contributory factor, I was also doing a lot of research and I joined the AES and purchased and read dozens of AES papers as well as other research papers from researchers all over the world,  will you answer my question now ?


----------



## KeithEmo

Thank you for the correct term there.....
  
 And, in the current market for high-def downloads, what we're talking about is the marginal utility of the difference that is believed to exist.
  
 Note that I'm not saying "perceived" either, because some people probably think they hear a difference, while others may in fact not even imagine they do. For example, some people who admittedly don't hear any difference whatsoever may believe that, after some future system upgrade, _THEN_ they will hear a difference; other may simply be buying them to impress their friends. 
  
 In fact, marketing studies done on commodities like dish detergent strongly demonstrate that some people will always buy "the premium product" - presumably because, even when no claims or evidence are presented, they simply believe that the "premium" version is "better". Likewise, some people will always buy "the economy version" - because they imagine it's saving them money or somehow otherwise identifies them as "thrifty". (The LABELS sell the various products - the facts are basically incidental.)
  
  
  
 Quote:


watchnerd said:


> Actually, it doesn't prove anything about what they can audibly discriminate. It just demonstrates where their preferred marginal utility lies on a price curve.


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > So tell me, was your "epiphany" as a result of doing home-run blind tests? Is this what resulted in your "revelations"?
> ...



Right, so you never did a professionally organised, blind test & yet you had an "epiphany" - do you not think you might owe it to yourself to do a proper test?

I didn't see any question asked of me!


----------



## nick_charles

mmerrill99 said:


> I didn't see any question asked of me!


 
  
  
 here it is----------------------------------
  
  If by normal listening you mean the kind of narrative we see here such as


> The Source presents a flatter, up-front perspective with lots of image width, whereas the Yggy offers lots of depth and proportionate width. Yggy layers the performers front to back and separates voices and instruments in a way the Source does not. I prefer the Yggy here."


 
  
 ********This has greater validity than someone who compares an MP3 file created from a WAV file and converted back to a WAV file to the source WAV file and scores 15/15 using Foobar ABX plugin comparing two samples of the exact same length , perfectly time-aligned and with the same output format and concludes that for that sample the MP3 converted back is audibly different - really ?***********

  
 ------------------------------------------------------
  
 The clue is in the use of the ?
  
 So do you consider
  


> The Source presents a flatter, up-front perspective with lots of image width, whereas the Yggy offers lots of depth and proportionate width. Yggy layers the performers front to back and separates voices and instruments in a way the Source does not. I prefer the Yggy here."


 
  
 more valid than 

  


> someone who compares an MP3 file created from a WAV file and converted back to a WAV file to the source WAV file and scores 15/15 using Foobar ABX plugin comparing two samples of the exact same length , perfectly time-aligned and with the same output format and concludes that for that sample the MP3 converted back is audibly different


 
  

 ?


----------



## charleski

keithemo said:


> Simply, if you want to see if someone can really discern a difference; then, before the test, tell them that there is in fact a difference which you expect to be audible to them. Then have them do their very best to discern that difference. If, after the test, you find that their answers correlate to random guessing within statistical limits, then you will have proven that they are really unable to discern a difference. And, if their performance suggests statistically that they did in fact discern a difference, then you will have proven that a difference exists.


 
 While I'd agree it would be worthwhile to include a formal assessment of expectation bias in these tests, there are results out there in which subjects clearly came in expecting to hear a difference.
  
Here's the results of Archimago's web-based test which netted 140 respondents and showed no ability to discriminate 16- from 24bit samples, even when broken down into various subgroups and looking only at experienced listeners. In Part III of the test he relates some of the subjective responses he received. While you can find ones such as 'This test confirmed what I'm sure about from a long time: that I cannot hear any difference from 16/44 to 24/96 or more files.' indictaing someone who clearly had a negative bias, there are plenty who displayed positive bias: 'Easier than expected. I found the classical pieces chosen to be very revealing' and 'Much easier than expected. I was expecting to have to play the tracks a number of times to discern any differences, but I played each track through only once and could pick up the differences within seconds', and even 'I would not expect 24 bit to be different to 16bit, personally, but had a clear preference for A, B and B which surprised me' (A,B,B turned out to be the 16bit tracks).


----------



## castleofargh

keithemo said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
 to me this is blue pill vs red pill question. are we willing to risk our enjoyment, to go and search for the truth. and it should be a personal choice.
 would I want to know if my GF was sleeping with another guy? I can just turn a blind eye and never try to find out. but IMO that kind of stuff works only for the totally ignorant. if you're already at the point where you have to ask yourself the question, doubt is in to slowly ruin everything. and only the search for truth will get rid of doubt. a least that's how I am.
  
 and seeds of doubt for the audibility of highres, you can't get enough of those:
 - people failing blind tests would be a good
 - DSD always using the term "analog" to get the old guns who hate digital to jump and in and pay. little do they know that in almost all DSD albums, they're just using oversampled PCM and pulse modulation, the very thing most of those guys hate and will always reject. this is indeed happiness through lies.
 - the staircases graphs we have seen for years to "demonstrate" how more samples would improve the signal. and all the people who believe it and use it to justify why they hear better sound in highres.
 - the graphs about improved noise floor, that forget to show how the device doesn't output music with anything remotely close to that SNR. so that people can still believe it matters and resolution of the file is the limiting factor of their system. when it's not.
 - the rigged tests like tidal where they EQ the low res to make it sound inferior. so now the guy can stop questioning, his brain got the answer, even though it's a total lie.
 - the rigged test like pono where you get superstars to listen to music... in a car... and have highres pitched against mp3. MP3!! not CD. so where is the highres test? and they didn't stop there, they didn't even dare to do that informal listening with 320mp3. they had to use low bitrate to be able to actually make a point and get audible difference.
 - all the "highres" albums that have no content past 22khz...
  
 and those are only the stuff that made me doubt. 
	

	
	
		
		

		
			




  
  


frodeni said:


> By "you", you cannot speak about me. I never claimed that I have ever heard any improvement going 24bit. Never.
> There were no limits to the laws of Newton, prior to E=mc². Please prove otherwise, before going "your arguments are BS" on me.
> 
> Seems like you deny basic logic, as the lack of equivalence in hypothesis testing. Maybe you rather care to prove this equivalence of yours, using sound logic?
> ...


 
 again for science it's perfect to always be skeptic and not take anything for granted. you're right. but for the real world you make no sens.
 should we walk secured to a rope because we don't understand gravity enough and never know when it's going to stop working? I'm sure you would have been great at killing most human discoveries over the centuries.
 should we dismiss statistics altogether because it's not a 100% thing?
 you present things as if less than perfect cannot be used to advance toward a conclusion, this is ludicrous. how many books at school would survive to your scrutiny? how many of your teachers would get fired for not being close enough to the factual truth? how would anybody learn anything?
 the way you talk about blind tests and ABX is only always an appeal to ignorance. only by doing some did I become aware of some of the flaws, everything must start somewhere. you don't wake up knowing stuff or being able to sculpt a statue. you try, you mess up and, you come back knowing a little bit more.


----------



## Don Hills

rrod said:


> ... Sadly I can't both know the truth about box B and not know the truth. ...


 
  
 The truth is that Schrodinger's cat is in box B...


----------



## watchnerd

nick_charles said:


> It was a contributory factor, I was also doing a lot of research and I joined the AES and purchased and read dozens of AES papers as well as other research papers from researchers all over the world,  will you answer my question now ?


 
  
 AES papers are awesome, aren't they?


----------



## Joe Bloggs

frodeni said:


> I have had enough for now. You guys are not going to grow any, and will remain at where you are now, if I check back in a year or five. I will only join in, if someone reasonable joins in. We probably end up having a conversation with 10-15 post in between any one of ours, telling us how stupid we are. That we know no science, even if that is exactly what we are discussing. That is how bad this sub-forum has become.
> 
> Bye now.




And I'm going to hold you to your word--we should not expect to see you here again for another year or five. Yet I see you making several follow up posts. Please stay true to your word and leave this thread--I have no interest in being "educated" by someone who's mind has become so "open" that it's falling... well I'll leave out the rest.

Your test "methodologies" have been repeatedly shown to produce "night and day" perceived differences between test objects that are totally identical. If you see no problem with that and would rather yadda about all the "false negatives" we're missing, I have a 900000USD empty box to sell you. Oh but I swear it's filled with phlebotinized unobtanium that will make your audio sound 1000x better!


----------



## KeithEmo

I agree - although there can be significant practical ramifications to the "blue pill and red pill dilemma".
  
 Assuming you were in pain, and the doctor gave you a pill that made it go away.....
 Would you want to know that it's a sugar pill - at which point it would stop working?
 (Or are you the sort of person who would say "Thanks, now that I know the pain is all in my head I can control it without the pill"?)
 Would someone else want to know?
 Would you tell dear old granny, who thought the sugar pill was taking away her pain, that it was a fake?
 (And the one nobody asks; if it really is all in your mind, then isn't there some way to consciously control it?)
  
 Now, obviously, some of us are more subject to these effects than others.
 We're all actually pretty much equally subject to expectation bias - what really varies is what we expect to begin with.
 If you believe everything you're told, then you expect it to work; if you're naturally skeptical, then you assume everything won't work until proven otherwise.
  
 I would like to say that it's as simple as money (if they're charging you for it, then you should know the truth, so you aren't paying for a sugar pill).
 But there's still a dilemma there.... and that is that we humans (most of us) have a programmed-in association between price and value.
 In other words, for MOST humans, if you pay $500 for the sugar pill, you REALLY have more expectation that it will work than if it costs $1.
 (So, if you pay more for it, it "really" works better. So, arguably, in a modern culture, if you want a placebo pill to work well, you HAVE to charge a lot for it.)
  
 Incidentally, my PERSONAL philosophy in all this is that unquestioning belief is more likely to harm me than help me...
 Like you, I also find myself unable to voluntarily "choose not to know something".
  
 I think there's also another factor involved in your later questions - like the one about statistics....
 And that is that things like statistics seem far less uncertain when you UNDERSTAND what they actually mean.
 (When you flip that coin once, the outcome is uncertain; but, if you flip it 1000 times, the overall results are certain - even though that certainty contains a range of possible answers.)
 I think a lot of the problem is that, for people who don't actually understand the science, it simply looks like "a different brand of magic".
 And, to someone who has that little actual knowledge, one "opinion" that the Earth is flat makes just as much sense as another "opinion" that it's round.
  
 Quote:


castleofargh said:


> to me this is blue pill vs red pill question. are we willing to risk our enjoyment, to go and search for the truth. and it should be a personal choice.
> would I want to know if my GF was sleeping with another guy? I can just turn a blind eye and never try to find out. but IMO that kind of stuff works only for the totally ignorant. if you're already at the point where you have to ask yourself the question, doubt is in to slowly ruin everything. and only the search for truth will get rid of doubt. a least that's how I am.
> 
> and seeds of doubt for the audibility of highres, you can't get enough of those:
> ...


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > I didn't see any question asked of me!
> ...




I didn't see this as a question to me - more of a rhetorical statement from you!
Your question is meaningless but I'm still amazed that you used home-based blind tests as the basis for an "epiphany" without ever checking yourself in a properly conducted, professionally organised, statistically validated blind test. 

I have no doubt that the same applies to most here who "once were magical thinkers" but now know what's "true"


----------



## RRod

mmerrill99 said:


> I didn't see this as a question to me - more of a rhetorical statement from you!
> Your question is meaningless but I'm still amazed that you used home-based blind tests as the basis for an "epiphany" without ever checking yourself in a properly conducted, professionally organised, statistically validated blind test.
> 
> I have no doubt that the same applies to most here who "once were magical thinkers" but now know what's "true"


 
  
 And how does one go about getting one's self into a properly conducted, professionally organized, statistically validated blind test?


----------



## mmerrill99

castleofargh said:


> to me this is blue pill vs red pill question. are .
> 
> again for science it's perfect to always be skeptic and not take anything for granted. you're right. but for the real world you make no sens.
> should we walk secured to a rope because we don't understand gravity enough and never know when it's going to stop working? I'm sure you would have been great at killing most human discoveries over the centuries.
> ...




The highlighted text is where your logic is off - what if the books are not just less than perfect, what if they are completely wrong - does this advance knowledge?
How do you judge the accuracy of the bABX tests that many here claim gave them "epiphanies"? How do you know the strength of the test - how do you know if it's capable of testing the subtle perceptual differences that are in question? How do you know that the test is "just less than perfect" & not randomly wrong? Do you have any basis on which to say a home run ABX test is "less than perfect" as opposed to "completely useless" when it comes to differentiating subtle differences of perception?


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > I didn't see this as a question to me - more of a rhetorical statement from you!
> ...



That's not my problem - I'm not the one claiming the validity of the test that gave me epiphanies. 
The problem, as I see it, is that people rely on such flawed tests & claim "epiphanies"


----------



## RRod

mmerrill99 said:


> That's not my problem.
> The problem, as I see it, is that people rely on such flawed tests & claim "epiphanies"


 
  
 The question you were asked was how one method of epiphany was "worse" than an other, when one actually makes some attempt at control of external influences. And so you punt by saying the only acceptable criteria for making any decision is to be in a proctored study. Ok, fine, then how about you also say that all these sighted "I can hear better detail and separation" posts are also completely bunk?


----------



## nick_charles

mmerrill99 said:


> Right, so you never did a professionally organised, blind test & yet you had an "epiphany" - do you not think you might owe it to yourself to do a proper test?


 
  
 I have done proper tests thanks, you might not consider them proper but that is hardly my problem. When I do DBT I use the same setup (currently PC/External DAC/Headphone amp/headphones) , the same tracks, the same platform Windows 7 - Foobar and the same protocol. I load the tracks into Foobar select the ABX plugin and do a set of trials. The tracks are in the same kind of container - generally wav or flac with the same bit-depth/sample rate if appropriate. I generally do 15 trials unless it is obvious that I cannot make a judgment several times running and am just guessing in which case I bail. During trials I try different small segments repeatedly. Tests are always done in one session. Please tell me in detail what level of control is missing that renders such tests invalid a priori.
  
 I have had a mix of being able to discriminate (13/15 or better)  , and failure to discriminate - I use the P < 0.05 criterion which is admittedly pretty generous.
  
*Things I have been able to discriminate *
 Outputs recorded from different CD players (not level matched)  
 Red Book vs High-res one sample but with a giveaway artifact 
 MP3 (VBR0)  vs wav (hot track "Nobody wants to" with marginal clipping before MP3 encoding)
 Low pass filter at 13K vs not on wav file "Walking with a Mountain"
  
  
*Things I have not been able to discriminate *
 Outputs from different CD players (level matched)
 AD captures of tracks recorded using different Analog cables (Copper, Solid Silver, Silver coated) 
 High res vs 16/44.1
 Low pass filter at 14K vs not on wav file "Walking with a Mountain"
 MP3 vs wav on several tracks
  
 These are just the ones I remember off hand. What do I conclude from these tests? Nothing much beyond a general sense of my capabilities. But as I did say my personal tests are just one element in my arriving at my current thinking. That I cannot discriminate between A and B does not of course extend beyond me. I've seen plenty of credible positive DBTs appearing in press and some questionable ones as well. Probably the most interesting being Blech and Yang's PCM vs DSD test which is worth buying from the AES - a small sample of their listeners did positively discriminate between the two encoding methods. On a broader question If I recall correctly you were defending amirm's home tests (jangling keys) elsewhere, you seem happy to accept the results of his home tests as valid but not those done by others  - why is that ?


----------



## nick_charles

mmerrill99 said:


> I didn't see this as a question to me - more of a rhetorical statement from you!
> Your question is meaningless but I'm still amazed that you used home-based blind tests as the basis for an "epiphany" without ever checking yourself in a properly conducted, professionally organised, statistically validated blind test.
> 
> I have no doubt that the same applies to most here who "once were magical thinkers" but now know what's "true"


 
  
  
 No it was very definitely a question and I very much want an answer as I have answered your questions you should have the courtesy and intellectual honesty to do the same.
  
 No, it is not a meaningless question. I am explicitly asking you if you give more credence to sighted evaluations than to what you call home-based blind tests - so do you ?


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > Right, so you never did a professionally organised, blind test
> ...


Ye the validity of your tests are your problem when you make claims of "epiphanies" based on the results of such tests 





> When I do DBT I use the same setup (currently PC/External DAC/Headphone amp/headphones) , the same tracks, the same platform Windows 7 - Foobar and the same protocol. I load the tracks into Foobar select the ABX plugin and do a set of trials. The tracks are in the same kind of container - generally wav or flac with the same bit-depth/sample rate if appropriate. I generally do 15 trials unless it is obvious that I cannot make a judgment several times running and am just guessing in which case I bail. During trials I try different small segments repeatedly.
> 
> I have had a mix of being able to discriminate (13/15 or better)  , and failure to discriminate - I use the P < 0.05 criterion which is admittedly pretty generous.
> 
> ...


Well, no you conclude two very important points - one that the test is providing you with accurate results - two that such results led you to an epiphany. Both of which I would consider important 





> But as I did say my personal tests are just one element in my arriving at my current thinking. That I cannot discriminate between A and B does not of course extend beyond me. I've seen plenty of credible positive DBTs appearing in press and some questionable ones as well. Probably the most interesting being Blech and Yang's PCM vs DSD test which is worth buying from the AES - a small sample of their listeners did positively discriminate between the two encoding methods.





> On a broader question If I recall correctly you were defending amirm's home tests (jangling keys) elsewhere, you seem happy to accept the results of his home tests as valid but not those done by others  - why is that ?


I was questioning the attitude that pertains to home ABX testing - the negative results are unquestioned (any number of false negatives are in these results) the focus is on eliminating false positives - that is the way the test is designed - flawed in my opinion. So when it is used inappropriately in home-based testing we get "epiphanies" - why? Because a test gave you a null result? All supporters of such tests claim that null results are of no importance - yet you (I don't mean you personally - it's the general you I'm talking about) have epiphanies. I find this double talk to be politician's speak.

I pointed out two examples of home-run positive ABX results ultmusicsnob's results posted on this & other forums & Amir's. Both of which were analysed & scrutinised by various forum members & follow up tests & cross-checks done. I used ultmusicsnob's descriptions of the effort needed to overcome the major barriers that are in-built in the test towards a null result. His efforts showed just what went into the training/attention & motivation needed. So, yes, when positive results from home-run ABX tests are presented & the scrutiny & analysis performed on them (as it usually is) & nothing wrong discovered, then there is some worth to these results.

I don't find that these results mean that only a few people can hear differences, I interpret the results as meaning that the test is so skewed towards a null (in so many ways) that only a few people have the dedication/motivation/training to overcome such odds stacked against them.

In other words the test has no sensitivity & only gross differences or very well trained/dedicated people will give a positive result. The fact that there is a great reluctance to analyse or even contemplate such a lack of sensitivity suggests that it attracts people of a certain mindset who see it as a useful tool to support their thinking.

The fact that it is usually presented to unwitting members as such a simple test, makes me even more convinced that it is a tool to support a mindset & not being used as a tool to get at "truth", as is so often the spiel. Added to that the usual pseudo-science schtick about DBTs, gold-standard & science, it begins to grate on any thinking person's sensibilities


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > I didn't see this as a question to me - more of a rhetorical statement from you!
> ...




I treat all impressions of audible differences that I read as anecdotes - some might be worth personally investigating & some not.
For my view on ABX home tests see my other reply


----------



## nick_charles

mmerrill99 said:


> Ye the validity of your tests are your problem when you make claims of "epiphanies" based on the results of such tests





> *As I said a contributory factor - please do not misquote me and I will endeavour to be accurate when quoting you *


 
  
 and my other question ?


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > That's not my problem.
> ...



Answered in my replies to ol' Nick. 
I believe an "epiphany" has the quality of changing one's whole view of an area (a semi-religous experience), not the same thing as the usual reports seen from sighted listening - "I like the sound of this device - here's what I hear when using it" - this ain't no epiphany, is it?


----------



## RRod

mmerrill99 said:


> Answered in my replies to ol' Nick.
> I believe an "epiphany" has the quality of changing one's whole view of an area (a semi-religous experience), not the same thing as the usual reports seen from sighted listening - "I like the sound of this device - here's what I hear when using it" - this ain't no epiphany, is it?


 
  
 I'm sorry but there are plenty of instances on this site of people saying "wow I was literally blown away by the change this op-amp made; this has made me rethink my entire setup." Let's leave out the no-true-scotsman's-epiphany argumentation.


----------



## mmerrill99

nick_charles said:


> > *As I said a contributory factor - please do not misquote me and I will endeavour to be accurate when quoting you *
> 
> 
> 
> ...



Huh? I didn't see another question!


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > Answered in my replies to ol' Nick.
> ...



Is the hyperbole of one equivalent to the life-changing, semi-religious hyperbole of the other?
On the one hand you have someone saying this is a huge improvement in my audio enjoyment - on the other side we have someone saying I have had an epiphany & this is the "truth" - I was once a "magic thinker"

Did you have an "epiphany" with blind testing, by any chance?


----------



## sonitus mirus

We have been around and around about this topic and I am confident nobody will budge. 
  
 My personal ABX testing has shown me that I am unable to easily hear differences where I once thought that the differences were quite obvious.  That is enough for me to place these results as an epiphany.  This does not equate to any truth, it is simply my own realization to this point.


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## RRod

mmerrill99 said:


> Is the hyperbole of one equivalent to the life-changing, semi-religious hyperbole of the other?
> Did you have an "epiphany" with blind testing?


 
  
 Not in vacuo, it also involved tons of reading on DSP, PCM encoding, lossy codecs (to learn how to spot artifacts), etc. You know, all that stuff that most human beings will NEVER do in their lives, even when they listen to lots of music. If we're talking about the ultimate decision on whether the human mechanism is capable of detecting any artifact between two systems, then I agree a large amount of rigor is necessary and both Type I and Type II error rates should be controlled. That doesn't mean we can't learn and test as individuals.


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > Is the hyperbole of one equivalent to the life-changing, semi-religious hyperbole of the other?
> ...



But no example of any reading on auditory perception! If you are doing a test that purports to test your auditory perception discrimination, would it not be wise to know something about the usefulness/appropriateness of such a test to it's intended function & it's sensitivity in this area? How auditory perception might be tested correctly & not reach for DBT because it sounds scientific ?

This is the great hole in the logic - a test of dubious quality being used to make decisions about what what we hear when listening normally. There are many weaknesses to ABX testing, all of which are only being admitted to by some members here when I & others bring them to attention - yet the test is clutched like a sort of comfort blanket for those that need assurance in their choices


----------



## RRod

mmerrill99 said:


> But no example of any reading on auditory perception! If you are doing a test that purports to test your auditory perception discrimination, would it not be wise to know something about the usefulness/appropriateness of such a test to it's intended function & it's sensitivity in this area? How auditory perception might be tested correctly & not reach for DBT because it sounds scientific ?
> 
> This is the great hole in the logic - a test of dubious quality being used to make decisions about what what we hear when listening normally. There are many weaknesses to ABX testing, all of which are only being admitted to by some members here when I & others bring them to attention - yet the test is clutched like a sort of comfort blanket for those that need assurance in their choices


 
  
 So you're assuming the "etc." didn't include anything on the subject. Nice. What other topics that I didn't list do you think are important to your argument at this particular time?


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > But no example of any reading on auditory perception! If you are doing a test that purports to test your auditory perception discrimination, would it not be wise to know something about the usefulness/appropriateness of such a test to it's intended function
> ...


The fact that neither you, nor Nick mentioned anything about auditory perception which I assumed would be foremost in your reading to evaluate the suitability of a test that purports to be a test of auditory perception discrimination. But, if you have done these readings, please let us know the titles.


----------



## RRod

mmerrill99 said:


> The fact that neither you, nor Nick mentioned anything about auditory perception which I assumed would be foremost in your reading to evaluate the suitability of a test that purports to be a test of auditory perception discrimination. But, if you have done these readings, please let us know the titles.


 
  
 As soon as you give us an exact regimen you think people need to undergo before they can legitimately compare audio formats or equipment. And not just "read UltMusicSnob's posts", an actual set of criteria you consider sufficient. Then you can haggle me about which exact books and online content I've read on how hearing works (funnily enough, not 20 minutes before these posts I ordered a new book pertaining to the subject). I will refrain from posting to you further until you have provided said regimen. ttfn.


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > Is the hyperbole of one equivalent to the life-changing, semi-religious hyperbole of the other?
> ...



But anyway, I see you state this"*If we're talking about the ultimate decision on whether the human mechanism is capable of detecting any artifact between two systems, then I agree a large amount of rigor is necessary and both Type I and Type II error rates should be controlled.*"
So do you mean by this that only for "ultimate testing" are Type I & Type II error rates needed & it's OK to do without for home tests? If so can you give us your view on how valid home tests are & how much do you think people are fooling themselves by doing such flawed home run blind tests?


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## Joe Bloggs

mmerrill99

In my book there are two types of audio tweaks:

1. Those where people argue about actual efficacy ad nauseum via ABX tests and denial of ABX test results (I leave it to the reader to point out which exact tweaks those are)

2. Those where the actual efficacy is out of question, in terms of actually making a difference. You know, changing speakers, headphones, physical mods, equalization, DSP... (but within which whole classes of tweaks have been dismissed out of hand by audiophiles as not making any *good* difference)

Personally, the reason why I am dismissive of tweaks that fall in class (1) is not because of any negative DBT test results of my own (I know how hard it is to carry out a properly controlled DBT test and so haven't really ever tried myself) but because of the vast differences and improvements I have obtained from class (2) tweaks alone, and knowing that I have a long, long way to go before I am even close to exhausting the possibilities in (2).

You know, stuff like this pro has done

http://www.head-fi.org/t/796791/the-most-reliable-easiest-way-to-eq-headphones-properly-to-achieve-the-most-ideal-sound-for-non-professionals/15#post_12302915
http://www.head-fi.org/t/796791/the-most-reliable-easiest-way-to-eq-headphones-properly-to-achieve-the-most-ideal-sound-for-non-professionals/15#post_12303368

One would think that you, being so in tune with sonic differences in components that we hardly acknowledge actually exist, must have an audio system that outperforms anything us plebians can ever hope to achieve. I on the other hand believe in quite the opposite: that if one were sat in front of Lunatique's system and then yours, the latter audition would last but mere seconds before one leaves the seat shaking his head. Because if one were to compare audio system design to forestry, an "audiophile" like you has simply never seen the forest for the trees.

In short: if you really knew how flawed the usual run-of-the-mill headphone listening system is from a psychoacoustic point of view (you know, that subject you've been pounding on regarding ABX testing for some reason), you would simply never have bothered to argue about the validity of blind testing [whatever it is you argue about] in the first place. :rolleyes:


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > The fact that neither you, nor Nick mentioned anything about auditory perception which I assumed would be foremost in your reading to evaluate the suitability of a test that purports to be a test of auditory perception discrimination. But, if you have done these readings, please let us know the titles.
> ...



I see
The guidelines are laid out in BS.1116 "Methods for the subjective assessment of small impairments in audio systems " but guidelines aren't fully prescriptive in how to do the tests & it takes someone with experience in this area to administer such tests to avoid unnecessary biasing of the results. An individual can do this as long as he has the motivation & goes to the bother that ultramusicsnob demonstrated in his description of what he did, how he did it & what he heard - none of which is contained in BS.1116. The only other way to have some view on whether a a ABX test was of any value (or just another person fooling themselves) is to have Type II error statistics built into the test procedure. MUSHRA includes such hidden anchors & references "MUltiple Stimuli with Hidden Reference and Anchor" that allows testing of the test. Without this we get ridiculous ABX tests like Arny Kreuger's test where he didn't even listen to the samples A or B, just hit buttons - if you don't know whether the test results are from a real test or a "fake" test then you are in no-man's-land as far as the validity of the results


----------



## RRod

mmerrill99 said:


> I see
> The guidelines are laid out in BS.1116 "Methods for the subjective assessment of small impairments in audio systems " but guidelines aren't fully prescriptive in how to do the tests & it takes someone with experience in this area to administer such tests to avoid unnecessary biasing of the results. An individual can do this as long as he has the motivation & goes to the bother that ultramusicsnob demonstrated in his description of what he did, how he did it & what he heard - none of which is contained in BS.1116. The only other way to have some view on whether a a ABX test was of any value (or just another person fooling themselves) is to have Type II error statistics built into the test procedure. MUSHRA includes such hidden anchors & references "MUltiple Stimuli with Hidden Reference and Anchor" that allows testing of the test. Without this we get ridiculous ABX tests like Arny Kreuger's test where he didn't even listen to the samples A or B, just hit buttons - if you don't know whether the test results are from a real test or a "fake" test then you are in no-man's-land as far as the validity of the results


 
  
 Since you are attempting to answer the question I shall comment:
  
 I am looking precisely for a prescription. And unless you are one of these people who has "experience in this area to administer such tests to avoid unnecessary biasing of results", then why should I take your view of UMS's supposed training with anything but the grain of salt you seem to give the experimentation the rest of us have done? You are holding his training up as an example but have not in any way coalesced his method into anything resembling a training regimen. To me it just seems like you're saying "he ostensibly passed so he must be doing it right, and anyone who doesn't pass is doing it wrong." Well that's not an argument I find particularly compelling. Note also that he wasn't part of an actual controlled study. And all of this is still of no use to the person sitting at his computer deciding if he needs a hi-res master of a song, because he isn't going to randomly land in a controlled study any time soon. He wants to know right *now* if its worth his money. What does he do? According to you, he throws his hands up. Neither a compelling nor useful narrative.
  
 I shall await actual details on the training you would specify for any of the common audio comparisons (format vs format, equipment vs equipment). Note this could be taken from a study already performed.


----------



## L8MDL

Re: "He wants to know right *now* if its worth his money."

From the perspective of an old guy, Fleetwood Mac's "Tusk" cost $15.98 in 1979, a cost we all balked at, even for a double album. It would cost $52.15 today, so I figure every hires download is a bargain. I purchase the highest resolution available.


----------



## RRod

l8mdl said:


> Re: "He wants to know right *now* if its worth his money."
> 
> From the perspective of an old guy, Fleetwood Mac's "Tusk" cost $15.98 in 1979, a cost we all balked at, even for a double album. It would cost $52.15 today, so I figure every hires download is a bargain. I purchase the highest resolution available.


 
  
 Well it's not just high-res, it's the whole gamut of audiophile accoutrements. And of course blind testing everything you might ever want to use isn't particularly feasible. But the point here is appreciating the effect of removing sighted perceptions, but some are saying that this is only worthwhile in the strictest of testing regimens. Otherwise just go with the feels.


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > I see
> ...


Well, it's typical to try to win an argument by asking for something which can't be delivered, as you well know. As I said, the guidelines are contained in that document - have you read it - what parts of it do you not comprehend? 

I used ultmusicsnob's posts to highlight & show the importance of certain aspects of elements mentioned in that doc - the necessity for training & what this actually means - not just doing a Phillips golden ears test but teasing out specific aspects in the sound during listening where specific differences can be consciously & definitely identified - not the usual stuff we see posted here "I thought I could hear a slight difference between devices or files but doing a blind test proved I couldn't" - no, it proved you don't know how to do a blind test & what is required to overcome the skew of blind tests towards a null result. 

The takeaway from UMS's posts is his dedication to finding these conscious, defined differences that he trains with until he gets a significant correct - only then does he do a blind test & his percentage correct drops. Not only is such focussed training required but the difficulty in finding the difference is spelled out - it was different for differentiating jitter as it was in differentiating between RB & high res - different "tells" needed to be found to be able to routinely differentiate between these audio samples. The tendency to drift into normal listening is also highlighted in his posts & the difficulty of retaining focus & taking breaks. It all shows a practical demonstration of a real world ABX positive test. 



> And unless you are one of these people who has "experience in this area to administer such tests to avoid unnecessary biasing of results", then why should I take your view of UMS's supposed training with anything but the grain of salt you seem to give the experimentation the rest of us have done? You are holding his training up as an example but have not in any way coalesced his method into anything resembling a training regimen.


This is rubbish. I have not seen any of you describe your ABX testing in similar detail to USMs - so if you have provided such detail then link me to it. I'm judging what I read & so far I haven't read anything from any of you giving any direction about what to do for improving the validity of a home-run ABX test. Maybe I missed these directions - got a link? 





> To me it just seems like you're saying "he ostensibly passed so he must be doing it right, and anyone who doesn't pass is doing it wrong." Well that's not an argument I find particularly compelling.


Well, as usual, you have twisted this around to suit your argument - his attention to the test procedure details I mentioned (which are also given as guidelines in BS.1116) demonstrate what is needed to do a valid ABX test. If you have other information that makes his tests invalid then please post them. 





> Note also that he wasn't part of an actual controlled study. And all of this is still of no use to the person sitting at his computer deciding if he needs a hi-res master of a song, because he isn't going to randomly land in a controlled study any time soon. He wants to know right *now* if its worth his money. What does he do? According to you, he throws his hands up. Neither a compelling nor useful narrative.


I can tell him what he shouldn't do & that is take a blind test which doesn't have the same preparation as UMS's. If he does go ahead & take one of the haphazard blind tests which get lots of support here & then announces his epiphany, I consider him a duped fool - led to this by other similarly duped fools or by those who know what they are doing



> I shall await actual details on the training you would specify for any of the common audio comparisons (format vs format, equipment vs equipment). Note this could be taken from a study already performed.


A strawman, if ever I saw one!


----------



## watchnerd

keithemo said:


> I think a lot of the problem is that, for people who don't actually understand the science, it simply looks like "a different brand of magic".
> And, to someone who has that little actual knowledge, one "opinion" that the Earth is flat makes just as much sense as another "opinion" that it's round.


 
  
 +1.


----------



## mmerrill99

rrod said:


> l8mdl said:
> 
> 
> > Re: "He wants to know right *now* if its worth his money."
> ...



No, as usual, you twist it or don't understand the subtleties of what I'm saying - I'm saying that enter a blind test without sufficient preparation & attention to the detail in doing such a test & you are guaranteed to get a null result i.e no better than guesswork.It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that. I accuse anybody who tries to suggest such a test on the basis that removal of sighted bias is the only thing that matters is either taking the person for a fool or is a fool themselves.

Yes, I would suggest that how one feels about the music over long term listening would be a crucial factor in evaluating it.


----------



## mmerrill99

watchnerd said:


> keithemo said:
> 
> 
> > I think a lot of the problem is that, for people who don't actually understand the science, it simply looks like "a different brand of magic".
> ...



If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"


----------



## RRod

mmerrill99 said:


> Well, it's typical to try to win an argument by asking for something which can't be delivered, as you well know. As I said, the guidelines are contained in that document - have you read it - what parts of it do you not comprehend? I used ultmusicsnob's posts to highlight & show the importance of certain aspects of elements mentioned in that doc - the necessity for training & what this actually means - not just doing a Phillips golden ears test but teasing out specific aspects in the sound during listening where specific differences can be consciously & definitely identified - not the usual stuff we see posted here "I thought I could hear a slight difference between devices or files but doing a blind test proved I couldn't" - no, it proved you don't know how t do a blind test & what is required to overcome the skew of blind tests towards a null result. the takeaway from UMS's posts is his dedication to finding these conscious, defined differences that he trains with until he gets a significant correct - only then does he do a blind test & his percentage correct drops. Not only is such focussed training required but the difficulty in finding the difference is spelled out - it was different for differentiating jitter as it was in differentiating between RB & high res - different "tells" needed to be found to be able to routinely differentiate between these audio samples. The tendency to drift into normal listening is also highlighted in his posts & the difficulty of retaining focus & taking breaks. It all shows a practical demonstration of a real world ABX positive test.
> This is rubbish. I have not seen any of you describe your ABX testing in a similar detail to USMs - so if you have such detail then do so. I'm judging what I read & so far I haven't read anything from any of you giving any direction about what to do for improving the validity of a home-run ABX test. Maybe I missed these directions - got a link?
> Well, as usual, you have twisted this around to suit your argument - his attention to the test procedure details I mentioned (which are also given as guidelines in BS.1116) demonstrate what is needed to do a valid ABX test. If you have other information that makes his tests invalid then please post them.
> I can tell him what he shouldn;t do & that is take a blind test which doesn't have the same preparation as UMS's. If he does go ahead & take one of the haphazard blind tests which get lots of support here & then announces his epiphany, I consider him a duped fool - led to this by other similarly duped fools or by those who know what they are doing
> A strawman, if ever I saw one!


 
  
 I have read it, and I see nothing that says "here's what you should do if you want to do a test of jitter" You have repeatedly held up UMS's posts as a paragon of proper training, but his descriptions are nowhere near being sufficient as a guideline for training. Yet you want us to be beholden to it as a standard for personal ABX testing (which you reject anyway, except in this one case evidently). You seem to find it unthinkable that any of the rest of us have spent hours trying to eek out artifacts, simply because we didn't bother to chronicle every feeling we had along the way as UMS seemed to do. Here's an example of what he listens for, supposedly:
 "I'm listening in the mids, if that helps any. The notes are "shaped" differently not in the bass extension or treble extension, but in the core of the piano attack, where it seems that the 'n' is focused, while the jr has a slightly 'flattened out' aspect. This was listening for the quiet chords right near the end again."
 Sorry but that is not a prescription for anything. Perhaps you think it is. I could say similar things about artifacts from castanets or harpsichords of my tests of Opus, but it's not a training regimen. I also like that you also invoke BS.1116 so readily but don't bother with how he verified the technical requirements from the document. Whose twisting things to suit their argument, eh? And nothing I asked for is a strawman: you have repeatedly made claims that adequate training is necessary.
  
 Going to have to be tomato-tomatoh at this point; only so long I want to be stuck in traffic.
  


mmerrill99 said:


> No, as usual, you twist it or don't understand the subtleties of what I'm saying - I'm saying that enter a blind test without sufficient preparation & attention to the detail in doing such a test & you are guaranteed to get a null result i.e no better than guesswork.It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that. I accuse anybody who tries to suggest such a test on the basis that removal of sighted bias is the only thing that matters is either taking the person for a fool or is a fool themselves.
> 
> Yes, I would suggest that how one feels about the music over long term listening would be a crucial factor in evaluating it.


  

 Bookman, is that you‽


----------



## sonitus mirus

If I believe I'm hearing a slight difference when making a sighted evaluation, any difference is completely eradicated somehow when attempting to do so when blinded.  That is the logic that is being used to counter any ABX that is not carefully proctored by a select few professionals.
  
 Did I misunderstand?


----------



## watchnerd

mmerrill99 said:


> If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"


 
  
 That's a pretty interesting statement, given I have a degree in applied physics, with a focus on signal processing, am an AES member, and (as a hobby) volunteer as a recording engineer for a local symphony.
  
 And I believe Emo is an electronics engineer who works for Emotiva.


----------



## nick_charles

mmerrill99 said:


> ...It's the nature of how auditory processing works





>





> *I have two degrees in Psychology, I took courses on perception, neurophysiology, neuropsychology and psychophysics. I've taken part in several perceptual testing experiments both as subject and as experimenter.*
> 
> 
> 
> ...


----------



## watchnerd

joe bloggs said:


> @mmerrill99
> 
> In my book there are two types of audio tweaks:
> 
> ...


 
  
 Exactly.
  
 Even if Category 1 are occasionally true to some small degree, the ROI in both time and money compared to optimizing Category 2, especially given the flawed nature of transducers and the complexities of acoustics, make it the pragmatic choice of the wise.
  
 Also, better recordings -- crap recordings still sound like crap on good gear (or sound even crappier), while good recordings are the opposite.


----------



## mmerrill99

rrod said:


> mmerrill99 said:
> 
> 
> > Well, it's typical to try to win an argument by asking for something which can't be delivered, as you well know. As I said, the guidelines are contained in that document - have you read it - what parts of it do you not comprehend? I used ultmusicsnob's posts to highlight t do
> ...


Exactly, it doesn't prescribe specifics - that's why it's called a GUIDELINE. It's intended as a guideline for those who genuinely want to try to ensure their test deals with most of the issues that have been identified as biasing factors in such perceptual testing. Of course it can't help those who have ulterior motives. 





> You have repeatedly held up UMS's posts as a paragon of proper training, but his descriptions are nowhere near being sufficient as a guideline for training.


By your needy request for exact descriptions of how to do something, you seem incapable of taking guidelines, using your intelligence & experience & translating them into a procedure for doing a specific test which adheres to these guidelines. Your statement is symptomatic of this need *"I see nothing that says "here's what you should do if you want to do a test of jitter" * Perhaps you are just trying to be awkward? 





> Yet you want us to be beholden to it as a standard for personal ABX testing (which you reject anyway, except in this one case evidently). You seem to find it unthinkable that any of the rest of us have spent hours trying to eek out artifacts, simply because we didn't bother to chronicle every feeling we had along the way as UMS seemed to do.


You twist & squirm every which way. I don't find it unthinkable but I have not seen you give any details of your ABX testing procedure which might be useful to somebody intending to do such a test. Neither have I seen you give any directions to anybody on the pitfalls of ABX testing & how they might best be avoided 





> Here's an example of what he listens for, supposedly:
> "I'm listening in the mids, if that helps any. The notes are "shaped" differently not in the bass extension or treble extension, but in the core of the piano attack, where it seems that the 'n' is focused, while the jr has a slightly 'flattened out' aspect. This was listening for the quiet chords right near the end again."
> Sorry but that is not a prescription for anything. Perhaps you think it is.


Firstly, for those who are not familiar with UMS's ABX test results - he repeatedly got positive ABX results which he posted (other members were of the opinion that he shouldn't be able to do this) & commented on how he did it & what he listened for - as I said a great example for all to read just what are the required preparations for valid ABX blind tests. The particular description being quoted here is of jitter files "n" & "jr" which contain different levels of injected jitter.

So, now to his description - what I take from it is that he hears a difference in attack of the mid frequency piano notes & goes on to describe one as "focussed" & another as "flattened out" - a difference in the shaping of the notes. I would interpret this to mean that the attack of the piano notes in "jr" is more diffuse - probably a temporal smearing? This gives me enough information of what to listen for, where to listen for it "the quiet chords near the end" & what frequency range I'm listening for - mid frequencies. What is it about this explanation that you find "not prescription for anything"? Again your neediness for a exact prescription of something 





> I could say similar things about artifacts from castanets or harpsichords of my tests of Opus, but it's not a training regimen. I also like that you also invoke BS.1116 so readily but don't bother with how he verified the technical requirements from the document. Whose twisting things to suit their argument, eh? And nothing I asked for is a strawman: you have repeatedly made claims that adequate training is necessary.


"I await actual details on training" is a strawman & obviously so to anyone reading this. It again displays your need for someone to tell you an exact prescription of training . If you are so needy you can find many examples of training in various blind tests but they are all different & depend on what is being tested - there's pre-screening, training & post-screening. If you know as much about blind testing as you are trying to portray then none of these questions should be needed 



> Going to have to be tomato-tomatoh at this point; only so long I want to be stuck in traffic.


Run, Forrest, run - Sorry, I couldn't help myself


----------



## mmerrill99

sonitus mirus said:


> If I believe I'm hearing a slight difference when making a sighted evaluation, any difference is completely eradicated somehow when attempting to do so when blinded.  That is the logic that is being used to counter any ABX that is not carefully proctored by a select few professionals.
> 
> Did I misunderstand?



Right so you don't understand that blind test is essentially a statistical test & that to achieve a better than random result requires more than what you simplistically stated.

So, yes you misunderstand


----------



## nick_charles

mmerrill99 said:


> This is rubbish. I have not seen any of you describe your ABX testing in similar detail to USMs


 
  
 When I do DBT I use the same setup (currently PC/External DAC/Headphone amp/headphones) , the same tracks, the same platform Windows 7 - Foobar and the same protocol. I load the tracks into Foobar select the ABX plugin and do a set of trials. The tracks are in the same kind of container - generally wav or flac with the same bit-depth/sample rate if appropriate. I generally do 15 trials unless it is obvious that I cannot make a judgment several times running and am just guessing in which case I bail. During trials I try different small segments repeatedly. Tests are always done in one session. Please tell me in detail what level of control is missing that renders such tests invalid a priori.


----------



## mmerrill99

watchnerd said:


> mmerrill99 said:
> 
> 
> > If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"
> ...




So you understand & know all the science that underpins perceptual testing? Please, both of you, be more prolific in your posts on this subject as I would love to learn some more about this field.


----------



## mmerrill99

nick_charles said:


> > *I have two degrees in Psychology, I took courses on perception, neurophysiology, neuropsychology and psychophysics. I've taken part in several perceptual testing experiments both as subject and as experimenter.*
> >
> >
> >
> > ...




Hmm, I believe you previously made this claim "*please do not misquote me and I will endeavour to be accurate when quoting you"*

So let's see my full quote "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that."

With your claimed expertise, is there something that you disagree with?

You mention that you have *"taken part in several perceptual testing experiments both as subject and as experimenter" *

Care to tell us anything about them? Did you follow any professional guidelines, procedures?


----------



## nick_charles

mmerrill99 said:


> "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that."


 
  
  Okay, full quote above, now what are your credential for making this statement ?


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music)
> ...




Nope you still misquote me - the full quote is "my full quote "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that."

Do I need credentials to make a statement? As I already asked - is there something specific you disagree with or is this just your pissing contest?


----------



## nick_charles

mmerrill99 said:


> "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that."
> 
> With your claimed expertise, is there something that you disagree with?


 
  
 Yes, its wrong, that is what I disagree with 
  
 I have myself detected small differences in stumulae and our knowledge of psychophysics has established thresholds of detection for a large number of different things as you would know , some thresholds are bigger than others and others are quite small - distortion in music can be detected down to about -45db - volume differences to about 0.2db , phase shift is not so sensitive, pitch and speed stability well just listen to piano music on a Rega turntable, crosstalk quite insensitive, channel delay a few miliseconds and so on


----------



## nick_charles

mmerrill99 said:


> Nope you still misquote me - the full quote is "my full quote "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music) & make itself known to our conscious mind - it just doesn't happen like that."
> 
> Do I need credentials to make a statement? As I already asked - is there something specific you disagree with or is this just your pissing contest?


 
  
 You are making a statement of fact. Where does this come from?. Is it from personal expertise or is it from some reputable source or did you just make it up?.
  
 If my undergrads made such statements I would certainly ask for at least citations


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > "It's the nature of how auditory processing works - differences (unless gross) don't jump out of a mix of fast changing audio signals (music)
> ...



I actually don't see what you are specifically saying is wrong in what I said - maybe you can be a bit more specific - you are mixing up formal testing for establishing thresholds with listening to a music signal in a home-based ABX test. 

You are stating that some differential thresholds are higher than others - am I saying anything different to this?

Are you actually suggesting that people should just sit down, do a blind test & that any differences will jump out & be consciously identified by them?


----------



## watchnerd

mmerrill99 said:


> So you understand & know all the science that underpins perceptual testing? Please, both of you, be more prolific in your posts on this subject as I would love to learn some more about this field.


 
  
 All the science of perceptual testing? Of course not.  As it relates to audio and psychoacoustics -- I've read probably 20-30 papers on the topic, attended lectures, conferences, etc.
  
 If you want to learn more, join the AES, read the multiple papers there on the topic.


----------



## mmerrill99

watchnerd said:


> mmerrill99 said:
> 
> 
> > So you understand
> ...



As I said "If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"


----------



## spruce music

mmerrill99 said:


> I actually don't see what you are specifically saying is wrong in what I said - maybe you can be a bit more specific - you are mixing up formal testing for establishing thresholds with listening to a music signal in a home-based ABX test.
> 
> You are stating that some differential thresholds are higher than others - am I saying anything different to this?
> 
> Are you actually suggesting that people should just sit down, do a blind test & that any differences will jump out & be consciously identified by them?


 

 Your last sentence sounds more like how sighted listening is done.  Just sit down, listen,  with no bother matching volumes and differences jump out into consciousness.


----------



## mmerrill99

spruce music said:


> mmerrill99 said:
> 
> 
> > I actually don't see what you are specifically saying is wrong in what I said - maybe you can be a bit more specific - you are mixing up formal testing for establishing thresholds with listening to a music signal in a home-based ABX test.
> ...



I take it you're a believer in the epiphanic effect of home-based ABX testing. Care to describe your testing procedure (You can skip the red herring of vol setting - all reasonable listening comparisons do this)


----------



## watchnerd

mmerrill99 said:


> As I said "If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"


 
  
 Which specific part of "not understanding the science" do you think I don't get?
  
 I've attended post-doc seminars lead by Poppy Crum, Principal Scientist at Dolby Laboratories, consulting professor at Stanford, on audio algorithm design using neuroscience and psychoacoustics.
  
 Also multiple lectures and workshops at the CCRMA, including several lead by JJ Johnston, who invented the AAC algorithm.
  
 So I'm dying to know what, specifically, you think I don't get, and what I said to make you think I don't get "it".....


----------



## mmerrill99

watchnerd said:


> mmerrill99 said:
> 
> 
> > As I said "If you believe for one minute that you or Emo "understand the science" then you are both completely lost in your own "brand of magic"
> ...


Huh?
Sorry, I went back to the original statement made by EMO from which you extracted your +1 quote - I see now after reading the context of your extracted text in EMO's full post that it didn't say what I read it to say. It's hard to keep up with fast moving threads. Apologies for any confusion


----------



## spruce music

mmerrill99 said:


> I take it you're a believer in the epiphanic effect of home-based ABX testing. Care to describe your testing procedure (You can skip the red herring of vol setting - all reasonable listening comparisons do this)


 

 Volume matching is no red herring it is a necessity for useful comparisons.  One which is very often ignored or done in a sloppy manner.
  
 Yes I believe in the efficacy of blind testing.  IBU standards would be best.  Lesser variations can be done in a manner which is more useful than sighted listening.


----------



## Hitec

spruce music said:


> Volume matching is no red herring it is a necessity for useful comparisons.  One which is very often ignored or done in a sloppy manner.
> 
> Yes I believe in the efficacy of blind testing.  IBU standards would be best.  Lesser variations can be done in a manner which is more useful than sighted listening.Here


 
  
 Great Read:
  
 http://www.newyorker.com/magazine/2013/01/28/music-to-your-ears


----------



## Hitec

"As a reasonable person, he says, “I don’t see the grand synthesis of one truth of hearing.” 
 In particular, he mistrusts the desire of the psycho-acousticians “to universalize—‘the brain
 does this, therefore that.’ Sure, everyone can agree that there’s a biological part of experience.
 For me, I’m much more interested in the questions about what we can and should change.
 I think actually that the fundamental insight of psycho-acoustics is incredibly profound for
 a humanist. Human ears aren’t natural reflectors of sound in the world. They are themselves
 these transducers that make reality—the perception of sound is not a mirror of nature. 
 Therefore, perception in a way makes sounds, and it makes sounds differently from a 
 microphone and a computer detecting vibrations out in the world.”
  
 Jonathan Sterne


----------



## RRod

hitec said:


> Great Read:
> 
> http://www.newyorker.com/magazine/2013/01/28/music-to-your-ears


 
  
 Nice read, though he kind of skips over the fact that you can do head tracking on headphones. The site for the research:
 https://www.princeton.edu/3D3A/BACCH_intro.html


----------



## castleofargh

@mmerrill you always seem to say that you treat abx and sighted evaluation both as anecdotes, but at the same time you've spammed us with 1 abx that just so happens to find differences where pretty much everybody else doesn't. it's not a serious work like you keep talking about as the only way to say anything, it's just one random guy. can't you read your own bias into this? it's all nice to suggest that we're deluding ourselves with our fake tests, but I don't see you offering any alternative. you just go and say "not good enough little padawan" to anything attempting to touch hearing. what a great way to never ever learn anything. and that goes back to my last post suggesting that you're not trying to help, but just to keep everybody in the dark asking to dismiss anything that doesn't reach your high expectations.
 where is you proof that abx is crap? that Arny didn't notice something in one test until someone said it was noticeable? yeah so ABX can still be biased by preconceptions. just like casual listening.
 the answer to that is pretty obvious, we must try to remove even that bias(and as many other biases as we can, like any science experiment would). many people do just that, giving more than 2 files, not giving away the different resolutions, and making sure to have at least 1 audible difference in one file, and 2 files that people should fail to discriminate, as scientific controls for the experiment. there is a software with 3 entries instead of 2, we can use that third one as a control. people willing to do the right thing can ask someone else to make the test for them(the very reason why blind test is better than sighted and why double blind is better than blind). but even without this there is no saying that we will always fail where we could have passed. you're just making up flaws much bigger than they are to fit your point of view of abx. where is the evidence that ABX is such a bad thing?
  please apply your legendary skepticism to your own judgment of abx and suggest ways to improve it or replace it, instead of trying to ruin it without more evidence of its flaws made by serious groups of scientists in a very controlled test etc etc blablablah the stuff you usually ask of us before accepting a statement. because your double game of killing abx with anecdotes while telling people that abx can't do better than be an anecdote, it's starting to feel more and more like you're just another guy with an agenda who just hided it better and a little longer than usual.


----------



## spruce music

hitec said:


> Great Read:
> 
> http://www.newyorker.com/magazine/2013/01/28/music-to-your-ears


 

 As soon as I can spare $50k for a BACCH SP I will look into it. 
  
 It appears it is based upon something similar to cross cancellation to give speaker sound much like binaural.  It is said to be demo'd with Chesky recordings.  Those are using two mics, usually MS or Blumlein.  I am wondering if it works with non-pure recordings.


----------



## Hitec

spruce music said:


> As soon as I can spare $50k for a BACCH SP I will look into it.
> 
> It appears it is based upon something similar to cross cancellation to give speaker sound much like binaural.  It is said to be demo'd with Chesky recordings.  Those are using two mics, usually MS or Blumlein.  I am wondering if it works with non-pure recordings.


 
 ..not sure what non-pure is, but I remember there was a note that recordings made from two mics, old and new recordings, could be made to work in this process..  Hey maybe, the price for the BACCH SP will drop and we could get this designed in a private home theater


----------



## jcx

how about the Smyth SVS Realizer - head tracking and personal HRTF measurement, calibration in real rooms, multichannel speaker systems - very realistic sensation of external speakers and room over headphones - has to be heard with a personal calibration to appreciate the lameness of almost all other headphone "soundstage"/"imaging" discussions when listening to conventionally mastered music intended for multichannel speaker playback - i.e. the 99% of commercially available recorded music
  
http://smyth-research.com/technology.html


----------



## spruce music

jcx said:


> how about the Smyth SVS Realizer - head tracking and personal HRTF measurement, calibration in real rooms, multichannel speaker systems - very realistic sensation of external speakers and room over headphones - has to be heard with a personal calibration to appreciate the lameness of almost all other headphone "soundstage"/"imaging" discussions when listening to conventionally mastered music intended for multichannel speaker playback - i.e. the 99% of commercially available recorded music
> 
> http://smyth-research.com/technology.html


 

 Yes, it seems like the better deal and price though still pricey.


----------



## dprimary

mmerrill99 said:


> I see
> The guidelines are laid out in BS.1116 "Methods for the subjective assessment of small impairments in audio systems " but guidelines aren't fully prescriptive in how to do the tests & it takes someone with experience in this area to administer such tests to avoid unnecessary biasing of the results. An individual can do this as long as he has the motivation & goes to the bother that ultramusicsnob demonstrated in his description of what he did, how he did it & what he heard - none of which is contained in BS.1116. The only other way to have some view on whether a a ABX test was of any value (or just another person fooling themselves) is to have Type II error statistics built into the test procedure. MUSHRA includes such hidden anchors & references "MUltiple Stimuli with Hidden Reference and Anchor" that allows testing of the test. Without this we get ridiculous ABX tests like Arny Kreuger's test where he didn't even listen to the samples A or B, just hit buttons - if you don't know whether the test results are from a real test or a "fake" test then you are in no-man's-land as far as the validity of the results


 

 These guidelines pretty much layout the procedure to do a double blind or ABX test. There seems to be steps missing in preparation of the source material, but I a read it quickly, skimming through what all looks like standard practice to me. 
 What exactly are your objections to blind testing? Who or what is an UMS? Please provide links, does anyone cite reference sources anymore?
  
 The tests that the person cannot tell any difference do not make much difference. Maybe they are not trying, they are not trained, don't understand what they should do, or any number of other reasons they do not hear any difference. If you expect most people to hear a difference then yes there could be a problem with the test.  I don't see much reason to test for things I expect everyone to hear. Though I have been known to turn on and off an acoustic calibrator just to see if anyone notices.
  
 For positive result it would be expected that everyone wants to review everything to the smallest detail to confirm that something was not overlooked, and/or so the test can be repeated by others. After trying 5999 different filaments when the 6000th one seems to work everyone is going to get excited. Does it really work? is just fluke? did they cheat? I can't get it to work, what am I doing wrong? These are expected responses.


----------



## mmerrill99

castleofargh said:


> @mmerrill you always seem to say that you treat abx and sighted evaluation both as anecdotes, but at the same time you've spammed us with 1 abx that just so happens to find differences where pretty much everybody else doesn't. it's not a serious work like you keep talking about as the only way to say anything, it's just one random guy. can't you read your own bias into this?


Did I say that based on UMS positive ABX results that I therefore claimed anything? No, I didn't - I just found his procedure & description of what is entailed in doing a reasonable ABX test 


> it's all nice to suggest that we're deluding ourselves with our fake tests, but I don't see you offering any alternative. you just go and say "not good enough little padawan" to anything attempting to touch hearing. what a great way to never ever learn anything. and that goes back to my last post suggesting that you're not trying to help, but just to keep everybody in the dark asking to dismiss anything that doesn't reach your high expectations.


This is your fallback complaint every time - I'm not giving you an alternative. Well, if you truly want to "learn" & are interested in the truth of the matter, the first step is recognising the weakness of your tests & not rejecting this analysis because it leaves you in a philosophical vacuum. It seems like you are an addict - you are being told what you're doing is wrong but you can't give it up because you can't envisage life without your addiction


> where is you proof that abx is crap? that Arny didn't notice something in one test until someone said it was noticeable? yeah so ABX can still be biased by preconceptions. just like casual listening.


I have said that home run ABX is crap - let's get that straight. I have laid out the weaknesses & there can be no proof because the test itself would need to be changed to allow the evaluation of false negatives but no one is interested in doing so - hence you are able to ask for proof, knowing full well that none is possible. As I said before this is a recognised issue in other blind tests MUSHRA - MUltiple Stimuli with Hidden Reference and Anchor - the hidden references & anchors are a major part of this test to ensure that it's sensitivity is measurable. This is not part of ABX testing. You can ignore this if you choose but please don't then try to tell me that there aren't alternatives or that these factors aren't important 


> the answer to that is pretty obvious, we must try to remove even that bias(and as many other biases as we can, like any science experiment would). many people do just that, giving more than 2 files, not giving away the different resolutions, and making sure to have at least 1 audible difference in one file, and 2 files that people should fail to discriminate, as scientific controls for the experiment. there is a software with 3 entries instead of 2, we can use that third one as a control. people willing to do the right thing can ask someone else to make the test for them(the very reason why blind test is better than sighted and why double blind is better than blind). but even without this there is no saying that we will always fail where we could have passed. you're just making up flaws much bigger than they are to fit your point of view of abx. where is the evidence that ABX is such a bad thing?


Listen you simply don't know how sensitive your home run ABX test is & yet you cling to it for comfort


> please apply your legendary skepticism to your own judgment of abx and suggest ways to improve it or replace it, instead of trying to ruin it without more evidence of its flaws made by serious groups of scientists in a very controlled test etc etc blablablah the stuff you usually ask of us before accepting a statement. because your double game of killing abx with anecdotes while telling people that abx can't do better than be an anecdote, it's starting to feel more and more like you're just another guy with an agenda who just hided it better and a little longer than usual.


Thank you for the "legendary" status - I'll wear it with pride - scepticism is the foundation of science so thank you!


----------



## mmerrill99

dprimary said:


> mmerrill99 said:
> 
> 
> > I see
> ...


UMS stands for ultmusicsnob - a ex-member of the forum who posted positive ABX test results that showed he routinely differentiated jitter files (prepared here) & also differentiated red-book from high-res audio. I'm not interested in th eresults or wish to debate them - what interested me & I thought it a great resource for anybody interested in home-run ABX testing was his description of how he did the training, how he did the tests, what worked for him, what didn't work & what he focussed on in his listening - a great education for anybody interested in the practicalities of ABX testing



> The tests that the person cannot tell any difference do not make much difference. Maybe they are not trying, they are not trained, don't understand what they should do, or any number of other reasons they do not hear any difference. If you expect most people to hear a difference then yes there could be a problem with the test.  I don't see much reason to test for things I expect everyone to hear. Though I have been known to turn on and off an acoustic calibrator just to see if anyone notices.


Yes, there are multitudinous reasons for a null result - it's why I query anyone who claims an "epiphany" from such a test or doesn't query the sensitivity of the test itself. And no, a null result is not treated as inconsequential - it's very much the whole point why people are constantly challenged to do a blind test - to "prove" that they really hear something. Many are lured by the apparent simplicity of this premise listen with "ears only" to "prove" if you can "really hear" a difference. It's a beguiling trap that most fall for. 



> For positive result it would be expected that everyone wants to review everything to the smallest detail to confirm that something was not overlooked, and/or so the test can be repeated by others. After trying 5999 different filaments when the 6000th one seems to work everyone is going to get excited. Does it really work? is just fluke? did they cheat? I can't get it to work, what am I doing wrong? These are expected responses.


----------



## Joe Bloggs

Why do I not find "UMS"'s methodologies enlightening in the same way as you do? (isn't it fun that you refer to the same guy so often that you're referring to him by shorthand now?) Too much flowery description of the perceived differences between the files (and nobody trained him to listen to those things--those just happened to be the things he thought he could hear. There is no evidence that, if a difference can be heard between the things he was testing between, that what he was listening for was the kind of thing that would best distinguish them.) and too little evidence that he got down basic controls on file preparation, volume matching and other things one needed to do to make sure that the differences heard are because of what was being tested for and not because of confounding variables. It was also conducted by one guy at home with no supervision and so by your own reasoning just as much bunk as anybody else's. The only reason you hold UMS's methodology as the one to follow while dismissing everyone elses' home tests seems to be because his results agree with what you're hoping for.

You also have no answer for this


joe bloggs said:


> mmerrill99
> 
> In my book there are two types of audio tweaks:
> 
> ...


----------



## JWolf

The problem with an ABX test is that if you find in favor of 24-bit, then you get bombed with things like "that cannot be correct" and "please provide your source files so we can see what's going on". It's all nonsense when you get people saying things like that when you post the results not in 16-bit's favor.


----------



## nick_charles

mmerrill99 said:


> I have said that home run ABX is crap - let's get that straight. I have laid out the weaknesses & there can be no proof because the test itself would need to be changed to allow the evaluation of false negatives but no one is interested in doing so - hence you are able to ask for proof, knowing full well that none is possible.


 
  
 Except for UMS and AMIRM whose home-tests are obviously fine ? and which you have staunchly refused to condemn ! You cannot have it both ways all home test are crap except the ones I agree with ?
  
 Your use of the term  false negative is incorrect. You have already been corrected on this and you admitted it
  


> Sure & the use of false negative is misleading because I'm not talking about the overall result of the test (or it's statistical power) - I'm really talking about how many trials in a ABX test would report no difference when a really audible difference was used in some trials (unknown to the participant). Other blind testing methods use these hidden references & anchors as a means of testing the test & validating it to some extent but ABX testing lacks this possibility.


 
  
 Your premise seems to be there was an audible difference but it was not heard, how can an audible difference not be heard ? That makes absolutely no sense whatsoever. If you are going to say "ah but it was such a big difference that it should have been heard" - how would you establish that the difference was of such a magnitude that it should have been heard ? unless you have some really solid data for the thresholds for detection for that pair of stimulae you are just out and out handwaving. In the case of UMS tests the best published evidence we have (and yes I have read Dunn and Hawksford) is that the type and magnitude of jitter he reported detecting were below currently established thresholds from research from amongst others the BBC , a Japanese state TV station and researchers at Dolby Labs. His tests were just Foobar ABX tests nothing special therefore you must reject them !


----------



## OddE

jwolf said:


> The problem with an ABX test is that if you find in favor of 24-bit, then you get bombed with things like "that cannot be correct" and "please provide your source files so we can see what's going on". It's all nonsense when you get people saying things like that when you post the results not in 16-bit's favor.


 
  
 -I wouldn't call it nonsense. Do keep in mind that established, recognized science indicate there shouldn't be an audible difference _caused by the bit depth alone_.
  
 Hence, if someone comes by, stating that they _do_ perceive a difference, it is quite natural that people are curious. After all, if the results hold up to scrutiny (and are reproducible!) - we'll have learned something new.
  
 After all, extraordinary claims require extraordinary evidence. As things stand today, an audible difference between two copies of the same master, one represented in 24 bit and the other in 16 bit would be extraordinary. That's all.


----------



## mmerrill99

joe bloggs said:


> Why do I not find "UMS"'s methodologies enlightening in the same way as you do? (isn't it fun that you refer to the same guy so often that you're referring to him by shorthand now?) Too much flowery description of the perceived differences between the files (and nobody trained him to listen to those things--those just happened to be the things he thought he could hear. There is no evidence that, if a difference can be heard between the things he was testing between, that what he was listening for was the kind of thing that would best distinguish them.) and too little evidence that he got down basic controls on file preparation, volume matching and other things one needed to do to make sure that the differences heard are because of what was being tested for and not because of confounding variables. It was also conducted by one guy at home with no supervision and so by your own reasoning just as much bunk as anybody else's. The only reason you hold UMS's methodology as the one to follow while dismissing everyone elses' home tests seems to be because his results agree with what you're hoping for.


Again, by intent or otherwise, you miss the point of UMS's posts (I wasn't the first to use the abbreviation UMS, btw). What's educational, is his description of the procedures necessary to have a chance of getting a non-null result. But, of course, this is only educational for those that have a willingness to learn



> You also have no answer for this
> 
> 
> joe bloggs said:
> ...



Do I need to answer?


----------



## mmerrill99

jwolf said:


> The problem with an ABX test is that if you find in favor of 24-bit, then you get bombed with things like "that cannot be correct" and "please provide your source files so we can see what's going on". It's all nonsense when you get people saying things like that when you post the results not in 16-bit's favor.



Well there is nothing wrong with analysis of results but why aren't null results equally analysed? I know the stock answer is that "null results are of no consequence" - well then please explain how people have "epiphanies" from such null results?

The second point is that after all possible analysis has been done & no reason found to reject the results we get a number of other excuses - one positive result is of little significance or the tester is a cheater & needs proctoring. So not only is the test itself skewed towards a null result but so is the mindset


----------



## Joe Bloggs

1. If one needs to "learn" how to create an unconfirmed positive test result that is riddled with question marks on actual methodology, one doesn't need to learn about blind testing at all.

2. "Do you need to answer"? Well, *if you are at all interested in forwarding the actual state of the art of your audio system* rather than raising strawman attacks on scientific members of the forum, then yeah, you owe it to yourself to answer this. But I guess you aren't, so you don't.


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > I have said that home run ABX is crap - let's get that straight. I have laid out the weaknesses
> ...


Again, whether intentionally or otherwise, you don't seem to comprehend - the ABX home-run tests are severely flawed by being biased towards delivering a null result. When a positive result is returned it should make people sit up & pay attention to what might have been done differently with this test to all the other ABX tests that returned a null result. In both these cases, UMS & Amir, the ability to extract a "tell" & self-training was one of the crucial factors & secondly the ability to stay focussed during the test (or take breaks & return). So, in both these cases, they have overcome the severe bias towards null result that the test defaults to. 



> Your use of the term  false negative is incorrect. You have already been corrected on this and you admitted it
> 
> 
> 
> ...


Because the subject is not focussed, is tired, has a negative bias towards what's being tested, etc - many, many reasons. Most tests don't give a 100% accuracy of correct answers & as the differential threshold gets nearer to 50% it becomes more difficult. So 75% or so is considered the normal differential threshold i.e. differences are consciously noticeable 3/4 of the trials. A differential threshold of 60% is considered subliminal i.e not consciously noticeable but subconsciously so - a feeling. 





> That makes absolutely no sense whatsoever. If you are going to say "ah but it was such a big difference that it should have been heard" - how would you establish that the difference was of such a magnitude that it should have been heard ? unless you have some really solid data for the thresholds for detection for that pair of stimulae you are just out and out handwaving.


That's why references & anchors are used in MUSHRA testing - these establish the sensitivity of the test/tester/equipment by using known impairments which have a threshold of differentiation established for the population. So let's take the case of volume matching - is it not the case that there is an established threshold above which a difference in volume will be noticeable? You claim to have done courses on ths so I'm at a loss why I should have to spell out such basic facts?





> In the case of UMS tests the best published evidence we have (and yes I have read Dunn and Hawksford) is that the type and magnitude of jitter he reported detecting were below currently established thresholds from research from amongst others the BBC , a Japanese state TV station and researchers at Dolby Labs. His tests were just Foobar ABX tests nothing special therefore you must reject them !


Well unless you can nominate what is wrong in his tests then, by your own criteria of objective analysis, you must either accept his results, continue in your analysis until you find the flaw in his testing or accuse him of being a liar!!


----------



## mmerrill99

joe bloggs said:


> 1. If one needs to "learn" how to create an unconfirmed positive test result that is riddled with question marks on actual methodology, one doesn't need to learn about blind testing at all.


So let me clarify what you are saying - you don;t recognise the pivotal role that training plays in blind testing/ That does fit in with the mindset that is happy to embrace & foster the "epiphany" moment that flawed blind testing is claimed to bring 



> 2. "Do you need to answer"? Well, *if you are at all interested in forwarding the actual state of the art of your audio system* rather than raising strawman attacks on scientific members of the forum, then yeah, you owe it to yourself to answer this. But I guess you aren't, so you don't.


Sure, there are small differences & there are big differences - so what? You worry about the big differences but I can tell you that your "big differences" are considered small by those who say that speakers & rooms are where the focus should be, not tweaking with the source. One man's ceiling is another man's floor - it's all a matter of perspective


----------



## Joe Bloggs

mmerrill99 said:


> So let me clarify what you are saying - you don;t recognise the pivotal role that training plays in blind testing/ That does fit in with the mindset that is happy to embrace & foster the "epiphany" moment that flawed blind testing is claimed to bring




No, I'm saying that an improperly controlled blind test is just as useless as a sighted test.



> > 2. "Do you need to answer"? Well, *if you are at all interested in forwarding the actual state of the art of your audio system* rather than raising strawman attacks on scientific members of the forum, then yeah, you owe it to yourself to answer this. But I guess you aren't, so you don't.
> 
> 
> Sure, there are small differences & there are big differences - so what? You worry about the big differences but I can tell you that your "big differences" are considered small by those who say that speakers & rooms are where the focus should be, not tweaking with the source. One man's ceiling is another man's floor - it's all a matter of perspective




*I* am the one who is saying that speakers & rooms are among the things you should be concerned about, in all cases YOU are the one who is claiming that we should all worry about jitter and sample depth of our sources. There's no switching of perspectives so yes, in terms of possible room for improvement for our respective audio systems, your ceiling has always been my floor. :rolleyes:


----------



## Joe Bloggs

And to say that blind tests are biased toward negative results? With all the ways a blind test can go wrong, giving clues to the listener that have nothing to do with the variable being tested, and all the ways that sample preparation itself can go wrong, again giving detectable differences that have nothing to do with the variable being tested, and finally, the overwhelming mental pressure on an audiophile to produce positive results or else be called a cloth-eared git? I would say that blind tests, as with all audio tests, are overwhelming biased toward *positive* results to start with!


----------



## nick_charles

joe bloggs said:


> And to say that blind tests are biased toward negative results? With all the ways a blind test can go wrong, giving clues to the listener that have nothing to do with the variable being tested, and all the ways that sample preparation itself can go wrong, again giving detectable differences that have nothing to do with the variable being tested, and finally, the overwhelming mental pressure on an audiophile to produce positive results or else be called a cloth-eared git? I would say that blind tests, as with all audio tests, are overwhelming biased toward *positive* results to start with!


 
  
 You must distinguish between single blind and double blind - a single blind with an experimenter has the potential for the issues of cues as you say but in Double-Blind neither the subject nor experimenter know which is which. In this context I believe that MM99 is talking about DBT (at least I hope so), when there is no experimenter we typically still call it DBT as long as there is no way the subject might find out the identity of the signals without being able to distinguish them in a test - the terminology is a bit sloppy.


----------



## mmerrill99

joe bloggs said:


> And to say that blind tests are biased toward negative results? With all the ways a blind test can go wrong, giving clues to the listener that have nothing to do with the variable being tested, and all the ways that sample preparation itself can go wrong, again giving detectable differences that have nothing to do with the variable being tested, and finally, the overwhelming mental pressure on an audiophile to produce positive results or else be called a cloth-eared git? I would say that blind tests, as with all audio tests, are overwhelming biased toward *positive* results to start with!




Tell me - have you ever done a blind test? 
What role does blind testing play in your development of your devices?


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > So let me clarify what you are saying - you don;t recognise the pivotal role that training plays in blind testing/ That does fit in with the mindset that is happy to embrace & foster the "epiphany" moment that flawed blind testing is claimed to bring
> ...


Ok, I agree with that but would add that 99% of home run blind tests are improperly controlled & therefore all should be treated as anecdotal reports.



> > > 2. "Do you need to answer"? Well, *if you are at all interested in forwarding the actual state of the art of your audio system* rather than raising strawman attacks on scientific members of the forum, then yeah, you owe it to yourself to answer this. But I guess you aren't, so you don't.
> >
> >
> > Sure, there are small differences & there are big differences - so what? You worry about the big differences but I can tell you that your "big differences" are considered small by those who say that speakers & rooms are where the focus should be, not tweaking with the source. One man's ceiling is another man's floor - it's all a matter of perspective
> ...


Ok, but changes to the quality of the source are fundamentally different in nature to changes to speakers & rooms


----------



## mmerrill99

nick_charles said:


> joe bloggs said:
> 
> 
> > And to say that blind tests are biased toward negative results? With all the ways a blind test can go wrong, giving clues to the listener that have nothing to do with the variable being tested, and all the ways that sample preparation itself can go wrong, again giving detectable differences that have nothing to do with the variable being tested, and finally, the overwhelming mental pressure on an audiophile to produce positive results or else be called a cloth-eared git? I would say that blind tests, as with all audio tests, are overwhelming biased toward *positive* results to start with!
> ...


Well as we are talking about ABX home tests then yes it is ostensibly a DBT!


----------



## Joe Bloggs

mmerrill99 said:


> joe bloggs said:
> 
> 
> > No, I'm saying that an improperly controlled blind test is just as useless as a sighted test.
> ...




...including UltMusicSnob's.


----------



## watchnerd

I'm going to attempt to see if  I can summarize the debate so far:
  
 Side A:
  
 1. Home ABX / DBT tests are crap because they're biased towards a null result
 2. Except in a few cases where some 1 or 2 people may have been very meticulous and reported being able to differentiate 16bit files from 24bit files, which over-rides the null bias
 3. Therefore there is some new evidence of an audible difference between 16bit files and 24bit files
  
 Side B:
  
 1. Home ABX / DBT test are methodologically flawed.  They may be curious or fun, but they don't meet peer reviewed publication standards for methodology, sample size, etc.
 2. This includes cases where some 1 or 2 people may have been very meticulous and reported being able to differentiate 16bit files from 24bit files
 3. Therefore there isn't any new credible evidence that meets peer review standards of an audible difference between 16bit files and 24bit files
  
 Is that a fair summary?


----------



## JWolf

odde said:


> -I wouldn't call it nonsense. Do keep in mind that established, recognized science indicate there shouldn't be an audible difference _caused by the bit depth alone_.
> 
> Hence, if someone comes by, stating that they _do_ perceive a difference, it is quite natural that people are curious. After all, if the results hold up to scrutiny (and are reproducible!) - we'll have learned something new.
> 
> After all, extraordinary claims require extraordinary evidence. As things stand today, an audible difference between two copies of the same master, one represented in 24 bit and the other in 16 bit would be extraordinary. That's all.


 
  
 There is the possibility that 24-bit can sound better  than 16-bit due to less errors in reconstructing the original waveform. But if the sample rate is higher than 44.1, it can sound better or just different and so yes, someone saying he/she can hear the difference is no reason to be ridiculed.


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > joe bloggs said:
> ...



As the default skew of the test is a null result because it is designed in that way to minimise false positives & he achieved a positive result, then unless you can identify the flaw in his result, I wouldn't consider his test flawed, no

The problem is that there are no controls to minimise false negatives - that's left up to the administration of the test & as most tests return false negatives we have no way of knowing how valid this result is - was it due to someone not bothering (like ArnyK) was it due to someone losing focus (& not even recognising this), was it due to not training, etc, etc.

So, no positive ABX results, unless found to be the result of flawed procedure, are by their very nature not flawed - the test is geared to eliminate all such false positives


----------



## mmerrill99

watchnerd said:


> I'm going to attempt to see if  I can summarize the debate so far:
> 
> Side A:
> 
> ...




I guess you have me in side B - so let me elucidate that position
1. We have no way of judging the validity of home ABX tests & nothing can be derived from them even if there are 1,000s of null results - it means diddly squat as 99% of them are probably flawed tests & never had a hope of differentiating differences which are audible when examined under properly controlled & administered rigorous tests 
2. When a home based positive ABX test result is reported it usually undergoes sufficient scrutiny & analysis to eliminate a flawed procedure or files. So if no flaw is found I accept this result as valid
3. I have little doubt that the same test & subject, if submitted to a formal, rigorously organised & professionally run DBT would achieve the same positive result
3. So, null results from home ABX tests are less than useful & certainly should not be the basis of "epiphanies" or be the test that is suggested to people to "prove" they are ACTUALLY hearing what they say they are


----------



## bfreedma

mmerrill99 said:


> Tell me - have you ever done a blind test?
> What role does blind testing play in your development of your devices?


 
  
 Another distraction to attempt to avoid the topic.  At some point, you're tactics are tiresome and pointless - for me, that point occurred many posts ago.


----------



## nick_charles

mmerrill99 said:


> Again, whether intentionally or otherwise, you don't seem to comprehend - the ABX home-run tests are severely flawed by being biased towards delivering a null result.





> *Where is your evidence for this assertion ?*





> Because the subject is not focussed, is tired, has a negative bias towards what's being tested, etc - many, many reasons.





> *Nope, if it is audible then it is heard - if it is not heard it is not audible - it's very simple really. Do you mean potentially audible instead?*





>





> Well unless you can nominate what is wrong in his tests then, by your own criteria of objective analysis, you must either accept his results, continue in your analysis until you find the flaw in his testing or accuse him of being a liar!!





> *They are home abx test so must be crap by your definition !*





> *Several alternatives exist including but not limited to - 1 UMS really did hear jitter that is considered in peer reviewed papers to be inaudible - 2 The jitter levels were higher than was thought by the creator 3 There was some other artifact or clue created when the samples were created 4 It was a wind up 5 some part of UMS playback chain is unusually sensitive to jitter - who knows ? But nobody else has been able to replicate his results here or elsewhere - I think the files might even still be available - an analysis of the data in the files does not reveal any differences considered audible *


 
  
  
 I have reported positive ABX tests and the conditions under which they were achieved  - why are mine less valid than the tests of UMS and AMIRM ?


----------



## mmerrill99

bfreedma said:


> mmerrill99 said:
> 
> 
> > Tell me - have you ever done a blind test?
> ...



This is completely on topic - do you not think it's a very relevant point - how many audio designers use DBTs in their development? The very audio devices you ask the consumer to prove he can hear a difference in using by using ABX tests - how many designs have actually been ABX tested before going out the door?

If only a few use ABX tests then what is this great debate about - sighted testing, being the predominant method of listening tests for audio device designers/developers seems to be the proven de facto standard for evaluation!!


----------



## mmerrill99

nick_charles said:


> > *Nope, if it is audible then it is heard - if it is not heard it is not audible - it's very simple really. Do you mean potentially audible instead?*
> 
> 
> You are just playing semantic games now - yes, you know what I'm saying - it is in the category of impairments that are above the differential threshold of audibility
> ...


----------



## nick_charles

mmerrill99 said:


> This is completely on topic - do you not think it's a very relevant point - how many audio designers use DBTs in their development? The very audio devices you ask the consumer to prove he can hear a difference in using by using ABX tests - how many designs have actually been ABX tested before going out the door?


 
  
  
 How about you do some research on this and report back ? It is a quite interesting question and you raised it.


----------



## nick_charles

.


----------



## L8MDL

nick_charles said:


> .




^^^^the very best post I've seen in this round and round and round discussion which has gone nowhere since the first post.


----------



## bfreedma

mmerrill99 said:


> This is completely on topic - do you not think it's a very relevant point - how many audio designers use DBTs in their development? The very audio devices you ask the consumer to prove he can hear a difference in using by using ABX tests - how many designs have actually been ABX tested before going out the door?
> 
> If only a few use ABX tests then what is this great debate about - sighted testing, being the predominant method of listening tests for audio device designers/developers seems to be the proven de facto standard for evaluation!!


 
  
 I'll admit, sighted testing is the de facto standard for sellers of high end cables, jitter reduction devices, and other products unlikely to be differentiated via controlled testing....


----------



## nick_charles

l8mdl said:


> ^^^^the very best post I've seen in this round and round and round discussion which has gone nowhere since the first post.


 
  
 Thanks,  I was inspired by John Cage !


----------



## mmerrill99

And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!





bfreedma said:


> mmerrill99 said:
> 
> 
> > This is completely on topic - do you not think it's a very relevant point - how many audio designers use DBTs in their development? The very audio devices you ask the consumer to prove he can hear a difference in using by using ABX tests - how many designs have actually been ABX tested before going out the door?
> ...


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > This is completely on topic - do you not think it's a very relevant point - how many audio designers use DBTs in their development? The very audio devices you ask the consumer to prove he can hear a difference in using by using ABX tests - how many designs have actually been ABX tested before going out the door?
> ...


 Well I started by asking JoeBloggs -let's see what his reply is? Care to hazard a guess?


----------



## OddE

jwolf said:


> There is the possibility that 24-bit can sound better  than 16-bit due to less errors in reconstructing the original waveform. But if the sample rate is higher than 44.1, it can sound better or just different and so yes, someone saying he/she can hear the difference is no reason to be ridiculed.


 
  
 -That would require our ears to be able to detect errors which until now have proven to be orders of magnitude below our threshold of detectability - which would be a most surprising finding indeed.
  
 (Or, in the case of higher sample rate - that the candidate's hearing extends significantly past what is normal, which would also be a most surprising finding.)
  
 It is, however, my impression that ridicule is not stowed so much upon the initial claim as on the subsequent failure/disinterest/outright refusal to provide evidence of said claim.


----------



## spruce music

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 

 I would guess designers of mainstream gear don't do any listening as part of the design/development process.


----------



## OddE

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 
  
 -My guess would be that if they did, it would be best to keep mum about it and rather claim to use their ears. Goes down much better with the target clientele, after all.


----------



## mmerrill99

spruce music said:


> mmerrill99 said:
> 
> 
> > And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!
> ...


I believe you are just playing semantic games, like Nick


----------



## mmerrill99

odde said:


> mmerrill99 said:
> 
> 
> > And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!
> ...


 huh?


----------



## sonitus mirus

mmerrill99 said:


> huh?


 
  
 It probably means that if they were to conduct listening tests, the results would match many of the well-conducted ABX tests that us lowly commoners perform.  There would be no difference identified, or the difference would be nearly insignificant to matter to practically anyone.  That is not good for their business.


----------



## OddE

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 
  


odde said:


> -My guess would be that if they did, it would be best to keep mum about it and rather claim to use their ears. Goes down much better with the target clientele, after all.


 
  
  
 Quote:


mmerrill99 said:


> huh?


 
  
 -Your average, run-of-the-mill, dyed-in-the-wool audiophile is likely to take a rather grim view of blind testing and the like, wouldn't you agree? (For starters, just consider how simply mentioning DBT will get you a slap on the wrist in the other subforums here on Head-Fi.)
  
 If I were designing audio gear, I'd base my designs on measurements; after all, in digital audio transparency is a relatively straightforward and attainable goal provided one is able to follow sound engineering (pun intended) practices.
  
 I'd probably still ask my marketing department (presuming I had the $$$ to hire one!) to wax lyrical about the ardous process of trial and error, acting on hunches, trusting my ears and whatnot to ensure the aforementioned clientele weren't put off by my designs, though.


----------



## jcx

Industry does do controlled, Blind listening tests when it is viewed as likely to give positive results, return on investment
  
 Harmon? - forgot Toole http://www.amazon.com/Sound-Reproduction-Psychoacoustics-Loudspeakers-Engineering/dp/0240520092 , Olive http://seanolive.blogspot.com/
  
http://www.aes.org/
  
http://www.amazon.com/Perceptual-Audio-Evaluation-Theory-Application/dp/0470869232
  
 and yes the preference protocols go beyond AB/X - but I think you will find overwhelming agreement that
 if AB/X doesn't give positive results then you are looking at something too small to work with
  
http://www.delta.dk/imported/senselab/AES125_Tutorial_T4_Perceptual_Audio_Evaluation_Tutorial.pdf presentation from the book authors


----------



## bfreedma

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 
  
 Now why would a business do something that would no doubt decrease the sales of their high priced components?  We can't even get most of the manufacturers to provide detailed measurements of products.


----------



## watchnerd

mmerrill99 said:


> I guess you have me in side B - so let me elucidate that position
> 1. We have no way of judging the validity of home ABX tests & nothing can be derived from them even if there are 1,000s of null results - it means diddly squat as 99% of them are probably flawed tests & never had a hope of differentiating differences which are audible when examined under properly controlled & administered rigorous tests
> 2. When a home based positive ABX test result is reported it usually undergoes sufficient scrutiny & analysis to eliminate a flawed procedure or files. So if no flaw is found I accept this result as valid
> 3. I have little doubt that the same test & subject, if submitted to a formal, rigorously organised & professionally run DBT would achieve the same positive result
> 3. So, null results from home ABX tests are less than useful & certainly should not be the basis of "epiphanies" or be the test that is suggested to people to "prove" they are ACTUALLY hearing what they say they are


 
  
 Actually, a feeling that home ABX tests have a null bias fits into the taxonomy of A.


----------



## spruce music

mmerrill99 said:


> I guess you have me in side B - so let me elucidate that position
> 1. We have no way of judging the validity of home ABX tests & nothing can be derived from them even if there are 1,000s of null results - it means diddly squat as 99% of them are probably flawed tests & never had a hope of differentiating differences which are audible when examined under properly controlled & administered rigorous tests
> 2. When a home based positive ABX test result is reported it usually undergoes sufficient scrutiny & analysis to eliminate a flawed procedure or files. So if no flaw is found I accept this result as valid
> 3. I have little doubt that the same test & subject, if submitted to a formal, rigorously organised & professionally run DBT would achieve the same positive result
> 3. So, null results from home ABX tests are less than useful & certainly should not be the basis of "epiphanies" or be the test that is suggested to people to "prove" they are ACTUALLY hearing what they say they are


 

 So you have a bias for accepting any positive results by assuming if the test were better the results would stand up.  While paying no attention to null results.  The reverse would seem to be more likely without looking at any particular result or procedure. In other words, the chances of a poor procedure by an amateur netting positive results is more likely than if done according to professional guidelines because professionals are more likely to control for sources of bias.
  
 You also seem to be making a mistake in how you view null results.  It is not uncommon.  A null result doesn't really prove something.  It simply means no results contrary to the null hypothesis were the result of a given procedure.  Subtle though important difference.


----------



## watchnerd

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 
  
 Harman uses blind testing during the development of some their Revel and JBL Pro monitors. 
  
 I suspect they don't do it for their 'lifestyle' products where they're not designing for audiophiles, have low price points, and looks and ease of use are more important..sound only needs to be 'okay'.


----------



## spruce music

mmerrill99 said:


> I believe you are just playing semantic games, like Nick


 

 No I am not.  Mainstream is the word you chose in an attempt to give your view validity.  Mainstream would mean Denon, Sony, Onkyo, Pioneer and such companies.  Or it might mean some of the pro audio brands.  My guess is they don't do listening tests for the great bulk of their products.  They design to a spec without listening.  
  
 Trying to place "audiophile" or boutique or high end companies as the mainstream or the prevailing norm as some sort of benchmark is a rhetorical argument.  They are the outliers.


----------



## watchnerd

jwolf said:


> There is the possibility that 24-bit can sound better  than 16-bit due to less errors in reconstructing the original waveform. But if the sample rate is higher than 44.1, it can sound better or just different and so yes, someone saying he/she can hear the difference is no reason to be ridiculed.


 
  
 Are you talking about a file bit depth or the bit depth of the DAC chipset?


----------



## JWolf

watchnerd said:


> Are you talking about a file bit depth or the bit depth of the DAC chipset?


 
  
 A bit of both. There is more of a gap between the bits in a 16-bit audio file and more chance of something not being correctly rendered between those bits. Also, if any of those bits are in error, there is more chance of what's reconstructed will; be in error. Wuth 24-bit, there are more bits so if a bit is wrong, then the error will be smaller as the next bit is a lot closer.
  
 Take the set of numbers 4 8 12 16 20 and 2 4 6 8 10 12 14 16 18 20. Now in the first set of numbers if 12 is wrong then the wave form could be wrong from 8 to 16. In the second set of numbers if 12 is wrong, then the error could only be as large as 10 to 14. So the error in 16-bit would be larger than the error in 24-bit. That can change the sound a lot more in 16-bit than in 24 bit.
  
 Another issue is that because 16-bit has more of a gap and if the waveform between the gap is not what the system thinks it is, there's a larger error. With 24-bit there's less of a gap and thus a smaller error or maybe none as there are more bits to use to reconstruct the waveform.
  
 I cannot say how true this is, but it's been said that we'd be best going from 16/44.1 to 20/88. That would give enough bits and enough sampling without being too large.
  
 Given that we have blu-ray which gives us more room in the same space as CDs, CDs should be phased out for blu-ray. Then we can have Hi-Res audio instead of CDs. But I would say the price of this new audio should not be any more expensive than CDs are now. It's just the cost of replacing CD players would be the major cost factor. But then look at TVs. Tube TVs are no longer made. So if you buy a new TV you buy LCD or LED or OLED.


----------



## spruce music

jwolf said:


> A bit of both. There is more of a gap between the bits in a 16-bit audio file and more chance of something not being correctly rendered between those bits. Also, if any of those bits are in error, there is more chance of what's reconstructed will; be in error. Wuth 24-bit, there are more bits so if a bit is wrong, then the error will be smaller as the next bit is a lot closer.
> 
> Take the set of numbers 4 8 12 16 20 and 2 4 6 8 10 12 14 16 18 20. Now in the first set of numbers if 12 is wrong then the wave form could be wrong from 8 to 16. In the second set of numbers if 12 is wrong, then the error could only be as large as 10 to 14. So the error in 16-bit would be larger than the error in 24-bit. That can change the sound a lot more in 16-bit than in 24 bit.
> 
> ...


 

 What you are trying to describe is quantization error.  It is smaller with 20 or 24 bit than 16.  Electronics aren't quiet enough to manage real 24 bit results.  20 bit is about it.  However, is quantization error in 16 bit audio large enough to be noticed vs 20 or 24?
  
 Another factor is dithering reduces quantization error.  Quantization error in 16 bit audio with dither is more like 20 bit levels and theoretically could be better than that.  An example it is possible to encode a signal below the -96 db floor of 16 bit audio if dither is used and recover that signal down to something like -120 db. 
  
 If things are working the largest quantization error would be half of the Least Significant Bit.  With dither it is smaller than that.


----------



## Don Hills

mmerrill99 said:


> I believe you are just playing semantic games, like Nick


 
  
 Pot, kettle, black...


----------



## Joe Bloggs

spruce music said:


> mmerrill99 said:
> 
> 
> > I believe you are just playing semantic games, like Nick
> ...






Spoiler: Hidden reply



Unfortunately I don't run FiiO, so my design preferences do not represent those of FiiO's. :rolleyes: But if FiiO were to carry out double-blind testing in their evaluation of audio designs (or simply ditched listening tests altogether as you suggest), it could only benefit them:

1. More money would be spent on audio changes that matter, cementing our reputation for affordably-priced gear with hi-end performance.

Right now, even despite the low going price of our products, there is a tendency towards design excesses and component choice excesses that I doubt makes an audible difference to the sound, while glaring problems (such as the X7 WiFi fiasco, and the E18 buzzing when bass boost is on...) were let out the door.

2. For our DAPs, more money could be spent on improving the UI (which is reputedly one of the best in the industry in China, but which I still find lacking) and processing-related issues (better CPU and RAM for rendering that better UI).

There would be fewer audiophile components in the players to tout, but I'm sure we can find other selling points to market. James did say to aim for "foobar2000 in your pocket". They aren't going to achieve that with their current design priorities. Even JWolf right here in this thread was pm'ing me with some improvement requests to the UI of our scroll-wheel-based X players. I don't think his suggestions can be implemented with our current priorities.

Heaven forbid, perhaps we can even ditch native 24/192 and DSD playback in favor of customizable convolver-based binaural HRTF processing of all audio sources. God knows there are enough confused Chinese audiophiles out there (or paid anti-shills?) claiming that our players don't do native 24/192 and DSD as it is... :rolleyes:


----------



## nick_charles

mmerrill99 said:


> - the ABX home-run tests are severely flawed by being biased towards delivering a null result.


 
  
 Seriously, once again what is the basis of the above assertion, where is your supporting evidence ? Is there a credible source for this ? You keep dodging this and it is a relevant question here.


----------



## castleofargh

mmerrill99 said:


> And you accuse me of deflecting, the irony! Any idea if mainstream designers of audio devices use blind trading during development? No? I thought not!


 
 wow you really went down today. I just quote this, but you've been dancing and avoiding to the point where you don't seem to know which street you're in. 
 are electrical engineer making devices by ear? are they building stuff trying to just copy the audible sound of another device? so why the hell would they need a tool that is used to assess audible differences?
 but even then, even if they conducted all sort of blind tests with parts or whatever, do you expect that to be advertised? "buy our stuff, it sounds almost the same as that other brand". that's gotta work well for sure...
  
 but as this is a file format topic, I would actually be curious to know who developed a file format without blind testing it against other formats. finding one would probably be a real rare thing. lossy formats can't even hope to do without it.
  
 anyway you have such a biased idea about ABX it's incredible. you try to look like you're just stating the obvious, but you really act like a hater. 
 I'm sure that to you I'm the one who's biased, but when we tell you that from our own experiences, false negative are far from as massive as you think, you don't really care to find out for yourself or to offer some super serious research on the matter that shows how we should run away from abx.  you just want to say that it's bad. where is your claim coming from? the anecdote about Arny, that's about it right?
 when we tell you that a negative result does not make for evidence of anything, you know what we think better than ourselves and say that we still use it all the time as evidence. but we don't. this is such a strawman.  we use the negative results to make choices for ourselves. if I can't for the life of me pass a blind test between 16/44 and 24/96(except a few online rigged ones that I stopped passing as soon as the error was removed), then even if there really is a difference that can be heard, should I care? if I fail when I try so hard, does it still deserve for me to pay the added cost of highres? well I decide that it doesn't. others will decide that it doesn, in the end it's a decision made on several factors, that isn't worst or better than picking a desert on a menu. I just have one more non conclusive result to help me make my decision. I don't tell people not to buy hires, I don't claim that it's impossible to hear a difference. I just made what I believe to be a more informed decision than when I just looked at the files and pressed play after watching Neil Young's mix of BS and misunderstanding about highres and mp3.
 you made fun of Nick Charles for using epiphany. but I've had change of mind from negative results, I really don't see what's strange about that. when I was expecting such obvious positive results, a failure to pass forced me to reconsider if I wasn't just splitting hair and pretend as if it was a major issue. it's eye opener even if it doesn't change lives and doesn't make me sure of much. it still makes me sure that the difference I was expecting isn't as much of a big deal as I thought and certainly not a "night and day difference" so many people will surely claim hearing, whatever the subject being discussed. and then there are all the times I thought something was a big deal or wasn't, and resulted in a 100% positive ABX. because hey be amazed, that happens all the time. when the result isn't what is generally accepted, the very first idea I have is that I messed up the test. because I'm not that confident of a person and I mess up something at least once a week. so I look for what I could have done wrong, and sometimes I call others for help or try to make up another sort of blind test(like putting files in my DAP and playing them at random, or asking someone to make me pass a blind test, or to provide me with test files in case I messed up. etc etc. we try, we learn. we grow. you know, life. mistakes will be done, we get back up and try again.
 but when we tell you that with practice we learn from our mistakes and improve even our crappy home brewed abx(the exact same way I improve the way I do measurements, with trials and errors, and by confronting my results with more serious results from other origins, and then look for what I did wrong). you don't even want to know about it, you don't care that we're not using ABX to force opinions onto others, but to engage them in double checking their opinions for themselves. to you bad is bad and that's the end of it. but at the same time you're fine with people making purchase decisions purely based on uncontrolled sighted evaluation... so if both options exist, and both are crap, how come you only ever fight abx? why don't I see you outside of sound science telling people how they can't make claims about what they heard because it's an anecdote and it can't be proved? about how flawed it is and how they should stop and do... what? nothing, sit on a chair and wait, as you offer no answer and no tool to help people make decisions. no you're not telling those people anything, they do what they do, this is life. but writing tens of pages in here to try and kill home made abx, for that you have time.
 even if both subjective approaches have flaws, they happen to have opposite kind of flaws, so wouldn't it be obvious to test both on a matter instead of just one when wondering about a personal situation? you're just on your hate horse trying to pretend like you're fair about all of it, but this is super BS. and because I encourage people to try and learn more about abx and blind tests, I'm the one who's biased... yeah sure. I must be such a monster and a fool to tell people to multiply their experiences. I'll never stop suggesting to people to try abx and learn about it, not until I find a better easy to use solution for individuals who are curious.
  
  
  
 Quote:


jwolf said:


> watchnerd said:
> 
> 
> > Are you talking about a file bit depth or the bit depth of the DAC chipset?
> ...


 

 I didn't think I would come to see this day, but you're almost the only one still on topic, and more than that, your attempt at explaining resolution works fine. we'll still fight until the end of times about how audible all that is and how much is "enough", but at least the measurement side of things we now seem to be talking about the same kind of reality.


----------



## gregorio

jwolf said:


> There is more of a gap between the bits in a 16-bit audio file ....


 
   
 No there isn't. In practice, the gap is exactly the same, roughly 6dB per bit, regardless of 16bit or 24bit.
  
 Quote:


jwolf said:


> .... and more chance of something not being correctly rendered between those bits.


 
  
 No there isn't! In practice, a waveform can be reconstructed virtually perfectly, regardless of bit depth! According to your theory of how digital audio works, 16bit (CD) should sound hundreds of times more accurate than 1bit (SACD), which should sound dreadful due to the massive "gaps" and errors ... In practice, does it?
  
 The difference between your theory of how digital audio works and how it actually works in practice is, as spruce music pointed out, dither (specifically, noise shaped dither). Without dither, 1bit digital audio would still contain the information necessary to perfectly reconstruct the waveform but it would be swamped by noise and virtually unrecognisable. With dither, all that noise can be moved to a part of the audio spectrum where you can't hear it. All that's left (which is audible) is that virtually perfectly reconstructed waveform (without any noise/error). With 16bit, that noise is already slightly below audibility but the application of noise-shaped dither is standard practice, taking the noise well away from audibility (to approx -120dB). Again, all that's left (which is audible) is a virtually perfectly reconstructed waveform (without any noise/error).
  


jwolf said:


> I cannot say how true this is, but it's been said that we'd be best going from 16/44.1 to 20/88. That would give enough bits and enough sampling without being too large.


 
  
 It's not true! All sorts of things have "been said" about digital audio which are not true, either out of ignorance or the desire to sell snake oil products. 20bit produces it's digital noise down at -120dB, pretty much the identical same point as CD (with noise-shaped dither), so no difference whatsoever in practice! 88kHz vs 44.1 is a different issue but with essentially the same outcome, no audible difference!
  
 G


----------



## Ruben123

The heat here is almost hell like lol, I have been missing some important facts about ABX'ing and this discussion in general.
  
 1. If you are fully able to differentiate 24 bit vs 16 bit (or 192k vs 44k) with your eyes open, there should be no reason you instantly cannot anymore with your eyes closed.
  
 2. ABX tests can made as long as you want. If you want to listen to twelve 24 bit songs fully, and then the same twelve songs in 16 bit fully, and then "guess" the X, or if you want to quickly switch between 24/16 bit, it is up to you. Only important is that you do it the way it must be done, follow the test rules.
  
 3. ABX'ing to general people is odd. People should know how to spot differences and what the differences could be, so they are able to "look" for them. For instance: 128kbps mp3 vs 320 could sound very alike, but only cymbals or some drums could sound different. If people dont know that, they could not focus on those differences and may not know where to focus on in the music so that subtle differences are not spotted. People who are going to do the test need to be informed correctly.
  
 4. People willingly to believe that 16 bit sounds the same as 24 bit, _could_ take the test less seriously than people who swear by the differences. They could manipulate the test, even if they dont want to. Unfortunately that is the reality.
  
 5. Many objectivists (spelled correctly??) dont own state of the art gear anymore because they are sure that basic gear (no transducers) should be no different (or even better than) from very expensive gear. I am one of them, which fits me greatly because I am a student without too much money. *Maybe, *an expensive studio quality set of gears with your speakers/headphones can actually make that very very subtle difference in which you can hear very subtle differences._ I am not talking about the science behind that, because I am too sure that 16/44 must be enough (probably even less). _Just very small differences could be (I dont know) be a little bit better audible on a very expensive studio quality system, again not talking about transducers.
  
 This points are *only *for the sake of the arguments here. I am fully sure anything above CD quality is nonsense and audiophile orientated brands are only looking for more "ghosts" to buy their "now even better" gear. But keep in mind please, that doing an ABX on your laptop with a transparent audio card and good headphones, does not give 100% confidence about your findings because much more expensive gear exists. 
  
 Again, I am an objectivist, but this over and over arguing with the same arguments being repeated over and over again, it's now time to bring this thread to an end with a conclusion. If the two camps cannot agree, then so be it. I have seen so many arguments being repeated that it makes me tired to even give a #### about it. All we need is *one *final conclusion, which can be posted and *locked, *so that no new discussions about this topic are needed any more. The conclusion may very well include arguments from both camps, as long as they are found generally correct/valuable.


----------



## RRod

jwolf said:


> A bit of both. There is more of a gap between the bits in a 16-bit audio file and more chance of something not being correctly rendered between those bits. Also, if any of those bits are in error, there is more chance of what's reconstructed will; be in error. Wuth 24-bit, there are more bits so if a bit is wrong, then the error will be smaller as the next bit is a lot closer.
> 
> Take the set of numbers 4 8 12 16 20 and 2 4 6 8 10 12 14 16 18 20. Now in the first set of numbers if 12 is wrong then the wave form could be wrong from 8 to 16. In the second set of numbers if 12 is wrong, then the error could only be as large as 10 to 14. So the error in 16-bit would be larger than the error in 24-bit. That can change the sound a lot more in 16-bit than in 24 bit.
> 
> ...


 
  
 You are leaving out a couple of important factors: The signal being encoded and the environment where the signal will play. These work together to determine if a given amount of quantization error is audible. Was going to go into techy detail but first we would need to agree that such factors are significant.
  
 Also understand that we're pretty much already getting the benefits of higher bit and sample specs on the ADC side of things. Higher specs allows engineers to, essentially, make an ideal mapping down to 16/44.1. So the question remains why force consumer-side electronics to have to deal with more throughput when the additional material is unnecessary given our music, listening rooms, and ears?


----------



## castleofargh

don't go ruin wolfy's post. the wild general idea isn't false. it's not super clear(but being who I am, it would be ironic if I started to blame people for expressing themselves poorly ^_^), and obviously the jump from technical, measurable difference to audibility keeps annoying me. but IMO he clearly made a big jump toward understanding bit depth a little bit better(or am I just seeing too much into this post?).


----------



## RRod

castleofargh said:


> don't go ruin wolfy's post. the wild general idea isn't false. it's not super clear(but being who I am, it would be ironic if I started to blame people for expressing themselves poorly ^_^), and obviously the jump from technical, measurable difference to audibility keeps annoying me. but IMO he clearly made a big jump toward understanding bit depth a little bit better(or am I just seeing too much into this post?).


 
  
 I didn't say it was false, but just looking at quantization error in a vacuum doesn't really get to the heart of the matter. But yes, rounding is, well, rounding.


----------



## watchnerd

castleofargh said:


> don't go ruin wolfy's post. the wild general idea isn't false. it's not super clear(but being who I am, it would be ironic if I started to blame people for expressing themselves poorly ^_^), and obviously the jump from technical, measurable difference to audibility keeps annoying me. but IMO he clearly made a big jump toward understanding bit depth a little bit better(or am I just seeing too much into this post?).


 
  
 It isn't false (albeit poorly described), but the quantization error + dither is below the threshold of hearing and thus not a reason to say that 24bit audio is a better listening experience.
  
 So it's moot.


----------



## dprimary

jwolf said:


> The problem with an ABX test is that if you find in favor of 24-bit, then you get bombed with things like "that cannot be correct" and "please provide your source files so we can see what's going on". It's all nonsense when you get people saying things like that when you post the results not in 16-bit's favor.


 

 You should expect to be questioned on it and the people that are open to it are going to want to verify and repeat your results. When you are claiming the audio equivalent of sustained fusion in a lab, You are going to get many questions and many people just outright dismissing it.
  
 When ever I hear something I don't expect the first thing I think is something is wrong, I didn't calibrate it correctly, bad signal source, mis-wired, just plain broken. 99.99% of the time it is one of those. One test out of 10,000 I find something interesting, Those are the things I want to revisit when I have time to set up a well defined test. Even if I carve out the time it does not mean I can divert resources and people to run the tests. I good portion of the ones that I do look into testing I can't find the difference so there is no reason to look into it more. The short of it is I will be very skeptical at most claims, yes you may hear a difference 99.99% chance something is just plain broken. Much of the consumer equipment is just plain broken from a design standpoint, it does not interface properly to other equipment.
  
 On the professional, research, and manufacturing level this is all settled science. They have already done tens of thousands of blind tests. The manufacturers are not going to share any of their tests from product development ever, if you are in the industry and independent from a manufacturer  you might get invited in some late stage testing.


----------



## Joe Bloggs

dprimary said:


> When ever I hear something I don't expect the first thing I think is something is wrong, I didn't calibrate it correctly, bad signal source, mis-wired, just plain broken. *99.99% of the time it is one of those. One test out of 10,000 I find something interesting,* Those are the things I want to revisit when I have time to set up a well defined test. Even if I carve out the time it does not mean I can divert resources and people to run the tests. I good portion of the ones that I do look into testing I can't find the difference so there is no reason to look into it more. The short of it is I will be very skeptical at most claims, *yes you may hear a difference 99.99% chance something is just plain broken.* Much of the consumer equipment is just plain broken from a design standpoint, it does not interface properly to other equipment.




If someone like you documented all the times you screwed up an ABX test and produced a false positive, that would totally put the lie to mmerrill99's claim of home ABX tests being biased toward false negatives...


----------



## JWolf

rrod said:


> You are leaving out a couple of important factors: The signal being encoded and the environment where the signal will play. These work together to determine if a given amount of quantization error is audible. Was going to go into techy detail but first we would need to agree that such factors are significant.
> 
> Also understand that we're pretty much already getting the benefits of higher bit and sample specs on the ADC side of things. Higher specs allows engineers to, essentially, make an ideal mapping down to 16/44.1. So the question remains why force consumer-side electronics to have to deal with more throughput when the additional material is unnecessary given our music, listening rooms, and ears?


 
  
 But is it really unnecessary? Some would say no. Are they wrong? Not at all. Personally, I do think that a well mastered recording at 24/96 can sound very nice. But a lot of today's highly compressed/over produced/distorting recordings it makes not enough difference to matter.
  
 As far as errors go, when you do have errors, 24-bit has a better chance of recovery.


dprimary said:


> You should expect to be questioned on it and the people that are open to it are going to want to verify and repeat your results. When you are claiming the audio equivalent of sustained fusion in a lab, You are going to get many questions and many people just outright dismissing it.
> 
> When ever I hear something I don't expect the first thing I think is something is wrong, I didn't calibrate it correctly, bad signal source, mis-wired, just plain broken. 99.99% of the time it is one of those. One test out of 10,000 I find something interesting, Those are the things I want to revisit when I have time to set up a well defined test. Even if I carve out the time it does not mean I can divert resources and people to run the tests. I good portion of the ones that I do look into testing I can't find the difference so there is no reason to look into it more. The short of it is I will be very skeptical at most claims, yes you may hear a difference 99.99% chance something is just plain broken. Much of the consumer equipment is just plain broken from a design standpoint, it does not interface properly to other equipment.
> 
> On the professional, research, and manufacturing level this is all settled science. They have already done tens of thousands of blind tests. The manufacturers are not going to share any of their tests from product development ever, if you are in the industry and independent from a manufacturer  you might get invited in some late stage testing.


 
  
 Asked how you did it maybe. But the way it's said, it's like you've lied about the results.


----------



## RRod

jwolf said:


> But is it really unnecessary? Some would say no. Are they wrong? Not at all. Personally, I do think that a well mastered recording at 24/96 can sound very nice. But a lot of today's highly compressed/over produced/distorting recordings it makes not enough difference to matter.
> 
> As far as errors go, when you do have errors, 24-bit has a better chance of recovery.


 
  
 What do you mean by "when you do have errors"? There is always quantization error; the question is if you can hear it above the material and in your given listening environment.  24/96 can indeed sound nice, and that niceness is maintained if you turn it into 16/44.1 with any decent decimator. Because believe it or not, most music albums out there don't have a lot of musical content below -70dBFS RMS and as humans we still really don't hear >20kHz all too well.
  
 So do you or do you not agree that the nature of the signal and the listening environment affect how well we can hear quantization distortion?


----------



## JWolf

rrod said:


> What do you mean by "when you do have errors"? There is always quantization error; the question is if you can hear it above the material and in your given listening environment.  24/96 can indeed sound nice, and that niceness is maintained if you turn it into 16/44.1 with any decent decimator. Because believe it or not, most music albums out there don't have a lot of musical content below -70dBFS RMS and as humans we still really don't hear >20kHz all too well.
> 
> So do you or do you not agree that the nature of the signal and the listening environment affect how well we can hear quantization distortion?


 
  
 This we don't hear above 20kHz has to stop. It's not about hearing past 20kHz. It's about getting the best quality sound from the frequencies you do hear.


----------



## RRod

jwolf said:


> This we don't hear above 20kHz has to stop. It's not about hearing past 20kHz. It's about getting the best quality sound from the frequencies you do hear.


 
  
 Well never mind we're switching to sampling rate rather than bit depth. Where can you show that cutting out high frequency content reduces quality in the audible range?
  
 You didn't answer my previous question about quantization.


----------



## spruce music

jwolf said:


> This we don't hear above 20kHz has to stop. It's not about hearing past 20kHz. It's about getting the best quality sound from the frequencies you do hear.


 

 Why?  We don't hear above 20 khz.  Bit depth isn't related to that.


----------



## Ruben123

Guys please.
  
 1. We cant hear above 20khz
 and
 2. many transducers fail to provide a good frequency response past 14khz with great roll off near 18-20khz.


----------



## mmerrill99

spruce music said:


> mmerrill99 said:
> 
> 
> > I guess you have me in side B - so let me elucidate that position
> ...


Hold on - we are talking about ABX testing run at home. I've already explained my position - If a positive ABX result is returned (this means it's statistically significant so it's not random guessing) & the test & tester have been scrutinised/analysed (as happens 99.9% of the time) & nothing found to explain the positive result, then yea I accept it. This is not a bias - this is how science works  





> While paying no attention to null results.  The reverse would seem to be more likely without looking at any particular result or procedure. In other words, the chances of a poor procedure by an amateur netting positive results is more likely than if done according to professional guidelines because professionals are more likely to control for sources of bias.


We are talking about ABX testing & you analysis doesn't hold - if it did then we would see more positive ABX results than null results - null results greatly outnumber positive ABX results.



> You also seem to be making a mistake in how you view null results.  It is not uncommon.  A null result doesn't really prove something.  It simply means no results contrary to the null hypothesis were the result of a given procedure.  Subtle though important difference.


I know the meaning of a null result & it's not me that interprets it incorrectly - I view home run ABX null results as not very interesting, anecdotal reports - a very high chance that the test is flawed. It's others who attach unnecessary meaning to these null results, attaching some life changing epiphany to them or using the home run ABX test as a means of trying to prove people aren't "actually" hearing what they claim. If there was real intent to search for the truth people then when suggesting to run a home based ABX test, the ways to try to overcome the inbuilt null bias of the test would be spelled out at the same time. I've seldom seen this done which makes me believe that these people have either been duped themselves & believe the ABX test is not biased or they know it is biased to return nulls & use this bias to support their worldview



castleofargh said:


> I'm sure that to you I'm the one who's biased, but when we tell you that from our own experiences, false negative are far from as massive as you think


Ok, this is from your experience, not from anything within the test which can give any statistics of false negatives. So, tell me how your experience leads you to this conclusion?

If you understand the ABX test then you will realise that it's heavily geared towards preventing false positives - that is at it's core - anybody denying this really doesn't understand the test. The other side of the coin - trapping/recording or considering false negatives is completely ignored within the test. If you disagree then show how the ABX test addresses false negatives.

I referenced other blind test protocols, MUSHRA which deal with false negatives with hidden anchors & references. Why do you think they do this? For fun? 

I understand your logic about try as hard as you can in an ABX test & you still can't hear any difference between RB & high-res so why should you be bothered with buying high-res? Sure but there are many questions that this gives rise to - why are you doing a ABX test - have you heard a difference in sighted listening that you want to "prove". What did you hear & have you isolated a particular section of audio that reveals the difference? Have you then trained yourself to routinely be able to identify this difference? If you can't answer yes to all these questions then you will almost definitely return a null result. That's why I linked to UMS's posts - it clearly shows that a specifically identified difference needs to be identified, isolated & self-training done until it can be identified routinely, prior to any blind test - otherwise you are doomed to hear no difference in the ABX test. As I said many times before - just going into a blind test with the expectation that any audible difference will jump out & make itself known to you, would only occur if the differences were so gross that they immediately attracted your attention. This doesn't happen with differences that aren't of a gross nature.

The lack of understanding of perceptual testing & the resultant lack of preparation needed for a valid ABX test is the main reason why Ihome based ABX testing is a joke. But it's not a joke - it 's a trick that many people are duped by & they think it s really showing them what they "actually hear" - it isn't



> why don't I see you outside of sound science telling people how they can't make claims about what they heard because it's an anecdote and it can't be proved?


Because I treat them as anecdotes & use my common sense to judge if I will investigate it myself - this is what makes sense to me. I don't expect or suggest that people prove to me what they hear - I judge my own hearing in my own system in the full knowledge that it's just my perception at this point in my journey & this can change as my system changes & reveals more or my identification of audio differences improves as I get more exposure to these differences





> I encourage people to try and learn more about abx and blind tests, I'm the one who's biased... yeah sure. I must be such a monster and a fool to tell people to multiply their experiences. I'll never stop suggesting to people to try abx and learn about it, not until I find a better easy to use solution for individuals who are curious.


So what have you learned about ABX & the pitfalls that commonly occur in such testing? Have you ever prescribed any procedures about identifying isolating, & training in specific differences in order that they can even begin to do a blind test which has any semblance of being valid?

Have you read MUSHRA? What did you understand about hidden references & anchors & their use in blind testing? What purpose do they have? Why are they needed? 

BTW, I'm sorry if I appear to be fair about all of this - I know it doesn't marry with your expectation bias i.e. anybody who points out the weakness of ABX testing is really a "hater of ABX", just pretending to be fair

I thought this was the sound science section of the forum? I had hoped that objectivity would be the predominant attitude here - not some emotionally charged accusations of a dubious nature?

However, I know that what I'm saying is as welcome as a fart in a spacesuit & some cherished sacred cows are being questioned but as i said, skepticism is a fundamental building block of scientific enquiry & investigation.


----------



## mmerrill99

joe bloggs said:


> dprimary said:
> 
> 
> > When ever I hear something I don't expect the first thing I think is something is wrong, I didn't calibrate it correctly, bad signal source, mis-wired, just plain broken. *99.99% of the time it is one of those. One test out of 10,000 I find something interesting,* Those are the things I want to revisit when I have time to set up a well defined test. Even if I carve out the time it does not mean I can divert resources and people to run the tests. I good portion of the ones that I do look into testing I can't find the difference so there is no reason to look into it more. The short of it is I will be very skeptical at most claims, *yes you may hear a difference 99.99% chance something is just plain broken.* Much of the consumer equipment is just plain broken from a design standpoint, it does not interface properly to other equipment.
> ...




Nope, I always said that positive ABX results usually get scrutinised/analysed in depth & if no reason is found for the positive result, then it isn't a false positive. So because of this very analysis we uncover false positives i.e they re eliminated from the valid results. 

The opposite is not true however - null results are not scrutinise/analysed & therefore we have no idea of the quantity/regularity of their occurrence - we have no measure of this!


----------



## nick_charles

mmerrill99 said:


> The opposite is not true however - null results are not scrutinise/analysed & therefore *we have no idea of the quantity/regularity of their occurrence* - we have no measure of this!


 
  
 Well, the tests we have been discussing therefore cannot by definition be said to be biased toward null results according to your own words.
  
 You have repeatedly said that such tests are biased to produce null results(paraphrase) but now you say "*we have no idea of the quantity/regularity of their occurrence"*
  
 Oh and you still have not answered my question regarding how you arrived at the conclusion that these test are _biased towards null results(paraphrase) _ , this is a genuine question and I am genuinely interested at how your world view was arrived at.


----------



## gregorio

jwolf said:


> [1] This we don't hear above 20kHz has to stop. [2] It's not about hearing past 20kHz. [3] It's about getting the best quality sound from the frequencies you do hear.


 
  
 1. Agreed. ... Why then are you bringing it up?
 2. If you're bringing up sample rates above 44.1/48 ... Hearing past 20kHz is the ONLY thing you are talking about!!
 3. Agreed. Those frequencies you can hear (which are below 20kHz) are just as perfectly reproduced at 44.1kHz (and with 16bits) as they are at 96kHz and incidentally, theoretically slightly better than they are at 192kHz.
  
 G


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > The opposite is not true however - null results are not scrutinise/analysed
> ...



Simple - At the test level, the more you tighten up on trapping false positives, false negatives become more likely. ABX testing is solely concerned with eliminating false positives

But I've already given you examples of why a trial (or number of trials) could produce a false negative - lack of focus/concentration. tiredness, lack of preparation prior to test in identifying, isolating a specific difference to focus on during the test, lack of self-training in differentiating these differences until they are regularly & routinely identified.

As I have said over & over - look at UMS's posts - his descriptions of doing the test very clearly state that if he drifts from the focus of listening for the specific "tell" he has already identified & trained himself to recognise, he gets random results. In other word if he listens in his normal way he gets random results i.e a null

But can you answer the same question I asked Casteofargh? Have you read MUSHRA? What did you understand about hidden references & anchors & their use in blind testing? What purpose do they have? Why are they needed?


----------



## nick_charles

Quote:


mmerrill99 said:


> Simple - the more you tighten up on trapping false positives, false negatives become more predominant. ABX testing is solely concerned with eliminating false positives


 
  
 You have still not answered my question : How did you arrive at the conclusion that ABX testing is biased to produce null results - ignore false positives I did not ask you about false positives at all - forget false positives completely -  why do you conclude that ABX tests are biased to produce null results  especially as you now say *we have no idea of the quantity/regularity of their occurrence* (wrt nulls)  at least please please please give me a citation I can go away and read !
  
 And stop using _false negatives_ again


----------



## mmerrill99

nick_charles said:


> mmerrill99 said:
> 
> 
> > Simple - the more you tighten up on trapping false positives, false negatives become more predominant. ABX testing is solely concerned with eliminating false positives
> ...


 I'm talking about home run ABX tests & by now, I believe I've explained it often enough my reasons for my conclusions.
At this stage we can just agree to disagree perhaps?


----------



## sonitus mirus

mmerrill99 said:


> I'm talking about home run ABX tests & by now, I believe I've explained it often enough my reasons for my conclusions.
> At this stage we can just agree to disagree perhaps?


 
  
 The reasons you have provided to form your conclusion seem illogical to me.


----------



## mmerrill99

sonitus mirus said:


> mmerrill99 said:
> 
> 
> > I'm talking about home run ABX tests
> ...


Fair enough but obviously they seem very logical to me & those on the fence can decide for themselves. It might, at least, bring a bit more appreciation of the pitfalls of home run ABX testing?


----------



## sonitus mirus

mmerrill99 said:


> Fair enough but obviously they seem very logical to me & those on the fence can decide for themselves. It might, at least, bring a bit more appreciation of the pitfalls of home run ABX testing?


 
  
 Yes, in this I agree.  When I first attempted the Tidal music subscription listening test, I was in my office at work and didn't believe I'd hear any difference between AAC 320 and FLAC.  I did not pass the test.  Only after I read about the Tidal test having audio issues that were measurable was I then able to take the test more seriously and successfully ABX it.   In some situations, my own bias can prevent me from passing an ABX.


----------



## MrLolendo

Hi,
 interesting discussion but very stupid. 
 Probably no one here knows how is working modulation.
 These frequency 44,1KHz - 192KHz are sampling rate! Not frequency bandwidth!
 Of course there is no point to have any sounds beyond 20KHz because human ear will not listen.
 Pls read something about Pulse-code modulation...
  
 Probably quality 24bit with sampling rate 192KHz is for listener useless because this quality is to high for human ears and brain. But wait. Why is this high quality even exist?
 Answer is simple. In digital world is always better to have information at the best possible quality because you are working with that. Artists has that quality for working purposes when they making changes. It always better to make changes at higher resolution picture than low-res. Or not?!   
  
 Marketing is that. If you have sound card with specification 24bit 192KHz but sound card only can working at that quality bud not really reproduce it. There need to be DAC that can reproduce it to analog signal to yours headphones. But of course that information is never written. head-fi.org right?!  It is like posting spec. power out of speakers. It is just stupid. But that is for different discussion.


----------



## dprimary

nick_charles said:


> Well, the tests we have been discussing therefore cannot by definition be said to be biased toward null results according to your own words.
> 
> You have repeatedly said that such tests are biased to produce null results(paraphrase) but now you say "*we have no idea of the quantity/regularity of their occurrence"*
> 
> Oh and you still have not answered my question regarding how you arrived at the conclusion that these test are _biased towards null results(paraphrase) _ , this is a genuine question and I am genuinely interested at how your world view was arrived at.


 
 Well is it _absence of evidence_ or an _evidence of absence. _We can measure resolutions and bandwidths humans do not seem to have the ability to detect. Unless you have a control test that you know the average middle aged person should be able to detect, I would expect most of the results to be nulls. Is that even a good idea? It is hard to say if the first test is so obvious will they try harder or less on the critical test when they really have to listen?


----------



## spruce music

dprimary said:


> Well is it _absence of evidence_ or an _evidence of absence. _We can measure resolutions and bandwidths humans do not seem to have the ability to detect. Unless you have a control test that you know the average middle aged person should be able to detect, I would expect most of the results to be nulls. Is that even a good idea? It is hard to say if the first test is so obvious will they try harder or less on the critical test when they really have to listen?


 

 I have seen some online tests where controls were used.  A small loudness difference not heard as loudness or a frequency response tilt not terribly noticeable.  The test takers detected those at or very near 100 % while getting null results on other parts of the test.  Such controls are certainly a very good idea.
  
 The idea home ABX may not have all the controls and therefore is no more convincing than sighted anecdote is of course ridiculous.  Simple blinding removes quite a bit of bias inducing elements that are known for certain to have an effect in sighted listening.


----------



## dprimary

> Originally Posted by *gregorio* /img/forum/go_quote.gif
> 
> No there isn't. In practice, the gap is exactly the same, roughly 6dB per bit, regardless of 16bit or 24bit.
> 
> ...


 
 Can this be put in the intro page of the science forum? It is not pointed out enough. If you want to surprise someone play them properly dithered 8bit audio, many that work in audio have preconception of what 8 bit sounds like. You play them something lacking the grit and graininess they expect just a higher level of noise (about that of a poor cassette)  and they are shocked.
  
 I think one of the engineers for Harman or Rane came up with a good saying for bit depth. "24 bit does not give headroom it gives foot-room" 0dBfs is the same at 4 bit, 8 bit, 16 or 24 bit, it gives you more room before you get down into the noise. There is not many places on the planet that ambient noise is lower then 16 bits of resolution.


----------



## castleofargh

mmerrill99 said:


> Spoiler: Warning: Spoiler!
> 
> 
> 
> ...


 
  
 I don't see how this goes against abx. it's like specs, because I can get some specs, or measure some stuff myself, I will still want to listen to the files like everybody else, and I may also want to try a few things, like rapid switching, listening to a full song critically, just let the album pass while doing something else, in case something may jump out of it as "annoying" or whatever...  including abx. and yes ABX is mainly used to confirm a difference. if it's enough for me to pass then I may consider. if not, then I conclude that it's no big deal. even if there may, of course, be a small difference that I failed to notice because the moon is up, I'm tired or whatever. to me it doesn't really matter, if it doesn't jump at my face in an abx, then it's a small stuff and that's what I wanted to make sure of. again I do not take a failed test as proof of no difference, ABX cannot allow that. I guess it's really about how you look into those tests. to me, it's the first test that tells me if a matter is worth investigating for my use.
 now about home made ABX, as it happens my own experience is that mistakes led me to hearing differences that weren't there plenty of times(I do try to use a control file or 2 that I also abx, it's not amazing as I know what I'm testing, but sometimes it lets me realize that the conversion made an audible difference. or that I wrongly renamed the files(no jokes I did that sooooo many times... blind test by stupidity). and plenty of silly little mistakes outside of the usual recommendations about how to test 2 formats(like convert them back to a common high format so that what is tested is the content and not the way the computer or DAC decodes 2 formats, and stuff like that). to mess up something in the protocol that leads to 2 identical sounds is kind of unlikely, most of the time mistakes lead to differences(which is to be expected). it happened to me when testing my EQ though. I was settings and if they mattered to me(linear, minimum phase....) where I ended up testing nothing because I had the stuff set as bypass like a noob. I noticed because I checked the files with diffmaker. else I would have indeed just made the wrong observation.
 anyway, control and multiplication of the tests are always good ideas, you will never see me going against that or claiming that my tests are perfect. but they are also not useless.
  
 also I'm all for better tests than ABX. I never thought ABX was all I will ever need. I'm kind of curious so I tend to try a little bit of everything I can (within my equipment limits sadly). when I get a new DAP, I always make a few folders with the highres version, the 16/44 conversion from it, and the that converted back to 24/96 and/or one as mp3 max VBR if I also want to test for the mp3 decoding. and instead of an immediate test, I just play that in shuffle(alone or with other albums) and try to guess what I'm listening to from time to time, and I note my result. that goes on without too much care for a week or 2. sometimes I just leave it there and keep on wondering from time to time when I remember to do it. it's also not a very meaningful test, full of flaws. but it's different from just vague opinion about my listening that just ends up being me agreeing with my preconceptions. I do what I can with what I have.
 sometimes I bother family or friends for a slightly better blind test. whatever goes.
 but killing ABX because there is MUSHRA, that's like saying that blind test is useless because we have double blind test. sure it's superior, sure it leads to better conclusions that's very obvious. but when I ask someone to measure a desk I want to buy, I don't require laser measured precision. a simple ruler can very much be of use as long a the accuracy needed isn't extreme.
 how am I supposed to set up a good mushra test or a double blind test? I'm the only guy around who knows how to prepare the tests, and I don't care too much that some other guys could pass my tests. what I want is to test my hearing.
  
 but that's me, I don't use my ABX results to prove anything, I don't even post my ABX results. I see that as a personal stuff like an aid to my opinion. but if something I can obviously identify was claimed not to be audible on headfi, then I would off course use my positive ABX results as evidence that I could hear a difference. and that obviously would lead to some people asking me for my results, maybe my files and what I used to convert them. and if I messed up, they will make me aware of it hopefully. and if I didn't, then yes indeed I will have brought evidence of an audible difference. even more so when other people will have also achieved positive results(because if I can, a all lot of people can, I'm no golden ear).
  
 so how useful and reliable is abx? it depends. it's a tool, some use it better than others, some just don't see a need for it. that's fine. but it can definitely be of use and is available to anyone a little curious. unlike a MUSHRA test or whatever that most people wouldn't even know how to prepare. those indeed I leave to people who know what they are doing, because they need expertise. an ABX test, the only expertise needed is not to draw stupid conclusions about a negative result. for the rest, you can just ask online for someone to make the files for you, and then discuss if something was wrongly done. if people weren't so touchy about pride and social rightness, we wouldn't ever have those arguments about why we only annoy people with positive results. it's the logic thing to do and the guy put under scrutiny should be happy to have so many people coming to help him make sure he didn't waste his time on his test.


----------



## castleofargh

dprimary said:


> > Originally Posted by *gregorio* /img/forum/go_quote.gif
> >
> > No there isn't. In practice, the gap is exactly the same, roughly 6dB per bit, regardless of 16bit or 24bit.
> >
> ...


 

 I guess it may be a little too specific, but you guys can harass currawong and maybe he's do it (I no tengo la magic power).
 is an experiment like this http://www.computeraudiophile.com/blogs/mitchco/16-44-vs-24-192-experiment-163/ flawed when it comes to demonstrate how increase bit depth does not alter the first 16bit(minus dither and stuff)?  just the diffmaker part. I don't understand what the software does well enough to be sure if it's a conclusive demonstration?


----------



## OddE

> Originally Posted by *dprimary* /img/forum/go_quote.gif
> There is not many places on the planet that ambient noise is lower then 16 bits of resolution.


 
  
 -Definitely not when proper noise shaping is employed as well!
  
 The thermal noise level at room temperature inherently present in 20kHz of bandwidth would (in a 2.2kOhm resistor, just for argument's sake) be approx. -120dBu. (Which, in a Benchmark DAC2, is approx. -144dB(FS) - the potential dynamic range of an undithered 24 bit signal)
  
 To take full advantage of 24 bits of dynamic range (if it did even really exist in recorded music), you'd need a very, very cold listening room, then - oh, and you'd need to be dead. (When in a quiet room, you will start hearing the noise of your body - blood flow, most notably - quite audibly, seemingly loud, even!)
  
 Not to mention of course, that even if this vast dynamic range was available to you, your ears - and no transducer I know of, short of a field howitzer - still wouldn't be up to it. Details, schmetails.
  
 But I digress. That's what you get for being bored while waiting for morning coffee.


----------



## gregorio

mrlolendo said:


> These frequency 44,1KHz - 192KHz are sampling rate! Not frequency bandwidth!


 
  
 Correct. Frequency bandwidth is roughly half the sampling rate. So with a sampling rate of 44.1kHz we've got a frequency bandwidth of roughly up to 22kHz.
  


mrlolendo said:


> Probably quality 24bit with sampling rate 192KHz is for listener useless because this quality is to high for human ears and brain.


 
  
 This is not correct though! 24bit/192 is NOT higher quality than 16/44.1, it is at best exactly the same quality! Those extra bits and additional samples do NOT increase quality, they maintain exactly the same quality but over a larger area. In the case of the sampling rate this "larger area" means that frequencies between 22kHz and 96kHz can be recorded/reproduced with the same quality as a 44.1kHz sampling rate can record/reproduce audio frequencies up to 22kHz. In the case of 24bit this "larger area" means that (in theory) we have a dynamic range which extends down to -144dB rather than the -120dB of 16bit (with noise-shaped dither). ... The big mistake made by many who are ignorant of how digital audio works is believing that doubling (or quadrupling) the sample rate doubles (or quadruples) audio bandwidth AND somehow magically also increases quality. There is no additional magic, there is only proven science!
  


> Originally Posted by *MrLolendo* /img/forum/go_quote.gif
> 
> But wait. Why is this high quality even exist?


 
  
 As I've just explained, there is no higher quality (within the hearing spectrum) but the reason it exists is purely and simply marketing. The vast majority of consumers are ignorant of how digital audio works in practice and therefore it's easy to fool them into believing 192/24 is higher quality and therefore consumers can be charged more for 192/24 content and the hardware and the services capable of reading content in that format. Everyone within the industry who markets to consumers wins, the only losers are the engineers, who don't get to charge more and have to live with the fact they are producing snake oil and of course the consumers, who are being scammed!
  


> Originally Posted by *MrLolendo* /img/forum/go_quote.gif
> 
> In digital world is always better to have information at the best possible quality because you are working with that. Artists has that quality for working purposes when they making changes. It always better to make changes at higher resolution picture than low-res. Or not?


 
  
 Ironically, 24/192 is actually slightly lower resolution (within the audible spectrum) and therefore no knowledgeable recording, mixing or mastering engineer would choose 24/192. However, that decision is generally not the engineers' choice, it's the decision of their clients who are either Artists (who generally have a poor understanding of how digital audio works) or record labels, who may or may not have a good understanding of how digital audio works but even if they do, it often doesn't matter because they are commonly far more interested in marketing potential than in audio quality/fidelity. The stated "fact" that 24/192 is the studio reference standard is at best a bending/perversion of the truth and at worst a bald-faced lie with the sole intention of scamming consumers!
  


dprimary said:


> If you want to surprise someone play them properly dithered 8bit audio, many that work in audio have preconception of what 8 bit sounds like. You play them something lacking the grit and graininess they expect just a higher level of noise (about that of a poor cassette)  and they are shocked.


 
  
 Agreed. And obviously, an even bigger shock to their preconceptions would be properly noise-shaped dithered 8bit audio, which would not only lack the grit and graininess they expect but would also have little/no audible noise!
  
 G


----------



## MrLolendo

gregorio said:


> Correct. Frequency bandwidth is roughly half the sampling rate. So with a sampling rate of 44.1kHz we've got a frequency bandwidth of roughly up to 22kHz.


 
 Sampling frequency need to be at least double than frequency source because of aliasing. That is fact. But it does not mean that source signal has always half bandwidth. It could be quarter or less. For example Source signal could has only 100Hz-1KHz bandwidth. Higher sampling frequency = more information to reproduce with AD/DA conversion. MP3 for example is using Sampling frequency dynamical to save space. You do not need to quantize lower bass frequency with 44.1kHz at 16 bit. That is part of smart convert algorithm. It is like "stereo" vs. "join stereo".
 MP3 songs are mostly cut at 16 KHz frequency. For most people it does not really matter because they do not hear anything above 16 KHz at reasonable decibels.


----------



## sonitus mirus

dprimary said:


> Can this be put in the intro page of the science forum? It is not pointed out enough. If you want to surprise someone play them properly dithered 8bit audio, many that work in audio have preconception of what 8 bit sounds like. You play them something lacking the grit and graininess they expect just a higher level of noise (about that of a poor cassette)  and they are shocked.
> 
> I think one of the engineers for Harman or Rane came up with a good saying for bit depth. "24 bit does not give headroom it gives foot-room" 0dBfs is the same at 4 bit, 8 bit, 16 or 24 bit, it gives you more room before you get down into the noise. There is not many places on the planet that ambient noise is lower then 16 bits of resolution.


 
  
 This has been posted before, but at around 4:30 into the video, various bit depths are demonstrated. (4bit, 8bit, 16bit)  The entire video is only 11 minutes long and easy to understand.
  
 https://www.youtube.com/watch?v=nLEhfieoMq8


----------



## watchnerd

Rhetorical question:
  
 If the quality difference between standard resolution and high resolution is so audible, why does Pono use "listening tests" involving celebrities in cars comparing vs MP3?  Or why does Tidal tweak the samples in their test?
  
 If I was a marketer working for either company and there was strong evidence of an audible difference I would use it rather than playing silly games like the above.  The fact that they don't is telling.


----------



## sonitus mirus

watchnerd said:


> Rhetorical question:
> 
> If the quality difference between standard resolution and high resolution is so audible, why does Pono use "listening tests" involving celebrities in cars comparing vs MP3?  Or why does Tidal tweak the samples in their test?
> 
> If I was a marketer working for either company and there was strong evidence of an audible difference I would use it rather than playing silly games like the above.  The fact that they don't is telling.


 
  
 Same for Schiit's policy of not touting about audible quality differences between their gear.


----------



## RRod

gregorio said:


> Ironically, 24/192 is actually slightly lower resolution (within the audible spectrum) and therefore no knowledgeable recording, mixing or mastering engineer would choose 24/192. However, that decision is generally not the engineers' choice, it's the decision of their clients who are either Artists (who generally have a poor understanding of how digital audio works) or record labels, who may or may not have a good understanding of how digital audio works but even if they do, it often doesn't matter because they are commonly far more interested in marketing potential than in audio quality/fidelity. The stated "fact" that 24/192 is the studio reference standard is at best a bending/perversion of the truth and at worst a bald-faced lie with the sole intention of scamming consumers!


 
  
 What do you mean by lower resolution?


----------



## gregorio

rrod said:


> What do you mean by lower resolution?


 
  
 Very high sample rates (rates >96/24) have a number of issues, all related to the fact that not only is there far more data to process with these very high sample rates but there's also far less time available to execute that processing. The first issue is when recording and the processing which takes place in the ADC: As you're obviously aware, an anti-alias filter has to be implemented to comply with Nyquist and avoid loss of resolution due to distortions (side-band images) being allowed into the system/recording. The digital anti-alias filters employed in pro ADCs during the decimation process typically achieve -120dB in the stop band (frequencies above the Nyquist point). However, at very high sample rates there is not enough processing time available to implement such a filter and attenuation above the Nyquist point is limited to about -80dB. In theory, this could potentially be audible but in practice is unlikely to be, nevertheless, it does technically/theoretically represent a "loss of resolution". During mixing, plugin processors also have to deal with this "far more data" and far less time to process it issue. Some more complex plugins simply cannot deal with this issue and avoid it by simply not supporting sample rates above 96kHz, others will operate at very high sample rates but often at the expense of some resolution/sound quality. Again, nothing particularly audible (generally) but the use of tens or even hundreds of plugins is common during mixing and therefore the potential exists for a cumulative effect, which obviously increases the likelihood of audible artefacts. Audible or not, this does again technically represent a reduction in resolution (relative to lower sample rates). I of course make no claims as to whether individual end listeners/consumers will be able to detect this reduction in resolution and even if they can, that they would necessarily recognise it as a loss in fidelity.
  
 In addition to all the above is the increased likelihood of causing IMD and also, although it doesn't directly affect resolution, of course DAW system resources are cut by half (or 4 times), thereby potentially reducing recording and mixing options. For all these reasons combined, no knowledgeable, self-respecting pro audio engineer (given the choice), would choose 192/24 (or 176.4/24).
  
 G


----------



## OddE

gregorio said:


> Very high sample rates (rates >96/24) have a number of issues, all related to the fact that not only is there far more data to process with these very high sample rates but there's also far less time available to execute that processing.


 
  
 -Audio applications do not utilize very high sample rates. (I currently have a couple of 210Msps/16-bit ADCs on my desk, part of a SDR project. That is a very high sampling rate.)


----------



## MrLolendo

Today's we have more than enough processing power for sound. *Gregorio *you are writing like we was at year 1995. More data = more quality and more option for adjustments. That is it. It is like 4K video, DTS-HD, games and everything digital. But one things is always important. If you have capable DA/AD converter and of course type like *direct-conversion ADC, successive-approximation ADC, ramp-compare ADC, integrating ADC, delta-encoded ADC, ...*


----------



## BiggerHead

It's a little similar to 4k in that you cannot see a difference in 2k vs 4k and to that extent they are both useless (I think the video situation is closer to marginal).
  
 It's not quite the same as 4k in that  4k video, does NOT mean more data.  Lossless compressed 4k even at 24 fps, is close to 1TB an hour!  Regular folks are not filling 4k bandwidth this year, which is just as well, since the only place they would be able to see the difference is still their wallet.


----------



## watchnerd

mrlolendo said:


> Today's we have more than enough processing power for sound. *Gregorio *you are writing like we was at year 1995. More data = more quality and more option for adjustments. That is it. It is like 4K video, DTS-HD, games and everything digital. But one things is always important. If you have capable DA/AD converter and of course type like *direct-conversion ADC, successive-approximation ADC, ramp-compare ADC, integrating ADC, delta-encoded ADC, ...*


 
  
 No, it's nothing at all like video.  
  
 For Nyquist to work, it has to be bandpass limited (i.e. has a cutoff at both ends).  Once your sample rate is 2x the highest frequency, you have enough to to completely create the original waveform.
  
 You don't get more detail or "rounder" analog waves by increasing the sample rate.  All you do is increase the top end range to allow for better filtering.


----------



## watchnerd

biggerhead said:


> It's a little similar to 4k in that you cannot see a difference in 2k vs 4k and to that extent they are both useless (I think the video situation is closer to marginal).
> 
> It's not quite the same as 4k in that  4k video, does NOT mean more data.  Lossless compressed 4k even at 24 fps, is close to 1TB an hour!  Regular folks are not filling 4k bandwidth this year, which is just as well, since the only place they would be able to see the difference is still their wallet.


 
  
 No, it's nothing like video.
  
 A 5khz sine wave is reproduced just as perfectly with a 44.1khz, 48khz, 88.2khz, 96khz, 176.4khz, or 192khz sample rate. To quote Lavry's paper "*The Optimal Sample Rate for Quality Audio*":
  
_"The sampling concept allows perfect capture and reconstruction of the signals as long as the sampling rate is at least twice as fast as the highest frequency contained in the signal to be recorded. There are no restrictions regarding the time at which the samples are taken, as long as the time interval between each adjacent sample is the same. When sampling (and reconstructing) stereo or surround, all the tracks are sampled simultaneously; which ensures that the proper time relationship between channels is maintained. The sample rate does not have ANY impact on the signal timing."_
  
 We're all so used to technology always making stuff better that it's hard for us to realize that audio is computationally easy and that the question of sample rates and bit depth is solved.
  
 Lavry advocates for a 60 khz sample rate just to make sure there isn't any cascading rolloff at the very tippy-top of the spectral range (not because it gives more detail) if you have a lot of AD to DA acting in a chain:
  
_"Good conversion requires attention to capturing and reproducing the range we hear while filtering and keeping out energy in the frequency range outside of our hearing. At 44.1 KHz sampling the flatness response may be an issue. If each of the elements (microphone, AD, DA and speaker) limit the audio bandwidth to 20 KHz (each causing a 3dB loss at 20 KHz), the combined impact is -12dB at 20 KHz.At 60 KHz sampling rate, the contribution of AD and DA to any attenuation in the audible range is negligible. Although 60 KHz would be closer to the ideal; given the existing standards, 88.2 KHz and 96 KHz are closest to the optimal sample rate. At 96 KHz sampling rate the theoretical bandwidth is 48 KHz. In designing a real world converter operating at 96 KHz, one ends up with a bandwidth of approximately 40 KHz."_
  
 As for bit depth, 16 bit dithered = 20 bit = 120dB dynamic range / quantization noise floor, which is plenty.  24bit, with no dither = 144 dB dynamic range, which is absurdly huge.


----------



## RRod

gregorio said:


> Very high sample rates (rates >96/24) have a number of issues, all related to the fact that not only is there far more data to process with these very high sample rates but there's also far less time available to execute that processing. The first issue is when recording and the processing which takes place in the ADC: As you're obviously aware, an anti-alias filter has to be implemented to comply with Nyquist and avoid loss of resolution due to distortions (side-band images) being allowed into the system/recording. The digital anti-alias filters employed in pro ADCs during the decimation process typically achieve -120dB in the stop band (frequencies above the Nyquist point). However, at very high sample rates there is not enough processing time available to implement such a filter and attenuation above the Nyquist point is limited to about -80dB. In theory, this could potentially be audible but in practice is unlikely to be, nevertheless, it does technically/theoretically represent a "loss of resolution". During mixing, plugin processors also have to deal with this "far more data" and far less time to process it issue. Some more complex plugins simply cannot deal with this issue and avoid it by simply not supporting sample rates above 96kHz, others will operate at very high sample rates but often at the expense of some resolution/sound quality. Again, nothing particularly audible (generally) but the use of tens or even hundreds of plugins is common during mixing and therefore the potential exists for a cumulative effect, which obviously increases the likelihood of audible artefacts. Audible or not, this does again technically represent a reduction in resolution (relative to lower sample rates). I of course make no claims as to whether individual end listeners/consumers will be able to detect this reduction in resolution and even if they can, that they would necessarily recognise it as a loss in fidelity.
> 
> In addition to all the above is the increased likelihood of causing IMD and also, although it doesn't directly affect resolution, of course DAW system resources are cut by half (or 4 times), thereby potentially reducing recording and mixing options. For all these reasons combined, no knowledgeable, self-respecting pro audio engineer (given the choice), would choose 192/24 (or 176.4/24).
> 
> G


 
  
 I guess I would find this surprising, especially given that DVD-A came out in 2000, so we've had at least 15 years to get ADCs to work well at 24/192 (or the necessary sigma-delta equivalent). Same with DAWs, especially for non-realtime applications. Still as you say, there's seems little reason to operate beyond 96ksps. Do you know of any specific examples comparing a signal split into ADCs at 96 and 192?


----------



## watchnerd

rrod said:


> I guess I would find this surprising, especially given that DVD-A came out in 2000, so we've had at least 15 years to get ADCs to work well at 24/192 (or the necessary sigma-delta equivalent). Same with DAWs, especially for non-realtime applications. Still as you say, there's seems little reason to operate beyond 96ksps. Do you know of any specific examples comparing a signal split into ADCs at 96 and 192?


 
  
 There is no engineering reason to invest R&D to make ADC "better" at 24/192 when the math of 24/96 is already more than good enough.
  
 You're better off spending the money making the ADC as good as possible in the 44-96 range.


----------



## RRod

watchnerd said:


> There is no engineering reason to invest R&D to make ADC "better" at 24/192 when the math of 24/96 is already more than good enough.
> 
> You're better off spending the money making the ADC as good as possible in the 44-96 range.


 
  
 The question isn't about making them better, it's if they are demonstrably *worse* at 192, which would indeed be a great thing to point out to people making claims that "bigger is always better." I'm just asking for an example, because as others have said there is hardware out there that runs at even higher effective bit rates.


----------



## gregorio

odde said:


> (I currently have a couple of 210Msps/16-bit ADCs on my desk, part of a SDR project. That is a very high sampling rate.)


 
  
 I'm not sure what application you're using those ADCs for. Are you saying that you can digitally process 16bits of data at a sample rate of 210mHz accurately? Say, accurately decimate that signal to 192/24 with appropriate anti-alias filters? Most pro studio converters operate at sample rates more like 11mHz or so but only with a handful of bits.
  


mrlolendo said:


> Today's we have more than enough processing power for sound.


 
  
 It's got nothing to do with computing power and everything to do with raw single chip computing speed, which has not changed significantly in many years.
  


mrlolendo said:


> More data = more quality and more option for adjustments.


 
  
 Congratulations for proving the already proven Nyquist-Shannon theorem is actually incorrect. I take it you do actually have such a proof and are not just making it up as you go along?
  
 G


----------



## OddE

odde said:


> -Audio applications do not utilize very high sample rates. (I currently have a couple of 210Msps/16-bit ADCs on my desk, part of a SDR project. That is a very high sampling rate.)


 
  


gregorio said:


> I'm not sure what application you're using those ADCs for. Are you saying that you can digitally process 16bits of data at a sample rate of 210mHz accurately? Say, accurately decimate that signal to 192/24 with appropriate anti-alias filters? Most pro studio converters operate at sample rates more like 11mHz or so but only with a handful of bits.


 
  
 -I do not use them for audio, but for SDR (software defined radio).
  
 The basic idea is that rather than tuning a receiver to a particular frequency and using dedicated hardware to demodulate whatever signal may be present on that frequency (or, more precisely - present within a given bandwidth, centered on said frequency), I use a high-bandwidth ADC to digitize a broad swathe of the radio spectrum, then use software algorithms to demodulate the desired signal.
  
 The (huge!) benefit is that basically, if you throw enough processing power at it, one SDR rig may replace any number of conventional radio receivers.
  
 To take one (trivial) example - I can easily decode the entire FM broadcasting spectrum, storing each station's broadcast in a separate flac file, using a cheap(-ish) ADC and some RF circuitry + a laptop computer to do the number-crunching.


----------



## nick_charles

gregorio said:


> It's got nothing to do with computing power and everything to do with raw single chip computing speed, which has not changed significantly in many years.


 
  
  
 Cough...Multicore processors...cough 
  
 Cough...advances in instruction code processing and architecture...cough
  
 Cough.. Gigaflops ..Cough


----------



## gregorio

rrod said:


> I guess I would find this surprising ...


 
  
 Dan Lavry first publicised the issue about a decade ago and backed-up his engineering assertions by refusing to make converters which supported 192/24. Unfortunately he couldn't maintain his refusal for long, as all the ADC/DAC chip manufacturers eventually only manufactured high quality chips which included the 192/24 format. Where honest comment from design engineers has been available, it has been confirmed that inclusion of the 192/24 sample rate on chips was a marketing dept decision rather than an engineering dept decision. Sometime after Lavry's paper, Digidesign/Avid (the manufacturers of ProTools and probably the most widely used higher-end pro audio converters) confirmed Lavry's engineering assertions, indicating a loss of fidelity at 192/24 and recommending it be avoided, except in some specialist applications. Later still, Benchmark, in response to an article about 192/24 in Stereophile, contradicted Stereophile's article and also asserted a loss of resolution/fidelity at 192/24, claiming that a loss was inevitable at sample rates above about 100kHz (with 24bit). There have been many highly respected/influential audio and design engineers in the pro audio field who have made similar observations/recommendations over the ensuing years. So much so, that it's now a non-issue, an accepted "given" within the educated pro audio community, unworthy of further discussion. The only time it even crops up any more is if some newbie makes an ill-informed statement about it. In the pro audio world, 192/24 is placed in much the same category as other audiophile snake oil myths.
  
 G


----------



## watchnerd

rrod said:


> The question isn't about making them better, it's if they are demonstrably *worse* at 192, which would indeed be a great thing to point out to people making claims that "bigger is always better." I'm just asking for an example, because as others have said there is hardware out there that runs at even higher effective bit rates.


 
  
 Yes, it's demonstrably worse:
  
 "[size=17.03px]192kHz considered harmful[/size]
  
_192kHz digital music files offer no benefits. They're not quite neutral either; practical fidelity is slightly worse. The ultrasonics are a liability during playback._
_Neither audio transducers nor power amplifiers are free of distortion, and distortion tends to increase rapidly at the lowest and highest frequencies. If the same transducer reproduces ultrasonics along with audible content, any nonlinearity will shift some of the ultrasonic content down into the audible range as an uncontrolled spray of intermodulation distortion products covering the entire audible spectrum. Nonlinearity in a power amplifier will produce the same effect. The effect is very slight, but listening tests have confirmed that both effects can be audible._"
 
https://xiph.org/~xiphmont/demo/neil-young.html


----------



## gregorio

nick_charles said:


> Cough...Multicore processors...cough


 
  
 Cough...Irrelevant...cough!
  
 I am of course not disputing that computing power has dramatically increased (Moore's law). It's increased by, as you say, more cores per chip, better coding/processing, etc. This increase in power is obvious but is largely dependent on splitting tasks into various smaller chunks which can then be processed far faster (say in parallel by two or more cores). In some cases this is not possible however, the processing tasks cannot be divided into smaller parts and executed faster/more efficiently, the recursive functions used in digital filters being one such case. In this and some other audio processing cases, overall computational power is irrelevant, what's relevant is the processing speed of an individual core. It has long been a practical observation within the pro DAW community that a system with fewer cores running at a higher speed is more efficient/powerful than more cores at a lower speed. A six core 3.33gHz mac pro significantly out performed an eight core 2.4gHz mac pro for example. This of course isn't necessarily true of other applications but is for audio processing.
  
 G


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## RRod

watchnerd said:


> Yes, it's demonstrably worse:
> 
> "[size=17.03px]192kHz considered harmful[/size]
> 
> ...


 
  
 Yes I am unsurprised that my consumer-grade playback setup can't handle 192 without noticeable IMD. What I don't have access to in my life is professional grade ADCs, which I would have assumed by 2015 could handle 192 without such issues on the mouth-end of the equation. If this assumption is wrong, then the reality would be great ammo to use against people who say that higher rates give more accuracy even in the audible range (which by theorem, of course, doesn't make much sense anyway, but many people are evidently not swayed by pesky theory). There seems to be some disagreement, though, about the capabilities of hardware in 2016.


----------



## JWolf

watchnerd said:


> Yes, it's demonstrably worse:
> 
> "[size=17.03px]192kHz considered harmful[/size]
> 
> ...


 
  
 That article is odd. 192kHz harmful? Not likely. What music goes past 96kHz? The rest is silence and how is silence harmful? xiph.org just have some odd people writing for them. They write some real ***** sometimes.


----------



## Guidostrunk

( ͡° ͜ʖ ͡°)


----------



## BiggerHead

watchnerd said:


> No, it's nothing like video.
> 
> A 5khz sine wave is reproduced just as perfectly with a 44.1khz, 48khz, 88.2khz, 96khz, 176.4khz, or 192khz sample rate. To quote Lavry's paper "*The Optimal Sample Rate for Quality Audio*":
> 
> ...


 
  
  
 You seem to have missed the entire shift in wit of my post.  I wasn't actually the one who claimed it was like video.  Read the post above it.


----------



## watchnerd

biggerhead said:


> You seem to have missed the entire shift in wit of my post.  I wasn't actually the one who claimed it was like video.  Read the post above it.


 
  
 I did.


----------



## watchnerd

jwolf said:


> That article is odd. 192kHz harmful? Not likely. What music goes past 96kHz? The rest is silence and how is silence harmful? xiph.org just have some odd people writing for them. They write some real ***** sometimes.


 
  
 Did you miss the part about IMD in the audible range?


----------



## castleofargh

jwolf said:


> That article is odd. 192kHz harmful? Not likely. What music goes past 96kHz? The rest is silence and how is silence harmful? xiph.org just have some odd people writing for them. They write some real ***** sometimes.


 
 and once more here you are looking down on a guy who not only understands digital formats, but has worked to create a few of them. where does that unwarranted confidence come from? your understanding of digital audio and DAC(that is a little better then before) leads you to decide what is right and what is wrong in an intuitive yet false system that would be more fitting to describe this audio system


 with more samples the errors between 2 points sure could be improved on that one.
 but low pass filters and why they're used, how amps and tweeters deal with ultrasonic noises/distortions, oversampling, sampling theorem, hearing thresholds. the inherent noises and distortions of the sound system limiting de facto the bit depth resolution, the limits of the recording gear and the noise in the studio, the fact that no musician is playing at 140db just so that you can enjoy the full dynamic of music...  all those are taken out of what is audio for your convenience, so that your idea of more points= more accuracy can still feel like the main point to you. 
 but just like you really couldn't care to have 500 points to draw Y=3X², having in infinity of points when within limited frequency range only serves (to a point) to deal with how we can't put the sampling theorem 100% in practice. so a little frequency range, for a gentler low pass filter, for example can go a long way. but soon after we have passed double the audible range, we get less and less benefits, more and more requirements, and soon enough it looks really foolish, both to those who understand how things really work, and to all those who cared to try and measure the change in actual resolution coming from a headphone or a pair of speakers.
  
  
 you don't realize it, but because of a few marketing idiots, we all end up victim of a race toward the useless increase of digital resolution, when we could benefit so much from other stuff, like getting better DSPs, better headphones, and of course proper recording and mastering that could really use a few standards for production quality. when guys like you fall for it and push for more resolution like that's what good soud was missing, you're proving the marketing guys that they're right. that we can be played with by providing us fake ideals and dreams instead of actually having to deliver on sound quality. just like tv screens racing toward 200hz and 4K, while they have for the most part no proper calibration, and a gamut that would make any guy in photography piss his pants. but hey as long as we feed dreams about bettererer resolution, surely people will believe and buy. how about we try to be more than walking wallets?


----------



## gregorio

jwolf said:


> That article is odd.


 
  
 "Odd" why exactly? "Odd" because it disagrees with your incorrect assumptions of how audio works? If so, then "enlightening" would be a better adjective, unless you are so convinced of your incorrect assumptions that your mind is completely closed to the truth.
  


jwolf said:


> 192kHz harmful? Not likely.


 
  
 And you have some actual evidence to back up this claim? Serious evidence to counter the evidence already presented? If not, then you are the one talking ****, a result of your incorrect assumptions, again!
  


jwolf said:


> What music goes past 96kHz?


 
  
 What has that got to do with anything? We're talking about 192/24 (and 176.4kHz) sample rates, no one has mentioned the even more ridiculous 384/24 rate.
  


jwolf said:


> The rest is silence and how is silence harmful?


 
  
 If it really were silence (above say 22.05kHz and below 96kHz) then many/most of those who distribute so called Hi-rez or hi-def audio would reject it on the grounds that it's indistinguishable from up-sampled 44.1kHz sample rate!
  


jwolf said:


> xiph.org just have some odd people writing for them.


 
  
 I personally don't care how odd they are, just how accurate the information they are presenting. Einstein was a bit "odd" but that doesn't make everything he said wrong.
  


jwolf said:


> They write some real ***** sometimes.


 
  
 I haven't read everything they've written so I can't say for sure that they never write "some real ****". I have read this particular article and found no "real ****". Again, are you calling it **** simply because it contradicts your incorrect assumptions, assumptions which you seem incapable or unwilling to question? If so, then your insults are all about your personal "real ****" and have nothing to do with xiph.org, which of course just further highlights your ignorance, plus makes you look foolish into the bargain! BTW, I'm not using the term "ignorance" as an insult, the audio world encompasses a wide number of complex fields and we are all ignorant to some degree.
  
 G


----------



## watchnerd

odde said:


> -I do not use them for audio, but for SDR (software defined radio).
> 
> The basic idea is that rather than tuning a receiver to a particular frequency and using dedicated hardware to demodulate whatever signal may be present on that frequency (or, more precisely - present within a given bandwidth, centered on said frequency), I use a high-bandwidth ADC to digitize a broad swathe of the radio spectrum, then use software algorithms to demodulate the desired signal.
> 
> ...


 
  
 Do you actually have this working?


----------



## KeithEmo

Actually, while I haven't been following this discussion in detail, I can in fact think of one major reason why ABX testing "favors" null results... although "favors" isn't technically what I'm referring to. By its nature, ABX testing is designed to avoid false-positives; no matter how much you are biased to think you hear a difference, ABX testing should prevent that bias from affecting the results. However, there is nothing inherent in ABX testing that would tend to prevent false-negatives. (In other words, ABX testing is immune to false-positives generated by bias, but is not immune to false-negatives due to bias... so, assuming that the real world contains biases in both directions, ABX testing will allow the negative biases to affect the results, but not the positive biases, and so it will skew the results statistically in favor of the false-negatives.)
  
 Quote:


mmerrill99 said:


> Fair enough but obviously they seem very logical to me & those on the fence can decide for themselves. It might, at least, bring a bit more appreciation of the pitfalls of home run ABX testing?


----------



## watchnerd

keithemo said:


> Actually, while I haven't been following this discussion in detail, I can in fact think of one major reason why ABX testing "favors" null results... although "favors" isn't technically what I'm referring to. By its nature, ABX testing is designed to avoid false-positives; no matter how much you are biased to think you hear a difference, ABX testing should prevent that bias from affecting the results. However, there is nothing inherent in ABX testing that would tend to prevent false-negatives. (In other words, ABX testing is immune to false-positives generated by bias, but is not immune to false-negatives due to bias... so, assuming that the real world contains biases in both directions, ABX testing will allow the negative biases to affect the results, but not the positive biases, and so it will skew the results statistically in favor of the false-negatives.)


 
  
 This is true.
  
 However, it also means that if you do get a persistent, repeatable positive result it has huge significance.


----------



## KeithEmo

re SDR
  
 I'm not a radio person, but I can tell you that commercial "SDR boxes" do exist, and are routinely advertised in TV industry trade magazines for monitoring broadcast and cable TV signals.
  
 If you google "SDR" you'll find all sorts of SDR devices...
  
 Quote:


watchnerd said:


> Do you actually have this working?


----------



## KeithEmo

Absolutely agreed...... and that's a good thing.
  
 However, I would point out that the slight tendency towards null results requires that one pay strict attention to the circumstances under which the test itself is presented. For example, if you invite a "hostile audience" to participate in an ABX test, they are quite likely to ignore any differences they might hear, actually believing that "they know what they should be hearing", and so ignoring differences that they really hear but presume are imaginary, and give you a false-null. And, especially in group situations, while an audiophile may report a false-positive in a sighted test because all his friends seem to hear something, people who don't expect to hear a difference will often be loath to be the only one who admits to hearing it.
  
 While these psychological issues are much more often used to skew the results in a sighted test, or one run by "a narrator", they can also strongly affect the results of ABX tests - especially if the differences in question are very small (in which case it's very easy for someone to convince themselves that they were just imagining a difference they actually heard).
  
 Quote:


watchnerd said:


> This is true.
> 
> However, it also means that if you do get a persistent, repeatable positive result it has huge significance.


----------



## OddE

watchnerd said:


> Do you actually have this working?


 
  
 -Not the 210Msps one (I just got the boards in the mail on Thursday) - but I have a working 130Msps unit in my ham shack. A Xilinx FPGA accepts the raw output from the ADC and performs a digital down conversion to baseband, after which the data stream is passed on to a Linux PC for demodulation. (Passing the full 2Gbps or so to a computer directly would be a bit on the taxing side for the poor computer)


----------



## mmerrill99

watchnerd said:


> keithemo said:
> 
> 
> > Actually, while I haven't been following this discussion in detail, I can in fact think of one major reason why ABX testing "favors" null results... although "favors" isn't technically what I'm referring to. By its nature, ABX testing is designed to avoid false-positives; no matter how much you are biased to think you hear a difference, ABX testing should prevent that bias from affecting the results. However, there is nothing inherent in ABX testing that would tend to prevent false-negatives. (In other words, ABX testing is immune to false-positives generated by bias, but is not immune to false-negatives due to bias... so, assuming that the real world contains biases in both directions, ABX testing will allow the negative biases to affect the results, but not the positive biases, and so it will skew the results statistically in favor of the false-negatives.)
> ...



Yes, Keith that's exactly the point & there's no need to show evidence to prove this - it's patently obvious - the test is designed to prevent false positives while being blind to false negative - that's what I mean by the results being "skewed" towards false negatives. In well run ABX tests, this factor is addressed to some extent by pre-screening of participants, by training participants & by being careful in the administration of the test - - it's also why there are hidden anchors & references in other blind test protocols, like MUSHRA & there are many. 

It's not only negative bias (nocebo) that invariably gives false negative results but all the other reasons I've outlined - tiredness, lack of training, lack of focus, etc. With any test, it's essential that we have a measure (or at least some idea) of the sensitivity of the test - to know what skew there might be in the test towards one result or another. My point has always been that when we get into the home run ABX tests these weaknesses & this skew is multiplied many-fold, simply because a) there's no protocol being used to ensure the test is being done correctly & b) there are no internal controls that could tell us if a test was done correctly.

I find any test suspicious that ignores this & operates on the basis of "if you do get a persistent, repeatable positive result it has huge significance" in other words, "we secretly know that the test is so skewed that any time it does actually produce a positive result, we pretend it is significant but then we find an excuse to reject the result, anyway" Look at Amir's positive results & UltMusicSnobs positive results to see what is meant here by " huge significance"


----------



## sonitus mirus

mmerrill99 said:


> Yes, Keith that's exactly the point & there's no need to show evidence to prove this - it's patently obvious - the test is designed to prevent false positives while being blind to false negative - that's what I mean by the results being "skewed" towards false negatives. In well run ABX tests, this factor is addressed to some extent by pre-screening of participants, by training participants & by being careful in the administration of the test - - it's also why there are hidden anchors & references in other blind test protocols, like MUSHRA & there are many.
> 
> It's not only negative bias (nocebo) that invariably gives false negative results but all the other reasons I've outlined - tiredness, lack of training, lack of focus, etc. With any test, it's essential that we have a measure (or at least some idea) of the sensitivity of the test - to know what skew there might be in the test towards one result or another. My point has always been that when we get into the home run ABX tests these weaknesses & this skew is multiplied many-fold, simply because a) there's no protocol being used to ensure the test is being done correctly & b) there are no internal controls that could tell us if a test was done correctly.
> 
> I find any test suspicious that ignores this & operates on the basis of "if you do get a persistent, repeatable positive result it has huge significance" in other words, "we secretly know that the test is so skewed that any time it does actually produce a positive result, we pretend it is significant but then we find an excuse to reject the result, anyway" Look at Amir's positive results & UltMusicSnobs positive results to see what is meant here by " huge significance"


 
  
 Why not have someone provide random files of unknown qualities and see if anyone can identify differences between any of them.  The test files could be lossless, high bitrate lossy, lower bitrate lossy, 24bit, 16bit, 8bit, etc.  At least we could all agree that some of the files were easier to identify than others, and that if any audible differences were present, with some of the files it would be more challenging for the testers to hear these.  Then maybe do an extended ABX listening test between a few of the files that were not statistically shown to be different in an ABX to see if anyone gets fatigued or experiences nausea over time.  (or whatever ill effects that people claim to experience)


----------



## mmerrill99

sonitus mirus said:


> mmerrill99 said:
> 
> 
> > Yes, Keith that's exactly the point
> ...



One major problem with this approach - in such a test, the participant knows that the test has certain files which should be audible - the motivation & focus will be very different doing this test compared to a test of different cables where the meme is that nobody has ever conclusively blind tested cables. Putting hidden anchors or reference trials in among other trials is a totally different approach.

As you can see, there's more psychology to blind testing than most want to admit to!


----------



## sonitus mirus

I only really care if I can hear a difference or not.  If I have to change the way I listen or need to carefully hear the files for 20 minutes before making a decision, it's just too close for me to care.  I listen to music all day long most days and I enjoy it immensely.  I'd be concerned if I was unable to hear any differences at all with ABX, but that is not the case.  I can consistently hear differences at a specific level of quality.  That is my limit and it is really that simple for me.


----------



## mmerrill99

sonitus mirus said:


> I only really care if I can hear a difference or not.  If I have to change the way I listen or need to carefully hear the files for 20 minutes before making a decision, it's just too close for me to care.  I listen to music all day long most days and I enjoy it immensely.  I'd be concerned if I was unable to hear any differences at all with ABX, but that is not the case.  I can consistently hear differences at a specific level of quality.  That is my limit and it is really that simple for me.



Sure, exactly my point - I decided what I like by listening normally - I don't need a flawed ABX test, riddled with psychological traps to "prove" whether I find A better than B (after 16 trials). So I'm riddled with sighted bias - so is life - live with it - it's a lot more fun to open oneself to possibilities rather than constantly limit oneself requiring the myth of surety. It's that simple for me.


----------



## sonitus mirus

mmerrill99 said:


> Sure, exactly my point - I decided what I like by listening normally - I don't need a flawed ABX test, riddled with psychological traps to "prove" whether I find A better than B (after 16 trials). So I'm riddled with sighted bias - so is life - live with it - it's a lot more fun to open oneself to possibilities rather than constantly limit oneself requiring the myth of surety. It's that simple for me.


 
  
 If you don't try and remove any bias, you might like one DAC over another for any number of reasons that have nothing to do with your hearing.  Same with any other audio evaluations.  ABX works for other areas of science, it would be ignorant to pretend that any ABX flaws with audio perception are so egregious to make them invalid or on the same level as listening normally.   I think you are wrong, and you know this already, so you can have the last word if you wish and I will read it and move on for now.


----------



## mmerrill99

sonitus mirus said:


> mmerrill99 said:
> 
> 
> > Sure, exactly my point - I decided what I like by listening normally - I don't need a flawed ABX test, riddled with psychological traps to "prove" whether I find A better than B (after 16 trials). So I'm riddled with sighted bias - so is life - live with it - it's a lot more fun to open oneself to possibilities rather than constantly limit oneself requiring the myth of surety. It's that simple for me.
> ...


For sure, I MIGHT (& this is an important distinction) but I find all this sighted bias stuff is overplayed to the extent where the unwary would be led to believe that this ALWAYS happens. 

I'm far happier to take a chance on a sighted bias affecting my choice than the opposite - many hidden factors affecting my choice when doing a home run ABX test. 





> ABX works for other areas of science, it would be ignorant to pretend that any ABX flaws with audio perception are so egregious to make them invalid or on the same level as listening normally.


How many branches of science would suggest that home run ABX tests have any validity whatsoever? This invoking of science is so far from what a home run ABX test is about that is laughable





> I think you are wrong, and you know this already, so you can have the last word if you wish and I will read it and move on for now.


Sure, but I think many of your colleagues in this section admit that home run ABX tests are flawed & far from science but many don't accept this & still cling to this test for the psychological succour that they seem to need.


----------



## jcx

I've used foobar2000 ABX plugin to verify things can be heard even by my 50+ year old ears - positive results are useful, null results less so
  
 but essentially null ABX is still at the base of any "threshold" testing scheme whatever protocol being used
  
 so its useful to make it a competition, provide incentive to show positive results despite the listening controls - US$10k went unclaimed in over a thousand trials of "amplifier sound" when level matched, EQ'd
  
https://www.google.com/#q=richard+clark+amplifier+challenge
  
 maybe many of the challengers were naïve, but nothing prevented training with the public test provisions, ruled out truly golden ears from participating


----------



## Don Hills

watchnerd said:


> Do you actually have this working?


 
  
 Just to add to what Keith and Odd said, SDR is not only commonplace but ubiquitous and cheap. For example, the TV tuner USB dongles available for $20 or less are SDR based. Some of them can be repurposed as general purpose wideband receivers and spectrum analysers by using the right software. Odd's boards sound like a step up from these...


----------



## BiggerHead

odde said:


> -To be fair, Nyquist-Shannon has further requirements than just sampling rate > 2x analog bandwidth, most notably that the signal in question must be perfectly band-limited, which in turn implies infinite duration.
> 
> I have not heard of anyone being able to demonstrate that easing on the band-limit criterion (within reasonable bounds) produces audible artifacts, though.


 
  
 I searched the word "infinite" in hopes somebody had already said this.
  
 This really true and this that isn't just an implementation technicality as has been implied.  If you sample a 19 khz frequency at 40kz you will get an amplitude fluctuation cycle of about 1 khz (I'm pretty sure we don't perceive that frequency as a result but I could be wrong). Since the frequencies almost match you can sample near both nodes in the first cycle and it will take about 10 cycles for the sampling phase to catch up to the peaks.  So you can already see that it takes time to "find" the amplitude.  In principle from enough time you can determine the correct amplitude, at least if you have only a single frequency to deal with and if you make some clever signal processing to figure it out.  I don't know how fancy these dacs are but in practice I think what you end up is a signal the ear will respond to at 19khz (if you're so young) but with a few db  reduced amplitude.  
  
 Add more frequencies and working out the correct amplitudes for all of them gets harder, and in fact for an artbitrary continuous frequency spectrum, it will require infinite time and processing power, assuming the signal really is band limited and thus can be described by *non-changing *frequency spectrum.  However  we perceive changes which theoretically require infinite frequency to represent (instead of infinite time), but practically just mean the sound doesn't go on forever, but actually stops.  Of course neither sampling nor our ears are limited to a purely frequency domain description.  We perceive the time domain, which trivially allows this. (I guess there's a bunch of aliases for the frequencies in that stopped sound).
  
 So there is a little room in the margins here to feed the psychoses.  Nyquist doesn't actually say that 40khz PERFECTLY , EXACLTY reproduces real music.  It does not.  It does come very close and I think the only practical effect is probably the treble roll off mentioned a few times in the upper couple of khz, hence the use of slightly higher than 40 as stated.
  
 So can we hear this 1ms amplitude envelope in impact?  I doubt it, and anyway, crank it up to 44khz or 48 and this is about wiped out as has been stated.  
  
 One thing I wonder though, if we band pass filter at 20 khz before recording, and there is thus nearly zero extra information by going to 192khz or whatever, then wouldn't a decent lossless compression keep the same file size anyway?  It becomes just like upsampling, but not bandwidth increase.  No gain, but no harm?


----------



## old tech

Anyone read this explanation of CD v LP in how stuff works?  It surprises me how a site which purports to be an expert resource can produce utter nonsense pseudoscience.
  
http://electronics.howstuffworks.com/question487.htm


----------



## ralphp@optonline

old tech said:


> Anyone read this explanation of CD v LP in how stuff works?  It surprises me how a site which purports to be an expert resource can produce utter nonsense pseudoscience.
> 
> http://electronics.howstuffworks.com/question487.htm


 

 I've read the exact same nonsense in many high end audio magazines as well. Which is why I like the "Sound Science" sub-forum much better than the "High-end Audio Forum" sub-forum


----------



## cjl

biggerhead said:


> I searched the word "infinite" in hopes somebody had already said this.
> 
> This really true and this that isn't just an implementation technicality as has been implied.  If you sample a 19 khz frequency at 40kz you will get an amplitude fluctuation cycle of about 1 khz


 
 That's not really true actually - even if the samples miss the peaks, there's still only one band limited signal that fits the sample points, and that signal does not have any amplitude fluctuation. The differences OddE are talking about are substantially subtler than what you're imagining here.


----------



## BiggerHead

With perfect bi


cjl said:


> That's not really true actually - even if the samples miss the peaks, there's still only one band limited signal that fits the sample points, and that signal does not have any amplitude fluctuation. The differences OddE are talking about are substantially subtler than what you're imagining here.


 
  
 With one frequency and infinite bit depth yes.  Effectively you lose bit depth (so there is information loss), but that's assuming you process it that well.  In practice I'm almost certain you don't, and you get the treble roll off.


----------



## watchnerd

old tech said:


> Anyone read this explanation of CD v LP in how stuff works?  It surprises me how a site which purports to be an expert resource can produce utter nonsense pseudoscience.
> 
> http://electronics.howstuffworks.com/question487.htm


 
  
 Thanks for ruining my day.


----------



## icebear

watchnerd said:


> Thanks for ruining my day.


 
  
 Isn't it fun to read just the beginning, a graph showing 5 big stair steps 
	

	
	
		
		

		
		
	


	



 for the CD and the text mentions 44,100 samples. (20 steps in the graph for DVD-A and a sample rate for 192,000).
 That's enough reason to crack up isn't it


----------



## limpidglitch

mmerrill99 said:


> For sure, I MIGHT (& this is an important distinction) but I find all this sighted bias stuff is overplayed to the extent where the unwary would be led to believe that this ALWAYS happens.
> 
> I'm far happier to take a chance on a sighted bias affecting my choice than the opposite - many hidden factors affecting my choice when doing a home run ABX test.
> How many branches of science would suggest that home run ABX tests have any validity whatsoever? This invoking of science is so far from what a home run ABX test is about that is laughable
> Sure, but I think many of your colleagues in this section admit that home run ABX tests are flawed & far from science but many don't accept this & still cling to this test for the psychological succour that they seem to need.


 
  
 I'm puzzled by this discussion. What you're saying about a bias for negative results equates to saying "if you don't look, you won't find", which surely is about as platitudinal as they come.
  
 If I, in bright daylight, gaze into the sky in the general direction of where Mars is, and say: "I can see no sign of life there", should NASA take notice? No.
  
 If a deaf person fails a double blind listening test, should we take notice? No.
  
 If you pass a sighted listening test, should we take notice? Let me answer like this: While a deaf person can't pass a double blind listening test, they can easily pass a sighted one.
  
 If there is any bias here worth noticing, it's on the reporting side. Someone who's convinced that they can hear a difference between, e.g. LAME vb0 and lossless, is unlikely to bother about posting a negative result, but all the more likely to post a positive one, if they even bother about testing it at all. The more inquisitive minds, like those here and at HA, while mostly content with lossy audio, are also perfectly aware of the theoretical possibility of hearing a difference. Therefore they use ABX as a way of checking whether that little warble they just noticed upon casual listening was in fact an encoder artefact, or is also present in the lossless version. To some extent there even goes a sort of sport in this, not unlike that Philips Golden Ears challenge. There is little point in posting an endless stream of negative results, we all know these differences are hard to notice, but conversely, there is all the more creds in store for whoever manages to pass. All in all, it seems to me that there should be a considerable overrepresentation of positive results, yet they are still incredibly thin on the ground.


----------



## watchnerd

mmerrill99 said:


> For sure, I MIGHT (& this is an important distinction) but I find all this sighted bias stuff is overplayed to the extent where the unwary would be led to believe that this ALWAYS happens.
> 
> I'm far happier to take a chance on a sighted bias affecting my choice than the opposite - many hidden factors affecting my choice when doing a home run ABX test.


 
  
 If you want to be subjectivist, rock on, it's your life, your time, your hobby, etc.
  
 But why spend all this time debating your personal preference on a science board?


----------



## Joe Bloggs

biggerhead said:


> With perfect bi
> 
> 
> cjl said:
> ...




Yeah you get a treble rolloff. What you don't get is the 1kHz beat cycle you speak of:


From top to bottom:
1. A 21500Hz sine wave sampled at 16/44.1 (closer to Nyquist than the example you gave, and at limited bit depth).
2. (1) upsampled to 384kHz. In the case of the resampling filter used in Audacity 21500Hz is well past the passband of the upsampling anti-alias filter and this frequency is about 33dB down in amplitude.
3. (2) amplified to full magnitude, showing that the beating seen in the linear-interpolated 44.1kHz waveform (i.e. an incorrect visualization) does not exist in the output analog waveform (here approximated with the 384kHz waveform)


----------



## Joe Bloggs

BiggerHead The start-stop discontinuities you speak of need to be treated differently. Perceptually speaking, a pure tone with an abrupt start, starts and stops with a click. The click can be heard even if the tone itself is supersonic. That's because the start and the stop are broadband phenomena.



And so it is that the 21500Hz tone encoded in 44.1kHz, when upsampled using a filter that does not fully pass frequencies so close to Nyquist (i.e. doesn't *hear* the frequency that well), passes the broadband info at the start and end:



...almost like how our ear would hear it...

To take this to the extreme, a 44100Hz sample train with all zeroes except a 1 in the middle is not a 11025Hz tone starting and stopping. It is a Dirac impulse (containing all frequencies that the sample format can represent in equal proportion).

And the human ear does not "trivially allow detection of changes which theoretically require infinite frequency to represent". It's an acoustic fact that the eardrum won't move in response to a sufficiently high frequency. Then too, the hair cells that transduce sound vibrations into nerve impulses also each have their own finite "frequency response".


----------



## headdict

icebear said:


> Isn't it fun to read just the beginning, a graph showing 5 big stair steps
> for the CD and the text mentions 44,100 samples. (20 steps in the graph for DVD-A and a sample rate for 192,000).
> That's enough reason to crack up isn't it



You are right. DVD-A has exactly 21.768707483 steps. They calculated it wrongly.


----------



## icebear

headdict said:


> You are right. DVD-A has exactly 21.768707483 steps. They calculated it wrongly.


 

 Oh shiit ... I can even count right from a graph 
	

	
	
		
		

		
		
	


	




, I missed 1.768707483 steps


----------



## mmerrill99

limpidglitch said:


> mmerrill99 said:
> 
> 
> > For sure, I MIGHT (
> ...


No, I'm not JUST saying that. What I'm saying is that:
- firstly you need specific listening skills to pass an ABX test, skills that aren't needed for normal listening
- these skills are the isolation & identification of specific differences you can narrow down to a specific segment of audio
- you need to then train yourself in this difference first before you do the ABX blind test itself because it's a statistical test & any more than 4 wrong trails in 16 trials & you are outside the normal .05 level of significance i.e you are considered guessing
- Yes, ABX tests can be passed by "feeling" you way through the trials but the focus is generally towards the conscious identification of audible differences

Given these criteria, it's no wonder that home run ABX tests are dubious misleading as most don't know what sort of test they are doing - they are beguiled by the simplistic slogans that we see in evidence here.

It's not a test of whether something is audible - it's a test of whether someone has the motivation, skills to do the preparation necessary & the focus needed to pass multiple trials to a statistical level of significance



> If I, in bright daylight, gaze into the sky in the general direction of where Mars is, and say: "I can see no sign of life there", should NASA take notice? No.
> 
> If a deaf person fails a double blind listening test, should we take notice? No.
> 
> If you pass a sighted listening test, should we take notice? Let me answer like this: While a deaf person can't pass a double blind listening test, they can easily pass a sighted one.


Why do people always refer to sighted listening & it's potential flaws when they are trying to defend the flaws inherent in home-run ABX testing?
The point is that few of us have the skills I nominated above - the skills needed to do a valid ABX test, so as far as the test is concerned, it reduces us all to the level of a deaf person 



> If there is any bias here worth noticing, it's on the reporting side. Someone who's convinced that they can hear a difference between, e.g. LAME vb0 and lossless, is unlikely to bother about posting a negative result, but all the more likely to post a positive one, if they even bother about testing it at all. The more inquisitive minds, like those here and at HA, while mostly content with lossy audio, are also perfectly aware of the theoretical possibility of hearing a difference. Therefore they use ABX as a way of checking whether that little warble they just noticed upon casual listening was in fact an encoder artefact, or is also present in the lossless version. To some extent there even goes a sort of sport in this, not unlike that Philips Golden Ears challenge. There is little point in posting an endless stream of negative results, we all know these differences are hard to notice, but conversely, there is all the more creds in store for whoever manages to pass. All in all, it seems to me that there should be a considerable overrepresentation of positive results, yet they are still incredibly thin on the ground.


I'm not talking about how many null results are reported, I'm talking about the number of people doing personal ABX testing & are fooled into thinking that there is no audible difference in A Vs B, simply because they don't have the skills necessary to pass an ABX test


----------



## sonitus mirus

At what point do reasonable people decide that it may not be about specialized training and hours of practicing an abnormal method of listening, but that any differences are so tiny to be practically insignificant?


----------



## watchnerd

sonitus mirus said:


> At what point do reasonable people decide that it may not be about specialized training and hours of practicing an abnormal method of listening, but that any differences are so tiny to be practically insignificant?


 
  
 When you spend more time listening to reference tracks and tweaking your system than you do actually listening to music, especially new music.


----------



## Guidostrunk

Seriously. If you have to go though so much to possibly hear a difference. What's the point? What enjoyment is there in putting yourself through something so rigorous? I can't see how anyone would enjoy music that way. Especially if it's that critical. 

Me personally, have never been able to really differentiate between Spotify premium, and flac that I have in my library , repeatedly trying to hear a difference using a song that I'm very familiar with. Some days I think I would , and others I didn't think there was. I've tried it blind with my back turned to my rig, having my wife swap the song randomly, and failed miserably. Even under the guise, that flac is supposed to be better. I don't even bother downloading flac anymore. 

I think people get caught up way to much in rates and bits, and miss out entirely on what the whole purpose of being in this hobby is. Enjoying music. At least that's why I got into it anyway. 
With all that said. It doesn't make any of my statements truth. It's just my opinion on the subject. I truly think that better sound in any file is dependent on the recording itself. Of course, another opinion.


----------



## mmerrill99

sonitus mirus said:


> At what point do reasonable people decide that it may not be about specialized training and hours of practicing an abnormal method of listening, but that any differences are so tiny to be practically insignificant?




When you are trying to do one of those spot-the-difference picture tests, the difference doesn't jump out at you & you would probably also say that it's therefore so tiny as to be practically insignificant - when one spots the difference, that viewpoint quickly changes & one can't help but seeing the obvious difference when you look at the pictures again


----------



## mmerrill99

guidostrunk said:


> Seriously. If you have to go though so much to possibly hear a difference. What's the point? What enjoyment is there in putting yourself through something so rigorous? I can't see how anyone would enjoy music that way. Especially if it's that critical.
> 
> Me personally, have never been able to really differentiate between Spotify premium, and flac that I have in my library , repeatedly trying to hear a difference using a song that I'm very familiar with. Some days I think I would , and others I didn't think there was. I've tried it blind with my back turned to my rig, having my wife swap the song randomly, and failed miserably. Even under the guise, that flac is supposed to be better. I don't even bother downloading flac anymore.
> 
> ...


I agree with you - I'm quite happy to live with my possible sighted biases & listen to devices & decide over a period which I deem sufficient to get a handle on the sound of the device - I don't need to "prove" to myself or anybody else what I hear but some do have this need & often try to enforce it on others through demands for ABX testing - I'm just addressing this. 

So just to reiterate a point that many miss - one is doing such an ABX test because you have heard a difference in a sighted test - this is usually the case.
So just because the test enforces a difficult approach in order to be able to pass the test, doesn't now mean that the difference is insignificant & practically inaudible - you've already easily "sensed" that there's an audible difference but you haven't consciously isolated & identified the specific difference which is what the test requires you to do.

And now we will get the same objection to what I just said by someone suggesting that you don't have to find a specific difference you can just pass an ABX test by "sensing" which file is better - I've already answered this 100 times before!


----------



## Joe Bloggs

mmerrill99 said:


> And now we will get the same objection to what I just said by someone suggesting that you don't have to find a specific difference you can just pass an ABX test by "sensing" which file is better - I've already answered this 100 times before!




I missed the part where you answered this in any coherent manner.

ABX tests make it as easy as humanly possible at present to sense a difference that can actually be sensed, as opposed to one that you fool yourself into thinking you hear.



joe bloggs said:


> mostlygordon said:
> 
> 
> > Probably a question for James or any Fiio staff lurking… If I play digital audio with no EQ, does the X5 feed the digital stream to its DAC with no processing?
> ...


----------



## gregorio

Quote:


mmerrill99 said:


> I find all this sighted bias stuff is overplayed to the extent where the unwary would be led to believe that this ALWAYS happens.


 
   
 Ironical that you should bring up the "unwary"! Only the unwary, those unaware of the science of how perception operates, would believe that this does not "ALWAYS happen". The only rational question, if one accepts the science, is, on an individual basis, how much sighted bias influences the perception, rather than if!
  
 Quote:


mmerrill99 said:


> This invoking of science is so far from what a home run ABX test is about that is laughable


 
  
 It's only "laughable" if one doesn't understand what science is. Science and the scientific method, is fundamentally just the reduction of bias from our understanding of our environment ... no more and no less! To many, who don't really understand it, science is the act of proving absolutes. To these people, science is very flawed, because it is very rarely able to absolutely prove anything. Most scientific understanding is based on a consensus of opinion (within the scientific community) of the probability that a particular theory is correct. This judgement call on the "probability" is based on the preponderance and quality of the evidence. In other words, science does not claim to have absolute, perfect answers, the only thing science claims is that it provides the best answers that we, as biased  human beings, are capable of at a particular instant in time.
  
 As ABX tests, even home run ones, are a serious attempt to remove bias from the acquisition of knowledge, they are by definition "science". It's therefore only appears "laughable" to those who either don't know what science actually is or to those who have a personal agenda of refuting science to dupe others for some personal gain!
  


mmerrill99 said:


> I'm far happier to take a chance on a sighted bias affecting my choice than the opposite - many hidden factors affecting my choice when doing a home run ABX test.


 
  
 ABX tests are not perfect, no rational person would say otherwise. An ABX test does not provide an absolute proof of anything, it just provides evidence. The only question is; what is the quality of that evidence? Generally, an ABX test designed by scientists and performed under laboratory conditions is better quality evidence than a home run ABX test. Likewise, a home run ABX test is generally better quality evidence than a home blind test, which in turn is better quality evidence than a sighted test, which in turn is better quality evidence than purely anecdotal evidence. All but the most rabidly irrational audiophiles would agree with the basic concept of the hierarchy of evidence quality. For example, even if they eschew ABX tests they would generally put their own (albeit sighted) tests above purely anecdotal evidence. For the more rational, home ABX testing isn't flawless but it is the least flawed practical method of acquiring the best quality evidence.
  
 Your rational is baffling, you seem to be making the typical creationist, flat-earth, climate change denier argument: Science is imperfect and therefore worthless. Those who are more rational believe that the fact science is imperfect is irrelevant because however imperfect it is, it's still significantly less imperfect than any other method we have. Most ABX deniers are effectively just irrational hypocrites. For example, chemotherapy only provides a limited probability of success, it is the best answer of which science is currently capable but highly imperfect. Applying your rationale, if you were diagnosed with cancer and prescribed chemotherapy, you would have to refuse it, based on it being such imperfect science. Your rationale dictates that instead you must rely on leeches, incantations or some other treatment based solely on human bias/superstition. Would you really refuse chemo or would you be a hypocrite to the argument you have been presenting here?
  
 G


----------



## limpidglitch

mmerrill99 said:


> No, I'm not JUST saying that. What I'm saying is that:
> - firstly you need specific listening skills to pass an ABX test, skills that aren't needed for normal listening
> - these skills are the isolation & identification of specific differences you can narrow down to a specific segment of audio
> - you need to then train yourself in this difference first before you do the ABX blind test itself because it's a statistical test & any more than 4 wrong trails in 16 trials & you are outside the normal .05 level of significance i.e you are considered guessing
> - Yes, ABX tests can be passed by "feeling" you way through the trials but the focus is generally towards the conscious identification of audible differences


 
  
 Special skills aren't required, and that's part of the beauty of it: anyone can do it.
  
 As long as you keep in mind what you're testing for, everything's fine.
 You test your own personal abilities, on a specific sample, at a specific time. Things might easily change, one doesn't extrapolate from a single data point.


----------



## castleofargh

mmerrill99 said:


> sonitus mirus said:
> 
> 
> > At what point do reasonable people decide that it may not be about specialized training and hours of practicing an abnormal method of listening, but that any differences are so tiny to be practically insignificant?
> ...


 

 and that's why when doing such a visual test, if the purpose was efficiency and not just waste a minute of someone's time, you would look at one pic on a screen and have a switch to almost instantly change for the other picture on the same screen.  and after 2 or 3 rapid switches, the differences would just at your face, no matter what previous knowledge you would have.
 so that's why we need a switch to better find differences.
 and if the brightness and contrast of the 2 pics was very different, then we would again have a harder time telling what is different, so matching those values just like matching the loudness are very important when the purpose is to find the differences with the best accuracy.
 and then you wouldn't look at those 2 pictures with the differences already surrounded with a red pen, because then you have no idea if you have found them or if you were just told where and what they were. that would be pretty pointless right. and that's why we don't understand the purpose of a guy doing a sighted evaluation. because whatever his conclusions, the guy himself will never know if he was guided to his conclusion, or if he really heard a different sound.
  
 and all that is why any blind test makes more sense than a casual listener trying 2 file formats knowing at all time what he's doing.


mmerrill99 said:


> I agree with you - I'm quite happy to live with my possible sighted biases & listen to devices & decide over a period which I deem sufficient to get a handle on the sound of the device - I don't need to "prove" to myself or anybody else what I hear but some do have this need & often try to enforce it on others through demands for ABX testing - I'm just addressing this.
> 
> So just to reiterate a point that many miss - one is doing such an ABX test because you have heard a difference in a sighted test - this is usually the case.
> So just because the test enforces a difficult approach in order to be able to pass the test, doesn't now mean that the difference is insignificant & practically inaudible - *you've already easily "sensed" that there's an audible difference but you haven't consciously isolated & identified the specific difference which is what the test requires you to do.*
> ...


 
 thank you for making it so clear that you don't understand how the human brain works.
 in the end you're just advocating for the usual "I felt it so it must be real", kind of logic. the very reason why subjectivists are more often wrong than objectivists. and the very reason why blind tests are required for any serious testing of our senses.
  


> I agree with you - I'm quite happy to live with my possible sighted biases & listen to devices & decide over a period which I deem sufficient to get a handle on the sound of the device - I don't need to "prove" to myself or anybody else what I hear but some do have this need & often try to enforce it on others through demands for ABX testing - I'm just addressing this.


 
 as for this...  what nonsense are you pretending to address here?
 we never require abx evidence from someone who did nothing and kept his ideas to himself. we ask for evidence when a guy comes in making claims. if he makes a claim he must have a way to prove it right? else why would he make it a claim? but if you believe abx isn't enough, be free to provide results from double blind tests performed on yourself at MIT by some overqualified scientists. we really don't want to force ABX onto you when you it's so bad and you have so much better to know what is factual and what isn't...
 we ask for abx because it's easy and available to everybody. so when the guy makes weird claims but won't even spend 30minutes on such a test, it tells us to also not waste 30minutes on him.
 now if he tries to provide some matter of evidence, even if he has difficulties, if he shows the will to look for evidence, or accept that he shouldn't have made a claim without some proof, then me and most people here, will be more than happy to provide all the help possible and listen to his situation.
 you really put the reality upside down with that quote.


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > And now we will get the same objection to what I just said by someone suggesting that you don't have to find a specific difference you can just pass an ABX test by "sensing" which file is better - I've already answered this 100 times before!
> ...


This is the great pretence in ABX testing. If it was just so easy & a matter of sitting down & doing an ABX test, why do you think the guidelines for doing blind testing emphasise the importance of training? Why do they have a large document prescribing the approach recommended for doing blind testing? Of course, if you are not interested in doing a test about audibility you will obviously proceed in this manner as it will return a null result & QED you are vindicated 



> joe bloggs said:
> 
> 
> > mostlygordon said:
> ...


Of course it's expectation bias that blind testing is trying to address. I really think you need to re-evaluate your information about blind testing


----------



## mmerrill99

gregorio said:


> Quote:
> 
> 
> mmerrill99 said:
> ...


If a bias ALWAYS operates then what's the point in doing a blind test because when we listen to the same things sighted we will be right back to our biased listening & hear exactly the same as before. Obviously this is incorrect - what you want to say is that our expectation bias can influence what we hear. The problem with that position is that we don't have expectation bias about everything we hear so we therefore are not biased in everything we hear. 


> mmerrill99 said:
> 
> 
> > This invoking of science is so far from what a home run ABX test is about that is laughable
> ...


Oh dear, science isn't defined by how serious your attempt is - please go back to the philosophy of science 101. Just because you & others have deluded yourselves about home run ABX tests & blinkered your viewpoint that all it does is remove knowledge, doesn't mean that it is in any way scientific. You really do fall into the cargo-cult definition of science.


> mmerrill99 said:
> 
> 
> > I'm far happier to take a chance on a sighted bias affecting my choice than the opposite - many hidden factors affecting my choice when doing a home run ABX test.
> ...


And why would that be, then?





> Likewise, a home run ABX test is generally better quality evidence than a home blind test, which in turn is better quality evidence than a sighted test, which in turn is better quality evidence than purely anecdotal evidence.


Sorry, your just stringing words together now - what's a sighted test & what's the difference to anecdotal evidence? What makes one better quality than the other? 


> All but the most rabidly irrational audiophiles would agree with the basic concept of the hierarchy of evidence quality. For example, even if they eschew ABX tests they would generally put their own (albeit sighted) tests above purely anecdotal evidence. For the more rational, home ABX testing isn't flawless but it is the least flawed practical method of acquiring the best quality evidence.


The quality of a test depends on its attention to all the factors that may influence the outcome - just dealing with one known factor while at the same time ignoring the influence of newly created biasing factors does not make for a test of any quality whatsoever, no matter what type of spin or attempt to couch it in "science" is made 



> Your rational is baffling, you seem to be making the typical creationist, flat-earth, climate change denier argument: Science is imperfect and therefore worthless. Those who are more rational believe that the fact science is imperfect is irrelevant because however imperfect it is, it's still significantly less imperfect than any other method we have. Most ABX deniers are effectively just irrational hypocrites. For example, chemotherapy only provides a limited probability of success, it is the best answer of which science is currently capable but highly imperfect. Applying your rationale, if you were diagnosed with cancer and prescribed chemotherapy, you would have to refuse it, based on it being such imperfect science. Your rationale dictates that instead you must rely on leeches, incantations or some other treatment based solely on human bias/superstition. Would you really refuse chemo or would you be a hypocrite to the argument you have been presenting here?
> 
> G


Nope, I'm pointing out the weaknesses of ABX testing & the absolutely flawed use of it in a home run environment. I'm shocked that people use terms like "epiphany" & "eyes being opened" in regards to the results from such a flawed test


----------



## Joe Bloggs

mmerrill99 said:


> Of course it's expectation bias that blind testing is trying to address. I really think you need to re-evaluate your information about blind testing




Of course it's not! :rolleyes:


----------



## mmerrill99

limpidglitch said:


> mmerrill99 said:
> 
> 
> > No, I'm not JUST saying that. What I'm saying is that:
> ...


And there you show the "great illusion" - are you the Wizard of Oz? Yes, if you want to guarantee a null result (except for gross differences), then your advise is perfectly suited



> As long as you keep in mind what you're testing for, everything's fine.
> 
> You test your own personal abilities, on a specific sample, at a specific time. Things might easily change, one doesn't extrapolate from a single data point.


Well, if you have already heard a difference sighted & you wish to analyse this - then what I have outlined above is necessary for doing a valid ABX test to answer this question. Of course if what you want to do is not this then what is the point of the ABX test?


----------



## Joe Bloggs

mmerrill99 said:


> Well, if you have already heard a difference sighted & you wish to analyse this - then what I have outlined above is necessary for doing a valid ABX test to answer this question. Of course if what you want to do is not this then what is the point of the ABX test?




Why would a difference so easily heard in sighted testing disappear into thin air except with your supposedly rigorous training if it were real? :rolleyes: You just need to point out the same things that were so blindingly obvious to your ears, now blinded.

And don't tell me you have gone over this a hundred times--we've gone over every piece of ground with you a thousand times. All ten of us. :rolleyes:


----------



## mmerrill99

castleofargh said:


> mmerrill99 said:
> 
> 
> > When you are trying to do one of those spot-the-difference picture tests, the difference doesn't jump out at you
> ...


Really, 2 or 3 switches & the answer jumps right out at you? You haven't done many real tests, have you? Maybe they were all kiddies ones you did?

Now, try to do a spot-the-difference test for a movie - it's pretty impossible unless you can find a frame where the two movies show the difference & you can flick between them to spot the difference. This is the equivalent of what is being asked for in an ABX test of audio


> so that's why we need a switch to better find differences.
> and if the brightness and contrast of the 2 pics was very different, then we would again have a harder time telling what is different, so matching those values just like matching the loudness are very important when the purpose is to find the differences with the best accuracy.
> and then you wouldn't look at those 2 pictures with the differences already surrounded with a red pen, because then you have no idea if you have found them or if you were just told where and what they were. that would be pretty pointless right. and that's why we don't understand the purpose of a guy doing a sighted evaluation. because whatever his conclusions, the guy himself will never know if he was guided to his conclusion, or if he really heard a different sound.


I presume you mean guided to his conclusion by "expectation bias"? What if he has an expectation that A will sound worse than B (because A looks cheap & B doesn't) & he finds the opposite in sighted listening? Where is he being guided into his conclusion? 



> and all that is why any blind test makes more sense than a casual listener trying 2 file formats knowing at all time what he's doing.


Flawed blind testing makes no sense unless you are pushing a particular mindset


> mmerrill99 said:
> 
> 
> > I agree with you - I'm quite happy to live with my possible sighted biases
> ...


Nope, again the subtlety of what I said escapes you - given two potentially flawed ways of listening (sighted Vs home run ABX blind testing), I will choose the one that has the least flaws i.e sighted listening - I consider it more rational & logical to do this - dare I say it, more scientific



> > I agree with you - I'm quite happy to live with my possible sighted biases
> 
> 
> 
> ...


All this talk of "claims, "evidence" "proof" - people in this hobby relate anecdotes of what they hear - judge the anecdotes for what they are - stop trying to pretend that you are in any way scientific with your cargo-cult science


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > Well, if you have already heard a difference sighted & you wish to analyse this - then what I have outlined above is necessary for doing a valid ABX test to answer this question. Of course if what you want to do is not this then what is the point of the ABX test?
> ...



There are a number of possible reasons for this but the main one is that the person hasn't isolated the exact difference heard & identified a section of audio where this difference can be routinely identified. Until he does this he isn't ready for an ABX test as this is what is needed to do a "genuine" ABX test. 

Does this mean that there isn't a difference that we can hear, no? This binary view of audio that uses a flawed test as the basis for its conclusions, is obviously badly mistaken.


----------



## Joe Bloggs

mmerrill99 said:


> There are a number of possible reasons for this but the main one is that the person hasn't isolated the exact difference heard & identified a section of audio where this difference can be routinely identified. Until he does this he isn't ready for an ABX test as this is what is needed to do a "genuine" ABX test.




If the tester had come across a real difference he could hear in sighted testing, why wouldn't he have picked it out as his basis for the blind test?



> Does this mean that there isn't a difference that we can hear, no? This binary view of audio that uses a flawed test as the basis for its conclusions, is obviously badly mistaken.




This well tested view of audio that uses as good a test as there has ever been as the basis for its conclusions is obviously well correct. :rolleyes:

Ooh look, I can do adjectives! :rolleyes:


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > There are a number of possible reasons for this but the main one is that the person hasn't isolated the exact difference heard & identified a section of audio where this difference can be routinely identified. Until he does this he isn't ready for an ABX test as this is what is needed to do a "genuine" ABX test.
> ...


Because the ABX test is a statistical test that requires at least 16 trials for any statistically significant result. We don't normally listen forensically - it's a different approach to listening - so we can come to conclusions about two different devices/tracks sounding different without identifying a very specific difference. Doing an ABX test requires a very different approach than our normal listening & tests us in different ways to our normal listening - ways that you seem to not understand or not accept.

I've given ultmusicsnob's posts of his positive ABX results as an example of just what's needed to do a valid test - half-hearted listening or listening for a somewhat vague & non-specific difference is a sure way to get a null result, as he outlines 

Let's look at Archimago's recent ABX tests of MQA where he gets 8/10 correct trials -he didn't do 16 trials (the recommended minimum) & many would reject this ABX test as invalid for the reason. But look at what he says about this "However, I did try to be more deliberate and "serious" than last time and took a number of minutes at the start to listen between the two files and pick a place to start where I thought I could familiarize myself with the slightly different sound."In other words he "trained himself" for this test

In his last ABX test of a 16/44 file VS MQA file (but not decoded through MQA) he achieved 7/10 correct (one less than above) & he concludes this "a quick and dirty ABX focused on a short passage just with standard DirectSound output at 16/44 setting (oops, forgot to tell foobar to use ASIO!). *Evidence that it's audible but not an obvious difference*."

So, he has admitted that he wasn't serious about this test, he used DS output & yet he states "Evidence that it's audible but not an obvious difference"

Please, this is the "science" you talk about? This is the "evidence" you are presenting? This is the "scientific approach" you are suggesting? This is kindergarten posing -"look I do _tests_ & get _evidence_ on which I base "conclusions" - see how scientific I am?" 



> > Does this mean that there isn't a difference that we can hear, no? This binary view of audio that uses a flawed test as the basis for its conclusions, is obviously badly mistaken.
> 
> 
> 
> ...



Please, just because you state this doesn't add any credence to it as having any veracity!


----------



## Joe Bloggs

> Please, this is the "science" you talk about? This is the "evidence" you are presenting? This is the "scientific approach" you are suggesting?




In other words, you utterly reject the empirical fact that one gets a big chance at scoring those nonsignificant results via a series of coin tosses?



mmerrill99 said:


> Please, just because you state this doesn't add any credence to it as having any veracity!




Do any of your statements lend credence to themselves in the first place? :rolleyes:



> *Does this mean that there isn't a difference that we can hear, no? This binary view of audio that uses a flawed test as the basis for its conclusions, is obviously badly mistaken.*




_"Please, just because you state this doesn't add any credence to it as having any veracity!"_

Doesn't seem to stop you from making hundreds of such statements though :rolleyes:


----------



## Joe Bloggs

Given how easily one passes "sighted" AB tests and the credence you lend toward them, why should I accept any blind ABX test results where the tester got even one wrong out of a hundred?

All the tester has to do is to carry the obvious differences he "heard" while sighted--and carry them over to the blind tests! :rolleyes:


----------



## mmerrill99

Ah well, let's bring this regression into pantomime posts to a close - it has become farcical


----------



## Joe Bloggs

Yes, what a farce :rolleyes:


----------



## gregorio

mmerrill99 said:


> Nope, I'm pointing out the weaknesses of ABX testing & the absolutely flawed use of it in a home run environment.


 
  
 Flawed? Yes. Absolutely? No! Thanks for proving with your reply exactly what I was asserting. Either you've read and understood science 101, in which case you are trolling or you need to read it and stop being a hypocrite!
  
 You're free to run your sighted tests if that's what convinces you or hell, just accept marketing "evidence" if you wish but don't come here trying to push your "science is flawed, therefore it's worthless" BS. Save it for audiophiles, creationists and flat-earthers who'll praise for your eloquence, rather than laugh at you!
  
 G


----------



## mmerrill99

gregorio said:


> mmerrill99 said:
> 
> 
> > Nope, I'm pointing out the weaknesses of ABX testing
> ...



What is evident from the replies here is the overwhelming desire to fit me into some category that can then be denigrated & dismissed.

Here's my summary of the position I see represented here - some of you state that "yes, ABX blind testing is flawed but not as flawed as sighted listening" So now it's just a matter of establishing which results are more flawed. These would be the results that I would give less credence to. I have given all my reasons for why home based ABX "tests" are, in my opinion, the more flawed.

I don't really see anything being advanced about the flawed nature of knowledge/sightedness. Some don't even seem to know that expectation bias is what's at play here. 

I asked a simple question about expectation bias - is it ALWAYS at play in sighted listening - if it is then the results of a blind test are irrelevant as once you stop listening blind & revert to sighted listening you will also revert back to your biased impression of what you hear. 

If expectation bias isn't ALWAYS at play in sighted listening then there is some sighted listening that isn't affected by this bias i.e it is just as valid as blind listening. People don't want to admit to this because they have adopted an unthinking binary view of the matter. 

So now we have some blind listening is flawed & some sighted listening is fine. 

You just have to work out your own way through this non-binary situation.

What I have been arguing against is the lazy-thinking, view that blind testing must be better than sighted listening


----------



## spruce music

mmerrill99 said:


> What I have been arguing against is the lazy-thinking, view that blind testing must be better than sighted listening


 
 And that is where you are making your mistake.  One you argue around the world to dodge having to admit that it is a mistake. 
  
 The central issue which you will no doubt refuse to acknowledge once again is the effects of bias.  Even if some sighted listening can be unbiased, sighted is the most easily biased method, the method subject to the most routes for bias to creep in, and gives the least consistent results.  The simplest act of blinding removes or reduces at least some of those.  Yet your curious reasoning is akin to sighted listening is biased, blind listening can be biased too, so sighted listening is at least as good as blind listening.  Sorry, your spiel does not make any sense. It is a naked rationalization to reach a conclusion you preferred to reach at the outset.


----------



## Joe Bloggs

mmerrill99 said:


> I don't really see anything being advanced about the flawed nature of knowledge/sightedness. Some don't even seem to know that expectation bias is what's at play here.




No, expectation bias isn't even half of what blind testing is fighting against. We get "I never expected it to make any difference, but now that I heard it with my own ears, this [paperweight audio scam doodad] really works!" all the time.

When one runs a sighted "test", anytime you begin to imagine A possibly having a certain characteristic over B, your brain starts looking for all sorts of "evidence" to confirm this "observation" in a self-enforcing cycle. The initial tipping point may have been caused by you slightly unseating your headphones or moving in your seat, changing the acoustics from A randomly over B; thereafter, no such actual difference in stimuli is needed anymore as your brain takes over.

What the audio world has demonstrated to us over and over again is, even a person who was in flat-out denial about there possibly being any difference in two different colored digital cables of the exact same construction will have at least a 50% chance of fooling himself into hearing night and day differences between them at some point--unless he has a rock solid understanding of both the physical facts of the two items being identical, AND the illusory effects of self-priming confirmation bias playing havoc with his perceptions.

THAT's why blind tests are the only way to go.


----------



## mmerrill99

spruce music said:


> mmerrill99 said:
> 
> 
> > What I have been arguing against is the lazy-thinking, view that blind testing must be better than sighted listening
> ...


The "simplest act of blinding" is not what I would call an ABX test where you have to do at least 16 trials & for each one identify if X is A or B. But there you go ignoring all that & calling it the simplistic. 





> Yet your curious reasoning is akin to sighted listening is biased, blind listening can be biased too, so sighted listening is at least as good as blind listening.  Sorry, your spiel does not make any sense. It is a naked rationalization to reach a conclusion you preferred to reach at the outset.


Nope, sighted listening CAN be biased towards false positive results, ABX blind CAN be biased towards false negative results - it's a matter of personal judgement which is more appropriate for your needs


----------



## KeithEmo

This whole thread really is getting silly - so much so that it's starting to be impossible for it to accomplish anything.
  
 This thread is _NOT_ the latest journal of some science association. It is merely a thread for audiophiles to discuss a specific issue based on science rather than pure opinion, or pseudo-science, or "subjectivism". And, to be honest, I see part of the purpose of this thread to be to encourage people to perform their own tests, following reasonable scientific protocols, and eliminating as much obvious error as possible. So, yes, claims based solely on opinions should be discouraged, and it's perfectly reasonable to criticize specific procedures and methodologies, but, even though your favorite standards organization may have decided that an ABX test must have at least 16 runs to satisfy _THEIR_ idea of being a "good" ABX test, the fact remains that any double-blind test is still better than a sighted test, and even four or five runs is still more information than none.
  
 Yes, of course, the more times you perform a test, the more you can eliminate the effects of randomness, and so the more statistically significant your results will be. However, statistical significance itself is a continuum. Tossing a coin and getting three heads in a row is statistically different than you would expect from random chance, but it isn't _MUCH_ different; ten heads in a row is a lot further from what you would expect by random chance; and, when you apply things like standard deviation, you will find that you can quantify how much less likely ten heads in a row is than three heads in a row... but there isn't some line where "anything above x is significant". In fact, if you actually research the subject, you will find that the amount of deviation that's required to consider something significant itself varies depending on the test. (If you flip a coin 1000 times, a result that's more than 5% from the norm may be very unusual; however, if you count the traffic through a certain intersection between 4 PM and 6 PM, you may find that a 20% variation from day to day is quite normal. And, if a hundred people each flip that coin five times, and 75 of them come up with four or more heads, then, in the aggregate, that is indeed a significant result.)
  
 My point is that, just because your favorite standards organization has decided that an ABX test requires at least sixteen runs to satisfy the criteria they've chosen for that particular type of test does _NOT_ mean that an ABX test with only five runs "isn't valid" or "isn't worth anything"... It simply means that the results of a single test with five runs are much _less_ conclusive, or _less_ meaningful, than the results with 16 runs. This is especially significant since we're talking about a group discussion here, and so we may get meta-data from multiple tests. So, for example, if twenty people each do an ABX test with five runs, and fifteen of them get a correct answer 4 times out of 5 or better, then, collectively, that data may actually be quite significant... because, while the odds of a single person getting 4 out of 5 may not be especially significant, the odds of fifteen people out of twenty each getting 4 out of 5 may be much higher - and so that result may be much more significant.
  
 Therefore, to put it bluntly, I think the goal of rational discussion is much better served if we _DON'T_ discourage everyone from performing a test just because their methodology, while reasonable, doesn't live up to some arbitrary standard... I'd much prefer to see a lot of people performing "pretty good" tests for themselves, than see them discouraged from doing so... as long as we all understand the limitations and significance of all the results..... (And, to put it even more bluntly, the discussion itself will help people with no experience to learn how to determine for themselves the difference between a really good test, and a pretty good test, and one that really is total junk - or pseudo-science.)
      
 To me, the only appropriate criticism of doing an ABX test with only five trials would be:
  
 "Five trials is a relatively low number, which means that, even though you've got the right idea, and your results are suggestive, you'd need a larger number of runs to produce a compelling result, and to more completely rule out the possibility that your results were due to random variations in results produced by random chance."


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > I don't really see anything being advanced about the flawed nature of knowledge/sightedness. Some don't even seem to know that expectation bias is what's at play here.
> ...


And what guides you to "imagine A possibly having a certain characteristic over B" - just random thoughts?



> What the audio world has demonstrated to us over and over again is, even a person who was in flat-out denial about there possibly being any difference in two different colored digital cables of the exact same construction will have at least a 50% chance of fooling himself into hearing night and day differences between them at some point--unless he has a rock solid understanding of both the physical facts of the two items being identical, AND the illusory effects of *self-priming confirmation bias* playing havoc with his perceptions.


Ah, "]self-priming confirmation bias" what is this a confirmation of - an expectation, perhaps?



> THAT's why blind tests are the only way to go.
> 
> Of course, you won't find many industry insiders talking about this, because it's apt to shut down 99% of the industry's revenue if everybody took this seriously. I only say these things here because at this point there's not a snowflake in hell's chance of anybody other than you taking this discussion seriously anymore.


OK, so let's all have a laugh & agree to disagree, right?


----------



## L8MDL

I put my Pono on random and read every message in this thread while it played a varied selection of cd and hires selections through my Sennheiser HD600's with a Fiio E12. Each time I thought it was playing a hires file, I looked for the "blue light ". I got it right 64 out of 72 times. Does this prove anything?


----------



## Joe Bloggs

mmerrill99 said:


> And what guides you to "imagine A possibly having a certain characteristic over B" - just random thoughts?




It was right on top of your head--*"The initial tipping point may have been caused by you slightly unseating your headphones or moving in your seat, changing the acoustics from A randomly over B;"*, although random mood swings can serve just as well



> Ah, "]self-priming confirmation bias" what is this a confirmation of - an expectation, perhaps?




Not an expectation in the sense of your expectation of whether there is a difference or not going into the test--but the expectations that can be set up by the random process described above--always favoring a difference. Once a difference appears because of some random fluctuation, your brain latches onto it, expands on it and continues to insist on it long after the initial difference has disappeared into the noise.


----------



## mmerrill99

joe bloggs said:


> mmerrill99 said:
> 
> 
> > And what guides you to "imagine A possibly having a certain characteristic over B" - just random thoughts?
> ...




If it's just random events, as you suggest, then we would see just randomness in people's sighted listening & no particular device/protocol would be favoured more than any other!!


----------



## mmerrill99

l8mdl said:


> I put my Pono on random and read every message in this thread while it played a varied selection of cd and hires selections through my Sennheiser HD600's with a Fiio E12. Each time I thought it was playing a hires file, I looked for the "blue light ". I got it right 64 out of 72 times. Does this prove anything?



I bet there are many here who will argue that your test proves nothing because it was not rigorous enough - we don't know if you actually were able to see the blue light out of the corner of your eye prior to making your determination, we don't know if you already know which of your tracks are high-res & which RB, etc. 

So when a test returns positive results the rigour of the test is examined but when it returns null results my querying of the underlying rigour of the test is questioned.


----------



## KeithEmo

Expectation bias is _ALWAYS_ in play.... most of what we experience in our lives is affected to some greater or lesser degree by what we expect to experience. Expectation bias is in fact a general term; in audio, it could mean that we expect a more expensive product to sound better, so we imagine it does; it could also mean that we like that new song by our favorite group just a little bit more than if we didn't know who was playing it; and it can be the opposite - it can mean that we hear a difference between two things, but, since we're convinced that they should be the same, we decide that we only imagined it. There are even arguably several different mechanisms involved. However, what they have in common is that they all cause our perceptions to be "skewed" in one way or another rather than being based entirely on reality.
  
 The whole point, in the specific context of identifying whether differences between specific audio products are audible or not, is that we want to eliminate any and all forms of bias so we can determine what the actual differences are in the product itself - and we can do this most effectively simply by not knowing which product we're listening to. (And, yes, it would be quite possible for a NEGATIVE expectation bias to lead to a false null. However, at least in this discussion, the goal of the test is usually to debunk a difference that someone claims is audible. And, if the difference only exists when we know which product we're listening to, then we can logically infer that the difference is in our mind and not in the actual product, and so that it is due to some sort of bias.)
  
 Quote:


mmerrill99 said:


> What is evident from the replies here is the overwhelming desire to fit me into some category that can then be denigrated & dismissed.
> 
> Here's my summary of the position I see represented here - some of you state that "yes, ABX blind testing is flawed but not as flawed as sighted listening" So now it's just a matter of establishing which results are more flawed. These would be the results that I would give less credence to. I have given all my reasons for why home based ABX "tests" are, in my opinion, the more flawed.
> 
> ...


----------



## mmerrill99

keithemo said:


> Expectation bias is ALWAYS in play.... most of what we experience in our lives is affected to some greater or lesser degree by what we expect to experience. Expectation bias is in fact a general term; in audio, it could mean that we expect a more expensive product to sound better, so we imagine it does; it could also mean that we like that new song by our favorite group just a little bit more than if we didn't know who was playing it; and it can be the opposite - it can mean that we hear a difference between two things, but, since we're convinced that they should be the same, we decide that we only imagined it. There are even arguably several different mechanisms involved. However, what they have in common is that they all cause our perceptions to be "skewed" in one way or another rather than being based entirely on reality.
> 
> The whole point, in the specific context of identifying whether differences between specific audio products are audible or not, is that we want to eliminate any and all forms of bias so we can determine what the actual differences are in the product itself - and we can do this most effectively simply by not knowing which product we're listening to. (And, yes, it would be quite possible for a NEGATIVE expectation bias to lead to a false null. However, at least in this discussion, the goal of the test is usually to debunk a difference that someone claims is audible. And, if the difference only exists when we know which product we're listening to, then we can logically infer that the difference is in our mind and not in the actual product, and so that it is due to some sort of bias.)



Keith, 
I'll put the question I posed before more precisely - if by ALWAYS you mean that we are therefore slaves to our expectation bias & the only way to avoid this is by blind listening, then there is no point in these tests as when we resumed listening normally (sighted) we will be back to hearing what our bias determines we hear.

If, however, by doing a blind test we now have removed the influence of the bias & we can now listen sighted but bias free, please explain the mechanism by which the expectation bias no longer operates

The final possibility is that the blind test changes our positive expectation to one of a negative expectation - i.e we imagine that the two device now sound the same because we couldn't determine any difference in the ABX blind test (no matter how flawed this test is)


----------



## JWolf

mmerrill99 said:


> I bet there are many here who will argue that your test proves nothing because it was not rigorous enough - we don't know if you actually were able to see the blue light out of the corner of your eye prior to making your determination, we don't know if you already know which of your tracks are high-res & which RB, etc.
> 
> So when a test returns positive results the rigour of the test is examined but when it returns null results my querying of the underlying rigour of the test is questioned.


 
  
 This is exactly why ABX is worthless. If it turns out that the results favor Hi-Res, then the results are questioned. STOP IT!


----------



## dprimary

Quote:
  
 Originally Posted by *L8MDL* /img/forum/go_quote.gif

 I put my Pono on random and read every message in this thread while it played a varied selection of cd and hires selections through my Sennheiser HD600's with a Fiio E12. Each time I thought it was playing a hires file, I looked for the "blue light ". I got it right 64 out of 72 times. Does this prove anything?


mmerrill99 said:


> I bet there are many here who will argue that your test proves nothing because it was not rigorous enough - we don't know if you actually were able to see the blue light out of the corner of your eye prior to making your determination, we don't know if you already know which of your tracks are high-res & which RB, etc.
> 
> So when a test returns positive results the rigour of the test is examined but when it returns null results my querying of the underlying rigour of the test is questioned.


 
 Maybe it proves something maybe it doesn't. Without the details it is meaningless.  Asking for more information is not shooting down the possibility that they hear a difference. I am hoping they had "hi res" and "low res versions of the same song. If not then it might prove they can remember which songs are "high res".
  
 If it is the same song is it from the same master? I have heard "high res" versions that only was it not the same master it was not the same mix and did not seem to even be the same take, of a song that was pretty well known.  
  
 If it is from the same source, how was it encoded? If it was encoded to MP3 at 128k using a 15 year old version of winamp. I think I might have good chance of picking out the MP3 having never even heard the original. Is it comparing raw 16/44.1 to raw 24/96? then that is completely different.
  
 Then you get to - how did the samples get prepped? are they the same level? was one clipped, eq'ed, distorted, compressed? There is plenty of ways make a mistake.
  
 If you are comparing a 16/44.1 MP3 at 256k to a FLAC 24/96 you are not really comparing 16/44.1 to 24/96 you are comparing lossy to lossless. Most people will not be able to detect a difference. A few will depending on the song. Even so not like you are going to be able to detect the difference riding on a subway.
  
 Comparing 16/44.1 AAC at 320k to 16/44.1 lossless is not impossible it is pretty difficult. You really have be completely focused in low noise room. If it is night and day as if often claimed then I would suspect the decoder to be broken or you are not testing it by listening but by some other means.


----------



## sonitus mirus

From a technical perspective, it may be possible for some listeners to identify the FR rolloff when playing a FLAC rip of a CD using 44.1kHz sample rate. (see Fig. 3 in the link below)
  
 http://www.stereophile.com/content/pono-ponoplayer-portable-music-player-measurements#XTOUiBXGPWaEDsOm.97
  
 This is simply a poor characteristic of the Pono player itself, which was presumably created to cater to folks using files with higher sample rates, but it makes any comparison with CD quality unfair with regards to the format.


----------



## dprimary

sonitus mirus said:


> From a technical perspective, it may be possible for some listeners to identify the FR rolloff when playing a FLAC rip of a CD using 44.1kHz sample rate. (see Fig. 3 in the link below)
> 
> http://www.stereophile.com/content/pono-ponoplayer-portable-music-player-measurements#XTOUiBXGPWaEDsOm.97
> 
> This is simply a poor characteristic of the Pono player itself, which was presumably created to cater to folks using files with higher sample rates, but it makes any comparison with CD quality unfair with regards to the format.


 

 That could be noticeable to some people. It is not up to the level of what I expect on a mid level piece of equipment. I was really interested in the Pono when it came out since it has balanced output and I need to interface to professional equipment most of the time. After many attempts to get complete specifications on the device or even find someone that had tested the balanced interface feeding pro gear I gave up and bought something else.


----------



## spruce music

dprimary said:


> That could be noticeable to some people. It is not up to the level of what I expect on a mid level piece of equipment. I was really interested in the Pono when it came out since it has balanced output and I need to interface to professional equipment most of the time. After many attempts to get complete specifications on the device or even find someone that had tested the balanced interface feeding pro gear I gave up and bought something else.


 

 Did you find a similar device with balanced out, and if so what was it?


----------



## watchnerd

jwolf said:


> This is exactly why ABX is worthless. If it turns out that the results favor Hi-Res, then the results are questioned. STOP IT!


 
  
 I'm confused. Stop what?


----------



## OddE

jwolf said:


> This is exactly why ABX is worthless. If it turns out that the results favor Hi-Res, then the results are questioned. STOP IT!


 
  
 -Guess why?
  
 A result favoring hi-res flies in the face of established science. Hence, one would be more inclined to ask questions as to methodology, source material and any biases which may have influenced the result than if the result was 'Wasn't able to tell any difference.'
  
 The flip side being that if questions were answered satisfactorily and results proved reproduceable, it would be a significant discovery.
  
 Sounds like a fair tradeoff to me.


----------



## gregorio

mmerrill99 said:


> I asked a simple question about expectation bias - is it ALWAYS at play in sighted listening - if it is then the results of a blind test are irrelevant as once you stop listening blind & revert to sighted listening you will also revert back to your biased impression of what you hear.


 
  
 I don't know for certain expectation bias is ALWAYS at play, although I strongly suspect that it probably is in the vast majority of cases, whether we are consciously aware of it or only sub-consciously. Your assertion that after an ABX test we will still have expectation bias would therefore be correct. However, the next part of your assertion is ridiculous!
  
 Let's take a simple example, let's say I'm testing two sets of speakers, A & B. Maybe I have an expectation that set A is better, maybe I don't know if set A is better but just hope they are. I do a sighted test and indeed I do have a preference for A. Then I do an ABX and discover that actually I can't discern any difference between them. After both the sighted and ABX tests I indeed still have expectation bias. However, my expectation is different, it has changed, it does NOT "revert back to my biased impression" of what I heard before the ABX test. In this example, if my expectation were unaffected in anyway by the result of the ABX then I would be a fool!
  
 The reality of the situation is not quite so absolute as my example indicates however. Anyone with a reasonable knowledge/experience of audio knows there are a lot of variables at play and even laboratory conditions ABX testing is not generally an absolute answer. In the example above, speakers A may, for example, have a slightly different dispersion pattern which may cause them to perform noticeably better (in an ABX test) than B, given different positioning or different test material or, my ABX test might have been flawed in some other way. ABX is simply additional (and all else being equal, more reliable) information, rather than absolute proof. I'm not sure why you find this so difficult to understand. Maybe you are just a fool, or maybe you do understand but are trolling or have some other agenda? Either way, your arguments are nonsensical/ridiculous and serve no purpose for either you or us. Again, why don't you try say the Cables forum, where your assertions are true, where ABX is indeed an impediment to the ethos and progress of that forum.
  
 G


----------



## mmerrill99

gregorio said:


> mmerrill99 said:
> 
> 
> > I asked a simple question about expectation bias - is it ALWAYS at play in sighted listening - if it is then the results of a blind test are irrelevant as once you stop listening blind
> ...


Right & there you have an example of expectation bias being nullified or changed. You are now listening sighted without this bias affecting what you hear. So why can't you do this prior to doing a blind test? Why do you have to do a blind test in order to achieve this - is there something magical about a blind test? No? It's just a case of being aware of your expectations. They are not the overriding, constant & ubiquitous influence that many here try to make them out to be.

Of course the other possibility is not that a blind test is nullifying your expectation bias but that it's changing it into a negative expectation bias i.e that you are now convinced & expect that you won't hear any difference between A & B in sighted listening - guess what, you won't.

And as I said already, I'm far more willing & at ease with the possibility that some subconscious influence is affecting my conclusions in sighted listening than introducing all the influences & possible new biases that blind listening entails 



> The reality of the situation is not quite so absolute as my example indicates however. Anyone with a reasonable knowledge/experience of audio knows there are a lot of variables at play and even laboratory conditions ABX testing is not generally an absolute answer. In the example above, speakers A may, for example, have a slightly different dispersion pattern which may cause them to perform noticeably better (in an ABX test) than B, given different positioning or different test material or, my ABX test might have been flawed in some other way. ABX is simply additional (and all else being equal, more reliable) information, rather than absolute proof. I'm not sure why you find this so difficult to understand. Maybe you are just a fool, or maybe you do understand but are trolling or have some other agenda? Either way, your arguments are nonsensical/ridiculous and serve no purpose for either you or us. Again, why don't you try say the Cables forum, where your assertions are true, where ABX is indeed an impediment to the ethos and progress of that forum.
> 
> G


Well you can call me a fool & a troll & my arguments nonsensical but I'll let non-biased readers (is there such a thing?) make up their own mind. Again, I'll say it, home based ABX testing is fraught with so many flaws that it is risible anybody would categorise it as scientific or of any worth


----------



## castleofargh

l8mdl said:


> I put my Pono on random and read every message in this thread while it played a varied selection of cd and hires selections through my Sennheiser HD600's with a Fiio E12. Each time I thought it was playing a hires file, I looked for the "blue light ". I got it right 64 out of 72 times. Does this prove anything?


 
 taken from stereophile:

  


> Fig.3 Pono PonoPlayer, frequency response at –12dBFS into 100k ohms with data sampled at: 44.1kHz (left channel cyan, right magenta), 96kHz (left green, right gray), 192kHz (left blue, right red) (0.5dB/vertical div.).


 
 it may not be the only reason, because some design choices on the pono are special. and maybe the highres one takes a little longer to load and gives you a hint, or as suggested, maybe some visual cue?. but the roll off when playing 16/44 files is big enough to potentially be audible. so as far as I'm concerned I believe you could identify the highres files. I'm just not sure it has much to do with the quality of the file itself.
  
 you could just upsample your 16/44 and try to see if you still can tell the difference as if it was highres music.


----------



## dprimary

spruce music said:


> Did you find a similar device with balanced out, and if so what was it?


 

 I gave up on a pocket sized device and bought apogee convertors that would connect to my existing phone, tablets, and computers. It was double the money but it has mic preamps and line inputs as well.


----------



## gregorio

mmerrill99 said:


> Right & there you have an example of expectation bias being nullified or changed. You are now listening sighted without this bias affecting what you hear. So why can't you do this prior to doing a blind test?


 
  
 Err, because we're human beings, constantly subjected to biases. OK, I realise this argument won't sway some audiophiles who appear to believe they are in fact not human beings but the next evolutionary step, with super-powers which allow them to hear things normal human beings can't and dismiss biases to perceive the world as it really is. Maybe I'm being unfair, maybe those audiophiles don't believe in evolution, it's only a scientific theory after all!
  


mmerrill99 said:


> i.e that you are now convinced & expect that you won't hear any difference between A & B in sighted listening - guess what, you won't.


 
  
 I am not convinced of anything! Did you actually read what I wrote or are you just taking quotes out of context to deliberately misrepresent, in order to support your agenda?
  


mmerrill99 said:


> Again, I'll say it home based ABX testing is fraught with so many flaws that it is risible anybody would categorise it as scientific or of any worth.


 
  
 Yes they are but AGAIN, you are deliberately missing the point. However many flaws home based ABX testing has, it has considerably fewer than the equivalent sighted test and is therefore of considerably more "worth" than an equivalent sighted test. Come on, this isn't a difficult concept to grasp!
  


mmerrill99 said:


> Well you can call me a fool & a troll & my arguments non-sensical ...


 
  
 OK then ... You are a fool and a troll and your arguments are nonsensical!
  
 G


----------



## gregorio

.


----------



## mmerrill99

gregorio said:


> mmerrill99 said:
> 
> 
> > Right
> ...


So let me get it straight what you are saying - prior to a blind test you are racked with biases affecting what you hear - after a blind test you aren't! How does that work?



> mmerrill99 said:
> 
> 
> > i.e that you are now convinced
> ...


It's your claim based on only one factor - that removing knowledge/sight of what you are listening to is less flawed. But I've already showed you that you are introducing a gaggle of other factors in an ABX test that simply don't exist when listening sighted. All these factors influence the outcome of the test so it's not less flawed - it's more flawed



> mmerrill99 said:
> 
> 
> > Well you can call me a fool
> ...


Thank you for showing that you have resorted to insults to try winning the argument - in other words, you have no argument


----------



## charleski

mmerrill99 said:


> Right & there you have an example of expectation bias being nullified or changed. You are now listening sighted without this bias affecting what you hear. So why can't you do this prior to doing a blind test? Why do you have to do a blind test in order to achieve this - is there something magical about a blind test? No? It's just a case of being aware of your expectations. They are not the overriding, constant & ubiquitous influence that many here try to make them out to be.


 
  
 If you don't do the blind test then you _simply don't know_ whether any difference heard in sighted listening is a result of bias.
  
 If I buy a shiny new gadget then I _really want_ to think it's better than the old one and that I didn't waste my money. I can't just wish that bias away. The only way to do that would be to perform a blind test. Even then, let's be honest, there are countless examples of people claiming they can still hear a difference when sighted even after they fail a blind test. This sort of bias is perfectly strong enough to make them reject the rational conclusion. People just don't like to think they've been duped, and are quite willing to co-operate in maintaining the delusion in order to assuage their pride. Being objective about this is not easy.
  
 All this talk of negative bias is just a rather desperate red herring. I've already pointed to a recent test of 24bit audio in which plenty of the participants had positive bias and fully expected to be able to discriminate properly, but failed to do so.


----------



## JWolf

watchnerd said:


> I'm confused. Stop what?


 
  
 If someone does an ABX, and is able to mostly identify Hi-Res, others jump on said person.


----------



## OddE

jwolf said:


> If someone does an ABX, and is able to mostly identify Hi-Res, others jump on said person.


 
  
 See my earlier comment (quoted below for convenience)
  


odde said:


> -Guess why?
> 
> A result favoring hi-res flies in the face of established science. Hence, one would be more inclined to ask questions as to methodology, source material and any biases which may have influenced the result than if the result was 'Wasn't able to tell any difference.'
> 
> ...


----------



## Joe Bloggs

mmerrill99 said:


> Thank you for showing that you have resorted to insults to try winning the argument - in other words, you have no argument




Step 1. Produce a non-stop barrage of pseudoscientific arguments, all the while being impervious to logic and actual scientific arguments. Ignore all well constructed arguments against your position and twist every seeming contradiction or unknown in the science to mean that the science doesn't have a leg to stand on. Always post in a smug know-it-all tone that irritates the hell out of anybody who really knows anything on the topic while appearing superior to an unsuspecting audience.
Step 2. Invite well-deserved insults from frustrated opponents who have nothing else to throw at a target that has made itself immune to all sensible arguments.
Step 3. Profit?

Shall I call this the "Head-Fi puddinghead school of argumentation" --a result of exploiting the Head-Fi legal loophole where you can forever politely go about being insultingly, deliberately dense in a scientific argument that should really call for SCIENCE--and never get called on it?


----------



## gregorio

mmerrill99 said:


> So let me get it straight what you are saying ...


 
  
 Translation: "Let me misrepresent what you are saying to better serve my agenda"!
  


mmerrill99 said:


> I've already showed you that ...


 
  
 Yes, an absolute perfect proof. What a truly great scientist you are.
  


mmerrill99 said:


> Thank you for showing that you have resorted to insults to try winning the argument - in other words, you have no argument


 
  
 No, thank YOU for inviting me to insult you, much appreciated. BTW; pot, kettle, black.
  
 G


----------



## Quadfather

The hdtracks.com versions of Rush catalog is superior to the recent,  lifeless CD remasters, but they may have had more care in the mastering.


----------



## bigshot (Oct 22, 2017)

On the internet there is always one other option... that someone is outright lying to make their point.

I had that happen with a lossy listening test I gave out to a hardcore audiophile who claimed that lossy always sounded bad. I gave him the ground rules- listening test only. He agreed. He took the test and came back the next day knowing exactly which one was the lossless track. But the way he identified the track showed me that he hadn't listened. He had the order of the tracks wrong and could only give me a time code reading. So I asked him to rank the other tracks from best to worst so he could tell me which ones sounded better than the others. I took advantage of the time it took him to reply to do a little quick googling of his username and found him on that same day talking in another forum about how to identify lossy tracks by looking at the waveform.

The next day he came back and haughtily refused to rank the lossy tracks, saying that it "wasn't worth his time" because the difference was "night and day". I asked if he had listened to all of them and not just opened them up in a sound editor per the rules, and he got huffy and swore that he didn't even own a sound app, and that could hear the differences clear as day. He finished up suggesting that I must be deaf. That's when I PM'ed him the cut and paste from the other forum. I pointed out that the date of his comments was the same day he posted the answers to his listening test and he was admitting that he did actually have a sound app. No reply. He never came back to that forum again.

It's OK to take people at their word if they are really looking for an answer. But if someone wants to prove a point regardless if it's true or not, you have a right to be a little suspicious of their claims.


----------



## RRod

Joe Bloggs said:


> Step 1. Produce a non-stop barrage of pseudoscientific arguments, all the while being impervious to logic and actual scientific arguments. Ignore all well constructed arguments against your position and twist every seeming contradiction or unknown in the science to mean that the science doesn't have a leg to stand on. Always post in a smug know-it-all tone that irritates the hell out of anybody who really knows anything on the topic while appearing superior to an unsuspecting audience.
> Step 2. Invite well-deserved insults from frustrated opponents who have nothing else to throw at a target that has made itself immune to all sensible arguments.
> Step 3. Profit?
> 
> Shall I call this the "Head-Fi puddinghead school of argumentation" --a result of exploiting the Head-Fi legal loophole where you can forever politely go about being insultingly, deliberately dense in a scientific argument that should really call for SCIENCE--and never get called on it?



https://en.wikipedia.org/wiki/Gish_gallop


----------



## frodeni

Wow! Hey guys, calm down!

There are a lot of people in this very forum, that stick to a very strict form of positivism. That is, that they believe that the world can be quantified and described, and reject the notion of subjectivity. Others, like me, belong in the tradition of the interpretive paradigm, in which knowledge is made by the intersubjectivity of subjects.

Sure, it is fine to disagree. But when people claim that people with different believes, assumptions, understanding, and knowledge, are stupid and dumb for having a different background, then go get some ice-cream, cool down. That is not cool.

To stay in the conversation in here, I would need a lot of ice cream. The level of insults are insane. Which is a pity. This is a hobby of mine, both listening and philosophy, so I would love a friendly and civil conversation about this topic.


----------



## Quadfather

frodeni said:


> Wow! Hey guys, calm down!
> 
> There are a lot of people in this very forum, that stick to a very strict form of positivism. That is, that they believe that the world can be quantified and described, and reject the notion of subjectivity. Others, like me, belong in the tradition of the interpretive paradigm, in which knowledge is made by the intersubjectivity of subjects.
> 
> ...



I am in the "I don't give a damn what others or God himself thinks of me or my opinions" camp.  LOL P.S. Full Terror Assault 3 was awesome this September.


----------



## castleofargh

frodeni said:


> Wow! Hey guys, calm down!
> 
> There are a lot of people in this very forum, that stick to a very strict form of positivism. That is, that they believe that the world can be quantified and described, and reject the notion of subjectivity. Others, like me, belong in the tradition of the interpretive paradigm, in which knowledge is made by the intersubjectivity of subjects.
> 
> ...


AFAIK the only people to ever reject subjectivity are some of the self proclaimed subjectivists. when we decide to take an objective approach to a problem, it is because we understand the difference between reality and our perception of it. so when we question feelings we answer subjectively, and when we question objective reality, we try to answer with an objective approach. it's not so much philosophy or positivism, and more about using the right tool for the right job.
and if we seem to insist on the objective aspect of sound so much, maybe it is because it is the purpose of that subsection of the forum...

as for your claim of claim about people being stupid and dumb, maybe you need a double dose of ice cream. when somebody goes overboard, any forum member can report the post and obviously moderation would deal with it. when nothing is done, you can conclude that at the very least, that nobody got offended enough to press a button. I feel that we can survive just fine under such anticlimactic circumstances.


----------



## bigshot

frodeni said:


> To stay in the conversation in here, I would need a lot of ice cream. The level of insults are insane. Which is a pity. This is a hobby of mine, both listening and philosophy, so I would love a friendly and civil conversation about this topic.




One thing to keep in mind is that if someone attacks the argument, they aren't attacking the person saying it. Challenging opinions is fine. Ad hominem attacks against the person with the opinions aren't. Some of us understand that, some folks have a weaker grip on that concept. With those folks, you have two options- give them a shot across the bow reminding them what they're doing isn't right, or just ignore them entirely and let them earn their way back into your attention.


----------



## frodeni

Quadfather said:


> I am in the "I don't give a damn what others or God himself thinks of me or my opinions" camp.  LOL P.S. Full Terror Assault 3 was awesome this September.



Well, if you do not care what others think, why be in a community with others, like a forum?



castleofargh said:


> AFAIK the only people to ever reject subjectivity are some of the self proclaimed subjectivists.



Care to describe the form of "subjectivity" you speak of here? Are you claiming to be an interpretivists, in which you claim that knowledge is made through the experience of the individual, and constructed by communicating with others? I sort of lost you on this one.



castleofargh said:


> ... when we decide to take an objective approach to a problem, it is because we understand the difference between reality and our perception of it.



Which is a dead old positivists stand. That subjectivitiy can be eliminated by reason, and that the world, including perception can be described and quantified. That emotions can be rationalized, and described in universal terms. The exact opposite of any interpretivist would tell you.



castleofargh said:


> ... so when we question feelings we answer subjectively, and when we question objective reality, we try to answer with an objective approach.



And this "objective" approach, is not made by humans? Is not based on a common intersubjective understanding of the world? Not made of language, that is nothing but a common understanding between people on what words means? The positivist claim to be able to describe the world, to quantify it, through their own subjectivity, then claiming it to be made without any subjectivity. How does that work?[/QUOTE]



castleofargh said:


> it's not so much philosophy or positivism, and more about using the right tool for the right job.



Well, if you make your picks based on a postivistic stand, choose your methodology solely based on positivism, which you do, then i disagree. When you claim to be able to tell me, what I hear or not, because you do not grasp the limits of your philosophy, then get rude an attacks me, which you have in the past, then it is all about flawed philosophy. Because that's what it is. Sure, you got valued points and angles, but there are colors in the rainbow. There is no black and white in any rainbow.

Your main flaw, is not grasping the limits of your reasoning. You may claim that Santa does not exists. Sure. No proof ever of him living on the north pole. But he enters a lot of living rooms for Christmas, so something exists. There are a ton of tales of figures about Santa. Which get really, really messy to explain with positivist terms.

The flip part, is when trying to use interpretive measures to explain physics or math. To try to find the universal mechanisms of nature. That is why most people, when given the right question, actually belongs in both camps. At least those who are sane.

For what this forum is supposed to cover, it a very strange thing to read that people get music reproduced, because they do not. They get sound waves reproduced. So how does that work, if both the fruits of the nature science, and the ability to infer sound waves were present? That inferring is subjective, how ever you might twist it. There is no human in your speaker. Nor any guitar. It is all in your and my head.



castleofargh said:


> and if we seem to insist on the objective aspect of sound so much, maybe it is because it is the purpose of that subsection of the forum...



Which makes this very forum one of the most obnoxious sub forums ever. As there is no real understanding of how to conduct proper science by proper scientific standards. Even for postivistic work. As in rigging test to achieve specific results, not the correct one. That by far, makes it a political, and purely political forum.



castleofargh said:


> as for your claim of claim about people being stupid and dumb, maybe you need a double dose of ice cream. when somebody goes overboard, any forum member can report the post and obviously moderation would deal with it. when nothing is done, you can conclude that at the very least, that nobody got offended enough to press a button. I feel that we can survive just fine under such anticlimactic circumstances.



Sure. Let's see how this works out. If there is room for the traditional discussion on scientific validity in this very forum. Which this very post is. If people are up to it. Or if it is just talk, while being offensive: Promoting a political agenda.

There are great articles at Wikipedia for both epistemology and ontology, which should enable people to find further great sources for this very topic. If you got access to a scientific library, there are more articles available than any normal human could ever manage to read.

So, can you take it, faced with someone who know their philosophy, and tells you, using traditional philosophy, that your argument is flawed? Do you dear to ask why? Or are you going to delete opposition. Again. Even opposition firmly based in the sciences?

In the community, there is a need to unite both camps. Getting to grips with the strength and weaknesses of both camps. As to try to gain both knowledge and understanding. (Would be great, if anyone could tell me why those two specific terms. Then there i probably more than one scientists in here.)

As for instance, why I cannot tell any difference of lossy and lossless compression, if I use my laptop as a source. Because I cannot. (As opposed to my PC) Why does the buffer setting influence the result as much as it does, because as an IT student, it really should not. But it does. Why? The sonic traits, clearly defined traits are all there.

How things is experienced and how the differences come to existence, is not understood at all. Also, what what is "organic" reproduction of sound? How does that manifest itself, as in distinct sonic traits? A common language is sorely needed.

If only people in this forum could drop the arguing, and just share in a friendly manner, quit the mocking. This forum could be a really interesting place. There is a lot of thought put into a lot of posts, and if presented in the spirit of making a great community in which we try to get things moving forward, things could turn really interesting.


----------



## jnorris

This is getting out of hand.  Here is the bottom line...
Recordings at 96k/24 do, unquestionably have the _potential_ for better sound.  Since these files are flat out to 48KHz, they _may_ have the _potential_ to carry oscillations or other anomalies that may harm the speakers.

Whether you can hear the difference is entirely up to the ears (and prejudices) of the beholder.

Moving on.


----------



## bigshot

You fellas sure are all worked into a lather! Maybe that's why you were banished to Sound Science. The thing is though, here at Sound Science we are all pretty nice folks and we're happy to chat about stuff calmly here. You aren't out in crazy town any more. Join our party and have fun.


----------



## JWolf

jnorris said:


> This is getting out of hand.  Here is the bottom line...
> Recordings at 96k/24 do, unquestionably have the _potential_ for better sound.  Since these files are flat out to 48KHz, they _may_ have the _potential_ to carry oscillations or other anomalies that may harm the speakers.
> 
> Whether you can hear the difference is entirely up to the ears (and prejudices) of the beholder.
> ...



The thing is, if there are going to be issues by hitting boundaries,  then having a spec that goes past those boundaries you don't have those issues. Also, in some 24/96 recordings, you do see the sound does go past 20kHz. Plus, with 24-bit, you have more bits to give more resolution. With 16-bit if you have bit error(s), you get more of the sound with 24-bits as you have more data to reconstruct the waveform.


----------



## Whazzzup

To me its obvious 192 kHz and 24 bit just sounds more full, intimate, and less sharp than 16 bit 44.1.


----------



## bigshot (Oct 25, 2017)

Jwolf, if all the added resolution of 24 bit is below the level where you can’t hear it, and all the additional superaudible frequencies are beyond your ears’ ability to hear, then you’re creating a hifi  for bats and dogs, not yourself. Redbook already is overkill.

Wazzup, the reason high sampling rates sound less sharp is probably because your equipment isn’t designed to deal with superaudible frequencies and they’re causing distortion down in the audible range


----------



## Quadfather (Oct 25, 2017)

frodeni said:


> Well, if you do not care what others think, why be in a community with others, like a forum?
> 
> 
> 
> ...





Well, if you make your picks based on a postivistic stand, choose your methodology solely based on positivism, which you do, then i disagree. When you claim to be able to tell me, what I hear or not, because you do not grasp the limits of your philosophy, then get rude an attacks me, which you have in the past, then it is all about flawed philosophy. Because that's what it is. Sure, you got valued points and angles, but there are colors in the rainbow. There is no black and white in any rainbow.

Your main flaw, is not grasping the limits of your reasoning. You may claim that Santa does not exists. Sure. No proof ever of him living on the north pole. But he enters a lot of living rooms for Christmas, so something exists. There are a ton of tales of figures about Santa. Which get really, really messy to explain with positivist terms.

The flip part, is when trying to use interpretive measures to explain physics or math. To try to find the universal mechanisms of nature. That is why most people, when given the right question, actually belongs in both camps. At least those who are sane.

For what this forum is supposed to cover, it a very strange thing to read that people get music reproduced, because they do not. They get sound waves reproduced. So how does that work, if both the fruits of the nature science, and the ability to infer sound waves were present? That inferring is subjective, how ever you might twist it. There is no human in your speaker. Nor any guitar. It is all in your and my head.



Which makes this very forum one of the most obnoxious sub forums ever. As there is no real understanding of how to conduct proper science by proper scientific standards. Even for postivistic work. As in rigging test to achieve specific results, not the correct one. That by far, makes it a political, and purely political forum.



Sure. Let's see how this works out. If there is room for the traditional discussion on scientific validity in this very forum. Which this very post is. If people are up to it. Or if it is just talk, while being offensive: Promoting a political agenda.

There are great articles at Wikipedia for both epistemology and ontology, which should enable people to find further great sources for this very topic. If you got access to a scientific library, there are more articles available than any normal human could ever manage to read.

So, can you take it, faced with someone who know their philosophy, and tells you, using traditional philosophy, that your argument is flawed? Do you dear to ask why? Or are you going to delete opposition. Again. Even opposition firmly based in the sciences?

In the community, there is a need to unite both camps. Getting to grips with the strength and weaknesses of both camps. As to try to gain both knowledge and understanding. (Would be great, if anyone could tell me why those two specific terms. Then there i probably more than one scientists in here.)

As for instance, why I cannot tell any difference of lossy and lossless compression, if I use my laptop as a source. Because I cannot. (As opposed to my PC) Why does the buffer setting influence the result as much as it does, because as an IT student, it really should not. But it does. Why? The sonic traits, clearly defined traits are all there.

How things is experienced and how the differences come to existence, is not understood at all. Also, what what is "organic" reproduction of sound? How does that manifest itself, as in distinct sonic traits? A common language is sorely needed.

If only people in this forum could drop the arguing, and just share in a friendly manner, quit the mocking. This forum could be a really interesting place. There is a lot of thought put into a lot of posts, and if presented in the spirit of making a great community in which we try to get things moving forward, things could turn really interesting.[/QUOTE]


frodeni said:


> Well, if you do not care what others think, why be in a community with others, like a



 I am in this thread to analyze others' views and to read through the scientific data, and together with my own subjective tastes and biases form my own opinions about what I prefer in terms of bit rates and kilohertz and such. Some of it is very interesting. If somebody disagrees with some opinions I form based on what's in this forum that's what I don't care about. If I like it, I like it.  If I don't, I don't.  It's that simple.


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## jnorris

I agree with you with a few qualifications.  It is not resolution that is improved by 24 bits, it is dynamic range.  And moving boundaries well above the range of hearing does eliminate the anomalies incurred by the 22K boundary, but it may let ultrasonics through that may have been removed from a 44.1/16 master.


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## 71 dB

jnorris said:


> Recordings at 96k/24 do, unquestionably have the _potential_ for better sound.
> 
> Whether you can hear the difference is entirely up to the ears (and prejudices) of the beholder.


Television sets that produce ultraviolet light too (*RGBUV* color) do unquestionably have the _potential_ for better picture quality.

Whether you can see the difference is entirely up to the eyes (and prejudices) of the beholder.


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## JWolf

bigshot said:


> If all the added resolution of 24 bit is below the level where you can’t hear it, and all the additional superaudible frequencies are beyond your ears’ ability to hear, then you’re creating a hifi  for bats and dogs, not yourself. Redbook already is overkill.



But how do you know you don't hear increased resolution? And while you may not hear the extended frequencies, the idea is for the frequencies you do hear to sound better.


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## JWolf

The thing is, the spec should have been 20-bits/48kHz and then we may not be arguing about this.


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## Quadfather

So far, I like the way 96 kilohertz 24-bit sounds. Another possibility with these higher bitrate recordings is that more care is taken mastering them back into digital format than the CD creators.


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## bigshot

JWolf said:


> But how do you know you don't hear increased resolution?



Great question! Two ways...

1) I understand the specific technical difference between 16/44.1 and 24/96. I also know the human thresholds of perception related to those specs. When I compare them, I can clearly see that all the added benefits of high rate sound is outside the range of human hearing.

2) I have compared 16/44.1 to 24/92 myself in a line level matched direct A/B switchable blind test using a ProTools workstation. And I couldn’t discern any difference at all.

Number 1 tells me you can’t hear it with a high degree of confidence. Number 2 tells me for sure that I can’t hear it. Ultimately, that’s all that matters to me. If you want to be sure too, just do a controlled ABX and you won’t have to worry about potential or theoretical sound quality any more either. It’s liberating!


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## bigshot

JWolf said:


> The thing is, the spec should have been 20-bits/48kHz and then we may not be arguing about this.



Why? How loud do you play your stereo? If you’re getting into the zone between 16 bit and 20 bit, you should be concerned about damaging your hearing.


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## 71 dB

JWolf said:


> But how do you know you don't hear increased resolution? And while you may not hear the extended frequencies, the idea is for the frequencies you do hear to sound better.


Because listening tests tell us that. Sounds that we do hear can't sound better (better in what way?), because 16/44.1 already can reproduce them completely.

People who think high res audio sounds better are victims of thinking more must be better, victims of placebo effect and victims of not understanding enough digital audio or human hearing.


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## jnorris

bigshot said:


> Why? How loud do you play your stereo? If you’re getting into the zone between 16 bit and 20 bit, you should be concerned about damaging your hearing.




The difference between 16 and 24 bit is not how loud it gets, but how quiet it gets.  It's the number of bits that are used to define the amplitude of the waveform.


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## RRod

JWolf said:


> The thing is, the spec should have been 20-bits/48kHz and then we may not be arguing about this.



Wasn't it going to be 14 bit at first?



JWolf said:


> But how do you know you don't hear increased resolution? And while you may not hear the extended frequencies, the idea is for the frequencies you do hear to sound better.



For integer formats, resolution and dynamic range are the same coin looked at on either side. If you take as example 16 bits mapped into 24 bits, you can either see that as padding zeros before the 1st bit, where the extra 0s are increased 'detail' or 'resolution', or padding after the 16th bit, where the extra 0s are increased 'dynamic range'. Saying you need one is saying you need the other.



Quadfather said:


> So far, I like the way 96 kilohertz 24-bit sounds. Another possibility with these higher bitrate recordings is that more care is taken mastering them back into digital format than the CD creators.



They're already in digital format... And well cared-for recording/mixing/mastering has always produced good results. The issue with your argumentation is that you are begging the question: you have assumed 24/96 is audibly superior. The whole issue is how you actually prove that in an unbiased way.


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## 71 dB

Quadfather said:


> So far, I like the way 96 kilohertz 24-bit sounds. Another possibility with these higher bitrate recordings is that more care is taken mastering them back into digital format than the CD creators.


I also love SACDs because they are consistently brilliantly prodused recordings (+ multichannel!), but that doesn't mean 24/96 is perceptually better than 16/44.1. It means good mastering/production is better than bad mastering/production.


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## danadam (Oct 25, 2017)

jnorris said:


> The difference between 16 and 24 bit is not how loud it gets,


So... are you saying that in the following video you are hearing the sound till the end, at your usual listening volume level? And that you need to hear something even quieter?


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## 71 dB

jnorris said:


> The difference between 16 and 24 bit is not how loud it gets, but how quiet it gets.  It's the number of bits that are used to define the amplitude of the waveform.



Yes, if you set the MSB to equal loudness. Let's assume your volume setting is such that MSB equals 100 dB. At 16 bit LSB equals 10 dB. That is above the threshold of hearing between about 300 Hz and 7000 Hz and below elsewhere. However, if you listen to music loud, the threshold of hearing goes up a lot temporarily. Also, the noise level in very quiet living room is about 30 dB, already masking your LSB! If that's not enough, that example is true if dither is not used, but dither is used on every CD! So, the "resolution" of CD is perceptually even better than that! At 20 bit LSB would equal -14 dB and at 24 bit LSB  -38 dB. Amplifiers alone have such a noise floor that any potential benefit of going beyond 16 bits is easily lost. So, it is a question of how quiet is should get, what is relevant in real life.


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## gregorio (Oct 25, 2017)

jnorris said:


> [1] Here is the bottom line... Recordings at 96k/24 do, unquestionably have the _potential_ for better sound.
> [2] Whether you can hear the difference is entirely up to the ears (and prejudices) of the beholder.



1. "Better", how?
2. Yes, assuming the beholder is not a human being.



JWolf said:


> [1] The thing is, if there are going to be issues by hitting boundaries,  then having a spec that goes past those boundaries you don't have those issues.
> [2] Also, in some 24/96 recordings, you do see the sound does go past 20kHz. [2a] Plus, with 24-bit, you have more bits to give more resolution.
> [3] But how do you know you don't hear increased resolution?



1. Good, glad we cleared that up, 44.1 it is then.
2. Yes, we do see it but we don't hear it.
2a. More bits does not give more resolution.
3. Because there is no increased resolution to hear! It's a common audiophile myth that more bits = more resolution. To understand what actually happens and why there is no more resolution try this post.



Whazzzup said:


> To me its obvious 192 kHz and 24 bit just sounds more full, intimate, and less sharp than 16 bit 44.1.



How do you know that the recording is not supposed to sound more sharp, less intimate/full and that 16/44 is therefore being reproduced more accurately?

G


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## frodeni

71 dB said:


> Because listening tests tell us that. Sounds that we do hear can't sound better (better in what way?), because 16/44.1 already can reproduce them completely.
> 
> People who think high res audio sounds better are victims of thinking more must be better, victims of placebo effect and victims of not understanding enough digital audio or human hearing.



The proof for 16/44.1 is made by using sine waves. Beyond that, we do not know very much. You do not know everything ever written, just because you know the alphabet.

Is 16/44.1 enough for vector reproduction? Do you have any proof of that? Is that a headset limitation? How do you know that it is not?


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## bigshot

jnorris said:


> The difference between 16 and 24 bit is not how loud it gets, but how quiet it gets.



Absolutely correct. But assuming the music you're listening to is normalized, a CD has a dynamic range of almost 100dB. Your body has a noise floor of about 15dB. That's why when people are in caves underground, they can hear their own blood flow through their ears. The quietest room you would ever find has a noise floor of 30dB. So assuming you're wearing headphones, you probably have a noise floor of 20dB or so. Therefore, in order to hear all the way down to the lower limit of 16 bits, you need to boost the volume 20dB. That makes the peaks 120 dB, which is the threshold of pain and the point beyond which you incur hearing damage in short term exposure.

Do you follow the logic there? In order to hear all the bits in a CD, you have to incur hearing damage.

It doesn't matter anyway, because recorded music rarely has a dynamic range wider than 45dB. And your own ears can only hear around 45dB of dynamics at one time. In order to hear louder or quieter, they need a few minutes to acclimate. That's why LP records sound fine. They have around 35 to 40dB on the average.


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## frodeni

gregorio said:


> 1. "Better", how?
> 2. Yes, assuming the beholder is not a human being.
> 
> 
> ...



1. Better at soundstage, particularly for vector reproduction. Better for articulation. That could be the case, I just cannot get to work for my part, due to loss of signal quality, using USB.
2. This is about inferring complex sound, not sine waves. If you got any scientific papers on complex sound, please share.

As for when the boundaries are hit, we got a fairly good understanding of this, for sine waves. As for S/N for music reproduction, it is really hard to tell sounds apart at great dynamic range. At least I struggle. But for complex sounds, it is hardly understood how that is experienced, and why some people got some gifts, while lacking others.

Until people got any form of proof for vector sound and complex issues on timing, by not using sine waves, nothing is even close to a given. Even in the nature sciences.


----------



## bigshot

frodeni said:


> 1. Better at soundstage, particularly for vector reproduction. Better for articulation.
> 2. This is about inferring complex sound, not sine waves. If you got any scientific papers on complex sound, please share.




1) Soundstage and sound location is a function of the mix and the acoustics between the transducers and the listener, not the recording medium. Usually when people say "better soundstage" they mean "better expectation bias". Articulation would be covered under distortion. All digital formats have inaudible levels of distortion.

2) All audible sounds, both simple and complex are *perfectly* reconstructed with 16/44.1 and can be represented as sine waves. All of them. Again, not being able to accurately reconstruct sign waves would fall under the category of distortion. See #1.

See how interesting it is to hang out in Sound Science! You learn something new every day!


----------



## RRod

frodeni said:


> Until people got any form of proof for vector sound and complex issues on timing, by not using sine waves, nothing is even close to a given. Even in the nature sciences.



You're going to need to define 'vector sound'. Also, the sine-wavy aspect of all this is due to Fourier analysis, and I task you to get a voltage function out of a mic that doesn't have a Fourier decomposition...


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## bigshot

I believe by vector sound he means sound location within the soundstage.


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## RRod

bigshot said:


> I believe by vector sound he means sound location within the soundstage.



So wanting to get better location by adding bits/samples rather than channels (or using headphone virtualization)? L'sigh.


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## bigshot

Yeah... maybe he assumes that bigger numbers means better channel separation of something.


----------



## 71 dB

frodeni said:


> The proof for 16/44.1 is made by using sine waves.



All possible signals are a sum of sine waves, so if you prove something for sine waves and the system is linear, the proof is valid for sum of sine waves too.



frodeni said:


> Beyond that, we do not know very much.



Well, clearly you don't know much, so you are partly correct.



frodeni said:


> You do not know everything ever written, just because you know the alphabet.



No, but if I can prove I can write every alphabet, it's easy to show I could write anything if I wanted.



frodeni said:


> Is 16/44.1 enough for vector reproduction? Do you have any proof of that? Is that a headset limitation? How do you know that it is not?



Vector reproduction? Is that a term coined by highres advocates? Music is signals consisting of frequencies between 20 and 20 000 Hz. 16/44.1 is enough for that. Don't believe me, do some ABX tests to see it yourself.


----------



## gregorio

frodeni said:


> 1. Better at soundstage, particularly for vector reproduction. Better for articulation.
> 2. This is about inferring complex sound, not sine waves. [2b] If you got any scientific papers on complex sound, please share.
> [3] Until people got any form of proof for vector sound and complex issues on timing, by not using sine waves...



1. Then no, more bits and/or higher sample rate make absolutely no difference whatsoever.
2. Huh? Complex sound is sine waves! What do you think "complex sound" is, if it's made of something other than sine waves?
2b. Sure, how about starting with the paper which mathematically proved the sampling theory and enabled digital audio to exist in the first place: "A Mathematical theory of Communication" - Claude Shannon, 1948.
3. No idea what you mean by "vector sound" but what do you suggest we use instead of sine waves? Before you answer, bare in mind that speakers and headphones can only reproduce sine waves and human ears only respond to sine waves.

G


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## 71 dB

bigshot said:


> Yeah... maybe he assumes that bigger numbers means better channel separation of something.



Temporal resolution? We have already established that temporal resolution of CD is practically infinite.

HD video is better than SD video. 4K video is better than 2K (on large screens). 8K video is better than 4K video (on VERY large screens). So, maybe 24/96 is better than 16/44.1? No, because 16/44.1 already is what 8K video is for pictures. 16K video isn't better than 8K video, because you have to be a hawk to see the difference.


----------



## frodeni

bigshot said:


> 1) Soundstage and sound location is a function of the mix and the acoustics between the transducers and the listener, not the recording medium. Usually when people say "better soundstage" they mean "better expectation bias". Articulation would be covered under distortion. All digital formats have inaudible levels of distortion.
> 
> 2) All audible sounds, both simple and complex are *perfectly* reconstructed with 16/44.1 and can be represented as sine waves. All of them. Again, not being able to accurately reconstruct sign waves would fall under the category of distortion. See #1.
> 
> See how interesting it is to hang out in Sound Science! You learn something new every day!



Well, if you did know the understanding of soundstage and imaging, then you would know that it is the placement of instruments in space, as experienced by the listener. As a term in a terminology, it belongs in the subjective field of science.

As for what you are speaking of, which probably is supposed to be the physical reproduction of soundwaves, there is no soundstage. At best the reproduction is very limited, there are some phase shifting, but given the nature of humans, that shift cannot be static. Nor can the amplitude difference be. And that is before the acoustics you speak of. 

And no, I did not learn anything from what you wrote, as I was told this in the 80s. I find the physics of it intriguing, and the lack of the physics being reflected by real world gear, even more interesting. But as they say in the military, if the landscape does not fit the map, there is something wrong with the landscape.

I am not sure what the right landscape is.



RRod said:


> You're going to need to define 'vector sound'. Also, the sine-wavy aspect of all this is due to Fourier analysis, and I task you to get a voltage function out of a mic that doesn't have a Fourier decomposition...



The very basic physics of hearing, is the theory of phase shift and amplitude shift between the ears, as to be able to position the origin of sounds. To achieve this for humans, the individuals distance between the ears, is a minimum to consider. So if a sound source is 10m away, 10 deg up, and 46 deg to the left, that will result in specific phase shift and amplitude shift, that is unique for the individual. (well, not exactly unique, given these parameters, but not even close to equal for all humans.) In vector sound reproduction, the phase and amplitude shift is calculated to simulate the physics of hearing, for the individual. If using a gyroscope, and done on the fly, the experienced source will be a fixed position in space. The calculation is done by distance vectors.

Also, movement plays a role, as artifacts moving towards you, at a certain speed, actually get a phase distortion, as to change in wavelength due to the movement. Just like a car coming at you, or moving away from you. Again, this can be done by vector calculation.

There is a whole hosts of things that can be added to the reproduction. Sometime in a not so distant future, someone will introduce vector sound. Hopefully, since I speak of it in public, they cannot patent it. They cannot patent it for headsets, nor automatic distance calculation between the cans using ultrasound, or any sort of waves. Because that is given in the public domain. The use of gyroscope or any type of device, to register head movement, to assist for vector reproduction, well, it is in the public domain now. It will happen. Particularly since the only real change in the industry, is a shift to mono recordings of individual sounds, while the rest of the infrastructure only needs minor adjustments.

AMD has an API for vector sound, but it only includes amplitude shifts. It has no reading of listeners dynamics at all, as in distance between the ears.

This sound tech, used in combination of see trough VR, combined with great positioning, makes my head spin with ideas. Not just for music. Particularly augmented reality. Why there is no rush in the industry to be the first at this tech, shows a complete lack of visionaries.

Given the insane variance using 16/44.1 for classic stereo, that variance indicates, to me, that we probably need more for vector sound. If we don't, that is great news, as vector sound will arrive earlier then. 

This also gives the reason for these in dummy head recordings do not work. In general. They are close to work, if you got the exact right head dimensions, something which is forgotten by the fans. And as with most things fans of this type of recordings, particularly those with heads that fit a certain recording, they do not listen to the facts presented.

As for your dragging Fourier into this, I miss your point. I have lived long enough, to have seen an ellipse being represented by multiple circles. Sure, if that is mathematically possible, then it is. Trouble is, that when things are really understood, or so we thought, the movement of the planets had nothing to do with circles. The answer was of a completely different nature. As is the complexity of sounds. Well, again, this is not really understood at all, and maybe the placement of sounds is derived by some Fourier like process by humans. We just do not know.

What we do know, is that music or sound reproduction is far more complex than a single sine wave. As is having a dog, cat, or hamster, but mixing the three?


----------



## JWolf

danadam said:


> So... are you saying that in the following video you are hearing the sound till the end, at your usual listening volume level? And that you need to hear something even quieter?



That video is worthless because it's LOSSY!


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## bigshot (Oct 25, 2017)

frodeni said:


> Well, if you did know the understanding of soundstage and imaging, then you would know that it is the placement of instruments in space, as experienced by the listener. As a term in a terminology, it belongs in the subjective field of science.



Designing the soundstage in the mix is a creative/subjective process, but reproducing soundstage in a home stereo is basically a technical matter of applied acoustics.  Soundstage is the illusion of an aural plane in front of the listener representing the source of the music. It's governed by the radius of the triangle between the two speakers and the listener, by the distance between the listener and the speakers, and by the balance of each sound element in the mix between the two speakers. It's also helped along in the mix by secondary distance cues, like reflected echoes off walls or mic distance, which create the illusion of depth. That stuff is baked into the recording though. Incremental improvements in playback quality can't affect secondary distance cues.

Weren't you saying that bitrates and sampling rates above 16/44.1 improved the soundstage? I don't know how sampling rate or bit rate could possibly affect that. Soundstage is improved by proper speaker placement and better room acoustics, not a lower noise floor and super audible frequency content.



frodeni said:


> As for what you are speaking of, which probably is supposed to be the physical reproduction of soundwaves, there is no soundstage. At best the reproduction is very limited, there are some phase shifting, but given the nature of humans, that shift cannot be static. Nor can the amplitude difference be. And that is before the acoustics you speak of.



I'm not sure what you're talking about here I'm afraid. Are you trying to say that the channel separation affects the placement of sound elements within the soundstage? It does sort of. A lot of channel overlap would degrade the soundstage. But 16/44.1 and 24/96 both have the same basic specs for channel separation- perfect channel separation to human ears. Are you talking about the secondary distance cues baked into the mix? Because all of the subtle distance cues within the audible range would be perfectly rendered in 16/44.1. More bit depth wouldn't improve upon that because the time factor and the frequency factor are related here. Your point is a little vague. Maybe you could organize your thoughts and present them more clearly.


----------



## frodeni

JWolf said:


> That video is worthless because it's LOSSY!



I gave the video a go, could barely hear it a -78db. With a ton of noise. (I probably play too loud as well)

People need to wake up. If you do not get the meaning of the argument, you need to do some recording on your own. A SN of 78db is pretty darn good. For most recording equipment, this is achieved by using post process noise reduction. Also, people need to do their own recordings using both lossy and compressed formats.

The physics and math in this case, simply means that there is not enough resolution by the mic, to out resolve 16bit. Few does. At least not as how this is currently being understood. There is nothing preventing a ADC to work better at a higher resolution, resulting in a better result over all, but that is not given the high res nature of it, on its own. You could argue that multiple readings increase accuracy, or that higher resolution result in the inaccuracy to fall outside of the wanted accuracy: Which is common practice for physics. But if there is only a range of say 90db to begin with, that range is perfectly possible to represent by 16bit, without any loss.

As for reproduction, it may be perfectly probable, that a higher bit dept could end up with a more accurate reproduction after the DAC, but that is a messy topic, related to the rendering improving due to complex interaction of components, not necessarily the quality of the material itself. Also, if will effect how noise is rendered, and the need for filtering.



bigshot said:


> Designing the soundstage in the mix is a creative/subjective process, but reproducing soundstage in a home stereo is basically a technical matter of applied acoustics.  Soundstage is the illusion of an aural plane in front of the listener representing the source of the music. It's governed by the radius of the triangle between the two speakers and the listener, by the distance between the listener and the speakers, and by the balance of each sound element in the mix between the two speakers. It's also helped along in the mix by secondary distance cues, like reflected echoes off walls or mic distance, which create the illusion of depth. That stuff is baked into the recording though. Incremental improvements in playback quality can't affect secondary distance cues.
> 
> Weren't you saying that bitrates and sampling rates above 16/44.1 improved the soundstage? I don't know how sampling rate or bit rate could possibly affect that. Soundstage is improved by proper speaker placement and better room acoustics, not a lower noise floor and super audible frequency content.
> 
> I'm not sure what you're talking about here I'm afraid. Maybe you could organize your thoughts and present them more clearly.



I speak of headsets only. I also speak of reproducing placement of sound, by using the physics of sound. Once you leave the speaker, and reread what I wrote for head-fi, then you should be able to grasp it.

Also, when speaking of the "illusion", that is the experience. People need to read up on how the physics work for that. Why we got two ears. And why two ears is limiting. In the wild, you will see animals turning their heads and even their ears, as to optimize the angle of the ears. The accuracy differs greatly, as a function of angle to the plane made by the ears.

Again, if you at say 45 deg to the left, 10 deg up, at a distance of 10 meter, the sound from that source will hit your ears differently. There is an inverse square law of energy and waves by distance. Time to travel a distance is linear. If you know the distance from the source, to each ear, you may calculate the amplitude difference and the phase shift, between the ears. This is the very basics of sound science and human hearing. The utter basics.

Vectors may be used to calculate the distance from the artifact, to each ear. There will be a distance difference, unless the sound source is right in front or behind the listener, which results in an equal distance.

Phase shifts are then calculated for each ear, using its linear nature. Again, the sound will hit the ears at a very slight shift in time, as there is a difference in distance to travel to each of the ears.

Amplitude difference are then calculated for each ear, using its inverse square law. As for phase shift, there will be a minute difference.

The listener then uses these two properties, to calculate the origin of the sound. At least, that is the current consensus. It has been that, for decades now. It is not like we understand what is going on, inside the ear or the brain, at least not to my knowledge. If anyone knows of good research on the topic, please share.

Why this is lost in the community, I simply do not know. People seem to think there is height to sound reproduction of speakers, but no. There isn't. I have yet to hear much opposition, when people claim to experience height. Also, there is no height from headphones, not for amplitude modulated soundstages. When I talk about the plane is different in height for say most 3-way speakers, and that I find it irritating when for instance a guitar shifts up and down in the physically soundstage due to the tone being played, that is actually exactly what I am supposed to experience. Some speakers solves this by doubling up on their 2nd and 3rd phases, but the physics of that, seems lost on people.

Sure enough, I just got berated by someone claiming me to know nothing, yet displaying no insight into basic physics of hearing and sound. Are people going to stay offensive and stupid, or is this forum ready to discuss the tech that is doomed to arrive soon? It will be based on this very physics, that is well known since at least the 70s.


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## bigshot (Oct 25, 2017)

frodeni said:


> I speak of headsets only. I also speak of reproducing placement of sound, by using the physics of sound.
> 
> In the wild, you will see animals turning their heads and even their ears, as to optimize the angle of the ears.



Ah. I see. The problem is, you can't turn your head to pinpoint locate sound with headphones. They don't have true soundstage. The shifts in time and phase have to be synchronized with the head movements. You can simulate it using something like a Smyth Realizer, but I don't have one of those myself. By themselves, headphones can only arrange the sound in a straight line through the center of the head. If you turn your head one way or the other, it's still the same straight line through your head. If you want, you can call that "headstage" because it's all right through the middle of your skull.

Speaker systems are different. The sound is physically in front of you because the speakers are physically arranged in front of you to the left and right. If you turn your head, you can locate the sound in physical space in front of you. That is true soundstage. Vector location is only possible if you can turn your head, and you can't do that with headphones that are attached to your noggin.

Neither of these things would be any different with 16/44.1 as opposed to higher bit depths. I thought that was what you were claiming, Maybe I read wrong.

As for speakers creating height, there are a couple of ways to accomplish that. The easiest way to do it in a two channel system is to simply add a second set of speakers at a higher level. The dispersion pattern of the speaker can affect the perception of height too. A horn loaded speaker with highly directional sound will sound narrower and more in a straight line than a speaker that disperses the sound in a wide radiated pattern.

In a multichannel speaker system, you can raise the center channel a little higher than the mains to raise the height of the soundstage. There are also DSPs that use the information about the speaker placement you enter in your AVR to calculate subtle time offsets in the channels to give the illusion that the soundstage is deeper and higher. The ultimate way to create height is to use Dolby Atmos, which adds a set of speakers at roof level to mesh with the 5.1 speakers. That allows sound to be placed precisely in a three dimensional sound field. A sound field is to soundstage as soundstage is to headstage. It's a progression from a straight line to a dimensional plane to a three dimensional space.

Again, none of this would be any different with 16/44.1 as opposed to higher bit depths.

We're not berating you by the way. We're just trying to figure out your terminology. It's different than the terms we normally use for these things. It may be a language difference. We'll all figure it out.


----------



## RRod

frodeni said:


> The very basic physics of hearing, is the theory of phase shift and amplitude shift between the ears, as to be able to position the origin of sounds. To achieve this for humans, the individuals distance between the ears, is a minimum to consider. So if a sound source is 10m away, 10 deg up, and 46 deg to the left, that will result in specific phase shift and amplitude shift, that is unique for the individual. (well, not exactly unique, given these parameters, but not even close to equal for all humans.) In vector sound reproduction, the phase and amplitude shift is calculated to simulate the physics of hearing, for the individual. If using a gyroscope, and done on the fly, the experienced source will be a fixed position in space. The calculation is done by distance vectors.
> 
> Also, movement plays a role, as artifacts moving towards you, at a certain speed, actually get a phase distortion, as to change in wavelength due to the movement. Just like a car coming at you, or moving away from you. Again, this can be done by vector calculation.
> 
> ...



So wait, you say phase and amplitude but don't know why Fourier matters? Just forget it then! Also, absolutely zero of the books/monographs I've read on positional audio / virtualization bother mentioning hi-res; take from that what you will.


----------



## frodeni

bigshot said:


> Ah. I see. The problem is, you can't turn your head to pinpoint locate sound with headphones. They don't have true soundstage. The shifts in time and phase have to be synchronized with the head movements. You can simulate it using something like a Smyth Realizer, but I don't have one of those myself. By themselves, headphones can only arrange the sound in a straight line through the center of the head. If you turn your head one way or the other, it's still the same straight line through your head. If you want, you can call that "headstage" because it's all right through the middle of your skull. ...



Great. Now you are getting it. Yes, for classic stereo reproduction using speakers there is some form of soundstage. And no, there is hardly any soundstage as by physical space, outside the head of the listener, when using headsets. Not for regular recordings.

What I am arguing, is simply record every instrument in mono, with as little room acoustics as possible, and simply place the instrument or artist on the fly, or by pre calculation by the dimensions of the head of the listener. And, yes, finally, someone who gets the point, that you would have to do this on the fly, as to correct for head movement, if the listener is to experience the sound source as fixed in physically space. That is perfectly possible, if using vectors and math, and a sensor in the headset for head movements.

As for 16/44.1 I do not know. Not for this application. Since I have never tried it, as to my knowledge, it does not exists. The results for the Smyth Realizer is very promising, as they do some of the math I describe here.

As for reproduction of height for speakers: There is no information of height in the recording. Usually, base is at the bottom, as most drums and low frequency sources are placed low, and so on. But the plane is fractured by the placement of the speaker elements. There is no height in stereo and speakers. Not is you are true to your senses. Sure, some people imagine there is, and good on them. I am a different character, and try to stay true to my senses. I want vector music, not the last century tech of today.

For speakers or stereo, there is a known technique of lowering voices as in volume, that are supposed to be in the background, but that is only relational to louder voices. The reproduction is still in the same plane, if you listen carefully. The amount of false experiences among listeners are like crazy. One thing is to be able to imagine something, which might be easier with some gear than others, another is to claim to be able to hear height being reproduced, when there is no information about height at all, that may possibly be presented by the gear in question. Again, I like your vocabulary.

Going vector on music, it is really simple to render for speakers. Actually, having tall speakers, or a four speaker setup with one set low and one set high, height could be reproduced. On the fly.

It is much like the structure and formatting division of html and css. A multi speaker setup for like a home theater would only need a special rendering, applying math for that particular type of setup. It would just place and emulate the sound based on the medium.


----------



## dprimary

Quadfather said:


> So far, I like the way 96 kilohertz 24-bit sounds. Another possibility with these higher bitrate recordings is that more care is taken mastering them back into digital format than the CD creators.


24 bit recording equipment didn't appear on the market till around 1997, so if it was recorded before then it is unlikely to have greater than 16 bits of resolution.


----------



## danadam

JWolf said:


> That video is worthless because it's LOSSY!


Sheesh, there's no pleasing them  I don't think it makes much difference, the signal is there if you crank up the volume or download the audio and amplify it in some audio editor. If you don't hear it at your normal listening volume level then it is not because youtube's compression somehow removed it. But anyway, know my good heart, I generated it again, so here is flac and here is mkv with that flac as audio track. Enjoy.

(To be clear, I don't delude myself that this will satisfy "audiophiles". I'm waiting to hear what is wrong with it now  )



frodeni said:


> I gave the video a go, could barely hear it a -78db. With a ton of noise. (I probably play too loud as well)


What noise?
In a quiet room, with headphones (HD 650) and volume *higher* than my comfortable listening level (2 o'clock on O2+ODAC) I also stop hearing anything at around -72, -78 dBFS, but I don't hear any noise. After that, if I crank up the volume to max and turn on O2's gain (3.3x) I stop hearing it after -96 dBFS, and still no noise.


----------



## bigshot

frodeni said:


> What I am arguing, is simply record every instrument in mono, with as little room acoustics as possible, and simply place the instrument or artist on the fly, or by pre calculation by the dimensions of the head of the listener.




How many musicians can you fit inside your head?!

That's an interesting idea, but it would require something like the Smyth Realizer with head tracking along with a multichannel master with a channel for every musician. (You would need stereo for each musician, not mono because some electronic instruments output stereo.) I get the same basic thing with my 5.1 speaker system and an SACD. I guess if you had to use headphones, not speakers that would be nice though.

By the way, there are tricks for broadening and raising soundstage through speaker placement. You can create depth too. It's not a standard setup like you see on home theater sites, but it works. Basically, you run two sets of mains with one set ahead and wider than the other set. Then you set the center channel along with the back set of mains at a higher height. The result is a fairly life scale soundstage for both orchestral music and jazz combos.


----------



## pinnahertz

frodeni said:


> I gave the video a go, could barely hear it a -78db. With a ton of noise. (I probably play too loud as well)
> 
> People need to wake up. If you do not get the meaning of the argument, you need to do some recording on your own. A SN of 78db is pretty darn good. For most recording equipment, this is achieved by using post process noise reduction.


Huh?


frodeni said:


> The physics and math in this case, simply means that there is not enough resolution by the mic, to out resolve 16bit. Few does. At least not as how this is currently being understood.


Hmmm…well, I looked up an old favorite mic of mine, the Shure SM81, it's about a 40 year old design.  Total DR is 118dB, so just shy of 20 bits.  There are many more mics with even wider DR.  Not sure what you mean.


frodeni said:


> There is nothing preventing a ADC to work better at a higher resolution, resulting in a better result over all, but that is not given the high res nature of it, on its own. You could argue that multiple readings increase accuracy, or that higher resolution result in the inaccuracy to fall outside of the wanted accuracy: Which is common practice for physics. But if there is only a range of say 90db to begin with, that range is perfectly possible to represent by 16bit, without any loss.


Yes, that's true for the final release, but in production there is a point to working at higher bit depth, especially when mixing and performing other DSP functions, which is why today's DAW internal processing is 64 bit floating point.


frodeni said:


> As for reproduction, it may be perfectly probable, that a higher bit dept could end up with a more accurate reproduction after the DAC, but that is a messy topic, related to the rendering improving due to complex interaction of components, not necessarily the quality of the material itself. Also, if will effect how noise is rendered, and the need for filtering.
> 
> 
> 
> ...


"Spatial Hearing: The Psychophysics of Human Sound Localization"
by Jens Blauert and John S. Allen

The above would be a good start.



frodeni said:


> Why this is lost in the community, I simply do not know. People seem to think there is height to sound reproduction of speakers, but no. There isn't. I have yet to hear much opposition, when people claim to experience height. Also, there is no height from headphones, not for amplitude modulated soundstages. When I talk about the plane is different in height for say most 3-way speakers, and that I find it irritating when for instance a guitar shifts up and down in the physically soundstage due to the tone being played, that is actually exactly what I am supposed to experience. Some speakers solves this by doubling up on their 2nd and 3rd phases, but the physics of that, seems lost on people.


If you were taking only about traditional stereo recordings, I'd probably agree mostly, though there can be a bit of "accidental" height effect.  However, there are ways to get height out of two-speaker stereo (and it's hardly anything new):
http://www.audiocheck.net/audiotests_ledr.php


frodeni said:


> Sure enough, I just got berated by someone claiming me to know nothing, yet displaying no insight into basic physics of hearing and sound. Are people going to stay offensive and stupid, or is this forum ready to discuss the tech that is doomed to arrive soon? It will be based on this very physics, that is well known since at least the 70s.


It's never really right to assume every one reading or posting in a public forum share the same ideas or attitudes.  It's really just a bunch of individuals.


----------



## Quadfather

dprimary said:


> 24 bit recording equipment didn't appear on the market till around 1997, so if it was recorded before then it is unlikely to have greater than 16 bits of resolution.



Lately I have been doing new classical and smooth jazz when I am not in a Lamb of God, Hellyeah, Mudvayne, heavy metal mood.


----------



## dprimary

What you are calling vector based recordings. Is pretty much Dolby Atmos object based mixing. Up to 128 tracks (objects) are placed in the mix, the system records the metadata of the placement. When this is played back in the calibrated theater the system will place the objects based on the layout of sound system to reconstruct the mix as it was mixed at the mix theater. The mix processor is over 24 bits likely 64 bit. In practice even  mixing together 16 bit audio would only be slightly noisier than the noisiest track. It extremely difficult to record a sound with a 96dB signal noise ratio. Most places on the planet are noisier then that. ISS might work.


----------



## Quadfather

dprimary said:


> What you are calling vector based recordings. Is pretty much Dolby Atmos object based mixing. Up to 128 tracks (objects) are placed in the mix, the system records the metadata of the placement. When this is played back in the calibrated theater the system will place the objects based on the layout of sound system to reconstruct the mix as it was mixed at the mix theater. The mix processor is over 24 bits likely 64 bit. In practice even  mixing together 16 bit audio would only be slightly noisier than the noisiest track. It extremely difficult to record a sound with a 96dB signal noise ratio. Most places on the planet are noisier then that. ISS might work.



ISS?  In school suspension...


----------



## dprimary

pinnahertz said:


> Hmmm…well, I looked up an old favorite mic of mine, the Shure SM81, it's about a 40 year old design. Total DR is 118dB, so just shy of 20 bits. There are many more mics with even wider DR. Not sure what you mean.



The signal to noise ratio however is 78 dB.


----------



## pinnahertz (Oct 25, 2017)

dprimary said:


> The signal to noise ratio however is 78 dB.


No, the signal to noise ratio depends on the SPL of the reference signal.  We're talking total DR, not S/N re: a reference level at 94dB SPL.   IT's only 78 if you're recording 94dB SPL.  Maximum is 136dB SPL.  So....that actually comes out to 120dB...sorry, my error.


----------



## dprimary

pinnahertz said:


> No, the signal to noise ratio depends on the SPL of the reference signal.  We're talking total DR, not S/N re: a reference level at 94dB SPL.   IT's only 78 if you're recording 94dB SPL.  Maximum is 136dB SPL.  So....that actually comes out to 120dB...sorry, my error.


What do you record that hits 136 dB SPL? of course you need be in room with noise floor less than 16 dB SPL. For something like a violin you will not be able to exceed 16 bit with a SM81.


----------



## bigshot (Oct 25, 2017)

My favorite mic is the Neumann U-87. If I remember correctly, it had a switch for loud stuff, but I think along with it's signal to noise, it came out about 110 dB or so.



dprimary said:


> What you are calling vector based recordings. Is pretty much Dolby Atmos object based mixing.



Has the Smyth Realizer been updated to work with an Atmos mix yet? That would be exactly what he wants.



dprimary said:


> What do you record that hits 136 dB SPL?



That's a shotgun mic isn't it? I used that for close miked drums. I imagine that a close miked snare might hit that in the impacts.


----------



## frodeni

dprimary said:


> What you are calling vector based recordings. Is pretty much Dolby Atmos object based mixing. Up to 128 tracks (objects) are placed in the mix, the system records the metadata of the placement. When this is played back in the calibrated theater the system will place the objects based on the layout of sound system to reconstruct the mix as it was mixed at the mix theater. The mix processor is over 24 bits likely 64 bit. In practice even  mixing together 16 bit audio would only be slightly noisier than the noisiest track. It extremely difficult to record a sound with a 96dB signal noise ratio. Most places on the planet are noisier then that. ISS might work.



You are talking about a multi speaker setup. Which is about as far away as you can get, form using headsets. While headsets would have few if none limits as to what may be achieved, speakers have a ton of limitations.



pinnahertz said:


> Huh?
> Hmmm…well, I looked up an old favorite mic of mine, the Shure SM81, it's about a 40 year old design.  Total DR is 118dB, so just shy of 20 bits.  There are many more mics with even wider DR.  Not sure what you mean.
> Yes, that's true for the final release, but in production there is a point to working at higher bit depth, especially when mixing and performing other DSP functions, which is why today's DAW internal processing is 64 bit floating point.



That is a $350 microphone. Listed spec:
Signal-to-Noise Ratio 78 dB (IEC 651)* at 94 dB SPL S/N ratio is difference between 94 dB SPL and equivalent SPL of self– noise A-weighted
http://cdn.shure.com/specification_sheet/upload/228/sm81-specification-sheet-english.pdf

SN is also heavily dependent on the amp used. As for high floating point precision, sure, that is a thing these days. Signal processing is not my field of expertise.



pinnahertz said:


> "Spatial Hearing: The Psychophysics of Human Sound Localization"
> by Jens Blauert and John S. Allen



Thanks. I looked it up and had a quick look at the 50 first pages. Appears to be a purely positivistic work, yet not stating so at all. Not really stating any assumptions at all, like a lot of this type of work. A bit of warning though, a lot of the assumptions within the field of psychology, are really just that: Assumptions. Reading these works requires caution as to what tradition they are done within, as to trace the validity of their claims. I like to read the abstract first, then the conclusion, but that will have to wait til I get time for it, at the University.

It is also a bit strange to be faced with the general consensus as of decades, that I just wrote about. Knowing how equipment sensitive this is at the reproduction stage, I hope they rather did some real life experiments. Any results is beyond page 50, which is what Google shared. If they tested peopled that had a trained ear for this, it would be possible to reverse the calculation into bits and hz for sampling. If you know what the findings were, please share.



bigshot said:


> How many musicians can you fit inside your head?!
> 
> That's an interesting idea, but it would require something like the Smyth Realizer with head tracking along with a multichannel master with a channel for every musician. (You would need stereo for each musician, not mono because some electronic instruments output stereo.) I get the same basic thing with my 5.1 speaker system and an SACD. I guess if you had to use headphones, not speakers that would be nice though.
> 
> By the way, there are tricks for broadening and raising soundstage through speaker placement. You can create depth too. It's not a standard setup like you see on home theater sites, but it works. Basically, you run two sets of mains with one set ahead and wider than the other set. Then you set the center channel along with the back set of mains at a higher height. The result is a fairly life scale soundstage for both orchestral music and jazz combos.



People can handle five tasks at once, some up to 7. For any design work, there is a consensus at five tasks at the most. That is why we got harmonies.

As for the need of stereo, no, that is a false statement. The gear having stereo output assumes regular stereo setup, which vector sound is not. Stereo output does neither fit into any 5.1 or similar scheme either. Yes, you will need a "channel" for every single sound, and no, that is not done in post processing like the Smyth. It is a form of presentation layer, that requires a form of computer. Like a smartphone or a PC. For music and static reproduction (not head spinning adjusted), the output might be processed once, and saved as regular stereo, for a given user-gear combo.

People seem to struggle with the fact that: No. There is no height to stereo. It is simply not there. Just as there is no stereo to mono. If you claim there is dept to stereo, please share with us, the physics at work. I used to think there was, until I got a fairly clean digital signal. That was an embarrassing moment, to say the least.

This focus on speakers is a bit misplaced. The room in which you listen is almost always the limiting factor, unless you pad every surface like nuts. Including the ceiling. Windows are a no go. Once you move into vector sound, speakers really start to struggle, no matter the room.


----------



## dprimary

bigshot said:


> My favorite mic is the Neumann U-87. If I remember correctly, it had a switch for loud stuff, but I think along with it's signal to noise, it came out about 110 dB or so.


The current U-87 has about 1 dB lower noise (then the  SM81) and 117 dB max SPL without the pad. They are insanely expensive now and they don't even include a mount.


----------



## dprimary

frodeni said:


> You are talking about a multi speaker setup. Which is about as far away as you can get, form using headsets. While headsets would have few if none limits as to what may be achieved, speakers have a ton of limitations.


Mapping all that information to a pair a drivers what do you expect to get? You can do any of this manually with about any digital mixer, just play with your matrixes and delays. Unless recorded with stereo or surround microphone you are just creating a soundfield. With a decent DSP you can automate it.


----------



## dprimary

frodeni said:


> SN is also heavily dependent on the amp used. As for high floating point precision, sure, that is a thing these days. Signal processing is not my field of expertise.



All microphones have self generated noise. There is plenty of preamps that have lower signal to noise than most microphones.


----------



## bigshot

frodeni said:


> This focus on speakers is a bit misplaced. The room in which you listen is almost always the limiting factor, unless you pad every surface like nuts. Including the ceiling. Windows are a no go. Once you move into vector sound, speakers really start to struggle, no matter the room.




But speakers do soundstage and directionality naturally. Headphones have to be processed into doing it. I understand what you're talking about, but that has never been a part of acoustic sound reproduction. Back in the days of the acoustic Victrola they recorded everything dry and used the horn in the phonograph to project the sound into the room to give it a natural presence. Room ambience helped make severely limited recordings sound so real and present they could make the hairs on the back of your neck stand up. Natural room ambience is an important part of the sound quality of a speaker system too. If you think that the ideal listening room is completely dry and dead, and all of the ambience should be baked into the mix, you're wrong about what makes a good speaker system. The acoustics of the room is as much a part of the sound quality as the acoustics of the speakers.

It sounds like you're theorizing about speakers the way people theorize about headphones. But they are quite different animals and have different goals, advantages and limitations. With headphones, you're removing the room from the equation to make balanced frequency response and super low distortion easier to achieve. With speakers you are taking advantage of the room acoustics to give ambient space for the sound to exist in and you're adding a kinesthetic punch to the sound. Two totally different approaches- both equally valid- and both suited for their own purposes. You could design a listening room to mimic the advantages of headphones, but it wouldn't exploit the unique benefits of speaker sound. And you can probably synthesize speaker sound with headphones, but you'll lose the advantages headphones have. And it would be like a digital reverb hall ambience vs. a real concert hall ambience. Close but no cigar.

You might want to google information about room treatment theory. That might help you understand how speaker systems are implemented. You'll find out that the goal of room treatment isn't to remove all reflections. It's to remove *detrimental* reflections so the disadvantages are minimized and natural room ambience can help the sound bloom and exist in three dimensional space. Speaker systems are optimized to create something bigger and better than the recording alone. They aren't about clinically presenting the recording and nothing else. Recording booths are about isolation because you want to focus on just the sound you're trying to get down in the recording. But speakers use the room to* enhance* the recording.

Not many people have heard a surround system optimized for music listening. Most of them are optimized for home theater. But a really good speaker system in a really good room does things headphones can't. And headphones do things speakers can't. It's best to use each for what it does best and not try to make one into the other.


----------



## ev13wt

bigshot said:


> That's a shotgun mic isn't it? I used that for close miked drums. I imagine that a close miked snare might hit that in the impacts.



More like 120. Maybe 125 a rim shot.


----------



## castleofargh

@frodeni
 yeah we're a political sub forum with an agenda. you completely busted us. and it's not just here, if you go in the "portable headphone amps" sub section, most topics are about portable headphone amps. coincidence? I think not!
it would be silly enough to bring up the arguments you came up with, but the irony really hits home when you target this subsection in particular for your fantasy war. the sub forum was created as an objectivist ghetto, to get rid of people asking about controlled listening too often for the empty claim kind of audiophiles to feel comfy. so this section opened and that sentence ended up in the Terms Of Service


> If what you want to post includes words/phrases like "placebo," "expectation bias," "ABX," "blind testing," etc., please post it in the Sound Science forum.


so thank you mister righteousness, but we're doing things a certain way in here because it's the only place allowing us to do them.

I can't be bothered to discuss every made up argument you have about everything or how you seem to mistake your opinion for a fact all the time, but this IMO it's emblematic enough of your contribution:


			
				frodeni said:
			
		

> As for your dragging Fourier into this, I miss your point. I have lived long enough, to have seen an ellipse being represented by multiple circles. Sure, if that is mathematically possible, then it is. Trouble is, that when things are really understood, or so we thought, the movement of the planets had nothing to do with circles. The answer was of a completely different nature. As is the complexity of sounds. Well, again, this is not really understood at all, and maybe the placement of sounds is derived by some Fourier like process by humans. We just do not know.


you set him straight, dragging Fourier into stuff about time and frequencies or any attempt to characterize them in a complex signal, silly @RRod.


at least here is a quote we can all agree on:


			
				frodeni said:
			
		

> Signal processing is not my field of expertise.


 yeah, no crap. luckily you don't have to know much of anything to come and force your views on a topic about digital audio and what's relevant or not.







bigshot said:


> My favorite mic is the Neumann U-87. If I remember correctly, it had a switch for loud stuff, but I think along with it's signal to noise, it came out about 110 dB or so.
> 
> 
> 
> ...


the Realiser A16 that will come out last summer (kickstarter and expected timelines ^_^) is indeed supposed to work with Atmos and a few other multi channel stuff. they have it all working but I don't know about the certifications yet.


----------



## RRod (Oct 26, 2017)

bigshot said:


> Has the Smyth Realizer been updated to work with an Atmos mix yet? That would be exactly what he wants.



Last I read it was, as a max 16-channel setup.


----------



## gregorio

frodeni said:


> The proof for 16/44.1 is made by using sine waves. Beyond that, we do not know very much.



Regardless of what we actually know, why would knowing anything beyond sine waves make any difference? What waves other than sine waves are you talking about and why? You keep avoiding this question!



frodeni said:


> As for your dragging Fourier into this, I miss your point. ... The answer was of a completely different nature. As is the complexity of sounds. Well, again, this is not really understood at all, and maybe the placement of sounds is derived by some Fourier like process by humans. We just do not know. ...
> What we do know, is that music or sound reproduction is far more complex than a single sine wave.



There is a seemingly constant trend your posts:

1. You keep going on about "what we know"/"what we don't know" and that various things are "not really understood". As often seems to be the case with audiophile arguments, what they really mean by these types of statements is that they themselves don't know and/or don't understand, or perhaps by "we" they mean; we audiophiles don't know/understand. What they seem to be getting drastically wrong is the assumption that if they don't know or understand something, then no one does. If that's not bad enough, the actual primary sources of what we (collective humanity) really know and understand about audio recording and reproduction is concentrated in two areas, science and the professional practitioners, the two areas most shunned by audiophiles!! So please, enough with the "we do not know"! If YOU do not know, then ask. What we don't like here on the science forum is people stubbornly arguing nonsense theories based on "we don't know", which are nonsense theories because in fact we do know! The reason for dragging Fourier into this is because Fourier discovered (and proved!!) a mathematical means of breaking down any (real) complex waveform into it's constituent sine waves. The very fact that you "miss the point" of "dragging Fourier into this" indicates a massive hole in your personal knowledge/understanding of the "complexity of sound" and therefore all sorts of resulting nonsense theories, but it is just a case of you personally not knowing/understanding, NOT a case that "we don't know"!

2. You keep saying that we're ignoring the physics and then you come up with some assertion which ignores the physics?! If that's not bad enough, you also ignore other vital elements, aspects of: proven mathematics, electrical engineering, sound engineering and the practical application of it all. Case in point: "_What I am arguing, is simply record every instrument in mono, with as little room acoustics as possible_" - This is of course complete nonsense, you clearly do not understand the physics of instrument sound production, mics, mic placement, acoustics and the interaction between all these elements and neither do you understand the practicalities of music performance.

3. Misdirection! Instead of responding to the points raised, you go off on some ridiculous tangent. Your argument for 16/44 not being enough is because it may not accommodate (in your opinion) a format which you've just invented and doesn't exist. And then off you go explaining that non-existent format? ...



frodeni said:


> Is 16/44.1 enough for vector reproduction? Do you have any proof of that?



You're asking us for proof that 16/44 is enough for a hypothetical method of reproduction which doesn't exist and which no one apart from you has ever even heard of because you just made it up? Hands up anyone who thinks this contains even the slightest hint of logic or rationality!

I can play that game too: Can YOU prove that 16/44 is not enough for teleportational reproduction? ... No? That's it then, point made, case closed, 16/44 is enough! Glad we cleared that up, next.

G


----------



## 71 dB

frodeni said:


> People seem to struggle with the fact that: No. There is no height to stereo. It is simply not there. Just as there is no stereo to mono. If you claim there is dept to stereo, please share with us, the physics at work. I used to think there was, until I got a fairly clean digital signal. That was an embarrassing moment, to say the least.



We are left-right symmetric, but not down-up symmetric. That's why sounds coming from negative elevations get coloured by our body differently than sounds coming from positive elevations. Having spatial cues in the sound according to how sound if filtered by our torso, head and pinna, it's possible to have some height information even in mono sound. A simplified model goes like this:

Negative elevations: Dip at 7-8 kHz
Zero elevation: Dip at 8-9 kHz
Positive elevation: Dip at 9-10 kHz


----------



## charleski

gregorio said:


> So please, enough with the "we do not know"!


I see this old thread has blown up again. But I think the real problem, as ever, is not that the subjectivists "do not know", but that they don't _want _to know. What they _do _want is to believe. I think we need to accept that there's little chance of converting a devoted adherent of faith-based audiophilia - they have too much invested in their never-ending search for audio nirvana. The best we can hope for is to raise the spectre of doubt in the minds of those who have not yet fully bought into the subjectivist dogma and can, perhaps, still be saved from the quagmire of perpetual dissatisfaction that fuels the 'high-end' industry.

There are a small number who have attempted to demonstrate real perceptual effects for ultrasonic/super DR audio, and I've covered their efforts in previous posts here. They have failed, but there's no shame in that and one should respect anyone seeking to perform genuine and honest experiments. Such things are, at least, interesting and worth arguing over. But arguing over someone's faith is a different matter, and faith-based audiophiles are just going to continue to entrench themselves further as a bulwark against apostasy. Some cancers can't be cut out, but there are still things you can do to stop them spreading.


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## frodeni

@gregorio in post: 13809060

You need to quote me properly, as not to falsify what you claim to be said. As in this post.

If you do not comprehend what is being said, you need to be careful at rephrasing quotes.

I struggled through your rant, but when you claim that recording individual instruments in mono to be "nonsense", there is not much reason left. Apart from a very few select instruments, like perk, this is done every day, all around the world. As it has, for ages.

Also, if you did not misquote me, you will clearly see, that I actually do not claim any opposition to Fourier at all. I actually accepted at face value, the claim that you made. I just happened to use a secondary paradigm, and more than one philosophy at a time. Which I am known for. It might not be known to you, but what might appear as evident from one angle, might be evidently false, from another. If that drives you nuts, then so be it. I cannot help it, but I am not that narrow minded.



gregorio said:


> R...The very fact that you "miss the point" of "dragging Fourier into this" indicates a massive hole in your personal knowledge/understanding of the "complexity of sound" and therefore all sorts of resulting nonsense theories, but it is just a case of you personally not knowing/understanding, NOT a case that "we don't know"! ...



If you read my statement closely, I actually pointed to a very well known theory of astronomy, which actually does exactly the same as Fourier, which is making a math expression, simulate another piece of math. As for how humans break down sound inside the brain, well, that is not known at all. Which was my main point. Thus it also belongs, for now, firmly in the interpretive paradigm. It is the only one that makes any sense. You might have proof, for all I know, as to how the brain actually calculates the sensation of sounds. If you do, that would simply be a sensation, and a huge breakthrough of epic proportions. That is why, I believe you got no such proof, or anything even remotely close to it.

If somebody suddenly can convert the brain into a computer, which you claim to do, then that is the one of the biggest milestones in human history. After all, you are bashing me in a rant, for claiming that this milestone simply is no being reached. Maybe you rather should quit the ranting?

If proof is what is being reflected over the last few pages, then the epistemology of this place is lacking. Not my cup of tea. Good to know. Bye for now.


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## pinnahertz

dprimary said:


> What do you record that hits 136 dB SPL? of course you need be in room with noise floor less than 16 dB SPL. For something like a violin you will not be able to exceed 16 bit with a SM81.





dprimary said:


> What do you record that hits 136 dB SPL?


Close mic'ed drum hits.


dprimary said:


> of course you need be in room with noise floor less than 16 dB SPL.


Room noise isn't specified as SPL, but there are 10dB studios in the world.  The one I last worked in was NC15 (we don't use NC anymore).


dprimary said:


> For something like a violin you will not be able to exceed 16 bit with a SM81.


Yup, and it's not even a challenge.


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## pinnahertz

dprimary said:


> The current U-87 has about 1 dB lower noise (then the  SM81) and 117 dB max SPL without the pad. They are insanely expensive now and they don't even include a mount.


You don't use a U87 for it's noise floor, though.  There are far quieter LDCs.


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## RRod

frodeni said:


> If you read my statement closely, I actually pointed to a very well known theory of astronomy, which actually does exactly the same as Fourier, which is making a math expression, simulate another piece of math. As for how humans break down sound inside the brain, well, that is not known at all. Which was my main point. Thus it also belongs, for now, firmly in the interpretive paradigm. It is the only one that makes any sense. You might have proof, for all I know, as to how the brain actually calculates the sensation of sounds. If you do, that would simply be a sensation, and a huge breakthrough of epic proportions. That is why, I believe you got no such proof, or anything even remotely close to it.
> 
> If somebody suddenly can convert the brain into a computer, which you claim to do, then that is the one of the biggest milestones in human history. After all, you are bashing me in a rant, for claiming that this milestone simply is no being reached. Maybe you rather should quit the ranting?
> 
> If proof is what is being reflected over the last few pages, then the epistemology of this place is lacking. Not my cup of tea. Good to know. Bye for now.



There have been decades of research on how humans perceive sound, and it's the reason why we've been able to get the necessary transparency bandwidth for audio data down to 128-256k! All that leans heavily on Fourier theory. Fine if you want to say "human hearing is complex", but that isn't proof that you need more bits or samples/second, which is of course the kind of thing people want to assume rather than prove.


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## gregorio (Oct 26, 2017)

frodeni said:


> [1] You need to quote me properly, as not to falsify what you claim to be said.
> [1a] If you do not comprehend what is being said, you need to be careful at rephrasing quotes.
> [2] ... when you claim that recording individual instruments in mono to be "nonsense", there is not much reason left.
> [3] .. you will clearly see, that I actually do not claim any opposition to Fourier at all. I actually accepted at face value, the claim that you made. I just happened to use a secondary paradigm, and more than one philosophy at a time ...
> [4] As for how humans break down sound inside the brain, well, that is not known at all. Which was my main point.



1. Pot, kettle, black. 1a. POT, KETTLE, BLACK!!
2. You're right, I have no reason and we should do as you say: We should indeed spend a year recording each symphony orchestra piece, recording each of the 100 or musicians one at a time in mono, in an anechoic chamber. It would cost a prohibitive fortune AND sound like utter crap but hey, you're the one with reason and I'm the one talking nonsense.
3. That's nonsense, void of all logic! Fourier is proven math. You either accept that 1+1=2, in which case there is no "secondary paradigm" or you don't accept that 1+1=2, in which case you can have whatever paradigm/philosophy you want but of course everyone will think you're nuts unless you've actually found a way to prove that 1+1≠2!
4. Here we go again, you don't know, therefore no one does. The whole thing is nonsense, because whatever the human brain does or does not do is TOTALLY IRRELEVANT! We are talking about the amount of data (bit depth and sample rate) required by a consumer distribution format to accurately reproduce an electric current. You ears cannot hear either digital data nor an electric current, so what has any of this to do with your ears and brain"??

G


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## pinnahertz (Oct 26, 2017)

frodeni said:


> You are talking about a multi speaker setup. Which is about as far away as you can get, form using headsets. While headsets would have few if none limits as to what may be achieved, speakers have a ton of limitations.


Headphones have limitations too.



frodeni said:


> That is a $350 microphone. Listed spec:
> Signal-to-Noise Ratio 78 dB (IEC 651)* at 94 dB SPL S/N ratio is difference between 94 dB SPL and equivalent SPL of self– noise A-weighted
> http://cdn.shure.com/specification_sheet/upload/228/sm81-specification-sheet-english.pdf
> 
> SN is also heavily dependent on the amp used. As for high floating point precision, sure, that is a thing these days. Signal processing is not my field of expertise.


Not really, the equivalent input noise on a mic pre is generally much lower than a mic. That is the goal of a good mic pre.


frodeni said:


> Thanks. I looked it up and had a quick look at the 50 first pages. Appears to be a purely positivistic work, yet not stating so at all. Not really stating any assumptions at all, like a lot of this type of work. A bit of warning though, a lot of the assumptions within the field of psychology, are really just that: Assumptions. Reading these works requires caution as to what tradition they are done within, as to trace the validity of their claims. I like to read the abstract first, then the conclusion, but that will have to wait til I get time for it, at the University.
> 
> It is also a bit strange to be faced with the general consensus as of decades, that I just wrote about. Knowing how equipment sensitive this is at the reproduction stage, I hope they rather did some real life experiments. Any results is beyond page 50, which is what Google shared. If they tested peopled that had a trained ear for this, it would be possible to reverse the calculation into bits and hz for sampling. If you know what the findings were, please share.


You asked for a recommendation, and you clearly can google...go nuts. I didn't ask for a book review.



frodeni said:


> People can handle five tasks at once, some up to 7. For any design work, there is a consensus at five tasks at the most.


With each task added proficiency drops significantly. People can typically only do one task well.


frodeni said:


> That is why we got harmonies.


Playing/singing harmony is a single task.


frodeni said:


> As for the need of stereo, no, that is a false statement. The gear having stereo output assumes regular stereo setup, which vector sound is not. Stereo output does neither fit into any 5.1 or similar scheme either. Yes, you will need a "channel" for every single sound, and no, that is not done in post processing like the Smyth. It is a form of presentation layer, that requires a form of computer. Like a smartphone or a PC. For music and static reproduction (not head spinning adjusted), the output might be processed once, and saved as regular stereo, for a given user-gear combo.
> 
> People seem to struggle with the fact that: No. There is no height to stereo. It is simply not there. Just as there is no stereo to mono. If you claim there is dept to stereo, please share with us, the physics at work. I used to think there was, until I got a fairly clean digital signal. That was an embarrassing moment, to say the least.
> 
> This focus on speakers is a bit misplaced. The room in which you listen is almost always the limiting factor, unless you pad every surface like nuts. Including the ceiling. Windows are a no go. Once you move into vector sound, speakers really start to struggle, no matter the room.


Any directional vector can be either simulated with heaphones or speakers or  reproduce with the physical speaker at that vector.  The simulation accuracy depends on an accurate  HRTF, which, unfortunately, is highly individualized,  making general HRTF
processing somewhat ambiguous.[/QUOTE]


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## bigshot (Oct 26, 2017)

frodeni said:


> @gregorio in post: 13809060
> You need to quote me properly, as not to falsify what you claim to be said. As in this post.



I have a little advice for you to help you participate in discussions better...

1) Listen to what people say and reply to the points people are making.

2) If someone is misquoting or misunderstanding your point, that is an invitation to you to make your point more clearly and to provide supporting arguments to make your case. It may be that you are being unclear because you aren't organizing your arguments into clear paragraphs with a clear opening statement, followed by supporting facts, and finishing up with a concise summation at the end.

3) If someone is deliberately mischaracterizing your points or engaging in ad hominem attacks, ignore them or speak past them to the lurkers, or address the people in the discussion who are debating fairly.

4) If you reply to someone by simply saying, "you are wrong and you're mischaracterizing what I say" without providing clarification and proofs to back up what you say, you might as well just concede defeat, because that's exactly what a post like that communicates to your readers. Especially if you end the post with "I'm out of here."

Hope this helps.


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## bigshot

pinnahertz said:


> You don't use a U87 for it's noise floor, though.  There are far quieter LDCs.



The studios I work with have really good mic pres and noise gates on the vocal mics.


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## pinnahertz

bigshot said:


> The studios I work with have really good mic pres and noise gates on the vocal mics.


You certainly can use a noise gate, but they are tricky beasts.  Better done in post, but a good quiet studio or vocal booth, they should not be necessary.  A good quiet mic pre, quieter than any mic in the locker, should be no big deal.  The ones I designed in 1980 are still quieter than any mic's self noise today, and I didn't do much that was "special". 

The "Worlds Quietest Studio Mic"...according to Rode, is their NT1A, self noise at 5dBA and 132dB of total DR.  It is, in fact, very quiet indeed.  Yet, still the limiting factor (not the mic pre) because its output is very hot putting it's 5dBA self noise well above the equivalent input noise of any decent mic preamp.  Of course, noise performance alone isn't mics are selected for use.  The NT1A has it's points, but isn't really preferred over many noisier mics because of other factors.

But, using an NT1A and a good mic preamp will push the limits of any 24bit ADC today, none of which actually have true 24 bit noise performance (except for one).


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## bigshot

I always used that mic when I was recording the Chipmunks. We used a gate because the noise would get VSO'ed up along with the voices and sound really nasty. It was all done on analogue tape even though it was a digital studio because the head guy didn't like digital VSO.

I don't remember what specific mic pres they were using, but I remember being told by the engineer that they were worth more than he was!


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## Strangelove424

What is it like to record the Chipmunks? I bet the mics are tiny.


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## bigshot

Ha! They work for peanuts!


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## gregorio (Oct 27, 2017)

pinnahertz said:


> Close mic'ed drum hits.
> Room noise isn't specified as SPL, but there are 10dB studios in the world.
> Yup, and it's not even a challenge.
> You don't use a U87 for it's noise floor, though.



Except for the last quoted sentence, I feel you're ignoring the actual practicalities/reality. Yes, a close mic'ed snare drum rimshot for example might produce a very high level, there are very low noise floor studios in the world and in theory you might be able to exceed 96dB dynamic range with a violin (not sure it wouldn't be a challenge though). BUT, even if you did have an exceptionally low noise floor live room, it's not so low a noise floor once you put musicians in there. In the case of say a rimshot or drum hit, the noise floor picked-up by that close mic is effectively incredibly high because a drum hit/rimshot typically does not occur in complete isolation, there would be spill from other instruments in the drum kit and typically we never use just a close mic, because it doesn't give a desirable/aesthetically pleasing result. In the case of a violin, the only way I can imagine of potentially exceeding 96dB dynamic range would be to very close mic it but again, in practise that is very undesirable.

In theory I'm sure you know what you're talking about but I don't know in practise, I've never tried it because it's either not possible in practise or it's aesthetically undesirable. Mics are chosen for their sonic characteristics, noise floor is only one of those characteristics and typically not the characteristic of primary concern and the same is sometimes true of the mic pre-amp. In other words, if my only goal as a recording engineer were to achieve the highest possible dynamic range, then maybe 16bit would in some cases not be enough but that is not my only goal in reality, in fact, it's quite a long way down the list of goals.

G

EDIT: "none of which actually have true 24 bit noise performance (except for one)." - There's one which does, how is that possible? Can you let me know which one or give me a link please, I'd like to read up on it.


----------



## frodeni

pinnahertz said:


> ... You asked for a recommendation, and you clearly can google...go nuts. I didn't ask for a book review. ...



I asked for if people knew of any good research on a topic, that implies, at least in any scientific forum I know about, that you know what your recommend and find it good yourself. I actually tired to look it up, in multiple online research databases, and it is not available on those. Given that you seem to indicate you just googled the book, and is not a book you recommend based on your own knowledge of it, I will not spend more time on it. Seems like wasted time. As a researcher, this is pretty irritating waste of time.



bigshot said:


> But speakers do soundstage and directionality naturally. Headphones have to be processed into doing it. I understand what you're talking about, but that has never been a part of acoustic sound reproduction. Back in the days of the acoustic Victrola they recorded everything dry and used the horn in the phonograph to project the sound into the room to give it a natural presence. Room ambience helped make severely limited recordings sound so real and present they could make the hairs on the back of your neck stand up. Natural room ambience is an important part of the sound quality of a speaker system too. If you think that the ideal listening room is completely dry and dead, and all of the ambience should be baked into the mix, you're wrong about what makes a good speaker system. The acoustics of the room is as much a part of the sound quality as the acoustics of the speakers.
> 
> It sounds like you're theorizing about speakers the way people theorize about headphones. But they are quite different animals and have different goals, advantages and limitations. With headphones, you're removing the room from the equation to make balanced frequency response and super low distortion easier to achieve. With speakers you are taking advantage of the room acoustics to give ambient space for the sound to exist in and you're adding a kinesthetic punch to the sound. Two totally different approaches- both equally valid- and both suited for their own purposes. You could design a listening room to mimic the advantages of headphones, but it wouldn't exploit the unique benefits of speaker sound. And you can probably synthesize speaker sound with headphones, but you'll lose the advantages headphones have. And it would be like a digital reverb hall ambience vs. a real concert hall ambience. Close but no cigar.
> 
> ...



Sure. Bose in particular has been playing on this aspect for years now. The problem is the physics of it. What you speak of, is adding to the recording, which is a form of distortion. For speakers, there is plenty of distortions in any normal environment, and for most acoustics designs, making these as flattering as possible, is a stated goal. But when the speaker itself and the room acoustics becomes a part of the musical mix, clearly altering the presentation, do we need high res music for that? I have even attended lectures at conferences for recording industry, covering just room acoustics. Read books on the topics. Even books on speaker building, like the calculations done on the reflex volume, and its affect on the rendering. Speakers are not rendering exactly what the recording is.

But sure, some argue about this "dryness" whatever that is. If you mean that nothing is added to the recording, then I am all with you. Some speakers play just fine, without the need for much to be added, and without a great need for extra reflections. Typically a studio speaker.

Moving beyond 16bit, if distorting the rendering like a normal listening situation with speakers do, why do you need more than 16bit? If you do not play above 90db, in pure technical terms, it should be all there, at 16bit. If there is an audible difference, and sure, that might be the case, what is causing it? There is also a ton of added distortion, for non-"dry rooms". Why this need for this super accurate rendering then?



RRod said:


> There have been decades of research on how humans perceive sound, and it's the reason why we've been able to get the necessary transparency bandwidth for audio data down to 128-256k! All that leans heavily on Fourier theory. Fine if you want to say "human hearing is complex", but that isn't proof that you need more bits or samples/second, which is of course the kind of thing people want to assume rather than prove.



I have no idea what you are talking about. A search in multiple science data bases return hits from the field of networking and on Google I get a lot of hits in the field of signal processing. A quick definition of the term, is not available, and not obtainable, without a considerable effort, even for me and the access I got through the University. I do not have the time for that. If this is vital info, please share.

Also, even though some theories and models have success at making the signal as imperfect as possible, yet still recognizable, it still is not The Explanation of how human hearing is working or what algorithms humans use. It perfectly well might be just a coincidence or a correlation. For all we know, there might be no universality to this at all.

And yet, I agree. That something is complex, is not proof of much. But if it is complex, then it is not simple. Also, if it is poorly understood, and there is limited knowledge, and universality is a huge unknown, what is known is really dependent on the epistemology applied. What is proof, depends on your epistemology and your ontology.

What is expected, currently, with the results at hand, is that positional accuracy, as done by hearing, will correlate with the findings of the bounds discovered thus far. But that is for the simple stuff. There might be combination of variables, or complexity, that suddenly reveals a different result, as a result of how the brain and senses work. Until that is a known, it will remain an unknown.

People also need to realize, that hardly any, if any, theory in physics is proven. They are just not proven false: Yet. The concept of knowledge is not a trivial one, and what constitutes a proof for anything, is a very blurry landscape. At least in science. That is why we, and at the risk of putting people of, we as in the science community, refer to the tradition we are a part of. It is essential.

Having seen and been taught some signal processing, what was constituted as a proof of no loss for image treatment, was not really my thing. I got great respect for what have been achieved, but no so much on what is considered proof for no loss on lossy compression. Particularly, when faced with proof that falsifies the results, the community answer is a huge letdown. Their math proofs usually holds up though, but as a tradition, it is flat out weak, if any subjectivity is introduced. 



gregorio said:


> Except for the last quoted sentence, I feel you're ignoring the actual practicalities/reality. Yes, a close mic'ed snare drum rimshot for example might produce a very high level, there are very low noise floor studios in the world and in theory you might be able to exceed 96dB dynamic range with a violin (not sure it wouldn't be a challenge though). BUT, even if you did have an exceptionally low noise floor live room, it's not so low a noise floor once you put musicians in there. In the case of say a rimshot or drum hit, the noise floor picked-up by that close mic is effectively incredibly high because a drum hit/rimshot typically does not occur in complete isolation, there would be spill from other instruments in the drum kit and typically we never use just a close mic, because it doesn't give a desirable/aesthetically pleasing result. In the case of a violin, the only way I can imagine of potentially exceeding 96dB dynamic range would be to very close mic it but again, in practise that is very undesirable.
> 
> In theory I'm sure you know what you're talking about but I don't know in practise, I've never tried it because it's either not possible in practise or it's aesthetically undesirable. Mics are chosen for their sonic characteristics, noise floor is only one of those characteristics and typically not the characteristic of primary concern and the same is sometimes true of the mic pre-amp. In other words, if my only goal as a recording engineer were to achieve the highest possible dynamic range, then maybe 16bit would in some cases not be enough but that is not my only goal in reality, in fact, it's quite a long way down the list of goals.
> 
> ...



I really like this post. It falls inline with my impression of how musicians typically work. I know people who could probably say exactly the same, almost word for word, for a greater part of this post.

Noise Reduction is used in post, if a mic picks up too much noise. Many find this a non-issue. Sometimes mics are used, even knowing they produce a lot of noise, for multiple reasons.  If you struggle with that, you need to listen to some Ed Sheeran tracks of late. Like "Supermarket Flowers". Just listen for it, and you will be able to pick it up all over the place, all over, as across the music industry. Just like in the old days, when Dolby C was used. If you know how that degraded the track, you can clearly hear the tell tale signs.

As for picking mics, there are a ton of videos on mics on the tube. The irony is that they are often picked for having a character opposite of what many audiophiles call for: They are often picked for how they distort the sound. For most musicians, it is more about expression. They sound expression they seek. Or what mic would fit a particular voice, as to make it sound great.

You will also find a lot of the bloggers or tubers posting on the gear they use. Almost any video using a mosquito, include some NR in post. The talk is more about the noise floor, and at least for portable gear, the mic amp is a huge factor as well.

I tried the high res on my portable recorder, and the result was worse than that of 16/44.1. Which is in line with the original claim of this thread. I found no benefit in post either, actually quite the opposite. Thing is, I have the exact same experience with my USB interface at home as well, which is a best selling USB interface for musicians. Sure, there is a difference, it is just for the worse.

As for math, high res recordings should sound as low res, but not in my case. To me, it is evident that something else is going on, like the interplay of gear, digital noise, or something.

It is not really that hard to tell a difference, but just because it comes with a label with a higher number, it is not necessarily better. I hear all these people speaking of their great high res experience, I just struggle to reproduce it.

Tidal introduced "Master" quality, using lossy (!) compression for high res. I just cannot get it to work, as with anything high res. Sure there is a difference. A difference for the worse. It even mess up the 16/44.1, which is supposed to be embedded, as it clearly is not lossless anymore. I would gladly pay for high res audio, but it better give me an improvement. That improvement is lost on me, on all my gear. I must be getting old or something.

Yet, the reported Tidal "improvements" is easy to achieve for 16/44.1 as well, just reduce the buffer to be as small, that it will produce a signal loss over USB, and a lot of the sonic traits are there. Sure, I can hear plenty of what is described as positive sonic traits, but these are what in my experience typically is followed by digital noise or signal loss. Also, a lot of other signs of noise, is present. There seem to be some improvement, but accompanied by a lot of negatives, making me even doubt if there is anything positives at all. What the real positives are, someone has yet to tell me.

I get an improvement from USB filtering and cables, but high res audio, it is still lost on me. Including when doing my own recordings.


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## pinnahertz (Oct 27, 2017)

gregorio said:


> Except for the last quoted sentence, I feel you're ignoring the actual practicalities/reality. Yes, a close mic'ed snare drum rimshot for example might produce a very high level, there are very low noise floor studios in the world and in theory you might be able to exceed 96dB dynamic range with a violin (not sure it wouldn't be a challenge though). BUT, even if you did have an exceptionally low noise floor live room, it's not so low a noise floor once you put musicians in there. In the case of say a rimshot or drum hit, the noise floor picked-up by that close mic is effectively incredibly high because a drum hit/rimshot typically does not occur in complete isolation, there would be spill from other instruments in the drum kit and typically we never use just a close mic, because it doesn't give a desirable/aesthetically pleasing result. In the case of a violin, the only way I can imagine of potentially exceeding 96dB dynamic range would be to very close mic it but again, in practise that is very undesirable.
> 
> In theory I'm sure you know what you're talking about but I don't know in practise, I've never tried it because it's either not possible in practise or it's aesthetically undesirable. Mics are chosen for their sonic characteristics, noise floor is only one of those characteristics and typically not the characteristic of primary concern and the same is sometimes true of the mic pre-amp. In other words, if my only goal as a recording engineer were to achieve the highest possible dynamic range, then maybe 16bit would in some cases not be enough but that is not my only goal in reality, in fact, it's quite a long way down the list of goals.
> 
> ...


I absolutely recognize that my example was more theoretical than practical.  My point was to refute that mic self noise and preamp noise was the limiting factor.

The manufacturer was Stagetec, http://www.stagetec.com/en/

They made a stand alone interface that used an ADC array/cascade that could do a real 144dB DR. I don't see it in their current product line up, and the current ADC isn't quite as good. They proposed an application where any input level, mic through line, could be handled without preamp gain adjustments. The current products look focused on large systems.


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## bigshot (Oct 27, 2017)

frodeni said:


> What you speak of, is adding to the recording, which is a form of distortion.



Yes I absolutely agree if you're using the term "distortion" in a non pejorative way. Any alteration of the signal is distortion. In the strictest sense, it's distortion for an engineer to apply EQ or reverb, or even to adjust balances in a mix. Anything that alters the signal is technically defined as distortion. But distortion isn't necessarily something bad. It can be a tool for improving sound and a method to better present the music. No engineer expects his mix to be played in the home in a dead room. The point of the mix is to come up with something that works for a range of different presentations. They may have tried to minimize room acoustics in the studio, but that doesn't mean that they don't intend for there to be room acoustics in home playback. Some forms of distortion actually *correct* error, like the timing calculations done by an AVR when you input the distances of the various speakers to the listening position. The distortion created by DSPs can actually improve the sound beyond what was heard by the engineers when the music was being mixed.

"Random distortion"... or "unwanted distortion" is a bad thing. But signal processing, even acoustic signal processing caused by mechanical means (room acoustics, horn loaded speakers, acoustic panels, etc.) can be a very good thing. When I first started out in the hobby back in the early 70s, I was fighting distortion from all sides- inner groove distortion, harmonic distortion, wow and flutter, record wear, rumble, noise, etc.- but today, most of those demons have been tamed. Reducing distortion in digital audio even further isn't likely to make any audible difference at all. The whole "purity of sound" theory in home audio is obsolete. Digital sound is as pure as human ears can detect. We don't have to keep splitting the atom to reduce inaudible distortion. No point to it. However, we can *use* distortion in the form of *signal processing* to make a huge improvement on perceived sound quality and fidelity.

The future of audiophile sound lies in signal processing. We've barely begun to tap into the possibilities.


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## Strangelove424

bigshot said:


> Ha! They work for peanuts!



Ha, the perfect performers! I'm actually kind of curious about it though. Are they the same particular performers, or is it all editing and not really matter what the original voices were like?


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## gregorio (Oct 28, 2017)

bigshot said:


> [1] They may have tried to minimize room acoustics in the studio, but that doesn't mean that they don't intend for there to be room acoustics in home playback.
> [2] Some forms of distortion actually *correct* error, like the timing calculations done by an AVR when you input the distances of the various speakers to the listening position.
> [3] The distortion created by DSPs can actually improve the sound beyond what was heard by the engineers when the music was being mixed.
> [3a] The best part of speakers is that there is a *natural* coloration to a real world installation that actually *enhances* the sound of the recording. In this case coloration is a good thing. I know a lot of people think good sound is just the sound, nothing but the sound, but with humans that isn't true. We want *present* sound- meaning it sounds like the music is in the room with us.



In general I agree with your post but there are a few points I disagree with:
1. No, this is virtually never the case. In TV/Film we tend to use more absorption to reduce room acoustics but never minimise them. The philosophy of music studio design is quite different though, typically with much less absorption and much more diffusion. In music production/mastering, room acoustics are therefore not minimised much beyond the average home environment but they are randomised to (hopefully) achieve a reasonably flat, neutral acoustic.
2. Yes but in this case what's being corrected is your personal speaker position. The ideal goal is to add a distortion which cancels out the distortion of speaker positioning, resulting in a distortion-less reproduction of the recording.
3. Now this is a dangerous assertion which falls within audiophile myth/misrepresentation! As far as fidelity is concerned, you CANNOT improve what was heard by the artists/engineers in the studio! You can of course change it, according to your personal tastes but what you end up with is lower fidelity, which is not an improvement, it's just a better match to your personal preference. IMO, it's very important to make this distinction, because much of the misleading audiophile marketing and myth is based on perverting this distinction. Effectively selling a preference as a higher fidelity improvement.
3a. This is essentially the same as point #3. The recording was released with the "natural colouration" of real world speaker installation and the human factor ALREADY baked in! It was mixed and adjusted on speakers by humans, according to their human perception and subjective opinion. It therefore already contains the exact amount of "present sound" intended and adding more colouration to compensate for "humans" is effectively double compensating. Now, maybe your subjective opinion differs to those of the artists and in my case it frequently does, I often feel that I would have done something somewhat differently, but I still want to hear what the artists themselves did. I don't want my system to automatically apply some adjustment to maybe more closely match my subjective preference/opinion, I want to hear those artists' preference/opinion.



frodeni said:


> [1] If you do not play above 90db, in pure technical terms, it should be all there, at 16bit.
> [1a] If there is an audible difference, and sure, that might be the case, what is causing it?
> [2] There is also a ton of added distortion, for non-"dry rooms". Why this need for this super accurate rendering then?
> [3] That something is complex, is not proof of much. But if it is complex, then it is not simple. ... What is expected, currently, with the results at hand, is that positional accuracy, as done by hearing, will correlate with the findings of the bounds discovered thus far. But that is for the simple stuff. There might be combination of variables, or complexity, that suddenly reveals a different result, as a result of how the brain and senses work. Until that is a known, it will remain an unknown.
> [4] People also need to realize, that hardly any, if any, theory in physics is proven. They are just not proven false...



1. Assuming noise-shaped dither, which is standard practise, then we're talking more in the range of 120dB, not 90dB AND, that figure is the figure above the noise floor of your listening environment, which is probably around 20dB (with headphones) or >30dB with speakers. Therefore, your statement should be "if you do not play above 140dB it should all be there at 16bit". Can your headphones actually output 140dB? If not, any talk of audibility is irrelevant because you obviously cannot hear what your equipment is not producing in the first place. I don't know of any headphones which can but let's say there are some, now we can talk about audibility and then we run into another even more serious problem, 140dB is well beyond the threshold of pain and well into the range of serious permanent hearing damage. These two factors, producing headphones with 140dB output and what it would do to you if you actually tried to listen to such an output, is why we don't need more than 16bit!
1a. A range of potential factors: A fault or deliberate design choice by a DAC manufacturer or in many cases, placebo or comparing different versions/masters.
2. No, music recordings are not made in dry rooms or designed for playback in dry rooms, as I explained above.
3. You are confusing two, effectively unrelated factors. One factor is the container format (16/44 or 48), the other factor is what we choose to put into that container. As an analogy, let's say that the container is a plate and what we choose to put on that plate is food. As far as the food is concerned, we don't fully understand the perception of taste. A plate only has two things to worry about though; not adding it's own flavour to the food and being big enough to contain any amount of food one person could eat. Provided the plate achieves these two requirements, the perception of taste and our understanding of it is completely irrelevant as far as the plate is concerned. It is of course entirely relevant to the food we put on the plate but that's a human choice, a factor unrelated to the plate itself. 16/44 is already a plate which is far bigger than could ever be required, how would a plate another 100 times bigger improve the food? And as far as adding it's own flavour is concerned ...
4. We're not talking about theories of physics. We're talking about proven mathematics, which you have been supplied with, mathematics which prove, using the analogy above, that the plate does NOT add it's own flavour. The difficulty was the implementation of that proven maths with technology/engineering but this difficulty has to be put into context. Even in the earliest days of consumer digital audio this "difficulty" was relatively (though not entirely), audibly insignificant but the astounding advances in digital technology in the last 30+ years means that not only are we way past any notion of audible significance but we can achieve this feat at astoundingly low cost, about $1.50 trade price for such a DAC chip. That doesn't mean that all DACs are audibly perfect because of course it's up to the individual DAC manufacturer how they choose to implement that proven math, whether they choose to go down the proven route of cheap, effectively perfect audio or take the different route of imperfect audio in order to differentiate their product.



frodeni said:


> [1] I really like this post. It falls inline with my impression of how musicians typically work.
> [2] Noise Reduction is used in post, if a mic picks up too much noise. Many find this a non-issue.
> [3] As for math, high res recordings should sound as low res, but not in my case.



1. You say that but it's completely contrary to your previous assertion, of recording everything in mono with minimised acoustics. I presume you've seen for example how a drummer typically works? Have you ever seen a drummer record the snare drum in mono, then record the kick drum in mono, then the hi-hats, then each of the toms, then each of the cymbals? No, that's both impractical and undesirable aesthetically. They play the whole drum kit in one go, it's recorded both with spot mics on some of the instruments and in stereo and often with a room mic, and then this is all mixed together in stereo, for an aesthetically pleasing result. Essentially classical music ensembles are recorded similarly and so are most other genres of acoustic music. Additionally, minimising acoustics is particularly preposterous because all acoustic instruments rely on acoustics, in some cases entirely! For example, an audience never hears the direct sound of a french horn, the bell of the instrument is pointed towards the rear wall of the concert venue and the audience only ever hears the reflections. Recording the direct sound and minimising the reflections/acoustics would result in something quite different to the expected/desired french horn sound.
2. NR is routine practise in Film/TV, in fact you'd be hard pressed to find any TV/Film without NR but not so in music. NR cannot perfectly differentiate noise from signal because the difference is essentially a human perception. Removing noise also damages the signal to some degree and is therefore avoided.
3. If it really is not just a trick of your perception (placebo) and providing you are not inadvertently comparing different versions/masters, then the only explanation is either that your DAC has been deliberately designed not to audibly perfectly implement the math for 16/44, or the higher sample rates contain ultra-sonic frequencies which are causing audible inter-modulation distortion downstream from your DAC.



pinnahertz said:


> I absolutely recognize that my example was more theoretical than practical.  My point was to refute that mic self noise and preamp noise was the limiting factor.



Not sure I understand, because in practice mic self noise and pre-amp noise often are the limiting factor or certainly contributory, along with the noise floor of the environment.

G


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## bigshot (Oct 28, 2017)

I have the big Toscanini box set. The recordings were all recorded in mono in studio 8H in New York. The room was too small to have much ambience, so the recordings are a bit dead sounding. When I run them through a DSP that simulates the Berlin Philharmonic hall, not only does the sound bloom into a proper hall ambience, it creates a sound that is passable as stereo. The difference between the sound of the original recordings and the processed playback is stark. Not subtle at all. And it's undeniably a huge improvement. Back in the day, they didn't have digital reverbs and multichannel sound. If they had, they certainly would have used it. Engineers are human. They work within restraints and make mistakes just like anyone else. If I can improve upon what they did, I sure will.

If people want to achieve really great sound, they have to be open to multichannel processing. A synthesized 5.1 DSP might not make something sound as good as discrete 5.1, but it can certainly make stereo and mono sound a lot better. Multichannel processing is the next step in audiophile sound. It's ironic that home theater people who don't care as much about natural sound ambiences are the ones embracing multichannel DSPs, while audiophiles continue with the religion of "inerrant word of God" when it comes to not applying the technology where it could do the most good.

The bread provides its own flavor to a sandwich while still containing the meat and lettuce. It's the same with ice cream cones, tortillas on a taco, icing on a birthday cake,.. and a room.


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## ev13wt

bigshot said:


> I have the big Toscanini box set. The recordings were all recorded in mono in studio 8H in New York. The room was too small to have much ambience, so the recordings are a bit dead sounding. When I run them through a DSP that simulates the Berlin Philharmonic hall, not only does the sound bloom into a proper hall ambience, it creates a sound that is passable as stereo. The difference between the sound of the original recordings and the processed playback is stark. Not subtle at all. And it's undeniably a huge improvement. Back in the day, they didn't have digital reverbs and multichannel sound. If they had, they certainly would have used it. Engineers are human. They work within restraints and make mistakes just like anyone else. If I can improve upon what they did, I sure will.
> 
> If people want to achieve really great sound, they have to be open to multichannel processing. A synthesized 5.1 DSP might not make something sound as good as discrete 5.1, but it can certainly make stereo and mono sound a lot better. Multichannel processing is the next step in audiophile sound. It's ironic that home theater people who don't care as much about natural sound ambiences are the ones embracing multichannel DSPs, while audiophiles continue with the religion of "inerrant word of God" when it comes to not applying the technology where it could do the most good.
> 
> The bread provides its own flavor to a sandwich while still containing the meat and lettuce. It's the same with ice cream cones, tortillas on a taco, icing on a birthday cake,.. and a room.



While I wouldn't go as far as simulating a huge hall, I do agree that that is a fun venture! 
Lots of things to explore!!


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## pinnahertz

gregorio said:


> Not sure I understand, because in practice mic self noise and pre-amp noise often are the limiting factor or certainly contributory, along with the noise floor of the environment.
> 
> G



I was responding to this from Post 128, "The physics and math in this case, simply means that there is not enough resolution by the mic, to out resolve 16bit. " The example of a Rode NT1A into a quiet mic pre recording high SPL signals, as high as possible without clipping the microphone's built in preamp, and with mic preamp gain set for optimum (like matching it's clip point to that of the mic), and reducing preamp gain to the minimum required, the final output DR could exceed 16 bits, more like 20+.  In practice you can hardly ever do exactly that, and room noise gets in the way, but it can actually be done, and the DR of that mic is greater than 16 bits, as is the DR of some other mics.  But that's why it's a theoretical example.  There are relatively few mics as quiet and high output as an NT1A, most are 15-20dB noisier.  And, as I alluded to, we don't usually pick mics based only on their self noise.  And the point was to show that some mics have plenty of "resolution", actually DR, to max out 16 bits, even though it's not done in practice.


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## bigshot

ev13wt said:


> While I wouldn't go as far as simulating a huge hall, I do agree that that is a fun venture!
> Lots of things to explore!!



I just wish AVR's had a standard plugin architecture so I wouldn't be limited to just the DSPs that came with my receiver.


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## gregorio

bigshot said:


> I have the big Toscanini box set. The recordings were all recorded in mono in studio 8H in New York.
> [1] The room was too small to have much ambience, so the recordings are a bit dead sounding.
> [2] When I run them through a DSP that simulates the Berlin Philharmonic hall, not only does the sound bloom into a proper hall ambience, it creates a sound that is passable as stereo. The difference between the sound of the original recordings and the processed playback is stark. Not subtle at all.
> [3] And it's undeniably a huge improvement.
> ...



1. The room couldn't have been too small to have much ambience. Even very small rooms can have considerable ambience but the room obviously had to be fairly large in order for a symphony orchestra to even get in.
2.  OK, so you're deliberately and very significantly changing how the recordings were designed to be heard.
3. You can by all means state that this very significant change you've applied sounds hugely better to you but you cannot state with any certainty that it is an improvement and you definitely cannot state it's an undeniable improvement.
4. And maybe today, Mozart would have composed for synths rather than for an orchestra but we don't know. What we do know is that they did what THEY thought was best with the resources they had available.
4a. But what the engineers did was approved, unless there was a fault at the pressing plant but usually even that would be picked up quite quickly.
4b. It's entirely up to you how you want to playback your recordings but all you're doing by very significantly changing the playback is satisfying your personal tastes, you are not necessarily "improving" anything. Admittedly, with such old recordings we cannot be sure what (freq content, etc.) has been lost in the storage and transfer and on a modern system we cannot be sure what it was intended to originally sound like on a shellac disk through a gramophone horn. Maybe there would have appeared to be more ambience and reverb when originally played back or maybe they intended it to sound drier than we're accustomed to today, for some artistic reason, in which case your "improvement" might be the exact opposite of the musical intention.

You appear to repeatedly make incorrect assertions about recording studios and acoustics and, about making improvements to recording playback when you have no idea whether it's an improvement or not. We have to be careful of this, particularly in the sound science forum and particularly because it's the root of so many audiophile evils. I have no objection to you playing back your music however suits you preferences, I just object to you automatically labelling it an improvement and object even more strongly to you calling it an "undeniable improvement". Just call it what it is; poorer fidelity but better relative to your preferences/tastes. If you're not willing to do this, then you can't complain when audiophile manufacturers market their products in exactly the same way!

G


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## bigshot

Look up the Toscanini Studio 8H recordings if you aren't familiar with them. They're infamous for their cramped, boxy sound. If you aren't familiar with them, I can understand how you wouldn't understand what I'm talking about.


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## gregorio

bigshot said:


> Look up the Toscanini Studio 8H recordings if you aren't familiar with them. They're infamous for their cramped, boxy sound. If you aren't familiar with them, I can understand how you wouldn't understand what I'm talking about.



You seem to be completely missing my point. Those recordings may very well be relatively poor, cramped and boxy but that is what was created by the artists, engineers and approved by Toscanini. If I were to listen to those recordings, I want to hear what was created and approved by Toscanini, warts and all. That's my preference though and you are entitled to change what Toscanini created/approved more towards your personal taste BUT, that change is therefore an "improvement" only in terms of your (and others') preference NOT in terms of fidelity, which is you've lowered rather than improved. Confusing an improvement towards personal preference with an actual improvement in fidelity is exactly the tactic used in many audiophile marketing materials, which I've seen you justly rail against on many occasions!

G


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## 71 dB

gregorio said:


> You seem to be completely missing my point. Those recordings may very well be relatively poor, cramped and boxy but that is what was created by the artists, engineers and approved by Toscanini. If I were to listen to those recordings, I want to hear what was created and approved by Toscanini, warts and all. That's my preference though and you are entitled to change what Toscanini created/approved more towards your personal taste BUT, that change is therefore an "improvement" only in terms of your (and others') preference NOT in terms of fidelity, which is you've lowered rather than improved. Confusing an improvement towards personal preference with an actual improvement in fidelity is exactly the tactic used in many audiophile marketing materials, which I've seen you justly rail against on many occasions!
> 
> G


Toscanini approved yes, but did he do it because he thought it is perfect or did he approve because they did their best and he had no choice? What would Toscanini say about what *bigshot* does? Changes are he would approve it. Historical recordings are often so bad in technical quality on our standards, that almost _anything_ is an improvement.


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## gregorio

71 dB said:


> Toscanini approved yes, but did he do it because he thought it is perfect or did he approve because they did their best and he had no choice?



Yes, with such old recordings chances are that Toscanini would be delighted by modern technology but maybe he'll still have wanted a much drier recording than bigshot does? Pre-WWII recordings are of course the most extreme example, which is presumably why bigshot chose it, but even with these most extreme examples, there's still an argument for hearing what the artists/engineers created, an argument which becomes stronger as we progress beyond WWII, into the age of stereo and much higher quality technology.

G


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## reginalb

gregorio said:


> ...What we do know is that they did what THEY thought was best* with the resources they had available...*
> G



You seem to wildly under-emphasize that point.


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## gregorio

reginalb said:


> You seem to wildly under-emphasize that point.



No, it wasn't wildly, it was deliberately because we don't know what they would have done with modern technology.

G


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## reginalb

gregorio said:


> No, it wasn't wildly, it was deliberately because we don't know what they would have done with modern technology.
> 
> G



But we can safely assume they would have done things quite a bit differently. And I don't think there's ever anything wrong with trying to play something back in a way that sounds more like it would in person, and doubt many musicians of any type would have a problem with that. 

Of course, at the end of the day, we should all listen the way we prefer, so you're not wrong to have your preference, but it's definitely not mine.


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## gregorio

reginalb said:


> And I don't think there's ever anything wrong with trying to play something back in a way that sounds more like it would in person, and doubt many musicians of any type would have a problem with that.



Unless the mix is either already optimised to sound like it would in person or it's specifically designed not to sound like it would in person!

G


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## KeithEmo

I usually don't chime in on this group - but sometimes I just hear a need for a bit of actual balance (as opposed to listening to two soapboxes on opposite corners of the public square).

First....... "humans only hear from 20 Hz to 20 kHz".
Are you sure?
I read at least one quite reputable seeming report that claims that "under lab conditions some humans can actually hear sounds as low as 10 Hz".
Now, to be honest, I don't recall the details, and I'm not prepared to argue the point.
However, I'm also _NOT_ willing to absolutely positively say that "since nobody can hear 15 Hz we should just throw it away".
I've also read the results of at least two studies that seem to show that we humans can sometimes discern minute timing differences - of the sort that a 16/44k recording cannot properly reproduce.
I don't know if that's real either.
However, since bandwidth and storage space are so cheap, I'd really rather not find out ten years from now that I've been throwing away something that turned out to be useful.

I would also point out that many of the "truisms" I see repeated over and over again aren't actually strictly speaking true.
"The time resolution of a signal is not specifically limited by its bandwidth." (Which means that a 44k recording can reproduce a time/phase difference between the left and right channels of only 5 microseconds.)
"Any signal can be reproduced with perfect accuracy as long as your sample rate is at least twice the highest frequency you need to reproduce". (Basic Shannon/Nyquist.)
Both absolutely true - _BUT ONLY FOR CONTINUOUS SINE WAVES_.

If I play an "impulse" - an arbitrarily short click - it will be audible because it contains frequency components that fall in the audible range.
And, if that impulse is short enough, a recording with a 44k sample rate will _NOT_ be able to reproduce it accurately - because the position of a _SINGLE TRANSIENT_ that falls between samples cannot be accurately resolved or reproduced.
(None of that other theory applies because we're not talking about a continuous sine waves here.)
Of course, a "pure transient" isn't a valid digital signal..... but how about the exact starting point of a non-continuous waveform that falls between two sample points?
Hmmmmmm........
Does that mean that, on our "CD quality recording", that click might be shifted in the sound stage because its position in time is_ NOT_ accurately reproduced?
And can human ears distinguish the difference?
I'm not sure.... and that's the whole point.
Not enough of _THOSE_ tests have been done to convince me either way. 
At least one AES-reported test seems to show that it is quite possible to make a test recording such that limiting the bandwidth to 20 kHz (compared to a high-resolution original) causes perceived positions of sounds in the sound stage to shift.
We're not talking about hearing things above 20 kHz; we're talking about an audible click that changes its perceived position in the sound stage when you limit the signal to 20 kHz bandwidth.
(So, if that click is the sound of a drumstick tapping the rim of a snare, it will sound like it's coming from a slightly different place on the stage - so maybe it won't line up with the rest of the sound coming from the drum perfectly.)
Now, maybe that will turn out to be a red herring..... but I'm not _SURE_ it will.

Another thing is that, in at least some cases, some of those "pointless" HD versions have turned out to sound better than the non-HD versions.
My guess is that the main reason is that they were remastered - and it was simply done better than the mastering on the regular version.
However, since any sample-rate conversion involves filtering, and so potentially a slight change in sound, we can never compare that 192k version to an "identical" 44k version.
And, it doesn't really matter if "they could have made a 44k version that sounds identical" if they _DIDN'T_. 
If I buy the better sounding 192k version, but my DAC won't reproduce 192k files, then I have to convert it - which is a nuisance and will quite possibly change the sound.
(So, even if the 192k version doesn't_ inherently_ sound better, I'm still better off being able to obtain an unaltered copy, and play it as is, without putting it through another conversion.)

The question of whether "they" could produce recordings at 44k that sound every bit as good as the ones they're selling at 192k is a different question than of whether we have a reason to be able to play 192k files.

People who keep track of my posts know that I an a firm _OPPONENT_ of snake oil.
However, while I'm not 100% sold on the benefits of high-resolution audio, I'm also not sold 100% on all of the arguments against it.
And, to be totally blunt, if this week someone happens to be selling a great sounding re-master of an album I like.....   then it's worth buying.
And, since the bandwidth and storage space required for a 192k file only cost a few cents more, why should I care one way or the other?
(Lots of things in this world aren't perfectly efficient or perfectly optimized.... but we tolerate the inefficiency for other reasons.)

And, yes, I do consider many of the original arguments _AGAINST_ 192k presented by Xiph as somewhat specious.
I see them as being analogous to suggesting that "it's bad to make cars that can go over 100 mph because some drivers have problems at that speed".
(Any piece of equipment that is going to have terrible problems if presented with ultrasonic audio components should have bandwidth limiting designed into its input circuitry.)

Incidentally.... a complete aside about the original article that sparked this whole discussion.
At one point Xiph claims that, because the frequency range of our eyes is limited, we cannot ever see the light coming from an IR remote control.
Hmmmmmm.
I used to have an IR LASER pointer - which put out about 1/4 watt at 720 nm (well into the color range of IR remote controls).
Guess what?
You _CAN_ see "near infrared" quite clearly if it's bright enough (so that claim isn't actually true).
(And, yes, we are talking about something that's dangerously bright.... )



bigshot said:


> Just as a bit of interesting trivia... 44.1K covers the full spectrum of frequencies that humans can hear- 20Hz to 20kHz, with a bit to spare. Higher sampling rates extend the frequency response higher, far beyond our ability to hear, but the core frequencies below 20kHz are rendered exactly the same at 44.1 as they are at 192. So whatever it is that you seem to think is clearly audible isn't audible with human ears. Perhaps a bat!
> 
> However, it is possible that your equipment isn't designed to deal with super high frequencies and is adding distortion down in the audible range. So if you are positive you are hearing a difference, it is almost certainly noise, not music.


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## KeithEmo (Oct 31, 2017)

I would take that a bit further.....

What if I adjust that recording to sound just the way* I* like it, and you adjust it to sound just the way *YOU* like it?
Now which version is "better" or "right"?
And which one would Toscanini have preferred?
Or would he hate both of them, and prefer something entirely different?
Or, perish the thought, maybe he actually would have _LIKED_ that awful original the most.

Either way, I much prefer to have it in its original form, rather than accept someone else's idea about what it should sound like.
(We pay the mastering engineer for his/her opinion - in the form of what that master sounds like.)
And most of us pay for equipment that is best able to deliver it to us exactly the way it's recorded.
However, since neither of us was there, we can never know that "Toscanini would have liked it a certain way".

If I'd edited it to sound different - in a way I much prefer - I'd save a copy..... and I'd proudly explain to people that I'd "adjusted it" in a way that I think sounds better.
_HOWEVER_, I would _NOT_ present my new modified version as "a better version of the original" - nor would I claim it was "more accurate" of "higher fidelity".
(It's kind of silly to say "you have a great recording of Toscanini - played in Carnegie Hall" if in fact the recording wasn't played in Carnegie Hall. What you have is a simulation of what it _MIGHT_ have sounded like_ IF_ it was played there.)

I would also like to add something here about equipment. If you really want to alter the way a recording sounds, there are dozens of audio editors, and thousands of plugins for them, all designed to alter the way music sounds in an amazing variety of ways. It makes very little sense to pay a lot of money for a specific piece of equipment, based on the fact that it alters the way things sound in some specific way you happen to like, when performing your alterations using software gives you so much more flexibility and a much wider variety of options. (The odds of the alteration produced by a particular piece of equipment being favorable to all your recordings are pretty slim.)  



gregorio said:


> You seem to be completely missing my point. Those recordings may very well be relatively poor, cramped and boxy but that is what was created by the artists, engineers and approved by Toscanini. If I were to listen to those recordings, I want to hear what was created and approved by Toscanini, warts and all. That's my preference though and you are entitled to change what Toscanini created/approved more towards your personal taste BUT, that change is therefore an "improvement" only in terms of your (and others') preference NOT in terms of fidelity, which is you've lowered rather than improved. Confusing an improvement towards personal preference with an actual improvement in fidelity is exactly the tactic used in many audiophile marketing materials, which I've seen you justly rail against on many occasions!
> 
> G


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## RRod

KeithEmo said:


> If I play an "impulse" - an arbitrarily short click - it will be audible because it contains frequency components that fall in the audible range.
> And, if that impulse is short enough, a recording with a 44k sample rate will _NOT_ be able to reproduce it accurately - because the position of a _SINGLE TRANSIENT_ that falls between samples cannot be accurately resolved or reproduced.
> (None of that other theory applies because we're not talking about a continuous sine waves here.)
> Of course, a "pure transient" isn't a valid digital signal..... but how about the exact starting point of a non-continuous waveform that falls between two sample points?
> ...



Not following you here, Keith. The theory says you must first bandlimit the signal, so your infinitely thin 'click' will stretch out to a sinc-like waveform and thus be amenable to sampling at the given rate. And if the filter you use to bandlimit is linear phase, then there is no time shifting of the peak.


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## bigshot (Oct 31, 2017)

gregorio said:


> You seem to be completely missing my point. Those recordings may very well be relatively poor, cramped and boxy but that is what was created by the artists, engineers and approved by Toscanini. If I were to listen to those recordings, I want to hear what was created and approved by Toscanini, warts and all.



That's a valid philosophy for sure. But it doesn't have anything to do with the quality of the sound. You might want to go one step further and listen to Toscanini on a mono tabletop tube radio, because that's what they intended as well.  It's possible that it might sound better that way than playing CDs on a modern sound system. Some of those recordings are pretty unlistenable on CD without helping them with DSPs. It's a shame too, because the performances are electric.

I wonder if the Honeymooners would be funnier on an 8 inch Dumont TV?



gregorio said:


> Yes, with such old recordings chances are that Toscanini would be delighted by modern technology but maybe he'll still have wanted a much drier recording than bigshot does?



Toscanini loved performing in Carnegie Hall, which is a very similar acoustic to the Berlin DSP on my Yamaha DVR. It's not quite as resonant as the Vienna and Chamber DSPs. Berlin is the one I use to correct the 8H recordings. By the way, these are post-war recordings (47-55 or so), not pre-war. They were recorded for radio broadcast.



KeithEmo said:


> Either way, I much prefer to have it in its original form, rather than accept someone else's idea about what it should sound like.



That's the great thing about using DSPs. I use the original recording on CD and process it on the fly to improve it. Cake- eat it too.



KeithEmo said:


> What if I adjust that recording to sound just the way* I* like it, and you adjust it to sound just the way *YOU* like it? Now which version is "better" or "right"? And which one would Toscanini have preferred?



TOSCANINI CAN BUY HIS OWN DAMN STEREO SYSTEM!


----------



## bigshot (Oct 31, 2017)

My favorite story about "original artists' intent" is about the great pianist Arthur Schnabel. His Beethoven was widely acclaimed as being the greatest of his era. But he refused to record. He explained it like this... "If I record the sublime Moonlight Sonata, I couldn't help but imagine someone sitting at the kitchen table eating a ham sandwich wearing an undershirt listening to the genius of Beethoven on my phonograph record. I think too highly of Beethoven to allow that."

They finally did convince him to record, but the sound quality is poor because he put too many constraints on the recording sessions. Digital technology can go a long way to correct that unless you want to abide by the grudging intent of the creator. Spectacular performances though. If you ever get the chance to hear them, make sure you get dressed in a suit and tie!


----------



## castleofargh

the 20hz to 20khz is the estimated range for the human species, not world record numbers... many young children can hear above 20khz, geezers almost never get past 15khz. still we stick to 20-20k as a reference.  the day I learn that musicians and sound engineers manage the ultrasound content by ear, I'll be willing to bother keeping it as "the artist intended it". 
most of us here have highres media and use them when they're good. we're against false arguments pretending audibility that is consistently debunked in controlled tests. and we're against the highres industry because they seem unable to keep themselves from lying about what they're selling. I personally have nothing against fidelity, I can hear it or I can't, I keep it or I convert to lower resolution to save space on my hard drive, those are personal choices. I don't know anybody here who's trying to force the all world to use only AAC. 

I don't get the infrared thing. maybe the laser was also emitting visible light? did you fact check that?  


as for the impulse, again I don't get it. I can't help but think of an impulse in term of sine waves. and from there, if it contains above half sampling frequencies, then of course we'll lose some. so let's say it's not about the false argument to disguise ultrasounds as something audible, and it's a legit concern about the transient of some instruments attack or whatever. if my own ears aren't applying the equivalent of a low pass on those frequencies any way, how come it's so hard to pass a blind test about highres?  that's something I can't wrap my mind around, if something is audible and 44.1khz fails to keep it audibly transparent, why is it so hard to pass a blind test vs highres using musical content?  my own conclusion is that we're never hearing more of a transient signal than its content within the audible range. talking about audible stuff while failing to hear them in a blind test, that's a contradiction.


----------



## KeithEmo

Yes......

So, I start out with a PHYSICAL equivalent of an impulse, perhaps by tapping my drumstick on the edge of my drum, I have a very short duration pulse. 
Most of the energy contained in it will be at inaudibly high frequencies, but enough will be at audible frequencies that it will be audible as a click.
And, due to my ears resolving the times when that click arrives at each, my brain will construct an apparent location for that click in the sound stage.
However, once you apply your band limiting, the energy from that impulse will be spread out over several sample periods..... and the precise location in time where it occurred_ INSIDE A SINGLE SAMPLE PERIOD _will be "lost".
(We will no longer be able to determine it.)

If I had a continuous sine wave appearing in both channels, even if they were a few microseconds "out of synch", I could extract that information, even at a 44k sample rate.
And shifting the relationship between them, even by a few microseconds, might shift the apparent location of my recorded instrument a few inches to the left or right.
(Because I can resolve a timing difference of far less than a single sample - and in fact one that is infinitely small - with continuous sine wave signals.)
However, with that single short impulse, I'm going to end up with a sort of blurred peak... and, since I don't know the shape of that peak, or exactly where it occurred, I will be unable to reconstruct exactly where the peak itself occurred.

We're stuck with having to band limit the signal....
The real question is whether, when I band limit what started out as a short sharp impulse, there will be_ NO AUDIBLE DIFFERENCE_.
(If I can hear the difference between that mechanical click and the band limited recording of it, then, in terms of audio fidelity, I will have failed to reproduce it exactly.
And that's true even if the difference is merely an apparent shift of a few inches in position in the sound stage.)
Some studies have shown that we humans can detect _VERY_ small shifts in timing..... far smaller than can be resolved at a 44k sample rate.

The question is simply this......
The idea that human hearing can distinguish _NOTHING_ outside the range between 20 Hz and 20 kHz was established using pure continuous sine waves.
But have we determined, beyond any doubt, that this is also true for any and all _OTHER WAVEFORMS - including impulses and even purely arbitrary waveforms_?
Have we proven that there is no sound which cannot be reproduced with "audibly perfect accuracy" in a system that is band-limited to 20 kHz - or have we only really established that "fact" for continuous sine waves?
As far as I know, very few tests have been conducted with anything other than pure, continuous sine waves of relatively long duration.



RRod said:


> Not following you here, Keith. The theory says you must first bandlimit the signal, so your infinitely thin 'click' will stretch out to a sinc-like waveform and thus be amenable to sampling at the given rate. And if the filter you use to bandlimit is linear phase, then there is no time shifting of the peak.


----------



## KeithEmo

The "infrared thing" is pretty simple..... and the output of the LASER is quite pure.

The sensitivity of our eyes to frequencies outside what we normally refer to as "the visible range" falls off rather rapidly - but not immediately.
The red sensors in our eyes do in fact detect what we call "near infrared" - but only if it's _VERY_ bright.
(We're talking about a level that would only be safe to look at for a few seconds at a time.)
A 200 mW 750 nm LASER focused on a 2 mm spot is many tens of thousands of times brighter than the LED on your remote control..... and is in fact still visible to most humans.
(It appears as a sort of pale pink.)



castleofargh said:


> the 20hz to 20khz is the estimated range for the human species, not world record numbers... many young children can hear above 20khz, geezers almost never get past 15khz. still we stick to 20-20k as a reference.  the day I learn that musicians and sound engineers manage the ultrasound content by ear, I'll be willing to bother keeping it as "the artist intended it".
> most of us here have highres media and use them when they're good. we're against false arguments pretending audibility that is consistently debunked in controlled tests. and we're against the highres industry because they seem unable to keep themselves from lying about what they're selling. I personally have nothing against fidelity, I can hear it or I can't, I keep it or I convert to lower resolution to save space on my hard drive, those are personal choices. I don't know anybody here who's trying to force the all world to use only AAC.
> 
> I don't get the infrared thing. maybe the laser was also emitting visible light? did you fact check that?
> ...


----------



## bigshot (Oct 31, 2017)

KeithEmo said:


> Yes......
> 
> So, I start out with a PHYSICAL equivalent of an impulse, perhaps by tapping my drumstick on the edge of my drum, I have a very short duration pulse.
> Most of the energy contained in it will be at inaudibly high frequencies, but enough will be at audible frequencies that it will be audible as a click.
> ...



The problem here is that you don't really know what sort of scale a sample or a drum tap occupies. It's all small, so your mind just lumps it all together. But the duration of a drum tap is several orders of magnitude bigger than the sample- it probably spans a thousand samples. And when it comes to the way that we locate sounds in space, it's not by discerning microscopic fractions of a second like a sample. It involves room acoustics and turning our head to be able to parse directionality. Your understanding of frequencies is the same. How high of a frequency is a drum tap? How high is 25kHz as opposed to 20kHz or 15kHz or 10kHz? What do they sound like? I'll give you a hint... they don't really sound all that different because they're all around the highest octave we can hear- and the difference between 10 and 15 is more than the difference between 20 and 25. Try and define what the numbers represent and you'll understand the physics better.

As a for instance... you're talking about super audible frequencies and the audibility of sound above 20kHz. If I remember correctly, the highest frequency ever recorded as being heard by human ears is somewhere between 22kHz and 23kHz. The frequency range doubles with each octave. So the 23kHz represents 1/13th of an octave. There are seven notes in an octave, so that only represents a half of a note. Humans hear 10 octaves at best. So the difference between hearing 20kHz and hearing 23kHz is almost nothing.

Taking that one step further... The difference between 10kHz and 20kHz is one octave. 1/10th of the sound we can hear. Seven notes at the bleeding edge of human hearing. Only a couple of musical instruments produce sound in that range, and most of it is inaudible due to masking. Controlled testing has shown that the presence of super audible frequencies adds nothing to the perceived sound quality of recorded music. They have also shown that most people don't even really care if all the frequencies above 10kHz are rolled off. Super audible frequencies just don't matter.


----------



## 71 dB

KeithEmo said:


> I usually don't chime in on this group - but sometimes I just hear a need for a bit of actual balance



Well, let's see what you have to say:



KeithEmo said:


> First....... "humans only hear from 20 Hz to 20 kHz". Are you sure?
> I read at least one quite reputable seeming report that claims that "under lab conditions some humans can actually hear sounds as low as 10 Hz".
> Now, to be honest, I don't recall the details, and I'm not prepared to argue the point.
> However, I'm also _NOT_ willing to absolutely positively say that "since nobody can hear 15 Hz we should just throw it away".


We can "sense" frequencies down to 0 Hz. You can definitely feel 10 Hz, but it's a sensation of vibration rather than a sound. The lowest frequency that is sound-like is about 16 Hz, but the sound pressure level must be very high. At 20 Hz the hearing treshold is about 75 dB. Very low frequencies can be stored on CD. There is no lower limit, well if we say we need at least one oscillation and the length of CD is 80 min (4800 seconds), the _theoretical_ lower limit is 0.00021 Hz. However, most CD-players filter frequencies below 5 Hz, so that's the practical lower limit for CD in general.




KeithEmo said:


> I've also read the results of at least two studies that seem to show that we humans can sometimes discern minute timing differences - of the sort that a 16/44k recording cannot properly reproduce.


Common misconception of digital audio. The time resolution of digital audio is not limited by sampling rate AT ALL. It is only limited by bit depth and 16 bits provide more temporal resolution you or me will ever need. Study digital audio and someday you will understand this.



KeithEmo said:


> However, since bandwidth and storage space are so cheap, I'd really rather not find out ten years from now that I've been throwing away something that turned out to be useful.


Whatever. I know I am not throwing away something useful because dogs and bats don't listen to music. 



KeithEmo said:


> I would also point out that many of the "truisms" I see repeated over and over again aren't actually strictly speaking true.
> "The time resolution of a signal is not specifically limited by its bandwidth." (Which means that a 44k recording can reproduce a time/phase difference between the left and right channels of only 5 microseconds.)
> "Any signal can be reproduced with perfect accuracy as long as your sample rate is at least twice the highest frequency you need to reproduce". (Basic Shannon/Nyquist.)
> Both absolutely true - _BUT ONLY FOR CONTINUOUS SINE WAVES_.


No. Applies to all _bandlimited_ signals. Any signal will do as long as it's bandlimited.




KeithEmo said:


> If I play an "impulse" - an arbitrarily short click - it will be audible because it contains frequency components that fall in the audible range.
> And, if that impulse is short enough, a recording with a 44k sample rate will _NOT_ be able to reproduce it accurately - because the position of a _SINGLE TRANSIENT_ that falls between samples cannot be accurately resolved or reproduced.
> (None of that other theory applies because we're not talking about a continuous sine waves here.)
> Of course, a "pure transient" isn't a valid digital signal..... but how about the exact starting point of a non-continuous waveform that falls between two sample points?
> Hmmmmmm...…..


It doesn't matter when the impulse happens. At a sample point or in between.* Does not matter!* The impulse is bandlimited and speads in time so that each sample point has a value corresponding the sinc function. This is also how we can "store" the timepoint of the impulse at insane accuracy limited only by quantization/dither noise. Study digital audio and learn. Don't fall into the trap of intuition, because digital audio goes a bit against intuition.



KeithEmo said:


> Does that mean that, on our "CD quality recording", that click might be shifted in the sound stage because its position in time is_ NOT_ accurately reproduced?
> And can human ears distinguish the difference?
> I'm not sure.... and that's the whole point.


The click happens at exactly correct moment for human ear. Temporal accuracy could be 1000 times worse and still it would be good enough.



KeithEmo said:


> Not enough of _THOSE_ tests have been done to convince me either way.


Nothing will until you learn more.


----------



## castleofargh

KeithEmo said:


> The "infrared thing" is pretty simple..... and the output of the LASER is quite pure.
> 
> The sensitivity of our eyes to frequencies outside what we normally refer to as "the visible range" falls off rather rapidly - but not immediately.
> The red sensors in our eyes do in fact detect what we call "near infrared" - but only if it's _VERY_ bright.
> ...


alright so it's borderline visible range and we compensate the lack of cells sensitive to that range with high level energy. so about the same idea as making 20khz audible for me even nowadays if I boost the volume to unsafe levels.


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## RRod (Oct 31, 2017)

KeithEmo said:


> Yes......
> 
> So, I start out with a PHYSICAL equivalent of an impulse, perhaps by tapping my drumstick on the edge of my drum, I have a very short duration pulse.
> Most of the energy contained in it will be at inaudibly high frequencies, but enough will be at audible frequencies that it will be audible as a click.
> ...



We have to be really careful here:
1) "Lost" is a strong word. You mean absolutely ALL information has been lost? Nothing, even stochastically, can be said about the peak location of this packet of data?
2) "Purely arbitrary waveforms" isn't meaningful. White noise looks pretty arbitrary, after all. Plus, you can generate LOTS (official mathematical term) of signals from adding up even a finite numbers of sine waves.
3) You seem to be assuming that you can hear the phase shifts of HF content even if you can't hear the HF content in isolation, which is quite an assumption.


----------



## amirm

KeithEmo said:


> I usually don't chime in on this group - but sometimes I just hear a need for a bit of actual balance (as opposed to listening to two soapboxes on opposite corners of the public square).
> 
> First....... "humans only hear from 20 Hz to 20 kHz".
> Are you sure?
> I read at least one quite reputable seeming report that claims that "under lab conditions some humans can actually hear sounds as low as 10 Hz".


That is true.  Unlike high frequency limit that is quite sudden and absolute, the low frequency sensitivity remains, albeit at very high thresholds.  Here is a composite of various research from the last 20 years or so on audibility of low frequency through ear alone (i.e body sensation excluded):







You can see this in Fletcher-Munson graphs:






WHile high frequency sensitivity on the right cliffs very quickly that of low frequency is sloped and keeps going lower.


----------



## KeithEmo

All of the theory about the difference being inaudible is based on the claim that we humans will be unable to audible distinguish between the original impulse and that new "sinc-like waveform".
However, oddly, I'm not aware of any studies actually done that prove that to be true.
I'm also not sure that the process of converting the impulse to the longer peak will be perfectly symmetrical in time.
(In order to tell if our short impulse occurred at the beginning of the sample, rather than the end, by looking at the longer waveform, we would have to make some assumptions about symmetry.)

I have read at least one study (it was an AES pub) that tested the audibility of "timing cues".
They seemed to have statistics to show that, when a given test sample was recorded at a 96k sample rate, then reduced to 44k, the positions in the sound stage of some sounds changed.
The authors interpreted this to suggest that we actually do notice timing cues of very short duration.

I'm not sure I have a definite opinion on this - but I am not convinced that it doesn't deserve a little more study before we start making assumptions.



RRod said:


> Not following you here, Keith. The theory says you must first bandlimit the signal, so your infinitely thin 'click' will stretch out to a sinc-like waveform and thus be amenable to sampling at the given rate. And if the filter you use to bandlimit is linear phase, then there is no time shifting of the peak.


----------



## KeithEmo

The problem is that not everything _CAN_ necessarily be "thought of as sine waves".... at least not if you don't have an infinitely long sample to work with.

Let's take a look at basic Nyquest theorem...... 
According to "the general truth", if I have three samples, they uniquely define a single sine wave (this is the justification for being able to reconstruct anything at less than half of the Nyquist frequency perfectly).
However, if you read the fine print, this is true for a continuous sine wave _ONLY_.
(Clearly, if you draw three dots on a piece of graph paper, I can draw any number of arbitrary waveforms that pass through all three.)
The normal logical response to my argument is that "if we band limit everything properly, then only one of those waveforms will fit properly".
However, again, this assumes sine waves..... after all, by definition, and filter we use will blur the signal in time.
(In other words, we CANNOT constrain my arbitrary waveform to the appropriate range of frequencies to deliver only one unique result without altering it in other ways.)

I should also point out that I'm talking in the theoretical realm here.
I will concede that it's quite possible that no such signal will ever occur in nature.
If that's true, then you might make a fair argument that "nobody will ever hear the difference".
However, from a purely scientific point of view, if a single test sample proves to be audibly changed, then you cannot claim that the difference is absolutely inaudible.

If you look very carefully, you'll find lots of other areas where the theory falls down....
For example, DACs don't actually deliver a sin-C function - they approximate one.
The theory also states that, when we band limit my impulse, the resulting waveform will extend into the infinite future and infinite past (at ever decreasing levels).
Yet, when we build an actual circuit, it never delivers an output that occurs before we input the sample.
Instead, what we get is an approximation; it is only accurate forward and back in time as far as the number of taps in our filter allow us to calculate it.
Our conversion from impulse to sin-C approximation would only be perfectly accurate if we had a filter with an infinite number of taps.... and an infinitely long signal.
(And, if the energy from my single-sample impulse is spread over twenty samples, then you're betting that our ears cannot resolve the difference between a one-sample signal and a 20-sample signal as long as the energy works out the same.) 



castleofargh said:


> the 20hz to 20khz is the estimated range for the human species, not world record numbers... many young children can hear above 20khz, geezers almost never get past 15khz. still we stick to 20-20k as a reference.  the day I learn that musicians and sound engineers manage the ultrasound content by ear, I'll be willing to bother keeping it as "the artist intended it".
> most of us here have highres media and use them when they're good. we're against false arguments pretending audibility that is consistently debunked in controlled tests. and we're against the highres industry because they seem unable to keep themselves from lying about what they're selling. I personally have nothing against fidelity, I can hear it or I can't, I keep it or I convert to lower resolution to save space on my hard drive, those are personal choices. I don't know anybody here who's trying to force the all world to use only AAC.
> 
> I don't get the infrared thing. maybe the laser was also emitting visible light? did you fact check that?
> ...


----------



## KeithEmo

1) 
I meant to suggest that _SOME_ of the information is lost... as opposed to none being lost.

2) 
In fact I can produce purely arbitrary waveforms. 
I can write a program that generates random 16 bit numbers and writes them to a file - then interpret that file as a wave file.
My point was that the idea of transforming a complex waveform into a sum of sine waves depends on a signal of reasonable length.
I'll give you a totally absurd example......
I have three consecutive samples, sampled at 44k; they are  127, 275, and 111.
Can you offer me a single sum of sine waves that is UNIQUELY associated with those three samples?
(Note that we are not assuming anything about the samples on either side of those three.)
In order to do a proper unique transform you would need to know something about what happened before and after.
(A drum hit is surely not a perfect impulse; but, equally surely, you will significantly alter the time/energy distribution when you band limit it.)

I think it would be interesting for someone to do a full run of tests.....
Let's see if people can distinguish between 44k and 96k sample rate with short runs or tone bursts of different waveforms.
And let's try it with impulses of different duration, band limited to different rates, and positioned at various points in space.

3)
I'm NOT assuming that you can definitely hear it. (I've only heard of one study - and it seemed inconclusive to me.)
However, I'm not willing to assume that you _CANNOT_ hear it either.

Let me clarify my overall sentiments on this subject......
I am not at all convinced either way.
However, I also believe in things like "margin for error".
It seems foolish to me to insist on limiting the bandwidth of a recording to 22 kHz "because we couldn't hear it if it was better".
When I print photos, I don't carefully limit the resolution to that which I'm sure will be visibly better... I use several steps better resolution "just in case".
And, when I buy a ruler, I don't make sure to buy a ruler that's no more accurate than I need.
Since both storage space and bandwidth are now so cheap, I'd rather use 96k instead of 44k, just on the off chance that 44k is less than absolutely audibly perfect.
(The cost of 24/96k vs 16/44k is really negligible when expressed as $/song.)

If, a decade from now, someone discovers that it takes a 46k sample rate to be totally indistinguishable from the original....
I'd much rather have a bunch of 96k recordings that are "unnecessarily good" instead of a bunch of 44k recordings that are "almost good enough".
I just look at it like cheap insurance.

Here's one test I'd like to see performed.
Start with an impulse - the same in both left and right ears.
Now start advancing the timing on one compared to the other.
We should expect the perceived position of the sound in space to shift.
I'd like to know the minimum shift, in microseconds, before it becomes noticeable.
Then I'd like to see if that changes at different sample rates.
When that impulse is band limited, it will be transformed into a wide shallow peak.... 
The width of the peak will depend on the sample rate - and so the band limit.
Perhaps human ears are better able to differentiate the timing differences with a particular peak shape or width.



RRod said:


> We have to be really careful here:
> 1) "Lost" is a strong word. You mean absolutely ALL information has been lost? Nothing, even stochastically, can be said about the peak location of this packet of data?
> 2) "Purely arbitrary waveforms" isn't meaningful. White noise looks pretty arbitrary, after all. Plus, you can generate LOTS (official mathematical term) of signals from adding up even a finite numbers of sine waves.
> 3) You seem to be assuming that you can hear the phase shifts of HF content even if you can't hear the HF content in isolation, which is quite an assumption.


----------



## KeithEmo

You didn't read carefully enough.....

I didn't say " a drum sample" or "a drumbeat".
I said I'm going to tap my plastic drumstick on the metal edge of the drum....
This should produce a single sharp "tick" - which I would expect to be quite short.
And, yes, if you hear a single click, you do generally get a sense of its location in space.



bigshot said:


> The problem here is that you don't really know what sort of scale a sample or a drum tap occupies. It's all small, so your mind just lumps it all together. But the duration of a drum tap is several orders of magnitude bigger than the sample- it probably spans a thousand samples. And when it comes to the way that we locate sounds in space, it's not by discerning microscopic fractions of a second like a sample. It involves room acoustics and turning our head to be able to parse directionality. Your understanding of frequencies is the same. How high of a frequency is a drum tap? How high is 25kHz as opposed to 20kHz or 15kHz or 10kHz? What do they sound like? I'll give you a hint... they don't really sound all that different because they're all around the highest octave we can hear- and the difference between 10 and 15 is more than the difference between 20 and 25. Try and define what the numbers represent and you'll understand the physics better.
> 
> As a for instance... you're talking about super audible frequencies and the audibility of sound above 20kHz. If I remember correctly, the highest frequency ever recorded as being heard by human ears is somewhere between 22kHz and 23kHz. The frequency range doubles with each octave. So the 23kHz represents 1/13th of an octave. There are seven notes in an octave, so that only represents a half of a note. Humans hear 10 octaves at best. So the difference between hearing 20kHz and hearing 23kHz is almost nothing.
> 
> Taking that one step further... The difference between 10kHz and 20kHz is one octave. 1/10th of the sound we can hear. Seven notes at the bleeding edge of human hearing. Only a couple of musical instruments produce sound in that range, and most of it is inaudible due to masking. Controlled testing has shown that the presence of super audible frequencies adds nothing to the perceived sound quality of recorded music. They have also shown that most people don't even really care if all the frequencies above 10kHz are rolled off. Super audible frequencies just don't matter.


----------



## KeithEmo

I'm trying not to take this to the level of the absurd......
The author who wrote the article on which the title of this thread is based made the assertion that "humans can't see the frequencies of light emitted by IR remote controls".
I just wanted to point out that his claim is yet another generalization which is usually, but *not* always, true.



castleofargh said:


> alright so it's borderline visible range and we compensate the lack of cells sensitive to that range with high level energy. so about the same idea as making 20khz audible for me even nowadays if I boost the volume to unsafe levels.


----------



## KeithEmo

I tend to get rather long winded... so I wanted to throw in a short answer here for those who've been following along.

Digital sampling theory says that I must first band limit the signal before sampling it.
However, the theory _DOES NOT_ say that the band limited signal will be audibly identical to the original.

A totally different set of studies tell us, pretty conclusively, that we humans are only able to detect continuous sine waves between 20 Hz and 20 kHz (with a few exceptions).
This strongly suggests that the presence or absence of sine wave components above 20 kHz will be audibly undetectable to humans.

However, specifically combining those two to claim that we cannot hear differences in digital signals above 20 kHz, even with non-sine-wave content, involves an inference.
(The specific inference is that, for purposes of human audibility, we can safely treat any digital signal as "continuous sine wave information".) 
I do not believe that this inference has been sufficiently substantiated to be accepted as "true".




RRod said:


> Not following you here, Keith. The theory says you must first bandlimit the signal, so your infinitely thin 'click' will stretch out to a sinc-like waveform and thus be amenable to sampling at the given rate. And if the filter you use to bandlimit is linear phase, then there is no time shifting of the peak.


----------



## bigshot (Oct 31, 2017)

KeithEmo said:


> I said I'm going to tap my plastic drumstick on the metal edge of the drum....
> This should produce a single sharp "tick" - which I would expect to be quite short.



I read you correctly. That sound you describe would register from attack to decay over hundreds and hundreds of samples. A sample is 1/44,000th of a second. That's 20 times faster than a camera shutter. It's a ridiculously small sliver of time. In the real world, you could never make any sound anywhere close to that fast. A little googling would solve all the questions you've got.


----------



## danadam

KeithEmo said:


> I'd much rather have a bunch of 96k recordings that are "unnecessarily good"


@amirm looked into some of them, you can see how "unnecessarily good" they are 



KeithEmo said:


> I'd like to know the minimum shift, in microseconds, before it becomes noticeable.


20 µs according to some (for 16/44, though personally I don't believe it matters).


----------



## KeithEmo

I would go with "very short", and I agree that you probably wont find many - if any - natural sounds that quick.
However, I have a high speed strobe that delivers 2 microsecond light pulses, and it's trivial to generate a one microsecond electrical pulse.

Incidentally, I took your advice and Googled some test results.....
You might find this one interesting....    
http://www.aes.org/e-lib/browse.cfm?elib=18296

It's an AES paper from 2016 titled: *A Meta-Analysis of High Resolution Audio Perceptual Evaluation*

They concluded that a small but statistically significant number of untrained listeners can in fact distinguish between high-resolution audio and CD quality.
Further, they concluded that the ability to hear the difference "improved significantly with training".
(Everyone should read it - the download is free.)




bigshot said:


> I read you correctly. That sound you describe would register from attack to decay over hundreds and hundreds of samples. A sample is 1/44,000th of a second. That's 20 times faster than a camera shutter. It's a ridiculously small sliver of time. In the real world, you could never make any sound anywhere close to that fast. A little googling would solve all the questions you've got.


----------



## RRod

KeithEmo said:


> I would go with "very short", and I agree that you probably wont find many - if any - natural sounds that quick.
> However, I have a high speed strobe that delivers 2 microsecond light pulses, and it's trivial to generate a one microsecond electrical pulse.
> 
> Incidentally, I took your advice and Googled some test results.....
> ...



So basically we have to debunk this meta-analysis? Done on a bunch of non-replicated studies? If so, there's a thread on another site that gives plenty of analysis.

Literally all anyone has to do to please me, and probably others on this thread, is to give a self-honest DBT assessment of their hi-res listening skills on a real hi-res music source on the system of their choosing. That's it.


----------



## bigshot (Oct 31, 2017)

KeithEmo said:


> I would go with "very short", and I agree that you probably wont find many - if any - natural sounds that quick.
> However, I have a high speed strobe that delivers 2 microsecond light pulses, and it's trivial to generate a one microsecond electrical pulse.



What do you plan to listen to on your home stereo? Because you'd never find anything like that in music.

By the way, I've checked out that paper in the past. "Meta analysis" means that it isn't a study... it's someone's interpretation of various studies. None of the studies cited in that meta analysis came to the same conclusion as that particular meta analysis. That paper is pretty much disregarded because it cherry picked and massaged statistics to come to the conclusion they did. If you use normal statistical analysis, it comes out that no one can perceive anything above 20kHz... which is basically what all the studies cited in that paper decided.

It's VERY common for people to decide what they want to believe is true and then look for facts to support their conclusion. The problem is, that isn't good science. It's better to assemble facts without a preconceived bias and then see what they tell you.


----------



## KeithEmo

I'm not suggesting that anyone has to debunk anything..... 
Nor am I suggesting that anyone buy anything.....
(And, since I haven't done any sort of carefully planned testing myself, I claim no definite opinion either way.)

However, from the information I've read about the various tests that have been performed, to me many of them seem to have been deeply flawed, or too limited, or badly thought out.
This meta-analysis, and the fact that it reached the exact opposite conclusion many of the individual studies reached, just reinforces my notion that the testing so far has been insufficient to prove the point either way.
We have an assumption - that there are no audible benefits to high-resolution audio.... and I simply don't believe that it's been tested thoroughly enough to believe or disbelieve that it's always correct.

I'll also admit to being somewhat puzzled as to why so many of the tests that were run were so deeply flawed.......
It seems obvious to me that a test which would provide relatively conclusive results could be devised without a lot of effort....
I agree with you that it_ SHOULD_ be easy enough to prove either way.



RRod said:


> So basically we have to debunk this meta-analysis? Done on a bunch of non-replicated studies? If so, there's a thread on another site that gives plenty of analysis.
> 
> Literally all anyone has to do to please me, and probably others on this thread, is to give a self-honest DBT assessment of their hi-res listening skills on a real hi-res music source on the system of their choosing. That's it.


----------



## 71 dB

KeithEmo said:


> However, the theory _DOES NOT_ say that the band limited signal will be audibly identical to the original.


That's actually a good point. We can't store signals 100 % identical to original. Analog is never identical. Digital is never identical. Nothing will ever be. I accept this fact of life and just enjoy the music. It's identical enough for me, gives me the _illusion_ of being 100 % identical.


----------



## KeithEmo

There is a distinct difference between "an interpretation" and a meta-analysis.
In a proper meta-analysis, you are still required to use standard statistical methods, and so the results are in fact still valid.
In fact, the idea is that, because you have a much larger sample, the results are more valid (of course assuming you don't cheat).
(Since they stated their methods, I'm sure a statistician somewhere can point out any invalid assumptions they made, or rules they violated, when making their calculations.)

Personally, I'm inclined to dislike "statistical significance" as a statement of fact.
As many here, I would much prefer actual facts, derived from the results of actual tests.
And it also seems to me that it shouldn't be that difficult to devise a test that would produce conclusive results. 
(But I find it sad that most of the tests I've seen published had glaring flaws..... like using "high-res" samples that weren't authenticated to actually be high-resolution at all.)

Unfortunately, there's also a valid issue with "the inability to prove a negative".
If some number of people can consistently hear a difference - then that proves there is a difference.
However, if any number of people consistently fail to hear a difference, we have no way to tell if no difference exists, or if a difference exists, but our test is insufficient to show it.




bigshot said:


> What do you plan to listen to on your home stereo? Because you'd never find anything like that in music.
> 
> By the way, I've checked out that paper in the past. "Meta analysis" means that it isn't a study... it's someone's interpretation of various studies. None of the studies cited in that meta analysis came to the same conclusion as that particular meta analysis. That paper is pretty much disregarded because it cherry picked and massaged statistics to come to the conclusion they did. If you use normal statistical analysis, it comes out that no one can perceive anything above 20kHz... which is basically what all the studies cited in that paper decided.
> 
> It's VERY common for people to decide what they want to believe is true and then look for facts to support their conclusion. The problem is, that isn't good science. It's better to assemble facts without a preconceived bias and then see what they tell you.


----------



## bigshot

Are you sure you aren't cherry picking? It sure seems like it.


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## pinnahertz (Oct 31, 2017)

KeithEmo said:


> I would go with "very short", and I agree that you probably wont find many - if any - natural sounds that quick.
> However, I have a high speed strobe that delivers 2 microsecond light pulses, and it's trivial to generate a one microsecond electrical pulse.
> 
> Incidentally, I took your advice and Googled some test results.....
> ...


Google some more.  There were several issues with that paper, not the least of which are that it is a study of other papers results, but not how they got them.  It also included in its data set a paper written in 1980 when there was no hi-res.  The other point to note is, for positive (able to distinguish) results they got 3.27 % better than placebo, and that's with a potentially rather "noisy" data set.   The "improved significantly with training" came from a single study (Meyer-Moran) which in itself has been questioned by both sides of the debate.

The high-res proponents will be thrilled and  shout with glee, but if this were a medical study the results would be dismissed as inconclusive.

In the 20+ years we've had 24 bit and higher than 44.1 sampling, 3.27% over placebo is the best we can do?  How disappointing!


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## bigshot

It basically took the expected statistical success rate at the far end of the bell curve and called that "training".


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## gregorio

KeithEmo said:


> I should also point out that I'm talking in the theoretical realm here.
> I will concede that it's quite possible that no such signal will ever occur in nature.



Unfortunately, you have to concede that it's quite IMPOSSIBLE that such a signal will ever occur in nature! Speaker and headphone drivers can only produce sine waves, only sine waves can travel through air and the human ear drum can only respond to sine waves. What sort of waves other than sine waves are you talking about and why would it make any difference if they cannot travel through air in the first place??

I agree that the processing required to band limit a signal is imperfect but ultimately we come down to whether those imperfections are of significant magnitude to even be reproduced in the first place and even if the answer to that question is yes, then if those imperfections are even vaguely audible.



KeithEmo said:


> (But I find it sad that most of the tests I've seen published had glaring flaws..... like using "high-res" samples that weren't authenticated to actually be high-resolution at all.)



You're talking about the M&M study but actually that wasn't a glaring flaw. The samples were chosen on the recommendation of audiophiles who stated those samples most obviously demonstrated the difference between high and standard res. The fact that some of those samples turned out not even to be hi-res in the first place not only demonstrates how easy it is to fool what audiophiles think they're hearing, even on their own systems, but was also (albeit maybe unintentionally) a good representation of the hi-res marketplace in general!

The meta-analysis paper has been heavily debunked here and elsewhere, not only is it a fine example of cherry-picked studies to obtain the desired result but the abstract and conclusion are not supported by the actual evidence presented in the paper!

G


----------



## 71 dB

Does anyone know if the companies selling high res audio products or high res audio content have tried to influence research? Just like Tobacco industry have tried to produce "scientific" research results good for their business and oil/coal companies push for "science" denying the climate change, I wouldn't be surprised if companies involved in the high res world try the same. Money talks. Money corrupts.


----------



## RRod

KeithEmo said:


> I'm not suggesting that anyone has to debunk anything.....
> Nor am I suggesting that anyone buy anything.....
> (And, since I haven't done any sort of carefully planned testing myself, I claim no definite opinion either way.)
> 
> ...



I agree that it's certainly possible that some set of people can hear some amount of HF content in some material. That paper is a bad vessel for proving the point.



KeithEmo said:


> I have three consecutive samples, sampled at 44k; they are 127, 275, and 111.
> Can you offer me a single sum of sine waves that is UNIQUELY associated with those three samples?



You cannot assume the reconstruction begins and ends with the samples: if the signal is bandlimited it cannot be time-limited and so you must assume some infinite support into which those samples fit:
.If you assume the rest of the "missing" samples are 0, then the proper reconstruction is the sum of 3 sincs with peaks determined by the sample values. Those sincs are an integral (over frequency) of all cosine waves from 0 to the bandlimit, so while we can't offer you a discrete sum we can offer you an integral.
.If you assume the rest of the missing samples are repeats of the 3, then the proper reconstruction is a sum of 3 cosine waves whose parameters are determined from the DFT of those 3 points. These 3 cosine waves are the unique solution within the bandlimit.


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## reginalb

gregorio said:


> ...You're talking about the M&M study but actually that *wasn't* a glaring flaw. The samples were chosen on the *recommendation of audiophiles...*


 [Emphasis added]

I'd say that letting audiophiles choose the track is a pretty glaring flaw.


----------



## KeithEmo

I agree with most of what you said. However, as far as flaws, I'm referring to every study I've read in detail. I don't recall a single one that started with ostensibly high-res content, confirmed that the content actually contained information above 20 kHz, confirmed that the playback system (including the speakers) could actually deliver that into the room, used a reasonable sample size, and hit a few other points that would qualify it as "a well designed test". Measurement microphones are available with a response that extends well above 20 kHz. It would have been simple enough to rent one, and document that there was actually content above 20 kHz  in the air of the room which the participants might attempt to hear, yet even that simple confirmation was lacking from every test whose protocol I read. (Imagine testing a new drug without first confirming that your "active sample" actually contained any of the drug you were purportedly testing!)

Heck, if someone really wants to claim that "only bats and dogs can hear it", then let's start by bringing in a couple of bats to confirm that they CAN hear it before presenting it to our human subjects.  

However, all kidding aside, let's start with:
- here's the content we chose as samples of "high resolution recordings"
- here's a spectrum analysis of that content showing  that it has a frequency response that extends above 20 kHz
- here's a spectrum analysis taken from a microphone at the listening position - showing that the speakers and the rest of the system are reproducing it accurately
- and here are the test results

Let me reiterate that I make no claim either way.... I just don't think any sort of what I would consider exhaustive or conclusive testing was ever done.
(And, no, making a generalization based on a single type of test - steady state sine waves - really isn't very conclusive.)

As for the M&M study...... glaring flaw?
Hmmmmm........
I don't care what audiophiles said or claimed.
I would consider testing whether people can hear something, without first confirming that it's there to hear, to be a rather glaring flaw.
(People are subject to all sorts of odd biases and preconceptions - so does it make any sense to use a certain test signal based on human recommendations? Really?)

I also need to make an aside comment about how "tests whose results we agree with may have a few unresolved issues" but "tests whose results we disagree with get debunked".
That meta-test was based on the same data, from the same flawed tests, that everyone wants to use as "proof" of the opposite claim.
Therefore, unless there are actual flaws in their statistical methodology, it is no more or less valid than the tests on which it was based.
(And, if the data is that inconclusive that it can be interpreted either way, then I would not call either set of results conclusive.)

Of course, it might turn out that only a few people can hear the difference....
or that only a few speakers can reproduce it...
But that's the subject of ANOTHER test.

And, if we determine that, THEN we can have an intelligent discussion about "whether it's important enough to matter if only 1% of people will ever hear it" or "if only three models of speakers can reproduce it".
(My suggestion would be to run the initial tests using electrostatic headphones - thus at least ensuring that we're delivering the test signal to the test subjects.)

I also think you have an excellent point about "the market".
I'm pretty sure that nobody actually wants to know thee truth.
The people whose income is based on selling high-res audio content have obviously convinced a significant segment of the market that it has value.
Therefore, they have no need for test results to prove it, and no reason to spend money on tests that MIGHT prove them wrong.
(Also note that, if it turned out that a few people, but only a few, could hear the difference, the result would be "scientifically significant" but a marketing disaster.) 
And, of course, there's no real motivation for anyone to expend the money and effort necessary to actually prove the negative with a real full scale test.
(Nobody makes money by convincing us to keep our current CDs and not bother to buy the new high-res remaster.)



gregorio said:


> Unfortunately, you have to concede that it's quite IMPOSSIBLE that such a signal will ever occur in nature! Speaker and headphone drivers can only produce sine waves, only sine waves can travel through air and the human ear drum can only respond to sine waves. What sort of waves other than sine waves are you talking about and why would it make any difference if they cannot travel through air in the first place??
> 
> I agree that the processing required to band limit a signal is imperfect but ultimately we come down to whether those imperfections are of significant magnitude to even be reproduced in the first place and even if the answer to that question is yes, then if those imperfections are even vaguely audible.
> 
> ...


----------



## 71 dB

KeithEmo said:


> (Nobody makes money by convincing us to keep our current CDs and not bother to buy the new high-res remaster.)


That's why it is iffy to believe anyone "making money" in the business. Some companies "remaster" their 50 years old dusty tapes over and over again increasing samplerate and bit depth every time milking dumb people who can't separate placebo effect from real audible differencies. Other companies produce new exciting music nobody has on any format. It's about choosing your business strategy. If you have new music to sell you don't need high-res because you sell music instead of bits.


----------



## KeithEmo

That was sort of my point.
In theory you're entirely correct - but in practice you are not.

According to the math, the reconstruction indeed doesn't "begin and end with the samples".
However, _IN PRACTICE_, there will be no output before the input is sent to the circuit.
Likewise, output will not be produced forever; at some point someone is going to switch it off.
(When we use filters, we pretend to "start before the beginning" by delaying the entire signal.... but the interval over which we do that is limited by our number of filter taps.)
Therefore, in practice, we never "properly and completely reconstruct the signal"..... all we do is to create an approximation with some arbitrary accuracy.
(And then we're right back to the original question..... "Is it close ENOUGH that we can't tell the difference?")

The problem, as I see it, is that our brains can do all sorts of impressive types of data manipulation and interpretation.
We all know what the waveform will look like if I tap a bell.... and we all know what it sounds like.
(And note that, while our theoretical reconstruction may extend back to t=minus infinity, the bell only started producing sound AFTER it was hit.)
And, ignoring all that, our brains will pick out the leading edge of the waveform and output a "perceived physical location" for that bell that's usually quite accurate.
It also demonstrates that our brain/ear combination seems to be somewhat more than "a simple steady state spectrum analyzer".
The fact is that "3D stereoscopic video" doesn't convey real 3D well at all (luckily we're willing to overlook the botched cues if the important ones are more or less correct).
Likewise, walk by the window of a room with music playing inside, and you can virtually always tell whether it's a stereo system or live musicians.
(So clearly the music reproduction system is FAILING to get SOMETHING right that our brains can pick out... my guess would be that it involves directionality and motion/triangulation.)

To go back to our bell example......
We all know what the envelope of a bell-hit waveform looks like (a sine wave that jumps to full volume within one or two cycles, then dies down gradually).
We also agree that, once we band limit it, it will be spread out in time.
And, depending on the filter we use, some of that spread will be backwards in time - in the form of "pre-ringing".
(And we must include rather than delete it or the overall energy content will not be correct.)
However, that pre-ringing changes the overall envelope of the waveform of our bell hit; specifically it makes the leading edge less sudden.
And, if our brains use the relative arrival times between the sharply rising beginnings of those envelopes to calculate position, then isn't making them more gradual (by adding a pre-ringing "slope") going to make those calculations less precise?
Could this be why a lot of people claim to find signals reconstructed by filters that introduce pre-ringing to "sound less natural"?

I think there's still an awful lot of research left to be done here.



RRod said:


> I agree that it's certainly possible that some set of people can hear some amount of HF content in some material. That paper is a bad vessel for proving the point.
> 
> You cannot assume the reconstruction begins and ends with the samples: if the signal is bandlimited it cannot be time-limited and so you must assume some infinite support into which those samples fit:
> .If you assume the rest of the "missing" samples are 0, then the proper reconstruction is the sum of 3 sincs with peaks determined by the sample values. Those sincs are an integral (over frequency) of all cosine waves from 0 to the bandlimit, so while we can't offer you a discrete sum we can offer you an integral.
> .If you assume the rest of the missing samples are repeats of the 3, then the proper reconstruction is a sum of 3 cosine waves whose parameters are determined from the DFT of those 3 points. These 3 cosine waves are the unique solution within the bandlimit.


----------



## KeithEmo

The only catch seems to be that there doesn't seem to be all that much new music that a lot of people find worth buying.
(And it's such a shame to have all that old music sitting there, gathering dust, when there's got to be some way to make a buck off of it.)

I would also point out that periodic remixes and remasters is not at all a new thing..... this is just the latest iteration of an ongoing tradition of periodically "reissuing" anything that's still popular.

On one hand, perhaps the new streaming services will allow them to earn money "per listener" instead of "per song".
Of course, the down-side is that it reduces the incentive to ever produce new and interesting music.



71 dB said:


> That's why it is iffy to believe anyone "making money" in the business. Some companies "remaster" their 50 years old dusty tapes over and over again increasing samplerate and bit depth every time milking dumb people who can't separate placebo effect from real audible differencies. Other companies produce new exciting music nobody has on any format. It's about choosing your business strategy. If you have new music to sell you don't need high-res because you sell music instead of bits.


----------



## KeithEmo (Nov 1, 2017)

I honestly doubt it.

Every single bit of marketing I've ever seen for high-res content is based on the assumption that it's better (presumably that means it sounds better).
Some people obviously believe this and some don't... but I don't recall ever seeing claims of proof presented in a marketing context.
(I mean an ad saying "23% of audiophiles prefer our "brand B" files over those others").
In order to sell more product, research would have to show a significant difference.
(They wouldn't sell more by "proving" that 5% of people heard a slight difference on the occasional song.)
Current markets, especially for luxury products, are based more on perception than fact....
In other words, a dozen Facebook posts from enthusiastic audiophiles will sell more copies than a well thought out and executed academic study.... because more people will read them.
(And studies are both expensive and likely to either prove the claims wrong altogether or support them to a very minor degree.)
I don't see how actual research would help to sell more product (unless they could show that a lot of people hear a significant difference).

If you look at your examples, you'll note that the tobacco companies didn't start their campaign until it was necessary for damage control.
They didn't start selling cigarettes by publishing reports about health benefits; the reports were published to counter claims of negative effects.
Likewise, oil companies didn't start selling their product "because it was good for the environment" (unless they could compare it to older options like coal).
Rather, they are trying to convince people that they're not doing anything too dangerously wrong.
In the sort of industry where people actually WANT the product to begin with, scientific studies serve limited marketing purpose.....

In short, I don't think you'll see anyone sponsoring studies either way...
The only way it will happen will be if the vendors need to "counter" a real widely publicized study that shows that high-resolution files are totally useless.
(And that will never happen.... because nobody has any incentive to pay for such a study.)

A similar, but slightly different, situation exists with speaker and interconnect cables.
In that case, the companies who sell expensive cables have every reason to sponsor tests that purport to prove that their products are better.
However, the companies who sell equally good $10 cables have neither the incentive nor the budget to sponsor tests proving the opposite.
(They'd have to sell a lot more $10 cables to make back what it costs; and they probably don't lose very many customers to those $10k interconnects anyway.)

In this case, people who are convinced they hear a difference will continue to purchase high-res downloads.
(They also offer the added value that some re-masters just sound better because of improved mastering... whether they happen to be high-res or not.)
And people who either don't hear a difference, or resent yet another reissue, will not.
But a published study is unlikely to change very many minds either way.
Which provides little incentive to sponsor one.

I would also note that there would have been little or no incentive to do the testing that proved that tobacco was dangerous if there hadn't been the possibility of huge settlements
to justify funding them. For example, if the tobacco companies were all out of business, and there was nobody to sue, then there would have been little incentive to prove them negligent.
In this current situation, we don't see "non-high-res vendors" trying to reclaim their customers from competitors selling high-res versions.
It's easier to add high-res files to your list of options than to try and convince people not to buy them.



71 dB said:


> Does anyone know if the companies selling high res audio products or high res audio content have tried to influence research? Just like Tobacco industry have tried to produce "scientific" research results good for their business and oil/coal companies push for "science" denying the climate change, I wouldn't be surprised if companies involved in the high res world try the same. Money talks. Money corrupts.


----------



## 71 dB

KeithEmo said:


> The only catch seems to be that there doesn't seem to be all that much new music that a lot of people find worth buying.


Music isn't marketed to people. The biggest popstars are, but that's it. You don't see adds for the newest ECM releases on telly, do you?



KeithEmo said:


> (And it's such a shame to have all that old music sitting there, gathering dust, when there's got to be some way to make a buck off of it.)


If people knew high-res makes no difference there wouldn't be a way to make buck off of it. So, they'd have to do something else, like produce new music.



KeithEmo said:


> I would also point out that periodic remixes and remasters is not at all a new thing..... this is just the latest iteration of an ongoing tradition of periodically "reissuing" anything that's still popular.


Yeah, and it can be good for new audiences. When I discovered King Crimson a decade ago, the 30th Anniversary editions of the albums were out. Now there's the 40th Anniversary editions I believe, but I am not upgrading.


----------



## Whazzzup

I bought some hi res just to check it out. Regardless what i thought, Im back at I tunes for half the cost and a huge library of current releases just makes sense. No they are not twice as good iTunes, however a few particular albums sounded fantastic from my dedicated network server.


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## RRod (Nov 1, 2017)

KeithEmo said:


> Therefore, in practice, we never "properly and completely reconstruct the signal"..... all we do is to create an approximation with some arbitrary accuracy.
> (And then we're right back to the original question..... "Is it close ENOUGH that we can't tell the difference?")


Right, we're never bandlimited. But that just means we don't go down to -infinity dB at Nyquist. We can do stuff like go down to -140dB from 0.95Ny to 1.05Ny, so once again we're back into the real world of audibility as you say. But we shoudn't act like we have just no idea where we are violating theory.



KeithEmo said:


> We all know what the waveform will look like if I tap a bell.... and we all know what it sounds like.
> (And note that, while our theoretical reconstruction may extend back to t=minus infinity, the bell only started producing sound AFTER it was hit.)


We're not just recording a bell, we're also recording its environment.



KeithEmo said:


> Likewise, walk by the window of a room with music playing inside, and you can virtually always tell whether it's a stereo system or live musicians.
> (So clearly the music reproduction system is FAILING to get SOMETHING right that our brains can pick out... my guess would be that it involves directionality and motion/triangulation.)


The speakers in the room at not set up for proper imaging at the windows. You are outside of the virtual environment looking in.



KeithEmo said:


> To go back to our bell example......
> We all know what the envelope of a bell-hit waveform looks like (a sine wave that jumps to full volume within one or two cycles, then dies down gradually).
> We also agree that, once we band limit it, it will be spread out in time.


It's already spread out in time, and the amplitude of the ringing will depend upon the samples with which the filter is convolved. Masking and other audibility issues must be considered here.



KeithEmo said:


> However, that pre-ringing changes the overall envelope of the waveform of our bell hit; specifically it makes the leading edge less sudden.
> And, if our brains use the relative arrival times between the sharply rising beginnings of those envelopes to calculate position, then isn't making them more gradual (by adding a pre-ringing "slope") going to make those calculations less precise?


The problem with positioning arguments is that if I go through the whole rigmarole of measuring your HRTF and making a virtualized sound environment therefrom, and do all this at 44.1kHz, you will get much better positioning than a plane-jane stereo reproduction of the bell or two bells at 96kHz.



KeithEmo said:


> Could this be why a lot of people claim to find signals reconstructed by filters that introduce pre-ringing to "sound less natural"?


Or they are socially encouraged to claim that's what they hear. Once again, how about these people do some blind tests?


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## bigshot (Nov 1, 2017)

71 dB said:


> Does anyone know if the companies selling high res audio products or high res audio content have tried to influence research? Just like Tobacco industry have tried to produce "scientific" research results good for their business and oil/coal companies push for "science" denying the climate change, I wouldn't be surprised if companies involved in the high res world try the same. Money talks. Money corrupts.



Peer review pretty much negates that. That's why most of the pseudo scientific sales pitch is published as advertising on web sites. People pick it up and start parroting it as if it actually is a peer reviewed study.



KeithEmo said:


> The only catch seems to be that there doesn't seem to be all that much new music that a lot of people find worth buying.



Perhaps you aren't looking for it. There's a lot of great music, it's just mixed in with a lot of not great music. That takes effort, but I find that effort spent on uncovering great music is much more productive than effort spent on chasing down theoretical sound quality.


----------



## gregorio

KeithEmo said:


> [1] I also need to make an aside comment about how "tests whose results we agree with may have a few unresolved issues" but "tests whose results we disagree with get debunked".
> [2] That meta-test was based on the same data, from the same flawed tests, that everyone wants to use as "proof" of the opposite claim.
> Therefore, unless there are actual flaws in their statistical methodology, it is no more or less valid than the tests on which it was based.



1. True to an extent but the tests I disagree with are those whose results are contrary to known and demonstrated scientific facts.
2. Again true to an extent. The Theiss study for example demonstrated no ability to discriminate but in a supplementary non-ABX experiment with 3 of the subjects, a very high ability to discriminate was demonstrated. There are several reasons to strongly dispute this supplementary experiment, not least that it wasn't double blind but the meta-analysis only includes the supplementary experiment results, not the main ABX experiments. And, if we're going to eliminate the M&M study on the grounds that some of the samples were not Hi-Res, as the meta-analysis does, then how come there are other studies included which had no Hi-Res samples at all because they were well before hi-res was even invented?

The problem we have here is also one of time/history. Is it possible to discriminate 96/24 from 44/16? Yes, it is! BUT, using the same criteria it's possible to discriminate 44/16 from 44/16! In it's early days, 44/16 had it's issues, today's 44/16 is quite different. If it were possible to do a direct comparison, I'd be very surprised if we could not fairly reliably discriminate early 44/16 from today's 44/16. The same principle applies to 24/96. There was a period of 5 or so years when it was entirely possible to differentiate between 24/96 and 16/44. A large number of plugin processors operated significantly better at 96kHz that at 44.1kHz (easily determined in DBTs) and in some cases this is still the case, some: Compressors, limiters, soft-synths, guitar amp/cab and other modelling plugins for example. So, 24/96 can be audibly different/better than 16/44 then, case closed! Hang on, not so fast! Filter implementation has improved significantly over the last 20 years, so has upsampling/downsampling conversion, due to far more available processing power and improved software algorithms. Today (and for nearly a decade) a plugin which benefits from a higher sample rate can simply up-sample to that rate, process and then down-sample again, with no audible loss/artefacts. There's no longer any audible benefit to actually recording or outputting the audio files at 96kHz, and 24/96 can no longer be discriminated from 16/44, unless a DAC/DAP manufacturer deliberately (or inadvertently) builds a difference into their design.  

G


----------



## frodeni

KeithEmo said:


> ...We all know what the envelope of a bell-hit waveform looks like (a sine wave that jumps to full volume within one or two cycles, then dies down gradually).
> We also agree that, once we band limit it, it will be spread out in time. ..
> 
> Signal processing is simply not my field of expertise. I simply do not know of the effect of this processing. Sorry, it is probably just me, but I am at least one guy, that do not know much about this effect. Do you have some simple explanation of this, with illustrations, that could help med get an idea of how this works, without the need for a ton of math? Also, what makes 16/44.1 "band limiting"? Does this affect all freq ranges?
> ...


----------



## gregorio

frodeni said:


> [1] Still, using my PC or laptop as a USB source, is like night and day, for my setup. Which makes hardly any sense, but which is the case. The dynamic range of 16bit is far wider than my listening level, and given the sampling rate, it is closing at twice what is supposed to be needed. This makes no sense to me, and something is lacking by our understanding of this.
> [2] A shared understanding of how this manifest itself as clearly detectable sonic traits, cannot be achieved by blind tests. Hardly any understanding may be reached by a blind test.
> [2a] If someone passes or fails a blind test, so what?



1. So if something makes no sense to you then mankind is lacking understanding? If something doesn't make sense to you maybe it's because you simply don't know or understand something which mankind/science does? There are at least two explanations I can think of which make plenty of sense and do not contradict science.
2. Of course it can. A blind (preferably DB) test can tell us if that/those "clearly detectable sonic traits" are in fact audibly detectable in the first place and if not, the shared understanding is effectively a delusion, if it is still detectable, then we've eliminated a whole set of potential causes and significantly narrowed down the search for the real answer.
2a. No big deal but if everyone who takes it does, as the sample size increases so does our confidence in the result accurately extrapolating to everyone else.

G


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## KeithEmo

I agree..... with some qualifications.

We have plenty of scientific data about the audibility of continuous sine waves by human beings..... 
However, I'm not convinced that the data for continuous sine waves can necessarily be generalized to other types of content. 

I think you also have an excellent point about the history factor.
The technology has progressed a lot in recent years.
Perhaps it makes sense to ignore or discard _ALL_ results from the past and perform some new tests using current technology.

Another flaw in many previous tests was that they were limited to certain specific pieces of equipment.
With that limitation, all you can claim to have proven is that the differences were or were not audible using that particular equipment.
Perhaps the differences are clearly audible with a certain speaker and totally obscured by another; perhaps they're only audible when a certain filter type is used by the DAC, or when a certain ADC is used to produce the test content.
(And, no, allowing a bunch of audiophiles to select their own equipment, based on their own preconceived notions, does not prove that equipment has been chosen that will actually reveal any differences present.) 
If we're attempting to prove or deny a negative, then at least a good cross section of equipment must be tested.

I've found the attention to detail disappointing in ALL of the previous tests I've read about....

Here would be my minimum requirements:
- present detailed information about the provenance of all test tracks used
- present detailed information about how the different test versions were derived 
   (I would want both samples to be derived from the same master copy, of higher resolution, down-sampled using the same software and filter options for both samples) 
- present a spectrum analysis of each track to show that it does in fact contain content above 20 kHz
- present a similar spectrum analysis, taken with a calibrated microphone at the listening position, to confirm that the "extended content" is actually making it to the listener's position
- present details about the equipment used (DAC type and brand, filter options chosen, speakers, amplifiers, etc.)
- repeat the test using a variety of equipment (to avoid situations where one particular piece of equipment masks the differences between different sample rates - or treats them differently).
- present details about the test environment (quiet, noisy, absorptive room, reflective room)
- perform the test with several different filter options (maybe you can hear the difference with an apodizing filter, but not with a flat-phase brick wall filter, or vice versa)
- I would want to include both headphones and speakers (I find electrostatic headphones especially revealing of small differences that may not otherwise be obvious) 

This is all pretty basic stuff "when designing a credible scientific test"....
If we want to claim a general conclusion, then we need to include enough variations to reasonably do so.



gregorio said:


> 1. True to an extent but the tests I disagree with are those whose results are contrary to known and demonstrated scientific facts.
> 2. Again true to an extent. The Theiss study for example demonstrated no ability to discriminate but in a supplementary non-ABX experiment with 3 of the subjects, a very high ability to discriminate was demonstrated. There are several reasons to strongly dispute this supplementary experiment, not least that it wasn't double blind but the meta-analysis only includes the supplementary experiment results, not the main ABX experiments. And, if we're going to eliminate the M&M study on the grounds that some of the samples were not Hi-Res, as the meta-analysis does, then how come there are other studies included which had no Hi-Res samples at all because they were well before hi-res was even invented?
> 
> The problem we have here is also one of time/history. Is it possible to discriminate 96/24 from 44/16? Yes, it is! BUT, using the same criteria it's possible to discriminate 44/16 from 44/16! In it's early days, 44/16 had it's issues, today's 44/16 is quite different. If it were possible to do a direct comparison, I'd be very surprised if we could not fairly reliably discriminate early 44/16 from today's 44/16. The same principle applies to 24/96. There was a period of 5 or so years when it was entirely possible to differentiate between 24/96 and 16/44. A large number of plugin processors operated significantly better at 96kHz that at 44.1kHz (easily determined in DBTs) and in some cases this is still the case, some: Compressors, limiters, soft-synths, guitar amp/cab and other modelling plugins for example. So, 24/96 can be audibly different/better than 16/44 then, case closed! Hang on, not so fast! Filter implementation has improved significantly over the last 20 years, so has upsampling/downsampling conversion, due to far more available processing power and improved software algorithms. Today (and for nearly a decade) a plugin which benefits from a higher sample rate can simply up-sample to that rate, process and then down-sample again, with no audible loss/artefacts. There's no longer any audible benefit to actually recording or outputting the audio files at 96kHz, and 24/96 can no longer be discriminated from 16/44, unless a DAC/DAP manufacturer deliberately (or inadvertently) builds a difference into their design.
> ...


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## KeithEmo

My point there was simply that, if our brains use characteristics of the envelope to determine location, and we then change the envelope, then we are likely to alter the results.
If we add pre-ringing to that recorded bell sound, we will alter its original envelope, which begins rather sharply, into one that ramps up more gradually.
I personally don't know how much that would affect how our brain interprets the location of the "real bell" - but maybe we need to find out.
(In analogous electronic test equipment, spreading out the envelope of a pulse does in fact make it more difficult to resolve the beginning time accurately.)

Recent research has shown that a lot of the analyses performed by our brains is remarkably complex - and often involves multiple parallel processes.
For example, when you see something, and move your eyes to look directly at it, it "seems" as if you recognize something and move your eyes towards it (as a single process).
However, according to recent research, your brain senses motion and "starts your eyes moving in the right direction" _BEFORE_ identifying the target or even precisely locating it.
Once your eyeball is already moving, results from other sections of the brain, which are more precise but take longer to complete, either refine the motion command, or countermand it.

Therefore, it's not unreasonable to wonder if, while your brain is identifying the pitch you're hearing by using your ear like a spectrum analyzer...
another section has already tentatively identified the approximate location of the sound based on the beginning edges of the sound envelopes.
(Which might suggest that altering the shape of those envelope leading edges might affect that part of the result.)

Note that I said MIGHT.......
I don't know if this will turn out to be true or not; but I'm not willing to rule it out until it's actually been tested.
(Note that this wouldn't be especially difficult to test - I just don't think it's been done yet.)

And, yes, this would call for a lot of testing.
For example, we could ask people to locate a bunch of objects in the sound field, and rate their accuracy ("point to where it sounds like the violin is coming from").
We could then ask them to repeat the test with test samples recorded at various sample rates and see if their accuracy is the same for each - or not.
If a given person can locate various instruments with greater accuracy when higher sample rates are used, then it would prove that information is lost when reducing the sample rate.
If not, then it would prove that, at least under those particular conditions, it really doesn't matter.



RRod said:


> ...................................
> 
> It's already spread out in time, and the amplitude of the ringing will depend upon the samples with which the filter is convolved. Masking and other audibility issues must be considered here.
> 
> ...................................


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## bigshot

I never cease to be amazed at how much time people spend thinking about the inaudible. I guess it's fun for them.


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## RRod

@KeithEmo If you get me a free set of HRTF measurements out of it, I'll participate in that test.


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## frodeni

gregorio said:


> 1. So if something makes no sense to you then mankind is lacking understanding? If something doesn't make sense to you maybe it's because you simply don't know or understand something which mankind/science does? There are at least two explanations I can think of which make plenty of sense and do not contradict science.
> 2. Of course it can. A blind (preferably DB) test can tell us if that/those "clearly detectable sonic traits" are in fact audibly detectable in the first place and if not, the shared understanding is effectively a delusion, if it is still detectable, then we've eliminated a whole set of potential causes and significantly narrowed down the search for the real answer.
> 2a. No big deal but if everyone who takes it does, as the sample size increases so does our confidence in the result accurately extrapolating to everyone else.
> 
> G



You need to read carefully what you just wrote. If want to get a paper rejected, this is all you need to write. This will get you flunked, on its own. People do get the blunder, right?


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## castleofargh

bigshot said:


> I never cease to be amazed at how much time people spend thinking about the inaudible. I guess it's fun for them.



I also feel like I'm an alien sometimes. I hate loud music(that's on me), so realistically when I end up with a noise floor 35 to 40dB below music(computer, street, annoying people...), that's pretty much the hifi moment of my day. and distortions tend to be in that area too, not that I can tell unless it really goes high.  then band limiting, 16khz is but a quiet remnant of sound, 17khz is gone at my listening level(in fact I also have a very local drop around 7.4khz). but just to be able to tell that 16khz is my limit, I have to be careful not to use most of my IEMs as they roll off even faster than I do. 
 I feel boxed inside a small frequency range and a small dynamic range for what I struggle to call fidelity with a straight face. and that's before looking at all the issues related to transducers and HRTF with headphones. 
so when I read comments about the better soundstage with a gazillion taps, people crying about the lack of dynamic "caused" by redbook, or time smearing, temporal blur and other "hi I'm Barry Allen and I'm the fastest audiophile alive" references, they all feel divorced from my audio reality or the very concept of magnitude.


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## bigshot

More dynamics and frequencies isn't audiophile... More BALANCED dynamics and frequencies are.


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## castleofargh

I thought "more" was always audiophile. ^_^


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## frodeni

castleofargh said:


> I also feel like I'm an alien sometimes. ...



I feel human.


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## jimmers (Nov 2, 2017)

castleofargh said:


> I also feel like I'm an alien sometimes. I hate loud music(that's on me), so realistically when I end up with a noise floor 35 to 40dB below music(computer, street, annoying people...), that's pretty much the hifi moment of my day. and distortions tend to be in that area too, not that I can tell unless it really goes high.  then band limiting, 16khz is but a quiet remnant of sound, 17khz is gone at my listening level(in fact I also have a very local drop around 7.4khz). but just to be able to tell that 16khz is my limit, I have to be careful not to use most of my IEMs as they roll off even faster than I do.
> I feel boxed inside a small frequency range and a small dynamic range for what I struggle to call fidelity with a straight face. and that's before looking at all the issues related to transducers and HRTF with headphones.
> so when I read comments about the better soundstage with a gazillion taps, people crying about the lack of dynamic "caused" by redbook, or time smearing, temporal blur and other "hi I'm Barry Allen and I'm the fastest audiophile alive" references, they all feel divorced from my audio reality or the very concept of magnitude.


Me too,
My "Audiophile" What moment was when I first read a DAC spec, 32bit at 384k samples per second.
So I went and bought a 16 bit ladder (mainly R2R) DAC for my Redbook rips and gave up on stupid numbers and went back to enjoying my music.


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## theveterans

jimmers said:


> Me too,
> My "Audiophile" What moment was when I first read a DAC spec, 32bit at 384k samples per second.
> So I went and bought a 16 bit ladder (mainly R2R) DAC for my Redbook rips and gave up on stupid numbers and went back to enjoying my music.



Same here. Redbook and 320 kbps MP3 sounded amazing on the 16-bit R2R ladder-on-a-chip DAC that I have.


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## bigshot

Ears are what matters. Not theories.


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## gregorio (Nov 2, 2017)

KeithEmo said:


> If we add pre-ringing to that recorded bell sound, we will alter its original envelope, which begins rather sharply, into one that ramps up more gradually.
> I personally don't know how much that would affect how our brain interprets the location of the "real bell" - but maybe we need to find out.
> 
> Therefore, it's not unreasonable to wonder if, while your brain is identifying the pitch you're hearing by using your ear like a spectrum analyzer...
> ...



An interesting and not at all unreasonable point. However, it's an invalid point and there are several reasons for me stating it's invalid:

1. I agree that pre-ringing effectively changes the envelope but for that pre-ringing to have any effect, it must be audible/detectable. Let's say hypothetically that it may not be consciously audible but is detectable, in terms of the brain's interpretation of location. If the location of the bell is not where I want it to be, I (as a mix engineer) can simply change the location, if it's not where I expect it to be, I would typically investigate why. Never have I found pre-ringing to be the cause of a bell (or any other sound) not being where I expect it to be.

2. Rather ironically, the 1997 Theiss study I mentioned previously (Phantom source perception in 24bit @ 96kHz digital audio) set out to test exactly what you are suggesting. As I mentioned, there was a supplemental test on general perceived sound quality performed under less formal circumstances and it's this test which is frequently quoted by those who have a hi-res agenda but the main experiments were formal DBX tests designed specifically to test localisation and resulted in the conclusion that: "_Analyses of the data showed that the hypothesis that localization accuracy improves with higher sampling rates above the professional 48kHz standard has to be rejected_". (I linked to the paper above so you can read the details for yourself.)

3. As is frequently the case, the actual reality of the behaviour of sound and the practical realities of recording it are ignored. Very rarely (and pretty much never for a commercial music release) would a single violin be recorded with a single microphone placed a few inches from the instrument. The transients and frequency content of instruments are however typically/always measured and quoted this way. What an instrument actually sounds like from such close proximity is different, often vastly different, from what is expected and what would be heard by the audience. We've got absorption and reflections to consider, which results in very significantly different transients, freq content and dynamic range from what we would measure just a few inches away. This is with with actual live acoustic sound, if in addition we factor in mic response, timing differences between mics and more than one violin, we've got transients smeared all over the place and by all over the place I'm talking tens of milli-seconds up to seconds, not the few micro-seconds which can be detected with test signal! And, this is assuming any transients still even exist from a listening position in the audience and in many/most cases they won't! All this applies to any instrument/sound, even a snare drum rimshot, although with a rimshot there would typically still be a transient, just a very time smeared and different transient from the one created.

One of the difficulties facing us from the position of hobbyists, is the available information. The research and knowledge which covers most of our hobby is typically not led by independent scientific research, it's led by industry and as such is often proprietary and not available. For example, the start of digital audio is arguably the Nyquist Theory in 1924 which actually belonged to AT&T but they allowed it to be published as a scientific paper. However, most of the testing, data and research is not published science and even when there is independent scientific research, it's often/sometimes lagging many years behind industry research and sometimes lacking crucial factors. Then there's people like me, who actually use the results of that industry research day in and day out. For example, I studied, critically compared, then bought and was using greater than 16 bit technology every day, a good 8 years before >16bit even became available to consumers. Another example, the k-weighted filter used in loudness normalisation was the result of a lot of rigorous testing (perceptual DBTs) by the ITU's members, such as the BBC, ORTF and many, many others, but none of that research is published anywhere as far as I'm aware and the ITU specifications which resulted from them have since been modified, after people like me used them every day and discovered the deficiencies/loopholes. On top of this, while some front line companies are effectively unbiased, some industry organisations represent a membership which includes powerful manufacturers and distributors and are not in practice always entirely unbiased, the AES being an example. And finally, as you have mentioned, there is often great financial incentive to fund and publish research which demonstrates a positive result (that hi-res provides a tangible benefit for example) but relatively little or none at all to demonstrate a negative. All of this results in a knowledge landscape which is often extremely difficult to navigate and therefore relatively easy to abuse!

G


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## bigshot

When people say things like "Let's say for the sake of argument..." or "Assuming that something that isn't audible may be unconsciously perceived..." for me, it's like stepping over the line into fantasy land.

I believe in the inaudibility of inaudible sound.


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## KeithEmo

I basically agree with everything you've said..... but I would add a few qualifications.

First, I would point out that, to an extent, there are two different discussions going on here. One is about whether there is _ANY_ audible difference between high-resolution and "ordinary" recordings; the other is about whether there is a _SIGNIFICANT_ audible difference. As a mix engineer, you undoubtedly listened to what you were mixing very carefully. However, I'll bet you didn't ask a dozen people to listen to the high-res and standard version of each, on a half dozen different DACs, and a dozen different brands of speakers, and ask each to tell you exactly where in space the bell seemed to be located - and whether it seemed to occupy a distinct location or its apparent location was slightly blurred. Therefore, I would certainly support your assertion that "you've never known it to make a significant difference" - but that falls short of the scientific assertion that "no human being will be able to audibly discern a difference".

I suspect that some people are reading this as a scientific inquiry, while others are reading it as a _practical _discussion about what's worthwhile (I'm in that first category). In terms of your last point, I suspect you're right, and very little - if any - music would actually allow for such a distinction. However, speaking as a scientist, if a single human, using a single test tone, can reliable hear a difference, then we must concede that "there is an audible difference"..... and the fact that it is so rarely audible that it doesn't justify spending extra for one type of recording or the other is a separate discussion altogether. 

Second, considering how the state of the art has changed, I'm not sure I would consider _ANY_ study performed with 1997 vintage A/D converters and DACs to be especially definitive.
Twenty years is a long time, and the technology really has changed significantly....  and a lot of equipment from back then really was audibly inferior to much of what we have now.

I entirely agree with you that the differences are, at most, very small.... certainly much smaller than many other factors.... and so quite probably are insignificant to most people.
(I would also comment that, considering how good a well mastered CD _CAN_ sound, it's sort of sad that so many recent ones sound so bad.)
And, no, I'm not personally convinced that I could hear the difference between a really well mastered 16/44k file and an equivalent 24/192k version of it.

I also agree that the information currently available is confusing, and seemingly often deliberately misleading...... 
Starting with scope images showing that different DACs deliver different outputs when presented with a totally invalid single-sample transient test signal.
I've been considering this as a scientific discussion rather than a practical one. 
(I'm betting that I could in fact think up a test signal where the difference might be audible - but it might not at all be representative of real live music).
However, I would agree that the differences have, at least so far, not been shown to be "significant - in a practical sense".
(I would also agree that most of the differences people claim to hear seem likely to have been based on bias rather than on reality.)

I would also note that it can be very difficult to perform what I would consider to be "fully detailed tests" on subjects like we're discussing.
I can start by trying to compare the 24/192k version of a certain album to the 16/44k version.
However, if I treat the 24/192k version as my master, then I must perform a sample rate conversion to generate the 16/44k version - which introduces another processing step - which includes filtering.
And, if I instead obtain both versions from someone else, then I have to wonder whether they have applied exactly the same parameters to both.
And I also have to wonder if the particular DAC I've chosen happens to perform slightly differently at different sample rates - for whatever reasons.
To be honest, this makes me doubt that anyone will ever bother to perform a rigorous scientific analysis of the subject.



gregorio said:


> An interesting and not at all unreasonable point. However, it's an invalid point and there are several reasons for me stating it's invalid:
> 
> 1. I agree that pre-ringing effectively changes the envelope but for that pre-ringing to have any effect, it must be audible/detectable. Let's say hypothetically that it may not be consciously audible but is detectable, in terms of the brain's interpretation of location. If the location of the bell is not where I want it to be, I (as a mix engineer) can simply change the location, if it's not where I expect it to be, I would typically investigate why. Never have I found pre-ringing to be the cause of a bell (or any other sound) not being where I expect it to be.
> 
> ...


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## bigshot (Nov 2, 2017)

KeithEmo said:


> there are two different discussions going on here. One is about whether there is _ANY_ audible difference between high-resolution and "ordinary" recordings; the other is about whether there is a _SIGNIFICANT_ audible difference. As a mix engineer, you undoubtedly listened to what you were mixing very carefully. However, I'll bet you didn't ask a dozen people to listen to the high-res and standard version of each, on a half dozen different DACs, and a dozen different brands of speakers.



There's a third option... There shouldn't be any audible difference on any recording.

I've supervised more mixes than I can count in some very good sound studios in Hollywood. The last step is to output the mix to 16/44.1 and for everyone involved to compare it to the original still in the board for final sign off. If I ever heard a difference between the two, I would have thrown up a red flag, as would have the engineers and talent. The equipment in the room is always carefully calibrated to be consistent and perfect. It represents the reference standard. We never spent much time worrying about how a mix would sound on uncalibrated equipment or DACs that performed out of spec because the range of error would be so broad, there would be no point. We approved 16/44.1 on the reference system and made sure it matched everyone's intentions. And the bounce down never sounded different at all.

I think you're operating beyond the range of reality. It's great to finesse the details, but they have to be perceivable. And the level of finessing that makes sense for a recording studio is greater than the level required to play back that recording in the home. I can see arguing for the need to keep noise floors down in a mix where you're boosting levels on multiple channels, but when I sit in my living room and listen to an album, audibly transparent is audibly transparent.


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## gregorio

KeithEmo said:


> I entirely agree with you that the differences are, at most, very small.... certainly much smaller than many other factors.... and so quite probably are insignificant to most people.



The point I was trying to make is that tests using constructed signals, dirac pulses or single sine waves are effectively constructed to be audible, they are designed to improve the possibility of differences being audible. With real music recordings the transients are smeared all over the place, there's all sorts of masking going on and noise from processing and/or the recording environment. If we can't hear artefacts even with clean specifically designed test signals, we can completely forget about it with commercial audio. I know that audiophiles often scream the opposite, maybe we can hear "things" with real music that we can't with test signals but beyond marketing and their own anecdotal evidence, there's absolutely no reliable or scientific evidence which supports that assertion, in fact it completely contradicts such a belief. For example, jitter determination has been quoted down to about 20ns with test signals but with music as the signal most test subjects were unable to discriminate jitter below 500ns and the lowest achieved was 200ns. We get a similar picture with pretty much anything we test.

G


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## KeithEmo

You'll get no disagreement there from me.

My point was simply that, from a scientific point of view, if I'm trying to prove whether a difference is audible or not, I am in fact going to do my best to construct a test signal where it will be audible..... because, if it is audible under _ANY_ conditions, then I have proven the assertion that "it is audible". If I fail entirely, after making a reasonably thorough and competent attempt to prove my assertion, then we can reasonably conclude that it _ISN'T_ audible under any test conditions we could currently devise. And, if I succeed, and it does turn out to be audible with some specialized test signal, then we can move on to determine whether it is audible under "reasonable and practical" conditions, and how much that should concern the average consumer. I also agree that it should be possible to construct a test that is more sensitive to any difference that actually exists than any sort of listening under "normal conditions" - because the whole point of a test protocol is to maximize your chances of a definite result. (And, while I've heard a few valid points, I tend to agree that most claims to the contrary are simply ways of rationalizing they they didn't get the "obvious positive result" they expected.) 

My honest assessment of the current status of this argument is this..... A significant number of audiophiles are convinced that the difference is so obvious that it should be easily audible. Based on this assertion, several studies have been performed, most of which have so far failed to produce any positive results. (But, of course, a lot of people who are simply "believers" aren't going to believe any results that conflict with their beliefs anyway.) However, because all of the studies I've read about have also been deeply flawed, I do not consider their results to be conclusive. If and when a properly designed and executed test shows positive results, then we can move on to wondering about whether the results are meaningful with normal music, in normal listening conditions. And, if a properly designed and executed test _FAILS_ to produce positive results, then obviously there will be no next stage.  

Historically, however, I remember a time when many people insisted that a good quality cassette recording was "indistinguishable from the original" - which I don't think most people would claim today. (Remember "Is it real or is it Memorex?"). I also remember when MP3's were touted as "being indistinguishable from the original" - because "the psychoacoustic research has all shown that nothing audible is being omitted from them". However, the technology changes, the quality of the master content we have available continues to improve - at least sometimes, and our expectations change. (Perhaps a good quality cassette recording was able to match the quality of a master tape; but that doesn't mean it can match the quality of a good quality modern digital master.) However, based on history, I'm not convinced that "there's no possible difference with high-def content, so we shouldn't even wonder". Personally, I would very much like to see results that can reasonably be considered to be conclusive - one way or the other - from a well designed and properly run test. However, I don't think we've reached that point yet... and I don't see any real movement in that direction.

As I've mentioned before, I don't think the sellers of high-res content will ever sponsor those tests - because the value of being proven right is outweighed by the risk of being proven wrong (and even being proven right - but by a narrow margin - would probably do more harm than good to their sales). Likewise, nobody has a vested interest in proving that high-res files aren't better (because nobody makes money by convincing you _NOT_ to bother to buy that next remaster).



gregorio said:


> The point I was trying to make is that tests using constructed signals, dirac pulses or single sine waves are effectively constructed to be audible, they are designed to improve the possibility of differences being audible. With real music recordings the transients are smeared all over the place, there's all sorts of masking going on and noise from processing and/or the recording environment. If we can't hear artefacts even with clean specifically designed test signals, we can completely forget about it with commercial audio. I know that audiophiles often scream the opposite, maybe we can hear "things" with real music that we can't with test signals but beyond marketing and their own anecdotal evidence, there's absolutely no reliable or scientific evidence which supports that assertion, in fact it completely contradicts such a belief. For example, jitter determination has been quoted down to about 20ns with test signals but with music as the signal most test subjects were unable to discriminate jitter below 500ns and the lowest achieved was 200ns. We get a similar picture with pretty much anything we test.
> 
> G


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## bigshot

Well if you want to prove that noise or distortion or super audible frequencies are audible (or at least perceivable) that isn't hard. Just put on some good headphones with a nice tight seal, get yourself a really powerful amp, and crank the volume to the max. You may end up deaf as a post, but enough volume will make just about anything perceivable. But that isn't the point. What matters is, "Is this an issue I should be concerned about when I go out shopping for home audio equipment?" If the answer is "no" then you're done and it's time to pay attention to something that really does matter.

When you start challenging yourself to hear things that are generally inaudible, that is a great time to step back and take stock of what you're actually interested in proving. Perhaps it's an ego thing... you desperately want to have golden ears so you can tell people that you need better equipment than them because you're special. Or maybe it's an intellectual interest that's gotten out of hand and gone down the rabbit hole of calculating everything out to the full decimal value of pi. Or maybe it's that you want to justify the money you've already spent on equipment that is audibly identical to much cheaper equipment. Whatever it is, making the inaudible audible isn't helpful to anyone. It doesn't help people who read your carefully written arguments choose audio equipment more wisely. It doesn't help scientists understand perceptual thresholds any better. They already know everything human ears can do. The only people arguing that ears have special powers that are heretofore unheard of are high end audio salesmen that want to play into your ego to sell you something you don't really need.

I have great respect for people who can take science and put it into practical perspective. Ethan Winer is one of those folks, and that's why I include his videos in my sig file. He knows both sides- the equipment side and the perception side- and he helps people understand what matters and what doesn't. There aren't a lot of audio equipment reviewers that fit in that category, but I can spot them when I see them and I listen to what they have to say. I don't have much patience at all for commentators who argue endlessly "purely in theory", because that kind of discussion has no end and it has no value. It's just a bunch of words being generated for their own sake. No one should be required to listen to that kind of stuff.


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## gregorio

KeithEmo said:


> Historically, however, I remember a time when many people insisted that a good quality cassette recording was "indistinguishable from the original" - which I don't think most people would claim today



Yes but "the original" (a studio reel-to-reel) wasn't itself even linear within the known limits of human hearing and although some may have said consumer cassettes were indistinguishable, no one (beyond maybe some marketers) would have said that cassettes were linear within the limits of human hearing, just that they were maybe good enough to be "indistinguishable". To show that cassettes were distinguishable, all we would have had to do was demonstrate that the non-linearities of cassettes within the limits of human hearing were detectable. The problem we have here is different though, 16/44 is linear within (and beyond) the known limits of human hearing. So, there is nothing there to detect! Therefore, before we could even get to the question hi-res being distinguishable we would first have to demonstrate that the known limits of hearing are incorrect and only then can we get to the question of whether the content which lies outside the currently known hearing limits is actually enough to be distinguishable. To prove/demonstrate both of these questions is a tall order, with a heavy burden of proof. While I agree that pretty much all the published tests have some flaw/s, tests which agree with the known limits of hearing have a lower burden of proof than those which contradict them. 



bigshot said:


> When people say things like "Let's say for the sake of argument..." or "Assuming that something that isn't audible may be unconsciously perceived..." for me, it's like stepping over the line into fantasy land.



I have to agree with KeithEmo on this one, you are confusing "inaudible" with "not consciously aware of", which are two very different things! As a creator/sound engineer much of my time is spent manipulating that which the consumer will not be "consciously aware of" but none of my time is spent on that which is inaudible.

G


----------



## castleofargh

KeithEmo said:


> You'll get no disagreement there from me.
> 
> My point was simply that, from a scientific point of view, if I'm trying to prove whether a difference is audible or not, I am in fact going to do my best to construct a test signal where it will be audible..... because, if it is audible under _ANY_ conditions, then I have proven the assertion that "it is audible". If I fail entirely, after making a reasonably thorough and competent attempt to prove my assertion, then we can reasonably conclude that it _ISN'T_ audible under any test conditions we could currently devise. And, if I succeed, and it does turn out to be audible with some specialized test signal, then we can move on to determine whether it is audible under "reasonable and practical" conditions, and how much that should concern the average consumer. I also agree that it should be possible to construct a test that is more sensitive to any difference that actually exists than any sort of listening under "normal conditions" - because the whole point of a test protocol is to maximize your chances of a definite result. (And, while I've heard a few valid points, I tend to agree that most claims to the contrary are simply ways of rationalizing they they didn't get the "obvious positive result" they expected.)
> 
> ...



 I don't see the history repeating argument being applied to the right period. but if I look at every single highres format that came since CD, then for sure I can hear Shirley singing


> They say the next big thing is here,
> That the revolution's near,
> But to me it seems quite clear
> That's it's all just a little bit of history repeating.


 


as for not being sure of much of anything, of course even a billion failures wouldn't definitely prove there is nothing audible, and at least from a scientific perspective, the door is never closed. but at a more realistic and practical level, when there is so little sign of finding solid evidence that music requires more to be noticeably transparent despite many years passing by, there's a moment where it starts to feel like all the Clinton Benghazi investigations. it started like something reasonable. the first 5 investigations could maybe have been seen as really concerned people demanding the truth. but soon it clearly felt more like a "witch pursuit thing"





 we have to admit that "the dignity of truth is lost with much protesting". so maybe it's time to





 until the day we actually get relevant reason to question that issue again. 

meanwhile insecure people will have to wait until DXD becomes the standard for streaming music. but let's not kid ourselves, you know that someone will then suggest more and complain about how artificial and lifeless DXD sounds.  we'll blow the planet up long before audiophiles admit to a digital resolution being enough for their old ears.


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## KeithEmo (Nov 3, 2017)

My problem with your assertion is very simple......

All of the facts we have about the known limits of human hearing relate to the very specific question of: Can we perceive the presence of continuous steady state sine waves of a given frequency?.
Therefore your assertion of "there's nothing to detect" is overgeneralized - based on the actual data.
You are generalizing the results of tests conducted using steady state sine waves with all other possible situations involving sound.

We have plenty of data to say with relative certainty that: "We know most humans cannot hear a 25 kHz continuous sine wave".
However, we do not have enough data to claim that: "Therefore, we know that humans cannot discern a 5 microsecond timing difference between two channels.

We CANNOT reasonably claim that: "There's nothing to detect".
The best we can say is that: "Based on the limits determined under other conditions, we suspect that the differences won't be audible under any of the conditions we're discussing".

On your second assertion: "To show that cassettes were distinguishable, all we would have had to do was demonstrate that the non-linearities of cassettes within the limits of human hearing were detectable." (Including my statement that we're going to do the test using content derived from reel-to-reel master tapes.)

1) If we determine that no difference is audible, it could be because we're seeing the limits of tape masters rather than of human hearing. (Perhaps cassettes can reproduce master tapes "audibly perfectly" because, even though they have serious flaws, master reel-to-reel tapes have the same flaws.... so a weakness in our test is masking the audibility of the flaws in cassettes.)

2) Here's an even worse possibility. What if both open reel tapes and cassettes have flaws that are individually inaudible, but they interact to produce audible artifacts (maybe the noise on the cassette modulates the noise from the master tape). If that were the case, it could turn out that both cassettes are "audibly perfect" for recording live music, but that cassettes are NOT "audibly perfect" for reproducing music sourced from master reel-to-reel tapes.

And, to carry that back to digital audio, what if our 16/44k CD can reproduce music in a way that's absolutely audibly indistinguishable from the original live performance, but it makes the background noise from the master tape sound "odd"? In that case, a test using tape-mastered samples might point out a flaw that is _NOT_ detectable with live music. (This is not as far out as you might think. Many digital VIDEO formats, including standard DVDs, do very well at reproducing visible details, yet alter the "background noise" and the "film grain" in very obvious and easily seen ways. This would be analogous to reproducing the music perfectly, but making the tape hiss sound different.... which would have to be considered to be "a perfect reproduction of the music" but NOT a perfect reproduction of the master tape. 



gregorio said:


> Yes but "the original" (a studio reel-to-reel) wasn't itself even linear within the known limits of human hearing and although some may have said consumer cassettes were indistinguishable, no one (beyond maybe some marketers) would have said that cassettes were linear within the limits of human hearing, just that they were maybe good enough to be "indistinguishable". To show that cassettes were distinguishable, all we would have had to do was demonstrate that the non-linearities of cassettes within the limits of human hearing were detectable. The problem we have here is different though, 16/44 is linear within (and beyond) the known limits of human hearing. So, there is nothing there to detect! Therefore, before we could even get to the question hi-res being distinguishable we would first have to demonstrate that the known limits of hearing are incorrect and only then can we get to the question of whether the content which lies outside the currently known hearing limits is actually enough to be distinguishable. To prove/demonstrate both of these questions is a tall order, with a heavy burden of proof. While I agree that pretty much all the published tests have some flaw/s, tests which agree with the known limits of hearing have a lower burden of proof than those which contradict them.
> 
> 
> 
> ...





gregorio said:


> Yes but "the original" (a studio reel-to-reel) wasn't itself even linear within the known limits of human hearing and although some may have said consumer cassettes were indistinguishable, no one (beyond maybe some marketers) would have said that cassettes were linear within the limits of human hearing, just that they were maybe good enough to be "indistinguishable". To show that cassettes were distinguishable, all we would have had to do was demonstrate that the non-linearities of cassettes within the limits of human hearing were detectable. The problem we have here is different though, 16/44 is linear within (and beyond) the known limits of human hearing. So, there is nothing there to detect! Therefore, before we could even get to the question hi-res being distinguishable we would first have to demonstrate that the known limits of hearing are incorrect and only then can we get to the question of whether the content which lies outside the currently known hearing limits is actually enough to be distinguishable. To prove/demonstrate both of these questions is a tall order, with a heavy burden of proof. While I agree that pretty much all the published tests have some flaw/s, tests which agree with the known limits of hearing have a lower burden of proof than those which contradict them.
> 
> 
> 
> ...


----------



## 71 dB

castleofargh said:


> meanwhile insecure people will have to wait until DXD becomes the standard for streaming music. but let's not kid ourselves, you know that someone will then suggest more and complain about how artificial and lifeless DXD sounds.  we'll blow the planet up long before audiophiles admit to a digital resolution being enough for their old ears.



Yeah. I can't wait to see the debates over whether the pre-ringing at 160 kHz ruins the soundstage or not.


----------



## frodeni

KeithEmo said:


> You'll get no disagreement there from me.
> 
> My point was simply that, from a scientific point of view, if I'm trying to prove whether a difference is audible or not, I am in fact going to do my best to construct a test signal where it will be audible..... because, if it is audible under _ANY_ conditions, then I have proven the assertion that "it is audible". If I fail entirely, after making a reasonably thorough and competent attempt to prove my assertion, then we can reasonably conclude that it _ISN'T_ audible under any test conditions we could currently devise. And, if I succeed, and it does turn out to be audible with some specialized test signal, then we can move on to determine whether it is audible under "reasonable and practical" conditions, and how much that should concern the average consumer. I also agree that it should be possible to construct a test that is more sensitive to any difference that actually exists than any sort of listening under "normal conditions" - because the whole point of a test protocol is to maximize your chances of a definite result. (And, while I've heard a few valid points, I tend to agree that most claims to the contrary are simply ways of rationalizing they they didn't get the "obvious positive result" they expected.)
> 
> ...



About this historical claim about tape recordings, the claim of "indistinguishable from the original" was experience based. And repeatable. If running the same test today, using the exact same setup, the experience would be about the same. That is what is reasonable to expect. There is a need to point out, that this was typically pushed by clerks in stores, with some speakers placed in stupid locations, with horrible room acoustics, signal wandering through multiple analogue cable hubs, and so on. The background noise was insane. Once I got my first 3-headed tape player, I was shocked to get a delayed echo of the music all the time. Once I then went to any store, listening carefully, it was always there, for any playback. For any 3-headed player available at the time. All you had to do, was altering the setup, making it possible to detect the flaw.

Insane use of theory, is an age old thing. Denying clearly audible flaws in music reproduction, has been around forever. Even people claiming to musicians, but unable to flaw a 64bit/s Mp3. Knowing how disharmonic the flaws was, I find it hard to accept that any of them people had an ear for harmonics at

At some point though, we will hit a roof, at which the distribution format is actually surpassing the ability of human hearing. Given the prevailing understanding in physics, it is difficult to understand this need for ultra high dynamic range, way beyond 16 bit. Also, the sampling rate, given the prevailing understand in physics, is difficult to align with the need for these ultra high sampling rates.



bigshot said:


> When people say things like "Let's say for the sake of argument..." or "Assuming that something that isn't audible may be unconsciously perceived..." for me, it's like stepping over the line into fantasy land.
> 
> I believe in the inaudibility of inaudible sound.



For any reasonable discussion on any piece of research, knowing the assumptions, the paradigm, the methodology, the methods, the setting, is critical if to achieve any recoverability. It is critical. If this is a fantasy land, then all valid research is a fantasy land.



jimmers said:


> Me too,
> My "Audiophile" What moment was when I first read a DAC spec, 32bit at 384k samples per second.
> So I went and bought a 16 bit ladder (mainly R2R) DAC for my Redbook rips and gave up on stupid numbers and went back to enjoying my music.



Ladder DACs are strange constructions. They may easily use components with say 1% accurate resistors, which gives you, at best, an accuracy of 100 to 1. That is like 8bit DR.

Then there is power supply ripple. Getting a ripple below 16bit? At least for high end ATX powersupplies, measuring anything significantly below 1000 to 1 for ripple, is unheard of. Again, not anywhere near 16 bit.

http://www.jonnyguru.com/modules.php?name=NDReviews&op=Story4&reid=527
_"In these pictures, I see about 8mV and 5mV max for the minor rails. Can't get much better than that. The 12V rail is the worst one at 20mV." (Jonny Guru)
_
The way I see this, we are simply not there yet. The tech most people use, cannot reproduce 16bit/44.1 with any real accuracy. Maybe higher res format helps reducing the flaw of current gear, sure. Personally, I prefer real world testing, not using tech at all, to establish reasonable bounds of what humans can hear. Plenty of research on simple audio, but if someone has done credible research on complex sounds, that research is lost in the noise, at least on me.

Trying to engage in conversation on the experience people have, listening to music using their gear, is a real letdown, at least to me. There is no common understanding, no formalization of any sonic traits, and no will to do so. That includes this forum. It makes it almost impossible to arrive at a common and shared understanding of any experience. It is like a sauce, not a source. A source as to move us forward, to gain greater intersubjective understanding. (remember, this is within the interpretive paradigm) Sure, its is a nice sauce, tasty and stuff. It is still just a sauce. This sauce thing, will drive most people entering this field from say a perspective of physics nuts.

Hopefully, moving into vector sound reproduction, will help establishing the bounds given by human hearing. As that hopefully will enable people to at least engage in conversation that make any sense, using the language they got at hand.


----------



## 71 dB

KeithEmo said:


> We have plenty of data to say with relative certainty that: "We know most humans cannot hear a 25 kHz continuous sine wave".
> However, we do not have enough data to claim that: "Therefore, we know that humans cannot discern a 5 microsecond timing difference between two channels.



Doesn't matter, because 16/44.1 audio can do 5 µs timing differences with complete ease. 



KeithEmo said:


> We CANNOT reasonably claim that: "There's nothing to detect".



Have you detected something? I haven't, but maybe that's because I listen to CDs for the music, not to detect limitations of 16/44.1 digital audio which I know are nearly impossible to detect at best.


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## KeithEmo (Nov 3, 2017)

I don't disagree with you at all...... but you have to differentiate between "pure science", "practical science", and just plain old "common usage".
And I would also say that many audiophiles look at that distinction differently than "regular people".
(And this distinction exists in many areas.)

However, you also need to admit to some distinctions in the other direction.

When I measure lumber, I really don't need to have measurements accurate to 1/100 of an inch.
However, I still spent an extra $15 to buy the LASER ruler that was accurate to 1/100" instead of the one that was only accurate to 1/10".
Now, why, if 1/10" is plenty accurate, would I do that?
Will it ever _REALLY_ matter?
Will I ever _NEED_ to measure something to 1/100"?
In fact, it almost certainly won't matter, but I still prefer to have more accuracy than I need rather than risk less, so I paid a little extra for "insurance".
Likewise, when I used to wear a digital watch, I used to pay $10 more for the one that was accurate to 30 seconds a month instead of two minutes.
(I wouldn't pay $1000 more; but I also wouldn't expend a lot of effort to convince people not to spend the extra $10.)
And, will a $100k Lexus really get me to the corner market any better than my $20k Nissan?

Well, a lot of audiophiles seem to think the same way.
They may fancy they can really hear a difference.
Or they may just like the added assurance that they don't have to wonder if there's something better out there.
Or they may in fact just be buying bragging rights.
I remember occasionally saying something like: "Your clock must be wrong, because I KNOW my watch isn't more than a half a minute off".
Well, some audiophiles derive comfort when, after hearing something odd on a recording, they can say: "It must be a bad recording because I KNOW my equipment sounds right."
And, to people who think that way, it's worth a bit extra (or a lot extra) to take the step from "audibly good enough for most people" to "audibly perfect".

To the marketing department at iTunes, being able to say: "95% of listeners think it sounds perfect" is quite good enough.
To the marketing folks at Tidal, who justify their existence, and make their living, based on the other 5%, it would not be good enough.

And, when you get up into "audiophile land" the landscape becomes even stranger....
And there is a fine line between "things that are audibly better", "things that are technically better, even though the improvement may not be audible", and things everyone is just imagining.

From the title of this thread, the current discussion seems to be about the scientific absolute.
The title includes the assertion that "24 bit audio and anything over 48k is not only worthless but actually bad".
It does NOT assert that "16/44k is plenty good enough for most people, so you're probably wasting your money to pay extra for anything better".
(I probably wouldn't argue at all with that second version. However, to me, this thread is dedicated to a far more aggressive, and to me overreaching, claim.)

If you can reasonably suggest that no living human will ever be able to hear the difference - then the discussion is over.
If you cannot - then it becomes a discussion about priorities rather than absolutes.

Would you really want a TV that reproduced colors and brightness so accurately that you ended up with a sunburn after watching The Martian?
Probably not.....
But an audiophile just might.

(And, if you ever happen to see any of my posts on threads dedicated to "whether high-res downloads are worthwhile" you'll find that I universally suggest that people read the reviews about any given re-master, and decide whether to buy it based on the actual virtues of a given offering. I would certainly not recommend buying a high-res remaster that doesn't sound better than the copy you already have. However, if the new 24/192k remaster sounds really good, I also wouldn't suggest _NOT_ buying it _BECAUSE_ it's 24/192k..... and I'm not going to bother to convert it to 44k after I buy it just to save a few cents worth of storage space - even if I don't hear any difference when I do. I absolutely wouldn't be willing to live without a DAC that supports 24/192k..... _NOT_ because it specifically sounds better, but simply because every conversion changes things, and I want to be able to play any file I come across as it sits.... it's just more convenient than being locked into some limitation that makes me do more work. And, yes, if they're selling a 44k version for $25 and a 24/192k version for $30, I probably will pay the extra $5 for insurance; after all, I'll bet they mastered it at 24/192k, so the 44k version went through an extra conversion, which may not be terrible, but I doubt it's going to improve anything - and may even have been deliberately tweaked to _NOT_ sound as good as the premium version. )



bigshot said:


> Well if you want to prove that noise or distortion or super audible frequencies are audible (or at least perceivable) that isn't hard. Just put on some good headphones with a nice tight seal, get yourself a really powerful amp, and crank the volume to the max. You may end up deaf as a post, but enough volume will make just about anything perceivable. But that isn't the point. What matters is, "Is this an issue I should be concerned about when I go out shopping for home audio equipment?" If the answer is "no" then you're done and it's time to pay attention to something that really does matter.
> 
> When you start challenging yourself to hear things that are generally inaudible, that is a great time to step back and take stock of what you're actually interested in proving. Perhaps it's an ego thing... you desperately want to have golden ears so you can tell people that you need better equipment than them because you're special. Or maybe it's an intellectual interest that's gotten out of hand and gone down the rabbit hole of calculating everything out to the full decimal value of pi. Or maybe it's that you want to justify the money you've already spent on equipment that is audibly identical to much cheaper equipment. Whatever it is, making the inaudible audible isn't helpful to anyone. It doesn't help people who read your carefully written arguments choose audio equipment more wisely. It doesn't help scientists understand perceptual thresholds any better. They already know everything human ears can do. The only people arguing that ears have special powers that are heretofore unheard of are high end audio salesmen that want to play into your ego to sell you something you don't really need.
> 
> I have great respect for people who can take science and put it into practical perspective. Ethan Winer is one of those folks, and that's why I include his videos in my sig file. He knows both sides- the equipment side and the perception side- and he helps people understand what matters and what doesn't. There aren't a lot of audio equipment reviewers that fit in that category, but I can spot them when I see them and I listen to what they have to say. I don't have much patience at all for commentators who argue endlessly "purely in theory", because that kind of discussion has no end and it has no value. It's just a bunch of words being generated for their own sake. No one should be required to listen to that kind of stuff.


----------



## gregorio (Nov 3, 2017)

KeithEmo said:


> My problem with your assertion is very simple......
> [1] All of the facts we have about the known limits of human hearing relate to the very specific question of: Can we perceive the presence of continuous steady state sine waves of a given frequency?.
> [1a] However, we do not have enough data to claim that: "Therefore, we know that humans cannot discern a 5 microsecond timing difference between two channels.
> [1b] We CANNOT reasonably claim that: "There's nothing to detect".
> ...



1. Not it's not, there's been all kinds of tests done, not just with single isolated sine waves. The aforementioned Theiss study used noise for example. Single sine waves are good for certain tests because they are more audible.
1a. Well firstly, there is no timing difference between left and right channels with 44/16, none at all, nada. So there is nothing there to hear, regardless of the limits of human hearing! Secondly, timing inaccuracies within a channel of 44/16 are about half a million times below your quoted 5 micro-sec determination!
1b. No, we CANNOT reasonably claim anything other than there's nothing there to detect!

2. We know very well how the uncorrelated noise floor of 16/44 works and the very fact that it is uncorrelated means that it cannot interact with and change the noise on the original recording. AND, that uncorrelated noise, in the band where our hearing is most sensitive is down at about -120dB, WHERE YOU CANNOT HEAR IT, at least not without destroying your hearing. BTW, the levels at which hearing is damaged was not ascertained by single sine waves!

2a. No, that's not analogous at all! It could be analogous if we were talking about a very lossy audio MP3, say a 96kbps MP3 but we're not, we're talking about 44/16 uncompressed and therefore completely inaudible and UNCORRELATED digital noise artefacts as explained in point 2!

Your arguments are becoming more and more unreasonable because we're talking about demonstrable measurements which are either so far from even the most optimistic limits of human hearing it's laughable or are even more laughable because there are no differences to detect!

G


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## KeithEmo (Nov 3, 2017)

I agree with you..... however this thread is not a discussion about enjoying music.
It is about some very specific assertions about what is and is not "humanly perceptible".

1)
No. 
16/44.1 audio can do 5 uS timing differences easily _ON CONTINUOUS SINE WAVES_.
(and not so well under some other conditions and with some other waveforms.)

2)
I agree with you.
However, this thread is _NOT_ about "whether high-res is worthwhile"...
It makes a very specific assertion (and it asserts, not that the difference is "nearly impossible to detect" but rather that the difference is "_IMPOSSIBLE_ to detect").

In fact, the title of the thread actually makes two assertions that directly contradict each other (as does the original paper).
The original Xaph Audio paper actually asserts that the high-res version will sound audibly WORSE because of interactions between equipment.
(Which claim must be based on the idea that there will be an audible difference after all.)



71 dB said:


> Doesn't matter, because 16/44.1 audio can do 5 µs timing differences with complete ease.
> 
> 
> 
> Have you detected something? I haven't, but maybe that's because I listen to CDs for the music, not to detect limitations of 16/44.1 digital audio which I know are nearly impossible to detect at best.


----------



## KeithEmo (Nov 3, 2017)

I should note something here......

When I'm talking about those "5 uS timing differences" I am _NOT_ talking about jitter or timing_ errors _between channels.

What I'm talking about is having the same sound recorded in both channels, with a time delay being added to one of the channels.

The "overall time resolution" of any digital recording of a continuous sine wave is essentially infinite.
If I take a 500 hz sine wave, and record it on a stereo CD, after delaying one channel by 5 uS, you will be able to easily resolve the 5 uS difference (on an oscilloscope).
The reason this works is because we can accurately reconstruct the two 500 Hz sine waves (in the two channels), and compare them.

_HOWEVER_, our brain uses differences in arrival time to "calculate" location.
Assuming I start with a sound located equally in both channels, I can "move" its apparent location from left to right by adding delay to one channel or the other.
(Our brain calculates that the source is closer to one ear or the other by comparing differences in arrival times at each ear.)

Now let's assume that I start with an unreasonably abrupt impulse (let's call it 5 uS).
(And, yes, I can easily generate a 5 uS sound pulse using various methods.)
Even though most of the energy in that impulse will be at inaudible frequencies, enough will extend into the audible range that we will hear it as a click.
And, if I delay that click in one channel or the other, it will seem to shift locations - between left and right.
However, because that click actually falls between two samples at 44k, we will _NOT_ be able to precisely reconstruct its time from our 44k sample rate recording.
When we apply our band limiting, that impulse will indeed be spread out into a longer waveform that extends over multiple samples.
And, by looking carefully at that new waveform, we will be able to infer where the original impulse occurred in time.

HOWEVER:
1) in order to do so we will have to make certain assumptions about the filter we used
(we'll assume that, if there is equal amplitude in two samples, then the pulse was equidistant in time between them - but this assumption relies on our filter spreading the energy symmetrically in time)
2) the new waveform will be very different than the original
3) more importantly, unlike the original, our new waveform will have a much more gradual envelope
3a) as a result, mechanisms that rely on sensing abrupt edges of waveforms will be less able to accurately "find" the beginning edge of the impulse
3b) current research seems to strongly suggest that our brains do in fact look for those "edges"
3c) this in turn suggests that turning a sharp impulse into a more gradual band limited waveform may compromise the accuracy with which our brains can determine its exact beginning
3d) and this, in turn, suggests that doing so may reduce the accuracy with which our brains are able to utilize this particular location cue
(if the starting time of the impulse cannot be determined distinctly, then we will have the equivalent of a blurry image when we attempt to compare them)

The result of all this MAY be that our brains end up being less accurate in their estimation of where sound objects are located in space.
The result could be that objects seem to be in different locations, or that we perceive the location of individual instruments as being less distinct.
(A similar effect occurs with "stereoscopic 3D video"; when the various depth cues conflict, even slightly, which they often do, the image seems "less distinct and less real".)

Note that, if you accept the current "spectrum analyzer" model of the human ear (with a bunch of "detector hairs", each of which is "tuned" to a distinct frequency).
The frequency which the "top hair" responds to will determine the highest frequency we can detect as a continuous sine wave.
However, that number says nothing about the _TIME RESOLUTION_ (how quickly, and how accurately, our brain can respond to _WHEN_ a particular hair was excited.)

Please note that I'm not specifically suggesting that this will turn out to be true...... however I don't assume that it isn't true either.
Recent research in how your brain calculates eye movement has shown that the mechanism is quite different than what we previously thought... and not especially intuitive to most people.
Therefore, I prefer NOT to make claims based on inferences based on old or incomplete information...



KeithEmo said:


> I agree with you..... however this thread is not a discussion about enjoying music.
> It is about some very specific assertions about what is and is not "humanly perceptible".
> 
> 1)
> ...


----------



## frodeni

gregorio said:


> ...
> 2. Rather ironically, the 1997 Theiss study I mentioned previously (Phantom source perception in 24bit @ 96kHz digital audio) set out to test exactly what you are suggesting. As I mentioned, there was a supplemental test on general perceived sound quality performed under less formal circumstances and it's this test which is frequently quoted by those who have a hi-res agenda but the main experiments were formal DBX tests designed specifically to test localisation and resulted in the conclusion that: "_Analyses of the data showed that the hypothesis that localization accuracy improves with higher sampling rates above the professional 48kHz standard has to be rejected_". (I linked to the paper above so you can read the details for yourself.)
> ...



Looking at that paper, it is pretty obvious, that this does not generate universal knowledge. The gear and setup is highly in question, particularly the use of speakers. Also, they do not describe all factors, particularly interconnects of gear and handling of digital noise. The results are only valid for this single experiment, and only using this particular setup. The sample is also rather small, particularly for a postivistic study.

As a study, it does not prove what is claimed at all. It violates the norms of great research:

_"Analyses of the data showed that the hypothesis that
localization accuracy improves with higher sampling rates above
the professional 48 kHz standard has to be rejected."_​
To arrive at that conclusion, they need to prove that this is case. Which they do not. The setup is flawed, at best, and the samples are not all humans ever and all humans to come. At best, this study would indicate that the 48KHz standard for the setup at hand, used in this way, for the people in the sample, reaps little benefit. Using a closed headset, you obtain a better "laboratory" than this, which says an awful lot.

If the hypothesis was that this was valid for all humans, they would only need to prove it once. But how? They would need to prove that the reproduction was perfect, which they cannot.

Also, the results are not repeatable, given all the basic flaws given this paper.

Why anyone would refer to this as a great piece of research, is beyond me. It simply is not. To me, this is not valid research. "Research" conducted this sloppy, is simply not valid.


----------



## pinnahertz

KeithEmo said:


> You'll get no disagreement there from me.
> 
> My point was simply that, from a scientific point of view, if I'm trying to prove whether a difference is audible or not, I am in fact going to do my best to construct a test signal where it will be audible.... because, if it is audible under _ANY_ conditions, then I have proven the assertion that "it is audible".
> If I fail entirely, after making a reasonably thorough and competent attempt to prove my assertion, then we can reasonably conclude that it _ISN'T_ audible under any test conditions we could currently devise. And, if I succeed, and it does turn out to be audible with some specialized test signal, then we can move on to determine whether it is audible under "reasonable and practical" conditions, and how much that should concern the average consumer.
> ...


Scientific research involving human perception rarely returns such highly polarized black/white results.


KeithEmo said:


> Historically, however, I remember a time when many people insisted that a good quality cassette recording was "indistinguishable from the original" - which I don't think most people would claim today. (Remember "Is it real or is it Memorex?").


Yes, but that was entirely marketing hype. At their absolute best cassettes were always distinguishable from a master quality input signal. Of course they rarely ever received that kind of signal, yet even using the absolute best equipment, the results were audibly flawed.  Please realize that professional tape was and is also distinguishable from its input signal, and cassettes are far, far below that in terms of performance. 


KeithEmo said:


> I also remember when MP3's were touted as "being indistinguishable from the original" - because "the psychoacoustic research has all shown that nothing audible is being omitted from them".


And they are today, but MP3s cannot be discussed as if it's a single fixed parameter unit. Early MP3 usage was deliberately at very low bit rates of necessity dictated by the available bandwidth at the time. While MP3 may not be the most efficient lossy codec, when sufficiently high bit rates are used indistinguishability can be achieved.
Few thought earlier low rate files in common use then were transparent.


KeithEmo said:


> However, the technology changes, the quality of the master content we have available continues to improve - at least sometimes, and our expectations change. (Perhaps a good quality cassette recording was able to match the quality of a master tape; but that doesn't mean it can match the quality of a good quality modern digital master.)


(Cassettes at their best never came close to matching a master of any kind)


KeithEmo said:


> However, based on history, I'm not convinced that "there's no possible difference with high-def content, so we shouldn't even wonder". Personally, I would very much like to see results that can reasonably be considered to be conclusive - one way or the other - from a well designed and properly run test. However, I don't think we've reached that point yet... and I don't see any real movement in that direction.


I actually agree with this. If there is any possible difference we should know what it is.  But we must first start with determining if there is a difference and under what conditions.  That has not been done well.  But when contrasted to all previous recording methods, all of which were easily and immediately distinguished from a clean input signal, after about two decades of use there remains no overwhelming statistical fallout of clear differentiation.  That fact alone is a strong indicator that differences, if any, are not significant.


KeithEmo said:


> As I've mentioned before, I don't think the sellers of high-res content will ever sponsor those tests - because the value of being proven right is outweighed by the risk of being proven wrong (and even being proven right - but by a narrow margin - would probably do more harm than good to their sales). Likewise, nobody has a vested interest in proving that high-res files aren't better (because nobody makes money by convincing you _NOT_ to bother to buy that next remaster).


Yes, of course HR sellers and manufacturers won't test this.  They're operating on expectation, which sadly is a far more powerful influence than fact.


----------



## bigshot

There is an interesting argumentative formula being used here. Perhaps even more interesting than the line by line pick apart format...



> I don't disagree with you. In fact I agree with you.
> 
> But... (then another long repetition of all the misconceptions that have already been answered and corrected several times before in the thread)



You guys have more patience with people who are only interested in talking for their own benefit than I do. Whenever someone says, "I'm not talking about practical application, I'm talking in theory here." I know we're in for a lot of repetitive back and forth. Pure theory doesn't need any anchor in reality. It doesn't need facts either. It just needs a whole lot of words. There are debates that are worthy of simply dismissing with the back of your hand. Circular arguments made by people who aren't listening and processing what people are saying in response aren't worth spending a whole lot of time over.


----------



## KeithEmo

I read through that study (well, I scanned it)...... and I agree with you that it was in general sloppy and had lots of procedural flaws.
It was also specifically intended to identify whether there was a correlation between sample rate and spatial resolution.
(And the results, as credible - or not - as we may consider them, at least suggested that there was no strong correlation to be found there.)

However, check out the section entitled "additional experiment into overall sound quality" (bottom page 13).
In that section, they presented a very limited number of test subjects (4) with samples of music at 48k and 96k.
After being presented with samples at both sample rates, the subjects were presented with unknown samples, and asked to identify what sample rate they were listening to.
This was a somewhat modified ABX sort of test.

The results of that test, and the conclusions based on them, were interesting....
One subject was correct 16 times out of 17 trials.
And two of the subjects were correct 61% and 68% of the time respectively.

In fact, while the presenters of the test concluded that they had failed to demonstrate any difference in perceived spatial positioning accuracy.....
They also concluded that the differences between 96k and 48k were "clearly audible".... to at least some listeners.

From their conclusions:
"While there is little doubt that subject 1 reliably heard a difference between HDDA and 48 kHz reproduction, results of subject 2 and 3 need
closer evaluation. The probability that the results obtained from both subjects were randomly guessed is 6%. It is reasonable to conclude that
even in the 96 kHz to 48 kHz sampling rate comparison there was a perceivable difference."

I would have to say that using that test as "proof" that there is no significance between 48k and 96k in terms of spatial cues,
while ignoring the fact that it also concluded that there were other "clearly audible differences" would be a sort of cherry picking  

(Bear in mind that, in order to establish that "humanly audible differences exist", we only have to produce a single test subject for which this is provably true.)



frodeni said:


> Looking at that paper, it is pretty obvious, that this does not generate universal knowledge. The gear and setup is highly in question, particularly the use of speakers. Also, they do not describe all factors, particularly interconnects of gear and handling of digital noise. The results are only valid for this single experiment, and only using this particular setup. The sample is also rather small, particularly for a postivistic study.
> 
> As a study, it does not prove what is claimed at all. It violates the norms of great research:
> 
> ...


----------



## KeithEmo

I would want to add one thing to what you've said......

Pure theory does not in fact require "an anchor in reality".
(Although it's usually a waste if not at least based on reality. After all, we're not discussing what unicorns prefer for dinner.).
However, if you want to present a theory as_ FACT_, then it does require some solid references to reality.

I agree that the assertion that "nobody can hear any difference" is at the level of a theory...
As is the assertion that "at least some people can hear differences"...
And, at the level of theory, we can freely discuss our reasons for believing that either theoretical claim will turn out to be factually true...
However, _NEITHER_ rises to the level of "fact" without proof.

I would also point out that this entire thread is mostly about theory.....
(For most of us, in practical terms, both the difference in price, and the difference in storage space and bandwidth, between regular and high res files are really inconsequential.)



bigshot said:


> There is an interesting argumentative formula being used here. Perhaps even more interesting than the line by line pick apart format...
> 
> 
> 
> You guys have more patience with people who are only interested in talking for their own benefit than I do. Whenever someone says, "I'm not talking about practical application, I'm talking in theory here." I know we're in for a lot of repetitive back and forth. Pure theory doesn't need any anchor in reality. It doesn't need facts either. It just needs a whole lot of words. There are debates that are worthy of simply dismissing with the back of your hand. Circular arguments made by people who aren't listening and processing what people are saying in response aren't worth spending a whole lot of time over.


----------



## castleofargh

KeithEmo said:


> I read through that study (well, I scanned it)...... and I agree with you that it was in general sloppy and had lots of procedural flaws.
> It was also specifically intended to identify whether there was a correlation between sample rate and spatial resolution.
> (And the results, as credible - or not - as we may consider them, at least suggested that there was no strong correlation to be found there.)
> 
> ...



I find this sort of experiment addendum amazing by how suspicious it is. to put it there to hang at the end of the paper for the lolz. "oh BTW we could have proved something important, but sorry, research those days only gets funds for stuff where we debunk ourselves, discoveries and positive results are so overrated". ^_^
I don't know if they had the most unfortunate circumstances, or if a second follow up paper was projected but they found some flaw in their testing method or never got funded. but I would have lost my mind if I had been with them at the time.


----------



## bigshot (Nov 3, 2017)

I was using the term with the casual definition, but if you want to get technical about it, Google is your friend... "difference between hypothesis and theory"

In scientific terms; A *hypothesis* is either a suggested explanation for an observable phenomenon, or a reasoned prediction of a possible causal correlation among multiple phenomena. In science, a *theory* is a tested, well-substantiated, unifying explanation for a set of verified, proven factors.

http://www.oakton.edu/user/4/billtong/eas100/scientificmethod.htm
Note number 4

The belief that things that have been tested and determined to be inaudible actually are inaudible is a theory. Making up reasons how you might possibly be able to hear something below that threshold is a hypothesis. Raising that to the level of a theory would require a whole lot of testing that validates it. So far, no tests I know of support that idea- only cherry picking and sales pitch.


----------



## 71 dB

[QUOTE="KeithEmo, post: 13825887, member: 403988"

1)
No.
16/44.1 audio can do 5 uS timing differences easily _ON CONTINUOUS SINE WAVES_.
(and not so well under some other conditions and with some other waveforms.)
[/QUOTE]

I admit I don't understand this. Why does it only work on continuous sine waves? What goes wrong with other signals?

I tested this on Audacity. I created pink noise at 96 kHz. Then I duplicated it and delayed the duplicate 1 sample, 10.4 µs. Then I downsampled the original and duplicated noises to 44.1 kHz. Now, of course the waveform of the delayed noise differs from the original, but that's deceiving and doesn't correspont the analog signal after DAC. Now, I upsampled them back to 96 kHz and the waveforms look identical again, only bandlimited. The delayed one is clearly one sample behind the original as expected. So, I delay the original by one sample and invert it. I then sum the signals to get the difference which is the error signal. There is some quiet noise with most of it's energy on ultrasonic range. I downsample the error signal and the level of it is about -88 dBFS. While very quiet, the error should be zero. I don't know what is going on. What if I have been wrong all this time about digital audio and ALL my 1500 CDs actually sound very bad!! HORRIBLE!!! DO SOMETHING!!! Is even 192 kHz enough??!!?


----------



## pinnahertz

KeithEmo said:


> I would want to add one thing to what you've said......
> 
> Pure theory does not in fact require "an anchor in reality".
> (Although it's usually a waste if not at least based on reality. After all, we're not discussing what unicorns prefer for dinner.).
> ...


And both statements are far too polarized and without situational qualification....so....


KeithEmo said:


> I would also point out that this entire thread is mostly about theory.....


Perhaps for some, not all.  Like I said, 20 years and still no hard proof....


KeithEmo said:


> (For most of us, in practical terms, both the difference in price, and the difference in storage space and bandwidth, between regular and high res files are really inconsequential.)




But the actual handing and playing of it...not quite so inconsequential.  For just one example, the largest base of installed home sound systems today includes an AVR.  Even many here listen to music material only via an AVR.  Most AVRs handle volume control, and certainly calibration EQ, at 24/48, meaning pretty much everything you hear is resampled (rate-up, rate-down, bit depth) except for film soundtracks.  How's that getting 24/96 or higher out to a transducer...that can't reproduce it anyway?  If everything you hear is funneled to 24/48, why bother with anything higher unless you like the remastering job? 

Oh... sorry... practicality raises it's ugly head again.  There are many other examples of the problem.

And...just because I'd expect somebody to ding me on it...yes, there are 24/96 and higher processors in the world, i.e. the Trinnov Altitude32 with internal 64bit FP @ 192kHz.  Got a  spare $30K?


----------



## bigshot

File size isn't inconsequential. I have a very large music library and carry a big chunk of it around with me on a 256 GB micro SD card. That's the biggest one they make, but it still only can hold a small fraction of the music on my music server in AAC 256 format. If I carried around 24/96 files, I would have to be constantly updating the card and shifting files back and forth.


----------



## KeithEmo

You bring up a very good point.... which is that the relevance of a lot of this depends on the market you're talking about. (And the marketing folks who are working to sell this stuff often try to ignore or cause their customers to ignore these questions.)

First, for many of those people, the difference between standard and high-res files probably isn't going to make any immediate difference.
However, as you alluded to, re-masters often sound better for other reasons, which are audible even on mediocre equipment. 
Therefore, we might at least hope than an interest in "high-quality remasters" might encourage better re-masters in general.
(If you want to pursue that point, the majority of listeners probably listen to their music playing from their phone on $20 ear buds.)

Second, many of us actually purchase our music, and keep it in our collection.
Therefore, many of those people who currently own a mediocre sounding AVR may someday own better equipment, on which the difference will be audible.
I'm sure glad I have 2000 CDs instead of 2000 albums from iTunes..... even if the lossy compression used by iTunes might have sounded OK on the equipment I had twenty years ago.
Therefore it does make sense to avoid investing a lot of money on something that you'll end up having to buy again later... or to buy the better version as "insurance".
(This equation will be very different for people who use a streaming service rather than actually own their music.)

Third, you're really asking the difference between "better for everyone" and "better for a few select audiophiles"?
(Of course, from a sales point of view, the sellers would like everyone to assume they'll hear a difference.)




pinnahertz said:


> And both statements are far too polarized and without situational qualification....so....
> Perhaps for some, not all.  Like I said, 20 years and still no hard proof....
> 
> 
> ...


----------



## KeithEmo

Obviously that's all relative.

My main music library is currently on a 6 tB drive... and I don't carry it all around with me.
(At most, I may make a portable copy of a few hundred albums to take with me when I travel.)
In terms of cost, a 6 tB drive currently costs about $150....
Clearly we each have very different priorities.... perhaps _FOR YOU_ AAC does make more sense.

I would also point out that I can always convert my 24/96k files into AAC if I need a small portable copy.
However, if my "master copies" were in lossy AAC, the reverse would not be true.



bigshot said:


> File size isn't inconsequential. I have a very large music library and carry a big chunk of it around with me on a 256 GB micro SD card. That's the biggest one they make, but it still only can hold a small fraction of the music on my music server in AAC 256 format. If I carried around 24/96 files, I would have to be constantly updating the card and shifting files back and forth.


----------



## reginalb (Nov 3, 2017)

bigshot said:


> ...256 GB micro SD card. That's the biggest one they make...



I'm sorry I'm always being a pedant in response to your posts, but...

...negative ghostrider, you can upgrade that now: https://www.amazon.com/Sandisk-Ultr...pID=41MQ9ndxA7L&preST=_SX300_QL70_&dpSrc=srch

And if you subscribe to Play Music All Access, and use that in an Android phone, you can upload another 50,000 songs from your collection on top of the stuff you can stream. I think you're an AAC guy? You could get a whole lot of 256k AAC on to a 400GB card, maybe 60,000 tracks, then you've got your 50,000 uploaded (I know you're a classic guy, but the GPM limit is number of tracks, so you could strategically upload your biggest stuff), plus all of that you can stream? There would be no end to the music you've got with you at that point.


----------



## frodeni (Nov 3, 2017)

KeithEmo said:


> I read through that study (well, I scanned it)...... and I agree with you that it was in general sloppy and had lots of procedural flaws.
> It was also specifically intended to identify whether there was a correlation between sample rate and spatial resolution.
> (And the results, as credible - or not - as we may consider them, at least suggested that there was no strong correlation to be found there.)
> 
> ...



Thanks. I missed that. Sloppy just became fraudulent.

It is funny how the lack of real scientific proof for this supposedly proven knowledge is utterly missing, and that nobody in here seem to be able to point to concrete research supporting their claim. Yet there is a substantial level of screaming about a lot of stuff supposedly being knowledge.

Guys, great if there is real scientific proof, but this fraud, is a fraud. This simply does not stick up to a through review. Not even a quick and simple one. Throwing this kind of "proof" at us, simply prove that you cannot read a scientific paper, as a scientist.

Sure, you might be right, but if one contestant gets 16 out of 17 correct, which is a 94,1% hit-rate, please explain the conclusion. Remember the number of contestants, before you answer.



bigshot said:


> I was using the term with the casual definition, but if you want to get technical about it, Google is your friend... "difference between hypothesis and theory"
> 
> In scientific terms; A *hypothesis* is either a suggested explanation for an observable phenomenon, or a reasoned prediction of a possible causal correlation among multiple phenomena. In science, a *theory* is a tested, well-substantiated, unifying explanation for a set of verified, proven factors.
> 
> ...



Well, who ever this guy is, he has some serious issues with the philosophy of science. The big question is how he supposedly proves anything, and what constitutes a proven theory in science. He would flunk any test at almost any university, presenting this.

Also, if this guy has written ANY paper at all, I cannot find a single accepted work of his, in any scientific database. Please point me to his work. If there is any.

I guess asking for the proof, as in real scientific form, suddenly became more pressing.

Does anyone have any work, that they have actually vetted themselves, that they can point to? To prove this claim that mankind _know_ this high def audio to be all but waste.

I guess claiming that, suddenly got a lot harder, once you got real scientists in here? Doesn't it? What is next? Banning the scientist for being too scientific, in a science forum?

The great irony is, that what is thrown as accusation at writers in this forum, actually plagues the papers that are being presented as proof. The exact same thing. How come people does not see the flaw in work at which they point to as proof? It is not like they can argue that they did not know about it.


----------



## bigshot

frodeni said:


> Well, who ever this guy is, he has some serious issues with the philosophy of science. The big question is how he supposedly proves anything, and what constitutes a proven theory in science. He would flunk any test at almost any university, presenting this.



Who are you referring to? You’re throwing up a whole bunch of smoke, but I can’t figure out what you’re saying.

Have you checked out the links in my sig file? They might explain things for you. They’re pretty clear and straightforward.


----------



## KeithEmo

It's not that "it doesn't work" - only that the constraints of the theory aren't what many people seem to think.

Assuming I start off with a few points, I can draw a lot of lines that pass through them.
So first I limit my solutions to lines that can represent audio signals - they must move from left to right and never backtrack (the "time arrow" moves from left to right).
There are still an awful lot of lines that can pass through those points since they can zig zag up and down any way I please.
So let's band limit our line - which means that we're requiring it to be smooth (sharp corners represent higher frequencies).
However, we still have a bunch of options.
However, if we limit it to CONTINUOUS SINE WAVES, only one valid option remains - so we have gotten back our original signal.

Essentially, in non-math terms, if we remove the requirement that a continuous sine wave be involved, we don't have enough information to get back the original signal.
Note that it gets a little fuzzy here because, at least in theory, and odd squiggly CAN be described as a sum of sine wave components.

The short answer to your question is that there are all sorts of errors involved.
There are actual numerical errors - where number shave been rounded and so precision has been lost.
There are also various "uncertainty errors" - such as the rule that you _CANNOT_ have a sharp filter that doesn't introduce time errors.
This isn't exactly an error - it's more like a constraint. Any filter that can be used to limit bandwidth _MUST_ introduce time errors. 
(You can hope to design filters where the errors cancel each other out - but you cannot avoid the errors themselves.)

Also, to put it bluntly, the errors can at least be minimized, and some programs do a better job of that than others.



71 dB said:


> [QUOTE="KeithEmo, post: 13825887, member: 403988"
> 
> 1)
> No.
> ...



I admit I don't understand this. Why does it only work on continuous sine waves? What goes wrong with other signals?

I tested this on Audacity. I created pink noise at 96 kHz. Then I duplicated it and delayed the duplicate 1 sample, 10.4 µs. Then I downsampled the original and duplicated noises to 44.1 kHz. Now, of course the waveform of the delayed noise differs from the original, but that's deceiving and doesn't correspont the analog signal after DAC. Now, I upsampled them back to 96 kHz and the waveforms look identical again, only bandlimited. The delayed one is clearly one sample behind the original as expected. So, I delay the original by one sample and invert it. I then sum the signals to get the difference which is the error signal. There is some quiet noise with most of it's energy on ultrasonic range. I downsample the error signal and the level of it is about -88 dBFS. While very quiet, the error should be zero. I don't know what is going on. What if I have been wrong all this time about digital audio and ALL my 1500 CDs actually sound very bad!! HORRIBLE!!! DO SOMETHING!!! Is even 192 kHz enough??!!?[/QUOTE]


----------



## KeithEmo

If you want to be picky, there are more "interesting things" about that paper.
For example, four of their seven subjects were employees of the company who manufactured the speakers they used (I might guess that they convinced the company they worked for to donate the speakers).
It also seems obvious, after reading a bit more carefully, that the intent was not at all "to study whether there was an audible difference between high-res and regular files".
The purpose was a rather narrow and scholarly experiment about one very specific aspect of the difference..... perceived position vs time resolution.
And they seemed to have proven their hypothesis there by delivering a "no result" (they demonstrated that they were unable to validate their original theory).
Also note that they were unable to pursue several avenues of research due to equipment limitations and time constraints.
(They were unable to follow up on one question because their DAC refused to function reliably at all.)
They were clearly constrained by time, budget, and equipment availability.
(It reads to me like a term paper...... and not like a "major industry study".)

Honestly, from how that last conclusion was added, it seemed to me as if they considered it to be more of a confirmation of an obvious expectation that as something they considered to be a significant result.
I also find it VERY suggestive that a single subject scored so much better than the others; it would be very interesting to find out why.
Intuitively, I am inclined to wonder if there is something to the claim put forth in some other papers that there is a "training aspect" involved....
(By which I mean that people who are trained in terms of what specific differences to look for are much more likely to notice them.)
This is quite common in many other areas..... so I don't find it especially controversial to suggest that it might apply here as well.



castleofargh said:


> I find this sort of experiment addendum amazing by how suspicious it is. to put it there to hang at the end of the paper for the lolz. "oh BTW we could have proved something important, but sorry, research those days only gets funds for stuff where we debunk ourselves, discoveries and positive results are so overrated". ^_^
> I don't know if they had the most unfortunate circumstances, or if a second follow up paper was projected but they found some flaw in their testing method or never got funded. but I would have lost my mind if I had been with them at the time.


----------



## frodeni

bigshot said:


> Who are you referring to? You’re throwing up a whole bunch of smoke, but I can’t figure out what you’re saying.
> 
> Have you checked out the links in my sig file? They might explain things for you. They’re pretty clear and straightforward.



This is getting dumb. Who did you quote?

Has he written any paper, that has passed any real scrutiny? Can you please point me to his work?

Did you just Google that or is this someone you actually know and a piece of work you actually vetted?


----------



## bigshot (Nov 3, 2017)

I quoted Google search. As far as I know he isn't a he.

Is Miriam Webster a person? https://www.merriam-webster.com/words-at-play/difference-between-hypothesis-and-theory-usage

Nyquist had a theory. So did Einstein, Galileo and Darwin. We operate on theories every single day of our lives. To quote Miriam (what a lovely name!) quoting Kenneth R. Miller, a cell biologist at Brown University, a theory "doesn’t mean a hunch or a guess. A theory is a system of explanations that ties together a whole bunch of facts. It not only explains those facts, but predicts what you ought to find from other observations and experiments.”

I'm happy I could help you learn something new about science today! You're welcome!

Have you checked out the links in my sig file yet? Great info there.


----------



## reginalb

frodeni said:


> Thanks. I missed that. Sloppy just became fraudulent.
> 
> It is funny how the lack of real scientific proof for this supposedly proven knowledge is utterly missing, and that nobody in here seem to be able to point to concrete research supporting their claim. Yet there is a substantial level of screaming about a lot of stuff supposedly being knowledge.
> 
> ...



Not sure exactly what you're saying. But anyway, you can't prove anything in science. You disprove or fail to disprove.


----------



## pinnahertz

KeithEmo said:


> You bring up a very good point.... which is that the relevance of a lot of this depends on the market you're talking about. (And the marketing folks who are working to sell this stuff often try to ignore or cause their customers to ignore these questions.)
> 
> First, for many of those people, the difference between standard and high-res files probably isn't going to make any immediate difference.
> However, as you alluded to, re-masters often sound better for other reasons, which are audible even on mediocre equipment.


I'd modify that: re-masters often sound different from the original.  Better is often in question.


KeithEmo said:


> Therefore, we might at least hope than an interest in "high-quality remasters" might encourage better re-masters in general.
> (If you want to pursue that point, the majority of listeners probably listen to their music playing from their phone on $20 ear buds.)
> 
> Second, many of us actually purchase our music, and keep it in our collection.
> Therefore, many of those people who currently own a mediocre sounding AVR may someday own better equipment, on which the difference will be audible.


Careful, now... you're jumping right back into the same kind of issue.  "mediocre sounding AVR"?  Assuming any cheap AVR sounds different from an expensive one?  How about we just not go there.  Audible sound quality differences in hardware is a separate thread for sure, and will be chock-full of biased opinions.


KeithEmo said:


> I'm sure glad I have 2000 CDs instead of 2000 albums from iTunes..... even if the lossy compression used by iTunes might have sounded OK on the equipment I had twenty years ago.
> Therefore it does make sense to avoid investing a lot of money on something that you'll end up having to buy again later... or to buy the better version as "insurance".
> (This equation will be very different for people who use a streaming service rather than actually own their music.)


I get what you're saying, but the implications are all wrong.  You imply that 20 year old gear sounds so bad you couldn't here lossy codec artifacts, AND that iTunes 256K AAC is not transparent.  I have to disagree with both in general, though of course anyone can find specific contrary examples.  If the gear was good 20 years ago and still operating in spec, and if the music was done well to begin with...and so on....

We must be at least aware that the results of any of this will never by binary, they'll be at least 3D data cube in which trends can be seen. 


KeithEmo said:


> Third, you're really asking the difference between "better for everyone" and "better for a few select audiophiles"?
> (Of course, from a sales point of view, the sellers would like everyone to assume they'll hear a difference.)


Sort of, but what I'm really asking is for some statistically valid trend that it's repeatably better for any group, and under practically valid conditions, so no anechoic chambers, no DIRAC pulses, square waves, and a test group that follows a good population cross-section.  We need to know first "if", then we can work on "why".


----------



## bigshot

I've done a direct A/B switched line level matched comparison between my AVR and my Oppo HA-1 and I couldn't hear a bit of difference. People assume that high end stuff sounds better just because it measures better. They invest themselves and their money into believing that. But it isn't necessarily true. It all goes back to the classic audiophool dodge... if you can't hear a difference then either your ears or your equipment is faulty... Yeah, and that suit of clothes the emperor is wearing sure does look fine.


----------



## frodeni

reginalb said:


> Not sure exactly what you're saying. But anyway, you can't prove anything in science. You disprove or fail to disprove.



That comes down to epistemology. How can you know that you cannot know anything? We all need to accept that cause and effect has its limits, unless we want to go insane.

As to what I am saying, is that they ignored their true positive data, and ignored following it up. The conclusion is clearly in violation with the data. That raises the biggest flag that can be raised, in any science.



bigshot said:


> I quoted Google search. As far as I know he isn't a he.
> 
> Is Miriam Webster a person? https://www.merriam-webster.com/words-at-play/difference-between-hypothesis-and-theory-usage
> 
> ...



Sure, you quoted one source and cited it to another. Keep up the good work! Looking more carefully into to it, they are even contradicting each other. Great work buddy.

Then you probably googled and found something in the Post. Then you claim to quote Miriam, citing the posts as link. You are just trying to make a point, with absolutely no real understanding about the topic at hand.

I am not schooled in the tradition of theory making, and would not even dare to quote what you just did, as I have no real knowledge of the field of theory building. That is usually only taught at the Phd level, if at all. Throwing quotes you do not comprehend in the slightest, simply does not make you a teacher. What I just learned, is that you just twice used quotes and citations wrong, which would clearly indicate that you have never written any paper at any real university. Let us hope that is the case, for the sake of science. Also, you google to find your info, and by throwing results out of Google, you think highly of yourself as teaching people. Demanding recognition. That is what I just learned.

I also learned not to expect any quote to be cited correctly. If demanding people to provide anything beyond throwing quotes from a google search, I apparently cannot expect people to elaborate on what they quote, or to have vetted the content, as they simply do not comprehend what they boast about at all.

Also, you quoted a news article, citing in all essence other news media, and a lady promoting her book for high school usage. You did not quote any academic paper on the matter. If you had done so, you would find the topic of what constitutes a theory or a hypothesis, or what is the right epistemology or ontology to be quite a contested field in the sciences. For very good reasons.

As for your sig file, I actually checked them links ages ago. They are not helping much either. Particularly as they violates the ethics set by most scientific institutes. There is a ton of others concerns as well, that these guys earns by their behavior. To me, this appear to be the equivalent of alternative science.

As for science and knowledge, this falls inline with a lot of the crowd in this place. It really makes me wonder what we really do know, and what is really the state in the healthy part of science, on the topic at hand. In-fact, by the all the research pointed to, even the phony ones, they all indicate that some people seem to be able to hear differences, not the opposite, as people frequently claim in this very forum.

But let us be fair, this is by no means a slam dunk. Not for all people. We are closing in on the border of what can be heard. And for my part, if I use a crappy USB source, I cannot even tell the difference between 16/44.1 and lossy compression. I just cannot explain why. The more I spend thinking about possible causes, being given papers like lately, I just do not see the due diligence throughout the gear chain. Not to move us forward, providing low cost high performance gear.


----------



## 71 dB

KeithEmo said:


> It's not that "it doesn't work" - only that the constraints of the theory aren't what many people seem to think.
> 
> Assuming I start off with a few points, I can draw a lot of lines that pass through them.
> So first I limit my solutions to lines that can represent audio signals - they must move from left to right and never backtrack (the "time arrow" moves from left to right).
> ...



I feel sick and tired today, so it's hard to think. The analog signal is a sum of sample-weighted sinc-functions with various delays. I don't get why this works only for sine waves.
Time errors shouldn't be dependent on delay. Why would a filter cause different time error for delayed signal? Doesn't make sense. That would require time-variant filters. Maybe it's all because the sinc -functions are actually windowed versions. In that case we can increase window size and reduce the error, make it as small as we want (need).


----------



## Don Hills

KeithEmo said:


> ...
> Essentially, in non-math terms, if we remove the requirement that a continuous sine wave be involved, we don't have enough information to get back the original signal.
> Note that it gets a little fuzzy here because, at least in theory, and odd squiggly CAN be described as a sum of sine wave components.
> ...



... so Nyquist and Shannon were wrong then?


----------



## frodeni

bigshot said:


> I've done a direct A/B switched line level matched comparison between my AVR and my Oppo HA-1 and I couldn't hear a bit of difference. People assume that high end stuff sounds better just because it measures better. They invest themselves and their money into believing that. But it isn't necessarily true. It all goes back to the classic audiophool dodge... if you can't hear a difference then either your ears or your equipment is faulty... Yeah, and that suit of clothes the emperor is wearing sure does look fine.



Sure, you did. What do you mean by AVR?

I happen to also use the HA-1. It is not the easiest beast to tame, and given what is needed for it, that probably applies to most DACs. Let's just say that I hear plenty of differences using mine.

What does that mean. Well it means that you claim to hear no difference as compared to some "AVR", while I could tell you about clear and distinct sonic traits that differs clearly, by the type of setup, using mine. I am not dismissing your experience based on mine, but your are dismissing mine, based on your experience.

Which just enforces the impression given as of late.


----------



## KeithEmo

I'm not jumping back into anything.....

Someone specifically asked whether there would be any reason whatsoever for someone to purchase higher quality content "even if they couldn't hear the difference on their current equipment".
My reply is that it still makes sense to purchase it, even if you don't hear any difference on your current equipment,_ IF_ you expect that you might later have equipment on which you _WILL_ hear a difference.
(Note that I didn't say anything about price; although I will certainly assert that some AVRs sound noticeably inferior to others.... I wasn't the one who suggested "AVRs" as the example ) 

I also make no assertion that _ALL_ 20 year old equipment is inferior.... although I would also note that significant advances in ADC and DAC technology have certainly occurred.
While avoiding generalities, I would suggest that it's not unreasonable to suspect that at least some equipment available today is better than _ANY_ available 20 years ago. 
I would also assert that many of the tests I see quoted repeatedly were performed on equipment that I consider suspect... and none of them seemed to have thoroughly verified the capabilities of the equipment they used.
When you perform an experiment, one of the first validation steps is to confirm and document that your equipment can in fact deliver the test stimuli it is intended to test.
So, if you want to test whether people can hear 30 kHz, you start by using a test microphone to confirm that you have 30 kHz actually physically present at the test location (and present in your test content).

Errrrr..... I disagree with your final assertion entirely.

If you want to ask "whether high-res audio makes sense for the average consumer" then by all means lets look at statistics cubes.
_HOWEVER_, if we're talking about a scientific claim about whether "the difference is audible", a _SINGLE_ well documented and repeatable example is enough to establish that it is.
(If a single human can reliably hear a difference, on a single file, on a single combination of equipment, then "it is audible by human beings".)
Then, once that "if" is established, we would want to know both "why" and "how many".

Of course, we absolutely require the best possible cross section of humans.......
(If it turns out that only left handed midget harp players from Burundi can hear it, we wouldn't want to miss that by not including at least one of them.)
From the data so far, it seems quite possible that "audiophiles who are certain they can hear a difference" may _NOT_ be the best possible subjects.
(I might suggest running a few tests using school-age children as test subjects - since younger humans have been documented as generally having better high-frequency hearing than older humans.)




pinnahertz said:


> I'd modify that: re-masters often sound different from the original.  Better is often in question.
> Careful, now... you're jumping right back into the same kind of issue.  "mediocre sounding AVR"?  Assuming any cheap AVR sounds different from an expensive one?  How about we just not go there.  Audible sound quality differences in hardware is a separate thread for sure, and will be chock-full of biased opinions.
> I get what you're saying, but the implications are all wrong.  You imply that 20 year old gear sounds so bad you couldn't here lossy codec artifacts, AND that iTunes 256K AAC is not transparent.  I have to disagree with both in general, though of course anyone can find specific contrary examples.  If the gear was good 20 years ago and still operating in spec, and if the music was done well to begin with...and so on....
> 
> ...


----------



## KeithEmo

No. I believe you will find that stated as a condition in their original papers (although I'm not certain there).



Don Hills said:


> ... so Nyquist and Shannon were wrong then?


----------



## bigshot (Nov 3, 2017)

I think they said that 44.1 sampling rate can perfectly reproduce any sine wave representing sound that the human ear can hear. That's good enough for me. I tend to listen to music with human ears.

It doesn't matter if I use fancy equipment or cheap equipment. I haven't ever seen any evidence that super audible frequencies are audible, so higher sampling rates don't have much purpose for my human ears. Feel free to believe that pigs can fly and you can hear things that other humans might not be able to, but there's plenty of evidence on Nyquist's side and on the side of audiologists who have established the thresholds of human hearing. The ball is in your court to either prove that Nyquist is wrong and the audible range isn't perfectly reconstructed; or that human ears are capable of things that no one tested them for before. I think both of those things are unlikely, but the best way to test that would be a simple line level matched, direct A/B switchable, double blind listening test between Redbook and high sampling/bit rate audio. Go to it tiger! Achieve that and you'll be the most famous audiophile in the world! Maybe they'll add a KeithEmo corollary to the Nyquist theory.


----------



## RRod

Wait, are we suddenly being cast as advocates of the Theiss paper, the most stinky of the papers from the hi-res meta-analysis? What's next: I'm shilling for Oohashi?

Find me someone who is just totally convinced of the superiority of hi-res, and I'll offer to sneak in their house and change all their material to 128k Opus or AAC. Anyone care to take wagers on whether any of them would notice without some kind of blue light or flashing text letting them know?


----------



## jimmers

KeithEmo said:


> ...
> (And, yes, I can easily generate a 5 uS sound pulse using various methods.)
> ...


Not really interested in the the arguments generally, I have 99+% of my music as CD or rips thereof so there ends my practical/investment interest in "high res".

I am interested as to what amplification and transducers you used to "easily generate a 5 uS sound pulse" and the measurement techniques and equipment you used to verify you had created a 5 uS sound pulse.
I assume by "pulse" you mean something like this:






If you can actually generate a 5 uS sound pulse wouldn't transmission through the air alter the profile?
And as for hearing it wouldn't you be hearing resonances in your hearing equipment triggered by it?

Just wondering...


----------



## Don Hills

jimmers said:


> ... Just wondering...



The point usually missed is that such a pulse is not valid in digital form. It cannot be accurately digitised - it contains frequencies (energy) above half the sampling frequency of the ADC at any real-world sampling rate. If you run such a pulse through an ADC and plot the resulting sample values produced, you get a graph that looks exactly like the graph you get after running the same invalid pulse in digital form through a DAC. And the timing precision of the pulse after passing through the ADC - DAC chain is, for all practical purposes, exact. 
Shannon and Nyquist showed that as long as you keep all components of the input signal below half the sampling frequency, you can reconstruct the original signal perfectly - not just in terms of amplitude, but in terms of temporal relationships too. They only addressed sampling, and assumed infinite resolution in amplitude. With a digital signal the precision is limited by the number of quantisation amplitude steps. The actual best time resolution for a 16 bit, 44.1khz PCM channel is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time.


----------



## dprimary

bigshot said:


> There's a third option... There shouldn't be any audible difference on any recording.
> 
> I've supervised more mixes than I can count in some very good sound studios in Hollywood. The last step is to output the mix to 16/44.1 and for everyone involved to compare it to the original still in the board for final sign off. If I ever heard a difference between the two, I would have thrown up a red flag, as would have the engineers and talent. The equipment in the room is always carefully calibrated to be consistent and perfect. It represents the reference standard. We never spent much time worrying about how a mix would sound on uncalibrated equipment or DACs that performed out of spec because the range of error would be so broad, there would be no point. We approved 16/44.1 on the reference system and made sure it matched everyone's intentions. And the bounce down never sounded different at all.
> 
> I think you're operating beyond the range of reality. It's great to finesse the details, but they have to be perceivable. And the level of finessing that makes sense for a recording studio is greater than the level required to play back that recording in the home. I can see arguing for the need to keep noise floors down in a mix where you're boosting levels on multiple channels, but when I sit in my living room and listen to an album, audibly transparent is audibly transparent.



This is the difference between recording engineers and audiophiles. When engineers output to 16/44.1 from 24/96 source and they hear a difference they know something is wrong and start looking to find what is broken or set wrong. When an audiophile manages to here something wrong, they view it proof they have super human hearing, not that something is broken.


----------



## dprimary

amirm said:


> That is true.  Unlike high frequency limit that is quite sudden and absolute, the low frequency sensitivity remains, albeit at very high thresholds.  Here is a composite of various research from the last 20 years or so on audibility of low frequency through ear alone (i.e body sensation excluded):



I would like to know how they test for low frequency sensitivity yet block out body sensation.


----------



## dprimary

pinnahertz said:


> Close mic'ed drum hits.
> Room noise isn't specified as SPL, but there are 10dB studios in the world.  The one I last worked in was NC15 (we don't use NC anymore).
> 
> Yup, and it's not even a challenge.



You are either measuring dB SPL or pascals. I always measure flat and add the curves later. We don't use NC anymore because NC masks the problem. You can easily have an NC 10 room and it be completely unsuitable for foley recording. Anytime someone uses NC or A weighting they are just trying to meet some compliance not solve the problem.


----------



## pinnahertz

KeithEmo said:


> I should note something here......
> 
> When I'm talking about those "5 uS timing differences" I am _NOT_ talking about jitter or timing_ errors _between channels.
> 
> ...


Actually that's not quite it.  The apparent location of a sound is dependant on the localization mechanism that is dominant, and that depends on the frequency of the sound.  "Duplex Theory".


KeithEmo said:


> Now let's assume that I start with an unreasonably abrupt impulse (let's call it 5 uS).
> (And, yes, I can easily generate a 5 uS sound pulse using various methods.)
> Even though most of the energy in that impulse will be at inaudible frequencies, enough will extend into the audible range that we will hear it as a click.
> And, if I delay that click in one channel or the other, it will seem to shift locations - between left and right.


The spectrum of a 5us pulse would be placed well above the frequency band where both ITD and ILD cross over, so it's location would be dominated by ILD (assuming it could be transduced and heard at all), which 5us of interchannel delay alone will not change.  The apparent location shift won't be quite what you expect, or allude to. In fact the spectrum of that pulse is above the audible range.  You'll hear a click if you generate it only because of the nonlinearity of any transducer.  The click you hear is a byproduct, a distortion.  In any case a 5us interchannel time delay is far, far below hearing angular position resolution.


KeithEmo said:


> However, because that click actually falls between two samples at 44k, we will _NOT_ be able to precisely reconstruct its time from our 44k sample rate recording.
> When we apply our band limiting, that impulse will indeed be spread out into a longer waveform that extends over multiple samples.
> And, by looking carefully at that new waveform, we will be able to infer where the original impulse occurred in time.


But because of its frequency including whatever a band limiting filter does to it, and the fact that the only thing changed is ITD, which is not the dominant localization method at that frequency, the apparent image shift will not happen that way...or at all.  It simply is not an issue, ignoring the fact that such a pulse does not occur in any acoustic event.  Hence, I strongly suspect, our inability to localize it based on ITD alone.


KeithEmo said:


> HOWEVER:
> 1) in order to do so we will have to make certain assumptions about the filter we used
> (we'll assume that, if there is equal amplitude in two samples, then the pulse was equidistant in time between them - but this assumption relies on our filter spreading the energy symmetrically in time)
> 2) the new waveform will be very different than the original
> ...


Again, you seem to be ignoring how spacial localization actually works.  Our brains do not always "look for those edges" to determine anything, especially when those "edges" never occur in life.  What you're getting at here is a possible limitation to digital systems that falls outside of the application.


KeithEmo said:


> The result of all this MAY be that our brains end up being less accurate in their estimation of where sound objects are located in space.
> The result could be that objects seem to be in different locations, or that we perceive the location of individual instruments as being less distinct.
> (A similar effect occurs with "stereoscopic 3D video"; when the various depth cues conflict, even slightly, which they often do, the image seems "less distinct and less real".)


No, your visual parallel with 3D video is incorrect.  3D suffers from a divergence in two simultaneous localization cues, convergence and focus position.  The operate together, but 3D projection demands they separate.  There is no parallel in audio. 


KeithEmo said:


> Note that, if you accept the current "spectrum analyzer" model of the human ear (with a bunch of "detector hairs", each of which is "tuned" to a distinct frequency).
> The frequency which the "top hair" responds to will determine the highest frequency we can detect as a continuous sine wave.
> However, that number says nothing about the _TIME RESOLUTION_ (how quickly, and how accurately, our brain can respond to _WHEN_ a particular hair was excited.)
> 
> ...


I think there is some serious grasping for straws going on here.


----------



## pinnahertz

KeithEmo said:


> I'm not jumping back into anything.....
> 
> Someone specifically asked whether there would be any reason whatsoever for someone to purchase higher quality content "even if they couldn't hear the difference on their current equipment".
> My reply is that it still makes sense to purchase it, even if you don't hear any difference on your current equipment,_ IF_ you expect that you might later have equipment on which you _WILL_ hear a difference.
> ...


And right after that you'll want to be sure you can get your transducer to generate that 30kHz signal, without changing it's level or distorting it, and get that signal all the way to the listening year.  That's where this really falls apart.  There are transducers that have some 30kHz response, but it's hardly flat and distortion free.  Until you get clean transducers you're test includes all their distortion products added to the signal.


KeithEmo said:


> Errrrr..... I disagree with your final assertion entirely.
> 
> If you want to ask "whether high-res audio makes sense for the average consumer" then by all means lets look at statistics cubes.
> _HOWEVER_, if we're talking about a scientific claim about whether "the difference is audible", a _SINGLE_ well documented and repeatable example is enough to establish that it is.
> ...


I don't think any scientist or engineer would accept a single example as anything but an anomaly.


KeithEmo said:


> Of course, we absolutely require the best possible cross section of humans.......
> (If it turns out that only left handed midget harp players from Burundi can hear it, we wouldn't want to miss that by not including at least one of them.)
> From the data so far, it seems quite possible that "audiophiles who are certain they can hear a difference" may _NOT_ be the best possible subjects.
> (I might suggest running a few tests using school-age children as test subjects - since younger humans have been documented as generally having better high-frequency hearing than older humans.)


…and you'll be generating the afore mentioned data array.


----------



## pinnahertz

dprimary said:


> You are either measuring dB SPL or pascals. I always measure flat and add the curves later. We don't use NC anymore because NC masks the problem. You can easily have an NC 10 room and it be completely unsuitable for foley recording. Anytime someone uses NC or A weighting they are just trying to meet some compliance not solve the problem.


I used SPL because I meant SPL.  I clearly stated, and you quoted, that we don't use NC anymore.  But we do use PNC, which is a modification of NC and is more stringent.  You can measure any way you want, but if you're working with constructing studios with architects and HVAC contractors, you need to work with PNC.  I seriously doubt an NC 10 room would be unacceptable for foley, especially given that most foley effects have nothing below 150 Hz.  We have filters for that.


----------



## dprimary

frodeni said:


> Great. Now you are getting it. Yes, for classic stereo reproduction using speakers there is some form of soundstage. And no, there is hardly any soundstage as by physical space, outside the head of the listener, when using headsets. Not for regular recordings.
> 
> What I am arguing, is simply record every instrument in mono, with as little room acoustics as possible, and simply place the instrument or artist on the fly, or by pre calculation by the dimensions of the head of the listener. And, yes, finally, someone who gets the point, that you would have to do this on the fly, as to correct for head movement, if the listener is to experience the sound source as fixed in physically space. That is perfectly possible, if using vectors and math, and a sensor in the headset for head movements.




This has been done many times, I find it to be exaggerated and fake sounding. Go to AES or NAB, somebody is always demoing some great new 3D or VR audio. I think it was Dolby demoing VR for headphones. It is always exaggerated. You want it sound real, place your microphones to capture what you want 360 hemispherical, spherical. Map the physical locations on the microphone. Place loudspeakers at the inverse of the microphones a few feet out and sit in the center. It will be real enough you will be looking over your shoulder.


----------



## jimmers

Don Hills said:


> The point usually missed is that such a pulse is not valid in digital form.



The point missed in this case is what I was talking about.

 It was nothing to do with ADC, DAC or any other DSP, it was purely about the generation, propagation  and reception of a 5 uS *SOUND *pulse*.*
BTW  a 5 uS pulse is a totally valid digitally, a pulse is a common digital output, I can make one with a single monostable multivibrator or a digital file ..0.0,0,0,1,0,0 ... read at 200 kHz, with either fed into an electro-acoustical transducer. What I was querying is what kind of electro-acoustical transducer could accurately perform the task, and how it would be received unadulterated by the listener.


----------



## dprimary

pinnahertz said:


> I used SPL because I meant SPL.  I clearly stated, and you quoted, that we don't use NC anymore.  But we do use PNC, which is a modification of NC and is more stringent.  You can measure any way you want, but if you're working with constructing studios with architects and HVAC contractors, you need to work with PNC.  I seriously doubt an NC 10 room would be unacceptable for foley, especially given that most foley effects have nothing below 150 Hz.  We have filters for that.



PNC is an improvement, but it is still a curve. I often have to come in after the fact to find out why there is a problem. HVAC contractors rarely meet the specification, so I have the requirements put in their contract. 
Most movies are overhyped audio, big footsteps pushed up 30dB with HVAC rumble and you have a problem. 
What is the old saying "In theory there is no difference between theory and practice. In practice there is."


----------



## dprimary

jimmers said:


> The point missed in this case is what I was talking about.
> 
> It was nothing to do with ADC, DAC or any other DSP, it was purely about the generation, propagation  and reception of a 5 uS *SOUND *pulse*.*
> BTW  a 5 uS pulse is a totally valid digitally, a pulse is a common digital output, I can make one with a single monostable multivibrator or a digital file ..0.0,0,0,1,0,0 ... read at 200 kHz, with either fed into an electro-acoustical transducer. What I was querying is what kind of electro-acoustical transducer could accurately perform the task, and how it would be received unadulterated by the listener.




If it is truly unadulterated, I would expect the listener to hear nothing.


----------



## Don Hills

jimmers said:


> ...  BTW  a 5 uS pulse is a totally valid digitally, a pulse is a common digital output, I can make one with a single monostable multivibrator or a digital file ..0.0,0,0,1,0,0 ... read at 200 kHz, with either fed into an electro-acoustical transducer. What I was querying is what kind of electro-acoustical transducer could accurately perform the task, and how it would be received unadulterated by the listener.



..0,0,0,0,1,0,0 ... is an invalid digital (audio) data stream. It cannot be generated by sampling an analogue signal. Forget about doing this digitally. 
As for accurately transducing this pulse to acoustic waves, I don't know of any commercially available transducers that have a flat response from DC to the MHz range. Even if you had one, attenuation of sound in air rises rapidly with frequency. The transducer would have to be placed against the listener's ear. 

By this point, you're totally off into the weeds. The ear simply doesn't have any mechanism by which to perceive 200 KHz signals.


----------



## jimmers (Nov 4, 2017)

KeithEmo said:


> I should note something here......
> 
> Now let's assume that I start with an unreasonably abrupt impulse (let's call it 5 uS).
> (And, yes, I can easily generate a 5 uS sound pulse using various methods.)
> ...





dprimary said:


> If it is truly unadulterated, I would expect the listener to hear nothing.



That's what I was questioning, maybe I was too subtle (?)

As for my other quoter, try reading what I wrote.

Edit:

Sorry I overlooked this was a "Discussion in 'Sound Science'" I arrived here via a link in another thread.

cue: 
"Back to Earth Back to Reality".


----------



## gregorio

KeithEmo said:


> HOWEVER:
> 1) in order to do so we will have to make certain assumptions about the filter we used
> (we'll assume that, if there is equal amplitude in two samples, then the pulse was equidistant in time between them - but this assumption relies on our filter spreading the energy symmetrically in time)
> 2) the new waveform will be very different than the original
> ...



An audiophile myth which you keep repeating and which I've explained is nonsense:
1. Agreed, assuming a linear phase filter but a minimal phase/apodizing doesn't, you only get post-ringing, not pre-ringing. 
2. What original?
3. What original?
3a. What beginning edge of the impulse?
3b. Not that I'm aware of. All the research I'm aware of suggests that ITD, measures/compares phase relationships of the signal reaching different ears. How can the brain measure/compare  edges when there are no edges?
3c. A suggestion based on edges which don't exist.
3d. A further suggestion based on edges which don't exist!

A sine wave does not have an "edge"! Modulated sine waves do not have an edge! Anything other than sine waves cannot exist; they cannot travel through air, even if they could, your ear drum cannot respond to them, an analogue current can only be sine waves and our transducers (mics, speaker drivers) can only respond to or recreate sine waves. So, what "original" waveform are you talking about in 2 & 3? You CANNOT be not talking about an original waveform, you can ONLY be talking about digital data which cannot exist as a waveform and then you're complaining that this data which cannot be a waveform becomes distorted when we try and turn it into a waveform, huh? Let's put it another way, let's say we invented a theoretical system and filter which could perfectly reconstruct your impulse, what then? What are you going to convert it into? You can't convert it into an analogue electrical signal because you'll loose your "edges", you cannot use some other method because your speakers/headphones will distort your edges, as will the air and your ear drums.



KeithEmo said:


> ... if the starting time of the impulse cannot be determined distinctly, then we will have the equivalent of a blurry image when we attempt to compare them



And how can we have anything other than a "blurry image"? ONLY "blurry images" can travel through air, ONLY "blurry images" can be recorded, reproduced and heard. Think about it for a moment! With a square wave you have an instantaneous rise time ("edge"), a speaker cone would have to be in two different places at the same instant in time to accurately reproduce it, the molecules in the air would have to be in two different places at the same instant in time to transfer that square wave, so would your ear drums and so would the electrons in the analogue electrical current.

G


----------



## gregorio

71 dB said:


> Why does it only work on continuous sine waves? What goes wrong with other signals? ... I tested this on Audacity. I created pink noise at 96 kHz.



I'm not sure how that's a test of "other signals", pink noise IS continuous sine waves.

G


----------



## Darren G (Nov 5, 2017)

gregorio said:


> And how can we have anything other than a "blurry image"? ONLY "blurry images" can travel through air, ONLY "blurry images" can be recorded, reproduced and heard. Think about it for a moment! With a square wave you have an instantaneous rise time ("edge"), a speaker cone would have to be in two different places at the same instant in time to accurately reproduce it, the molecules in the air would have to be in two different places at the same instant in time to transfer that square wave, so would your ear drums and so would the electrons in the analogue electrical current.
> 
> G



This^^^   Yes, and mics, transducers, and our ears do not have infinitely fast rise/fall times either, hence frequency limited.


----------



## 71 dB

gregorio said:


> I'm not sure how that's a test of "other signals", pink noise IS continuous sine waves.
> 
> G


The difference of an impulse and white noise is phase. Both have flat magnitude spectrums.


----------



## 71 dB

jimmers said:


> I can make one with a single monostable multivibrator or a digital file ..0.0,0,0,1,0,0 ... read at 200 kHz, with either fed into an electro-acoustical transducer.


Hmm.. …what analog signal bandlimited to frequencies below 100 kHz as required by sampling theory is going to give you ...0.0,0,0,1,0,0… ? Such analog signal does not exist.

If you put it to DAC, you get sinc-like ringing analog sound depending on your reconstruction filter. If you sample it back, you get a sinc-like sample values.


----------



## bigshot

CDs are for bums because they can't reproduce cosmic waves!


----------



## jimmers

71 dB said:


> Hmm.. …what analog signal bandlimited to frequencies below 100 kHz as required by sampling theory is going to give you ...0.0,0,0,1,0,0… ? Such analog signal does not exist.
> 
> If you put it to DAC, you get sinc-like ringing analog sound depending on your reconstruction filter. If you sample it back, you get a sinc-like sample values.





jimmers said:


> As for my other quoter(s), try reading what I wrote.
> .


----------



## frodeni

dprimary said:


> This has been done many times, I find it to be exaggerated and fake sounding. Go to AES or NAB, somebody is always demoing some great new 3D or VR audio. I think it was Dolby demoing VR for headphones. It is always exaggerated. You want it sound real, place your microphones to capture what you want 360 hemispherical, spherical. Map the physical locations on the microphone. Place loudspeakers at the inverse of the microphones a few feet out and sit in the center. It will be real enough you will be looking over your shoulder.



You missed the point, as to virtually reproduce the physics of sounds, by using a headset, compensating for the physics of the head of the listener, on the fly or pre recording. As for using mono recordings, that is how greater parts of the industry work, as it is, right now.


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## KeithEmo

You're making a very basic - and false - assumption.
The original analog signal is a series of measurements of the pressure of the air over time.
It has no constraints and no limitations.... and no band limitations... and no windows limitations.
It is NOT a SinC function... it is completely arbitrary.
Set off a string of fire crackers.
Each pop is a single pressure wave, which expands, and eventually hits your ears.
What follows is a whole bunch of odd little squiggles in pressure as that wave bounces around and interacts with other stuff.
If I had a "perfect oscilloscope" I could draw a "perfect" picture of it.
There is no sound whatsoever before the first pop.

Things like "windowing errors" and "Gibbs ringing" do not exist in the original.
They are ERRORS that result from the conversion into digital.

1) 
I'm going to hit a bell......... now.
In order to make an "accurate digital representation" of that signal - sort of......
You can sample it.
When you then reconstruct those samples, in order to do so perfectly, and get back a "perfect" version of the original (in terms of energy distribution) your reconstructed signal would have to extend backward in time.
Forget the practicalities of that.......
The original bell hit had ZERO energy before I hit the bell; your reconstruction does; therefore your reconstruction has an error.
(The bell was NOT ringing before I hit it... but, in your reconstructed signal there is ringing before the bell hit; therefore they are NOT the same.)
Therefore, the ONLY question is whether the error that we know exists is audible or not.

2)
A DAC does NOT use a SinC function.
The output of a DAC is NOT "a sum of sample-weighted sinc-functions with various delays"
The DAC (chip) outputs a stream on analog voltages - one for each sample you feed to it.
The DAC (chip) does not put out signal before it receives the fist signal (even if a "real" SinC reconstruction would require it to do so).

I'm not a mathematician....... but I believe that the "problem" is that the SinC function of a non-continuous waveform must extend forward and backward in time to infinity.
(The SinC function of a continuous sine wave needn't do that..... which is why the theory works perfectly for continuous sine waves.)

I can give you a more ridiculous - but still valid - example.......

Let's design the most ridiculous filter imaginable.
It will be a super-duper-hyper-narrow bandpass filer.
It will pass 400.0000000000 Hz, with a cutoff of a million dB per octave.
I'm too lazy to do the math, but you will find that, due to the tradeoff between time resolution and sharpness
....our fun filter will take SEVERAL SECONDS to ring up to approximately full output level once it receives a 400 Hz input signal
....and our fun filter will ring detectably for several seconds after the signal stops (it will actually ring forever, but I've made sure it will ring powerfully enough that it will be easy to see).

I now create a tone burst that is 40 cycles of a 400 Hz tone (it exists for 0.1 seconds).
If I play it from reasonably good speakers in an anechoic chamber it will seem to start and stop quite suddenly.
I can create my signal by taking the output of a signal generator set to 400 Hz and gating it at the zero crossing point to pass ten full cycles and then stop.
(I'm going to gate it using an FET for a switch.)

Now I'm going to send this signal to my fun filter.

The input of my filter will be a 0.1 second set of ten sine wave cycles of a 400 Hz tone.
The output will NOT.
It will increase in level gradually and decrease (continue to ring) for several seconds.
The INFORMATION it contains will be the same (which satisfies Nyquist and Shannon).
(Nyquist and Shannon don't actually specify how long I have to wait for all of my information to "accumulate" or "reconstruct".)
However, the FORM of that information will be very different...... which may or may not satisfy a human being.

Basically, at the risk of being intuitive, the information theory says that, as long as you follow certain constraints, the SAME INFORMATION will still be there.
HOWEVER, their definition of the term "information" isn't intuitively what you might think.
I could post this message in Braille, or in MIME encoding...... the same information would be there...... but it would LOOK quite different.
Likewise, Nyquist & Shannon make a statement about the information.... but not about how our signal SOUNDS, or whether it is AUDIBLY the same as the original.

(When we design DACs, we do our best to design the filters and such so that the output also SOUNDS audibly similar.
Besides following the constraints, we follow other constraints that are based on acoustics and human perception.
For example, we don't spread out a tick over ten seconds - because we know that, even if the information content is the same, it will SOUND different.



71 dB said:


> I feel sick and tired today, so it's hard to think. The analog signal is a sum of sample-weighted sinc-functions with various delays. I don't get why this works only for sine waves.
> Time errors shouldn't be dependent on delay. Why would a filter cause different time error for delayed signal? Doesn't make sense. That would require time-variant filters. Maybe it's all because the sinc -functions are actually windowed versions. In that case we can increase window size and reduce the error, make it as small as we want (need).


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## KeithEmo

Bingo.
But sampling theory does NOT say that "all analog signals must be band limited".
I can mechanically or electrically generate a signal whose voltage values at consecutive sample points are 0,0,0,0,1,0,0.
Rather, that is a limitation of what the input of the DAC will accept, and of the PARTICULAR sample rate you have chosen.
Essentially, by choosing to convert your analog signal into a digital signal at a specific sample rate, you have "signed up" for a series of requirements and constraints.
YOU have chosen to accept the constraint of band limiting your input to 100 kHz by choosing a sample rate that cannot successfully handle frequencies above that. 



71 dB said:


> Hmm.. …what analog signal bandlimited to frequencies below 100 kHz as required by sampling theory is going to give you ...0.0,0,0,1,0,0… ? Such analog signal does not exist.
> 
> If you put it to DAC, you get sinc-like ringing analog sound depending on your reconstruction filter. If you sample it back, you get a sinc-like sample values.


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## KeithEmo

You're falling into a trap that a lot of people fall into.........

"The analog signal is a sum of sample-weighted sinc-functions with various delays."
WRONG!
The analog signal is AN ANALOG SIGNAL.
It may be EXPRESSED, for certain purposes, as "a sum of sample-weighted sinc-functions with various delays".
You could restate this as "for any analog waveform, you can define a sum of sample-weighted sinc-functions which will recreate it with any given level of accuracy".

One of the "catches", in very simple intuitive terms, is this....
If you start with a very simple continuous sine wave, that starts in the infinite past, and extends into the infinite future, you can express it as a very simple SinC function (it's going to have a single component with a single coefficient - it's a simple sine wave).
However, as soon as you make it in the least bit more complex, then the set of SinC functions you're going to need to describe it also get more complex - very rapidly.
For example - you can describe one sine wave with a single function.
And you can describe two sine waves mixed together with two.
Yet, to define a square wave, you need an INFINITE number (because a square wave is made up of a base frequency plus ALL odd order harmonics - an infinite number of them).
(It's not at all that "in ONLY works on sine waves"; it's that, as soon as you move away from simple sine waves, the errors increase very quickly.... and, as you approach short transients, the errors become very significant.)

Another catch is this...
In order to make things work "perfectly" all we have to do is to bandwidth limit the input signal.
Let's assume that, if we can"limit the bandwidth perfectly" no audible errors will be introduced.)
However, in reality, there is no such thing as a perfect filter; this means that whatever filter we use to do our band limiting will also introduce errors.
In fact, there is an inverse correlation between time errors and the sharpness of a filter.......
If we choose a filter with a gradual slope, the time errors will be minimal, but the band limiting will be imperfect - and so we introduce aliasing errors.
And, if we choose a filter with a sharp slope, we can minimize aliasing, but at the cost of various time errors like ringing.
(In other words, the same theory that says this would all work perfectly if we had a perfect filter also says that there's no such thing as a perfect filter.)

While there are certainly filters that introduce more than the bare minimum of errors - there is in fact a minimum for a given situation beyond which no filter can ever be "more perfect".
Beyond that we're trading things off.
For example, a typical "apodizing filter" "trades pre-ringing for post-ringing".... which is a nice way of saying that it reduces pre-ringing (which is believed to be audible).... 
but at the cost of increasing post-ringing even more (but post-ringing is believed to be less audible - partly because post ringing occurs on most "natural" sounds, while pre-ringing does not).

In simplest terms.....
Start with a 100 Hz square wave.... which is a collection of every odd order harmonic of 100 hz.
If you were to try to reproduce it using a SinC you would need an infinite number of terms (impractical).
So band limit it.
It will now be incorrect because you have truncated the series......
You will also have introduced additional errors by whatever band limiting filter you used.
(Now you get to decide WHICH of those errors is audible.........  note that the errors we have now are not simple "extra harmonic components" like normal distortion.)



71 dB said:


> I feel sick and tired today, so it's hard to think. The analog signal is a sum of sample-weighted sinc-functions with various delays. I don't get why this works only for sine waves.
> Time errors shouldn't be dependent on delay. Why would a filter cause different time error for delayed signal? Doesn't make sense. That would require time-variant filters. Maybe it's all because the sinc -functions are actually windowed versions. In that case we can increase window size and reduce the error, make it as small as we want (need).


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## RRod (Nov 6, 2017)

KeithEmo said:


> You're falling into a trap that a lot of people fall into.........
> 
> "The analog signal is a sum of sample-weighted sinc-functions with various delays."
> WRONG!
> ...



Every time error has a corresponding frequency error. In the case of audible ringing, you hear it because it's at a frequency where you can hear it. This is why you don't hear the pre-ringing of a steep lowpass filter at, say, 20kHz, because you can't hear at or near 20kHz, at least not at amplitude of the ringing.


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## KeithEmo

First off.....

*"Nyquist" says:* A continuous time signal can be represented in its samples and can be recovered back when *sampling* frequency fs is greater than or equal to the twice the highest frequency component of message signal.
Note that it's talking about "continuous time signals" and not impulses.

Here's a paper that outlines some of the ways in which Nyquist is commonly mis-interpreted.....
https://www.audiostream.com/content...-about-it-tim-wescott-wescott-design-services

Also note that Nyquist talks about information ...... which is not exactly the same as "sound" or "audio quality".

I could take this posting, and convert it into Olde English font, or even Braille, and it would contain all of the original INFORMATION.
However, it would LOOK very different.
Likewise, I can convert a square 1 millisecond pulse into a ten second XinC waveform, and it will contain all of the original energy.
In fact, if it's done very well, it will even contain all of the original time information (you will be able to figure out where the original energy was located).
However, the waveform itself will be very different, and the difference will be obvious if you do anything other than look at 'the basic information it contains". 




bigshot said:


> I think they said that 44.1 sampling rate can perfectly reproduce any sine wave representing sound that the human ear can hear. That's good enough for me. I tend to listen to music with human ears.
> 
> It doesn't matter if I use fancy equipment or cheap equipment. I haven't ever seen any evidence that super audible frequencies are audible, so higher sampling rates don't have much purpose for my human ears. Feel free to believe that pigs can fly and you can hear things that other humans might not be able to, but there's plenty of evidence on Nyquist's side and on the side of audiologists who have established the thresholds of human hearing. The ball is in your court to either prove that Nyquist is wrong and the audible range isn't perfectly reconstructed; or that human ears are capable of things that no one tested them for before. I think both of those things are unlikely, but the best way to test that would be a simple line level matched, direct A/B switchable, double blind listening test between Redbook and high sampling/bit rate audio. Go to it tiger! Achieve that and you'll be the most famous audiophile in the world! Maybe they'll add a KeithEmo corollary to the Nyquist theory.


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## KeithEmo

Electrically I can generate that using a wide variety of devices. 
The simplest would be a type of logic gate called "a one shot".
I could also simply "dial it up" on most signal generators.
Getting it into the air from there would be a bit trickier..... I suspect a piezoelectric transducer could do it.

Alternately, I could generate it directly as a sound in the air by using some sort of electric discharge... or a tiny explosive charge.
I'm betting I could also generate something similar by mechanical means..... (A tiny little hammer hitting a tiny little metal plate -  well damped to suppress ringing.)

And, yes, it will be altered significantly by being passed through the air.
And, yes, speakers and microphones are indeed going to introduce a lot of ringing and other interesting alterations.
(I have heard of test microphones that are claimed to have a response that extends beyond 100 kHz.)

I would also note.... and this was my point..... that a single non-repeating impulse has components of _ALL_ frequencies in it. 
(Although it has the most energy at higher frequencies.)
Therefore, assuming we could deliver it to the air, that 5 uS pulse would have _SOME_ energy in the audible range (although perhaps not very much).
(For example, if I were to repeat my 5uS impulse every 1/500 second, there would be a 500 Hz modulation component... if it never repeats, in theory the components would extend down to 0 Hz, at every decreasing amplitudes.)





jimmers said:


> Not really interested in the the arguments generally, I have 99+% of my music as CD or rips thereof so there ends my practical/investment interest in "high res".
> 
> I am interested as to what amplification and transducers you used to "easily generate a 5 uS sound pulse" and the measurement techniques and equipment you used to verify you had created a 5 uS sound pulse.
> I assume by "pulse" you mean something like this:
> ...


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## KeithEmo

There you would be incorrect.

A non-repeating impulse contains energy at every frequency - including low frequencies.




dprimary said:


> If it is truly unadulterated, I would expect the listener to hear nothing.


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## KeithEmo

Honestly, I'm not grasping so much as suggesting that the situation is not as simple and definite as many people seem to prefer to believe.

I disagree with your assertion about my comparison to 3D audio... although you may well be right that we tend to ignore the weaker cue.
I agree that the dominant cue takes precedence..... but has anyone studied whether a dissonant less-dominant cue is totally ignored, or whether it has some small effect.
(With 3D video, you see both cues; with audio, do you not notice the less dominant cue at all, or does it act as a distraction, and render the use of the dominant cue less certain?
For example, does the dominant cue take longer for your brain to process because there is another conflicting cue present, or does the conflicting cue affect the results?)
(It seems to get rather complicated in practice)......

https://msu.edu/~rakerd/Hartmann et al, Transaural Experiments and a Revised Duplex Theory.pdf

Also, while a square wave only contains the primary frequency and odd order harmonics, a single impulse contains frequency components at every frequency - down to DC.
(Although, with a 5 uS impulse, the energy present at audible frequencies is going to be pretty low.)




pinnahertz said:


> Actually that's not quite it.  The apparent location of a sound is dependant on the localization mechanism that is dominant, and that depends on the frequency of the sound.  "Duplex Theory".
> 
> The spectrum of a 5us pulse would be placed well above the frequency band where both ITD and ILD cross over, so it's location would be dominated by ILD (assuming it could be transduced and heard at all), which 5us of interchannel delay alone will not change.  The apparent location shift won't be quite what you expect, or allude to. In fact the spectrum of that pulse is above the audible range.  You'll hear a click if you generate it only because of the nonlinearity of any transducer.  The click you hear is a byproduct, a distortion.  In any case a 5us interchannel time delay is far, far below hearing angular position resolution.
> But because of its frequency including whatever a band limiting filter does to it, and the fact that the only thing changed is ITD, which is not the dominant localization method at that frequency, the apparent image shift will not happen that way...or at all.  It simply is not an issue, ignoring the fact that such a pulse does not occur in any acoustic event.  Hence, I strongly suspect, our inability to localize it based on ITD alone.
> ...


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## KeithEmo

Hmmmmm..... so let's see.
If I start out with a short signal that occurs between two sample points, when I band limit it it will be transformed into a much more spread out SinC "equivalent" waveform.
But I'll still be able to tell when it occurred because that spread out waveform will be spread out across multiple samples (so I can see where 'the highest spot on the hill" is).
However, if I use an apodizing filter, then the pre-ringing will be converted into post ringing... so my "mill will no longer be symmetrical.
I wonder how I'm supposed to tell where my short impulse signal occurred (where between the two samples that "bracketed" it).
Your apodizing filter has removed the information we need in order to retrieve that time information.

In the sense that all waveforms can be described as some sort of complex combination of sine waves you are correct.
However, there are microphones and speakers that respond well past 20 kHz. 

I'm talking about the edges of the sound amplitude envelope.
(Which, in the case of a single short impulse, would be the same as the leading edge of the impulse itself.)
(And, yes, if I start with 0 VDC, then gate a single cycle of a sine wave, the beginning of that waveform will be "an edge".)



gregorio said:


> An audiophile myth which you keep repeating and which I've explained is nonsense:
> 1. Agreed, assuming a linear phase filter but a minimal phase/apodizing doesn't, you only get post-ringing, not pre-ringing.
> 2. What original?
> 3. What original?
> ...


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## 71 dB

KeithEmo said:


> In simplest terms.....
> Start with a 100 Hz square wave.... which is a collection of every odd order harmonic of 100 hz.
> If you were to try to reproduce it using a SinC you would need an infinite number of terms (impractical).
> So band limit it.
> ...



Incorrect to who? Batman? God? Even if 16/44.1 audio could miraculously play all of the infinite odd harmonics, your square wave would be quite low-pass filtered after headphones or speakers. Amplifiers have finite bandwitdth. Even if your speakers had a response up to 50 kHz, the radiation pattern would be interesting to say the least. Your ears would need to be "on the radiation axis" to get anything. High frequencies attenuate in the air. Your ears are a low-pass filter etc. What do you need harmonics above 20 kHz for? I don't need. The bandlimited version is correct for my ears.

The correct way to create bandlimited square waves is to sum harmonics up to 20 kHz. That way you don't need to use filters.


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## castleofargh

KeithEmo said:


> You're making a very basic - and false - assumption.
> The original analog signal is a series of measurements of the pressure of the air over time.
> It has no constraints and no limitations.... and no band limitations... and no windows limitations.
> It is NOT a SinC function... it is completely arbitrary.
> ...


in real life everything *does* have limits and band limiting of sort, what you say in introduction is clearly false. air isn't some superconductor of vibrations, a mic or the eardrum are band limiting the signal, I can't think of one thing that will agree with your statement. I'm limited by the speed of my movement when I go to hit the bell. the bell is clearly limited by how fast and how much it will flex under my hit. the all thing will for the duration of me still pushing into the bell, be subjected to a deceleration over time(from the bell resisting my movement, and then starting to resonate at it's own freq). maybe I can't define that movement entirely with my redbook sampling, components above F/2 would be one obvious reason to start with. 
all your examples are in my eyes, the Dirac pulse argument revamped again and again. of course you won't have an ideal steady state behavior while looking at the most extreme transient part of any system. hitting the bell, a square wave or anything of the sort amounts to showing stuff outside of Nyquist's theorem.  they let you win the fidelity argument. so if you want to discuss increasing fidelity, do that. but if you want to discuss audibility, then that's a big pile of empty rhetoric. 
if the question is really potential audibility, then a measurable error isn't motive to avoid it if as far as we know it's not audible. who cares about the missing ultrasonic components of a transient that a human brain probably never perceived in real life?  the "better safe than sorry" approach suggesting to act just in case, is caution when the occurrence is credible based on statistics and facts, but paranoia when we act based on nothing concrete. the more I learn each year, the more I give my vote for the later. 

I'll put my eternal question here: if something is missing as we can easily demonstrate, and is audible in music as many believe it to be. why is it so hard to pass blind test? that's the question we have to answer because a blind test is what will prove audibility.


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## reginalb

KeithEmo said:


> ...There are microphones and speakers that respond well past 20 kHz...



But are there singers whose voices produce such frequencies? How about instruments? I guess you could generate the sound electronically, so perhaps that's the next big thing in EDM, supersonic frequencies that nobody can hear! 

You could be the Emperor's New DJ.


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## JaeYoon

reginalb said:


> But are there singers whose voices produce such frequencies? How about instruments? I guess you could generate the sound electronically, so perhaps that's the next big thing in EDM, supersonic frequencies that nobody can hear!
> 
> You could be the Emperor's New DJ.


Gonna be some seriously empty crowds.
Imagine an EDM event, where all the people in crowds are like "where's the music at?".
The DJ says it's playing right now, and all the sounds are above 20 khz XD


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## KeithEmo

I think you will find plenty of instruments which produce harmonics that extend up well above 20 kHz (cymbals certainly do).
(Likewise, a synthesizer set to produce square waves can do so - depending on the synthesizer.)
The only question is whether humans can tell when those harmonics are missing or not.

Personally I'd rather risk recording some frequencies I can;t hear rather than missing some that I can.



reginalb said:


> But are there singers whose voices produce such frequencies? How about instruments? I guess you could generate the sound electronically, so perhaps that's the next big thing in EDM, supersonic frequencies that nobody can hear!
> 
> You could be the Emperor's New DJ.


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## ev13wt

JaeYoon said:


> Gonna be some seriously empty crowds.
> Imagine an EDM event, where all the people in crowds are like "where's the music at?".
> The DJ says it's playing right now, and all the sounds are above 20 khz XD




Thing is, there will that one guy that hear it and starts dancing. Then you have a bunch of people dancing. Then you have the dude that said: “And those who were seen dancing were thought to be insane by those who could not hear the music.” get up and bitch slap the DJ.


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## KeithEmo

I've got to mention something here.........
I simply don't understand why some people seem so resentful about this entire subject.

I've never driven my car over 90 mph..... yet I still see no reason why it's "bad" that my car "can" go that fast.
In general, in almost every other subject, most people agree that you're better off if your tools can actually deliver, not just adequate performance, but performance that's BETTER than necessary.
It goes by names like "safety margin" and "margin for error" and "headroom" and even "clearance".
And who would really buy a car that can only go 56 mph?

So, why, even if you believe that we can't hear above 20 kHz, is it so awful to allow some safety margin.
If I were recording bats, and found out that their cries extended to 46 kHz, I would buy a microphone whose response extended to 60 kHz; I wouldn't buy one that went up to 46.1 kHz.
So, why, even if humans can only hear to 20 kHz, doesn't it make equal sense to make recordings that extend "well above" 20 kHz..... just in case.... to leave a little safety margin?
Why would anyone specifically choose to use a sample rate that's "just barely good enough"?

There is a reason why the 44.1k sample rate was chosen for CDs.....
The reason is that, with the constraints of the technology at the time CDs were invented, the time/space constraint on CDs was considered to be important.
They couldn't have used the next-higher standard sample rate without reducing the storage time on a standard CD below one hour - which had been established as a target requirement.
Using the 44.1k sample rate, they were able to fit over an hour on a disk, and still deliver frequency response that was a tiny bit above the bare minimum necessary.
The 48k sample rate was already in use on DAT tapes, and was considered to be a sort of standard; they would have used that except that, if they had, they couldn't have fit an hour on a CD.
In fact, most movie audio (on DVDs) is still standardized at 48k.... and not 44.1k.

However, when you're talking about download FILES, that constraint simply doesn't exist.
(in fact, even originally, it was strictly tied to "fitting a complete album on a CD")


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## KeithEmo

You might want to check out something called "a mosquito ringtone"  :    http://www.freemosquitoringtone.org/
(You don't dance to them; school kids use them so the adult teacher can't hear their phone ring.)



ev13wt said:


> Thing is, there will that one guy that hear it and starts dancing. Then you have a bunch of people dancing. Then you have the dude that said: “And those who were seen dancing were thought to be insane by those who could not hear the music.” get up and bitch slap the DJ.


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## bigshot (Nov 7, 2017)

KeithEmo said:


> *"Nyquist" says:* A continuous time signal can be represented in its samples and can be recovered back when *sampling* frequency fs is greater than or equal to the twice the highest frequency component of message signal. Note that it's talking about "continuous time signals" and not impulses.



That's fantastic because the music I listen to with my stereo consists of waveforms that Nyquist can perfectly reproduce. This makes me happy and satisfies me. I wouldn't expect a car to drive on the ocean. It wasn't designed for that. It was designed to drive on roads. I wouldn't expect digital audio to reproduce impulses and square waves and other theoretical things we can't really hear. I would expect it to reproduce what it was designed to reproduce- music.



KeithEmo said:


> I think you will find plenty of instruments which produce harmonics that extend up well above 20 kHz (cymbals certainly do).
> (Likewise, a synthesizer set to produce square waves can do so - depending on the synthesizer.)
> The only question is whether humans can tell when those harmonics are missing or not.



No, they can't. Studies have shown that super audible frequencies have no impact at all on audio fidelity of recorded music. And auditory masking makes sure that you probably can't even hear some of the upper level harmonics in the audible range either. The truth is that ears don't have a brick wall filter. They fade out starting at around 15kHz or so. When you listen to music, the core frequencies are MUCH more important to perceived audio fidelity than the top octave. In fact, of the ten octaves or so that humans can hear, the top octave is the least important. Why to people expend so much energy worrying about the least important thing? Makes no sense.

You can rest easy knowing that CD sound is all you need. When you look for the truth and use your ears, OCD and pathological attention to detail isn't a problem any more. You have a lot more time and attention to spend on things that really matter- like appreciating great music. Music can be appreciated even without perfect fidelity. Acoustic Caruso 78s never fail to impress me. I can't imagine life without the Duke Ellington sides from the early 30s. And Bruno Walter's first act of Die Walkure from 1935 has never been bettered regardless of the sound technology advances. We are VERY fortunate to be living in an era of perfect recorded sound. Why go looking for theoretical imperfections? Appreciate how great digital audio is.

Focus on the road, not the clarity of the windshield.


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## KeithEmo (Nov 7, 2017)

In all fairness, I don't think "people expend a lot of energy worrying about the top octave".
It's simply that, regardless of our personal limitations, the asserted goal of high fidelity is to reproduce the music as accurately as possible.
Therefore, some of us just like the idea that our gear will do so.
(So far I haven't seen a thread bemoaning how selling us audio amplifiers with a frequency response past 20 kHz is a major scam.)
Why is this such a big deal ONLY when it comes to high-res files?

My reply is simply.........
What's the big deal about paying an extra $5 for a 96k file instead of a 44k file?
I'm not personally all that sure I'd hear a difference......
But I'd rather pay an extra five bucks to get the version that's "twice as good as I need" instead of the one that's "just barely as good as I need" (most of my other equipment is also a bit better than I strictly need).
And, no, I'm not totally convinced that it's quite good enough.
But, even if it is, what's so awful about paying a few cents more to buy a lot of extra safety margin?
And what's the big deal about "only" being able to fit 20,000 albums on a $150 hard drive?
(If I just read this thread, I'd think 96k files were selling for $500 each - like fancy power cables.)



bigshot said:


> That's fantastic because the music I listen to with my stereo consists of waveforms that Nyquist can perfectly reproduce. This makes me happy and satisfies me. I wouldn't expect a car to drive on the ocean. It wasn't designed for that. It was designed to drive on roads. I wouldn't expect digital audio to reproduce impulses and square waves and other theoretical things we can't really hear. I would expect it to reproduce what it was designed to reproduce- music.
> 
> No, they can't. Studies have shown that super audible frequencies have no impact at all on audio fidelity of recorded music. And auditory masking makes sure that you probably can't even hear some of the upper level harmonics in the audible range either. The truth is that ears don't have a brick wall filter. They fade out starting at around 15kHz or so. When you listen to music, the core frequencies are MUCH more important to perceived audio fidelity than the top octave. In fact, of the ten octaves or so that humans can hear, the top octave is the least important. Why to people expend so much energy worrying about the least important thing? Makes no sense.
> 
> ...


----------



## Whazzzup

I like a clean windshield, but thats my server is for


----------



## JaeYoon

KeithEmo said:


> In all fairness, I don't think "people expend a lot of energy worrying about the top octave".
> It's simply that, regardless of our personal limitations, the asserted goal of high fidelity is to reproduce the music as accurately as possible.
> Therefore, some of us just like the idea that our gear will do so.
> (So far I haven't seen a thread bemoaning how selling us audio amplifiers with a frequency response past 20 kHz is a major scam.)
> ...


@gregorio 
Explained in his 24bit vs 16bit thread that the difference is not twice as much!


gregorio said:


> Hopefully you're still with me, because we can now go on to precisely what happens with bit depth. Going back to the above, when we add a 'bit' of data we double the number of values available and therefore halve the number of quantisation errors. If we halve the number of quantisation errors, the result (after dithering) is a perfect waveform with halve the amount of noise. To phrase this using audio terminology, each extra bit of data moves the noise floor down by 6dB (half). We can turn this around and say that each bit of data provides 6dB of dynamic range (*4). Therefore 16bit x 6db = 96dB. This 96dB figure defines the dynamic range of CD. (24bit x 6dB = 144dB).
> 
> So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. This is the misunderstanding made by many. There are no extra magical properties, nothing which the science does not understand or cannot measure. The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and *nothing else*. This is not a question for interpretation or opinion, it is the provable, undisputed logical mathematics which underpins the very existence of digital audio.
> 
> So, can you actually hear any benefits of the larger (48dB) dynamic range offered by 24bit? Unfortunately, no you can't. The entire dynamic range of some types of music is sometimes less than 12dB. The recordings with the largest dynamic range tend to be symphony orchestra recordings but even these virtually never have a dynamic range greater than about 60dB. All of these are well inside the 96dB range of the humble CD. What is more, modern dithering techniques (see 3 below), perceptually enhance the dynamic range of CD by moving the quantisation noise out of the frequency band where our hearing is most sensitive. This gives a percievable dynamic range for CD up to 120dB (150dB in certain frequency bands).


----------



## castleofargh

KeithEmo said:


> I've got to mention something here.........
> I simply don't understand why some people seem so resentful about this entire subject.
> 
> I've never driven my car over 90 mph..... yet I still see no reason why it's "bad" that my car "can" go that fast.
> ...


to be clear, almost nobody cares that a file is at 44.1 or 48khz, take price increase and storage cost/limits out of the equation and I'd be fine with DXD on my computer. why not? if it has no negative impact why should I care which format it is? you won't find many people going "oh no I have better resolution without any negative impact, what will I do?" ^_^
 your argument that I understand, is that there is no point nowadays in risking it all on the lower format. it's simple an intuitive enough. also it's an opinion so it doesn't have to be right or wrong and we're all free to have our own.
but then I read stuff like this:





KeithEmo said:


> What's the big deal about paying an extra $5 for a 96k file instead of a 44k file?


  and I angrily raise my fist in front of my screen with about the same impact on you that high-res tends to have on my ears. I purchase music for the sound I perceive, not to look at a spectrum in owe. so when I fail to perceive a difference(consistently over several years), the 96k file is worth to me exactly the same, not 5$ more. on the other hand if every 3 albums the 44 or 48k version saves me enough to buy a 4th one. now that's audible evidence right there, and it will pass all blind tests.


----------



## bigshot (Nov 7, 2017)

KeithEmo said:


> What's the big deal about paying an extra $5 for a 96k file instead of a 44k file?



If the mastering is better it's worth it. If the mastering is the same it isn't. If the mastering is worse than the CD, it really isn't. I've found examples of all of these things. Buying a 24/96 track is no guarantee that it sounds better or even as good as the CD. It would be a lot more productive to do listening tests comparing different releases of specific albums to determine which one has the best mastering, than it would to do listening tests to see if we can hear inaudible sound.

This is the core problem with audiophiles. They focus on some tiny detail and ignore the elephant in the corner. They go down the slippery slope of justifying incremental improvements even when it gets to the point of absurdity. When I buy audio equipment, which isn't often any more because my system is tricked out now, I focus on getting something that does the job cleanly, efficiently and inexpensively. Once I get there, I don't worry about specs beyond my ability to hear. Every $5 I save from not having to buy high bit depths that are basically just the same thing stuffed with packing peanuts is $5 I can spend on more great music.

Spend money where it counts. If you have OCD and want to worry about things you can't hear, buying HD tracks won't solve your problem. It will only take you one more step down the rabbit hole.

And yes, people do worry about that top octave of human hearing more than any other. I see it here all the time with people who try to justify the octave beyond our ability to hear. Listen to what 17kHz sounds like and then tell me that is important in your music. Not bloody likely.

96kHz is one octave more than 44.1... seven measly notes that I can't even hear. Why should I pay $5 more for that? I can't figure out why I would possibly pay almost a dollar for each note I can't hear.


----------



## Don Hills (Nov 7, 2017)

KeithEmo said:


> ... There is a reason why the 44.1k sample rate was chosen for CDs.....



And it's not the reason that you said.
Kees Immink and others have made it quite clear that disc playing time was not the reason for the sample rate choice. It was dictated by the requirement to be able to store the digital data on videotape.


----------



## 71 dB

I think the requirement of "continuous time signals" in Nyquist sampling theory is misunderstood by advocates of high-res / analog audio.

It doesn't mean something like sine waves that started in the distant past and will continue forever. It means continuity in mathematical respect, that a function doesn't "jump" from one value to another in no time. That doesn't happen in real world. Instruments can produce only "continuous time signals". Acoustic waves can be only "continuous time signals". So, the only thing to worry about in digital audio is_ bandlimiting_. If we do that correctly, everything will be fine. Worrying about impulses (mathematical constructions which can't exist in reality) is absurd.


----------



## KeithEmo

I've seen both claims made at different times..... and the technical arguments do seem to favor the fact that 44.1k "fits well" with the data structure on videotape.
However, neither relates to optimizing the audio performance...... 
And, for that matter, neither relates to the practical limits of audio circuit design.

Before oversampling was invented, the reconstruction filter used on a CD player or DAC had to be flat to 20 kHz.
However, because of the 44.1k sample rate chosen, it also had to have significant attenuation at and above 22 kHz.
These two requirements, taken together, make the required reconstruction filters difficult to design, difficult to manufacture at reasonable cost, and prone to unpleasant side effects.
Selecting a much higher sample rate would have made is simpler to design higher-performance reconstruction filters and cheaper to build them.
It was a bad choice, in terms of design flexibility... even if you don't consider having a little bit of safety margin in terms of frequency response worthwhile.
Even if you really only consider response up to 20 Khz to be important..... a higher sample rate will still make it easier to achieve it with fewer filter design issues and side effects.
(Modern oversampling was basically invented as a solution to this problem.... and would have been unnecessary if a high enough sample rate had been chosen to begin with.)



Don Hills said:


> And it's not the reason that you said.
> Kees Immink and others have made it quite clear that disc playing time was not the reason for the sample rate choice. It was dictated by the requirement to be able to store the digital data on videotape.


----------



## Don Hills

KeithEmo said:


> I've seen both claims made at different times..... ( ) However, neither relates to optimizing the audio performance...... And, for that matter, neither relates to the practical limits of audio circuit design. ...



Anyone who wants to find more than they ever wanted to know about the history and technical details of CD, DVD and Blu-Ray design and development can visit http://www.turing-machines.com/indeximmink.html 
Start with papers 50 and 105. The origin of the "Beethoven's 9th" legend is covered. 

Regarding sample rate and bit depth, it's pretty clear that they were pushing the limits of what was available and affordable at the time to develop a consumer oriented playback system. Developing a new low cost recording system to enable a higher sample rate was a bridge too far. So, 44.1 KHz. You can wish all you want, but that's the way it was. 



KeithEmo said:


> ... Modern oversampling was basically invented as a solution to this problem. ...



The first Philips CD player used 2x oversampling.


----------



## bigshot

Lots of stuff there. Thanks!


----------



## JaeYoon

Don Hills said:


> Anyone who wants to find more than they ever wanted to know about the history and technical details of CD, DVD and Blu-Ray design and development can visit http://www.turing-machines.com/indeximmink.html
> Start with papers 50 and 105. The origin of the "Beethoven's 9th" legend is covered.
> 
> Regarding sample rate and bit depth, it's pretty clear that they were pushing the limits of what was available and affordable at the time to develop a consumer oriented playback system. Developing a new low cost recording system to enable a higher sample rate was a bridge too far. So, 44.1 KHz. You can wish all you want, but that's the way it was.
> ...


Someone is on fire! Those papers are very good reads!


----------



## 71 dB

KeithEmo said:


> It was a bad choice, in terms of design flexibility... even if you don't consider having a little bit of safety margin in terms of frequency response worthwhile.



It was a good compromise in it's day some 40 years ago and provided a huge improvement in audio transparency on consumer market for normal people.



KeithEmo said:


> Even if you really only consider response up to 20 Khz to be important..... a higher sample rate will still make it easier to achieve it with fewer filter design issues and side effects.
> (Modern oversampling was basically invented as a solution to this problem.... and would have been unnecessary if a high enough sample rate had been chosen to begin with.)


Yes, but they had their reasons to choose 44.1 kHz. Fortunately it is able to provide transparent sonic experiment for human ears despite of filter difficulties. It is easy for us who live the age of 384 kHz DXD recordings to complain about low sample rate, but digital technology was different in the late 70's. Let's be happy about the fact they didn't choose 32 kHz!

The problems of 44.1 kHz are totally overblown out of proportion by hi-res business and other besserwissers who don't understand properly the scale of the issue. For me 16/44.1 is a transparent audio format. If it isn't for your bat ears then bad luck man!


----------



## KeithEmo

I don't disagree with you at all......

I have no complaints at all that 44.1k was chosen as the sample rate for CDs in the 1970's (and horses were a remarkably effective and efficient form of transportation in 1820).
However, as you point out, we now live in an age where there are lots of other alternatives, many of which are better.
The fact that CDs were an excellent compromise in the 1970's doesn't mean that we should stop trying to find something better.

One obvious example is oversampling.
It's impossible to design a practical, cheap, and effective reconstruction filter to operate directly with digital audio sampled at 44.1k. 
Therefore, oversampling was invented as a way to sidestep this problem.
However, with modern technology, it would be much simpler to just use a higher sample rate.
Taking a 44.1k input and upsampling it to 192k is still somewhat complicated - and the process itself offers many choices and compromises.
It would be much simpler to distribute audio recorded at a 192k sample rate... 
And then convert it using a simpler DAC which didn't need oversampling to deliver good performance.

In short, the reasons that were "compelling" for choosing 44.1k in 1978 are simply no longer true. 
44.1k is currently being used as the sample rate of choice simply because "it's what our grandfathers used"... like horses... and gasoline powered cars.
(And that argument is becoming less compelling every year, as less and less music is played from CDs.
After all, unlike plastic discs, there's no particular benefit to standardize digital audio files at a single sample rate at all.)

I find it humorous how so many people seem convinced that "modern music producers" are "all trying to rip us off by foisting yet another audio format on us".
(I seem to recall people saying pretty much the same thing about CDs in the 1970's.) 



71 dB said:


> It was a good compromise in it's day some 40 years ago and provided a huge improvement in audio transparency on consumer market for normal people.
> 
> 
> Yes, but they had their reasons to choose 44.1 kHz. Fortunately it is able to provide transparent sonic experiment for human ears despite of filter difficulties. It is easy for us who live the age of 384 kHz DXD recordings to complain about low sample rate, but digital technology was different in the late 70's. Let's be happy about the fact they didn't choose 32 kHz!
> ...


----------



## Whazzzup (Nov 8, 2017)

im not big on oversampling and its smoothing effects. however AK 380 does sound pretty smooth and warm with it


----------



## castleofargh

KeithEmo said:


> I don't disagree with you at all......
> 
> I have no complaints at all that 44.1k was chosen as the sample rate for CDs in the 1970's (and horses were a remarkably effective and efficient form of transportation in 1820).
> However, as you point out, we now live in an age where there are lots of other alternatives, many of which are better.
> ...



sure, we all remember the years of arguments to prove an audible difference between CD and k7 tapes, or CD vs vinyls. it was "pretty much the same thing" like apples and oranges are pretty much the same. 
what I can concede is that the marketing was already full of crap. high res didn't invent misguiding people with different masters for example.


----------



## bigshot

KeithEmo said:


> I have no complaints at all that 44.1k was chosen as the sample rate for CDs in the 1970's (and horses were a remarkably effective and efficient form of transportation in 1820). However, as you point out, we now live in an age where there are lots of other alternatives, many of which are better.



Not "better" just "easier". The end result is the same. It's like compressed audio. Yes it's easier to just encode in PCM. Modern compression codecs are much more complicated, but the end result is the same. At high enough bitrates, the sound is audibly transparent.

All that really matters is what you can hear. If it's audibly transparent and the usability is convenient, it's the exact same thing. There's absolutely no reason to go for a higher sampling rate now. All it will get you is sound to make your dog sit up and take notice.


----------



## OddE

bigshot said:


> There's absolutely no reason to go for a higher sampling rate now. All it will get you is sound to make your dog sit up and take notice.



-That, I think, is an overly simplistic view.

You will also need a bigger data plan and a ditto hard drive. (Though at today's prices, the latter isn't of much concern even if 32/768 or DSD-4096 is your poison of choice...)


----------



## Whazzzup

I have a cat


----------



## castleofargh

Whazzzup said:


> I have a cat


when your cat goes from fur ball to Freddy Krueger for no apparent reason, now you know that it's because your music was low passed too soon.


----------



## bigshot

My dog hates the movie Das Boot. The low frequency submarine rumbles and depth charges make her hide under the couch.


----------



## bigshot

OddE said:


> -That, I think, is an overly simplistic view.



There's no advantage to having superaudible frequencies in your music. There are only disadvantages. See the link CD AUDIO IS ALL YOU NEED in my sig file.


----------



## Whazzzup

I like das boot even u12. my cat hides when there is thunder, growls when the door bell rings, and chirp purrs when i pick her up.


----------



## RRod

71 dB said:


> I think the requirement of "continuous time signals" in Nyquist sampling theory is misunderstood by advocates of high-res / analog audio.
> 
> It doesn't mean something like sine waves that started in the distant past and will continue forever. It means continuity in mathematical respect, that a function doesn't "jump" from one value to another in no time. That doesn't happen in real world. Instruments can produce only "continuous time signals". Acoustic waves can be only "continuous time signals". So, the only thing to worry about in digital audio is_ bandlimiting_. If we do that correctly, everything will be fine. Worrying about impulses (mathematical constructions which can't exist in reality) is absurd.



Bandlimited signals cannot be time-limited, and we don't capture any infinitely-long signals, so we're never truly bandlimited. "Correctly" thus means "with aliasing that is inaudible", which I think arguably we can attain.



castleofargh said:


> but then I read stuff like this:  and I angrily raise my fist in front of my screen with about the same impact on you that high-res tends to have on my ears. I purchase music for the sound I perceive, not to look at a spectrum in owe. so when I fail to perceive a difference(consistently over several years), the 96k file is worth to me exactly the same, not 5$ more. on the other hand if every 3 albums the 44 or 48k version saves me enough to buy a 4th one. now that's audible evidence right there, and it will pass all blind tests.



If the nominal end-product of mastering is a 24/96k file, then it should cost precisely $0 extra for said file, and said file shouldn't cost more than the CD mastering of said file because why should less work cost more money?


----------



## 71 dB

KeithEmo said:


> I don't disagree with you at all......
> 
> I have no complaints at all that 44.1k was chosen as the sample rate for CDs in the 1970's (and horses were a remarkably effective and efficient form of transportation in 1820).
> However, as you point out, we now live in an age where there are lots of other alternatives, many of which are better.
> ...



Just buy 24/96 downloads then if CD feels grandfather audio to you. Why 192 kHz? That's comical overkill just to distribute 20 kHz audio band when 60 kHz sampling rate would already allow relaxed reconstruction filters. 16/44.1 reached a level of transparency beyond which there's hardly anything to gain so why bother?


----------



## 71 dB

RRod said:


> Bandlimited signals cannot be time-limited, and we don't capture any infinitely-long signals, so we're never truly bandlimited. "Correctly" thus means "with aliasing that is inaudible", which I think arguably we can attain.


None of my bandlimited CDs play more than about 80 minutes making them time-limited. How is this possible? It's possible because we live in a "noisy" reality where some aspects of mathematical theories become irrelevant. Maybe the CD I played 10 years ago is still ringing in the universe at level -70000073284780000282370000230 dB, but that's irrelevant even at atomic level and the damn CD stopped playing in my ears 10 years ago.


----------



## 71 dB

RRod said:


> If the nominal end-product of mastering is a 24/96k file, then it should cost precisely $0 extra for said file, and said file shouldn't cost more than the CD mastering of said file because why should less work cost more money?



You charge _as much as you can _to maximaze your profit. People are willing to pay more for bigger numbers because they are ignorant about the non-existing benefits of high-res audio and that's why they are charged more.


----------



## OddE

bigshot said:


> There's no advantage to having superaudible frequencies in your music. There are only disadvantages. See the link CD AUDIO IS ALL YOU NEED in my sig file.



-You are preaching to the choir. (See the 2nd paragraph of my post, quoted below.)

My (poor, as it were) attempt at snark was simply to suggest that there were other possible side effects to hi-res than the discomfort of pets - like increased download and storage costs...



OddE said:


> You will also need a bigger data plan and a ditto hard drive. (Though at today's prices, the latter isn't of much concern even if 32/768 or DSD-4096 is your poison of choice...)


----------



## KeithEmo

I was thinking the same thing.

I've owned quite a few different DACs - both now and in the past - and many of them sound distinctly different from each other. 
Ignoring the question of which is _better_, or why, there are all sorts of audible differences between DACs (even between those that measure so well that their flaws per-se should not be audible).

I would agree that the THD, noise, and frequency responses of most modern DACs are so close to perfect that there _shouldn't_ be any audible difference....
Yet there are in fact consistent differences.
Obviously this suggests that, perhaps, we _aren't_ measuring _everything_ that matters.
(I've even owned DACs where different filter choices sounded audibly different - even though all of them were arbitrarily close to perfect according to the measurements.)

At a recent company event we had a station set up where listeners could compare Emotiva's current DC-1 DAC to a prototype of next year's model.
Excluding a few new features, the basic performance of both is "beyond reproach", and very similar - very flat frequency response, very low noise, very low THD.
Yet, even when perfectly level matched, and playing ordinary 16/44k audio selections, the vast majority of listeners noticed a difference between them most of the time.
(And, while most listeners preferred the new model, a few actually preferred the other one, yet ALL of them described the difference consistently in similar terms.
In other words, pretty much everyone, including a few who liked the older model better, heard and described hearing the same difference between them.)

I can only see two possible conclusions here:
1) there is something audible that we're failing to measure
2) we're operating under some false assumptions about the audibility of small differences in the stuff we _ARE_ measuring

Of course, our individual hearing varies significantly, as do our audio systems, and the content we listen to.
And, yes, in the above-mentioned demonstration, we did cherry pick specific music selections that served to emphasize the differences.
However, they were played from the same files, using the same computer and USB card, the same player program, the same powered speakers, and a switching device that uses mechanical relays - with the only difference being the DAC.
(Yes, we even used the same USB cables.)



SilverEars said:


> Which DAC do you use?


----------



## KeithEmo

Of course people who sell things are going to charge as much as they can.
And, yes, it should actually be cheaper to sell a copy of the 24/96k master than to go to the trouble to re-sample it to 16/44k.
(Although, arguably, the marginal cost of allowing the user to decide which one he or she wants to download is quite trivial.)

Note that the cost of distributing an album as a file download is also far cheaper than the cost of pressing, shipping, shelving, and picking a physical disc.
Incidentally, the cost to produce physical CDs, with jewel cases, and pretty colored label inserts, is about $2 per disc in 500 quantities. 
(In fact, I think an excellent case could be made that, all else being equal, it's a _LOT_ cheaper to distribute a 24/192k download to end users than a 16/44k physical CD disc.)
It's pretty obvious that current pricing is based mostly on "what the market will bear".... and "what the sellers are convinced the music license is worth".

You do, however, always have to keep the business model in perspective.
For example, the two current largest music streaming services _LOST_ money last year (apparently because their license costs are more than their profit margin).
This means that either their prices will have to go up, or they'll have to figure out how to improve efficiency, of the licensing costs they pay will have to go down - or they will go out of business.
At the moment "high-res downloads" are what we call "a market differentiator".... which means that either they're going to charge more for them, or hope to win customers from their competitors by offering them (if their competitors don't), or both.

Ignorance is ignorance...... and, after all, most of the differences they tout on TV commercials for various products are made up..... so why should we hope for better from the music industry?
High-res audio files are simply "this year's model".
Of course the buyer should think for themself.



71 dB said:


> You charge _as much as you can _to maximaze your profit. People are willing to pay more for bigger numbers because they are ignorant about the non-existing benefits of high-res audio and that's why they are charged more.


----------



## KeithEmo

And, by the way, what if your pet _LIKES_ the more accurate rendition of those animal recordings?



OddE said:


> -You are preaching to the choir. (See the 2nd paragraph of my post, quoted below.)
> 
> My (poor, as it were) attempt at snark was simply to suggest that there were other possible side effects to hi-res than the discomfort of pets - like increased download and storage costs...


----------



## KeithEmo

Actually your comment about the reconstruction filter is only partly correct.
From an engineering perspective, it would be easier to design a filter that was much shallower, and use a much higher sample rate to go with it.
(Most DACs internally oversample 44k input signals by as much as 8x.... which works out to a 384k "internal" sample rate.)

I would also like to point out that many of the concerns I see described here are both perfectly valid and also specious.... depending on the specific circumstances involved.
For example, my Internet download account is unlimited, and the movie I downloaded to watch last night took up more space than my entire music collection would - even at 192k (a typical Blu-Ray quality movie averages about 40 GB).
However, if I had a portable player with limited capacity, or was streaming audio to a phone with a data limit, I might well consider using high quality lossy compression.
My only concern there is that I be offered the option - rather than having someone else decide that they know with 100% certainty what I do and don't need.
I have no desire whatsoever to convince everyone to buy high-res downloads.
However, I am concerned that someone will convince someone else that 'there's no reason to sell them" because "they know better".
(Specifically because I'm not convinced that they really do know better.)

My Nissan Versa does a perfectly adequate job of getting me to work on time.
So why would anyone bother to buy a Lexus?
Personally, I would agree, which is why I drive a Nissan......
But I'm not trying to make a case for why Lexus should shut down their factory ("because nobody really needs one").



71 dB said:


> Just buy 24/96 downloads then if CD feels grandfather audio to you. Why 192 kHz? That's comical overkill just to distribute 20 kHz audio band when 60 kHz sampling rate would already allow relaxed reconstruction filters. 16/44.1 reached a level of transparency beyond which there's hardly anything to gain so why bother?


----------



## Whitigir

Simply put, real life sound waves have infinite bits .  True sin waves


----------



## RRod

71 dB said:


> None of my bandlimited CDs play more than about 80 minutes making them time-limited. How is this possible? It's possible because we live in a "noisy" reality where some aspects of mathematical theories become irrelevant. Maybe the CD I played 10 years ago is still ringing in the universe at level -70000073284780000282370000230 dB, but that's irrelevant even at atomic level and the damn CD stopped playing in my ears 10 years ago.



Well yes, exactly.


----------



## pinnahertz

KeithEmo said:


> At a recent company event we had a station set up where listeners could compare Emotiva's current DC-1 DAC to a prototype of next year's model.
> Excluding a few new features, the basic performance of both is "beyond reproach", and very similar - very flat frequency response, very low noise, very low THD.
> Yet, even when perfectly level matched, and playing ordinary 16/44k audio selections, the vast majority of listeners noticed a difference between them most of the time.
> (And, while most listeners preferred the new model, a few actually preferred the other one, yet ALL of them described the difference consistently in similar terms.
> ...


Interestingn conclusions. 

So, your "test" was at a company event, the comparison was fully sighted, and therefore the results fully biased.  Your results weren't 100%, but were they actually tallied?  And from THAT you get the differences are either not being measured or that small measurable differences are audible?

Could your methods BE any less scientific?


----------



## 71 dB

KeithEmo said:


> My Nissan Versa does a perfectly adequate job of getting me to work on time.
> So why would anyone bother to buy a Lexus?
> Personally, I would agree, which is why I drive a Nissan......
> But I'm not trying to make a case for why Lexus should shut down their factory ("because nobody really needs one").



Cars are different in many ways while listening tests show no reliable difference between 16/44.1 and high-res beyond placebo effect. That's why Lexus makes sense.


----------



## bigshot (Nov 9, 2017)

KeithEmo said:


> My Nissan Versa does a perfectly adequate job of getting me to work on time.
> So why would anyone bother to buy a Lexus?



A Lexus has better features. If you are buying a more expensive piece of equipment because it has additional features that you like, that is a good reason to spend more. I have an Oppo BDP103D player, and it has the exact same video and sound quality as my $100 Sony blu-ray player, but it plays DVD-A, SACD, MKV on thumb drives and it has Darbee video noise reduction. It has a whole slew of features that justify the price. When I bought it, they also made a BDP105D, which was the exact same player, except the specs were even further into audio overkill. I passed on that one. No point to throwing good money at sound I can't hear.



KeithEmo said:


> my Internet download account is unlimited, and the movie I downloaded to watch last night took up more space than my entire music collection would - even at 192k (a typical Blu-Ray quality movie averages about 40 GB).



You might want to look into Handbrake. If you plan to save that video you downloaded, you can compress it to MP4 format with no real quality loss. You just need to use the proper compression scheme. In fact, if you really understand the settings in Handbrake, you can end up with a MP4 version that is *better* than the uncompressed blu-ray rip.



KeithEmo said:


> At a recent company event we had a station set up where listeners could compare Emotiva's current DC-1 DAC to a prototype of next year's model. Excluding a few new features, the basic performance of both is "beyond reproach"



Too bad the testing procedures weren't!


----------



## KeithEmo

This was not designed nor run as a scientific test (although I sort of wish we had taken the time to do so).
Therefore, this is just a bit of anecdotal evidence.
Note that everyone who posts "I don't hear any difference" is also simply adding one more piece of anecdotal evidence as well.

I should also note that this discussion, and every test which I've seen cited (as flawed as they are), has related to the audibility of differences between different sample rates.
Does anyone here know of any actual test to determine whether there were audible differences between different brands or models of DACs?
If not, then we're ALL just relating various pieces of anecdotal evidence.

However, if you're curious, my "test" was "partially sighted".........
The switch box was in plain sight, but isn't designed to be easily read (the LEDs are relatively small and close together - and were not labelled).
The switch also involved a key-press on the associated computer (so it wasn't as simple as "watching the hand that flips the switch").
In some instances I was narrating the switches - "here's the DC-1...... now here's the DC-2....... now here's the DC-2 playing DSD".
At other times it was simply - "OK.... here's one.... now here's the other... do you notice any difference?"
This all also occurred in what I would call "a presentation situation"....
Both DACs were set up.... and, whenever a significant number of audience members accumulated, I played some sample files for them.
However, at least to me, it does seem significant that the vast majority of listeners claimed to hear a difference, and described the difference they heard in similar terms.
(Nobody was really attempting to TEST for an audible difference because we all found the difference to be relatively obvious - with a wide variety of program material.) 

And, yes, in a QUALITATIVE test such as this, if any number of participants can reliably hear a difference, then the existence of an audible difference has been established.
(They do NOT have to be able to reliably identify which is which - simply that they can reliably distinguish a DIFFERENCE when switching back and forth.
A proper A/B/X test is testing a more complex question - and providing more information in its results - than we are talking about here. 
This is simply a matter of "I'm playing two runs - are they the same or are they different?")



pinnahertz said:


> Interestingn conclusions.
> 
> So, your "test" was at a company event, the comparison was fully sighted, and therefore the results fully biased.  Your results weren't 100%, but were they actually tallied?  And from THAT you get the differences are either not being measured or that small measurable differences are audible?
> 
> Could your methods BE any less scientific?


----------



## bigshot

My evidence may be anecdotal but at least my test was controlled and wasn’t conducted by a company advertising their own products!


----------



## pinnahertz

KeithEmo said:


> This was not designed nor run as a scientific test (although I sort of wish we had taken the time to do so).
> Therefore, this is just a bit of anecdotal evidence.
> 1. Note that everyone who posts "I don't hear any difference" is also simply adding one more piece of anecdotal evidence as well.
> 
> ...



1. Yes, of course.
2. That's "sighted", not partially sighted.  
3. No, not at all.  What you have is called noise in your statistics.
4. A proper ABX test would provide you with actual scientific data.  What you have is more anecdotal and statistically noisy data.


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## KeithEmo (Nov 10, 2017)

re #3.

Err...... no....... because it's not a statistical test; it is a qualitative test.

In a qualitative test, if ANY SINGLE TEST SUBJECT can reliably distinguish a difference, then we have proven the existence of a distinguishable difference. This is an "all or nothing" result. If a single human, on the entire planet, can reliably tell the difference, then we have answered the question "can ANYBODY tell the difference" - in the affirmative. It makes no difference whether we find a billion other subjects who cannot - or a billion other subjects who can - because we have determined that at least one human can. Since the result we're seeking is not statistical in nature, the term "noise" is meaningless. (Of course, we might well choose to expand the test, and so to determine WHAT PERCENTAGE of humans notice the difference... but that is a separate question.)

The only purpose of statistical analysis would be, at the individual level, to determine if our test subject was actually able to distinguish a difference or was simply random guessing.

If I were doing this properly, I would test a whole bunch of subjects - individually. And, if a single one of them could reliably notice a difference, then I would stop the test - because I would have proven the positive assertion by producing an example of someone who could determine the difference. (Even an outlier is "part of the human population" - and so "proves the case".)

re #4.

An ABX test would provide lots of information.... including whether people could distinguish which SPECIFIC sample they were listening to and which they preferred. However, if you're asking a simple question, such as "is there a detectable difference or not", then you don't need such a complex test to do so. (If you're looking for a single result then you're always better off using the simplest protocol that provides it.) 

For example, all you need to do is to play a series of PAIRS of test samples. Assuming we're trying to determine whether "there are audible differences between A and B" we would play a random mix of pairs (A+A, B+B, and A+B). The test subjects would then be asked simply to state whether the two samples were the same or different. There is no need to determine if they can tell which is which - as long as they can detect whether they are the same or different.
Likewise, there is no need to know what criteria they are using to determine similarity.
If they can reliably identify which pairs were "mixed" and which are "the same" then we have established that the two different samples can be reliably distinguished. (Although, assuming that the subjects can reliably determine that the samples are different, FURTHER TESTS to determine how they're distinguishing them would be useful. And I'm sure the Marketing Department would love to know if they consider the difference worth paying extra for - or can even tell which is which.) Note that our initial test has NOT determined "which is better" nor "how big the difference is" - merely that a recognizable difference does in fact exist.

So, for example, if we're testing to see if different brands of DACs sound identical or not, all we need to ask is: "I've just played two samples for you; were they played on the same DAC or different DACs?" If the subject answers correctly a statistically significant percentage of the time - beyond random chance - then we've proven that there are audible differences. The test protocol is much simpler, and answers the actual question we're asking, with no extra complexity.

In this case "are two items identical or not" is a very simple criterion to test for.... The only real pitfall is that you might end up with false-null results if your samples or test protocol are badly chosen. For example, the difference may be undetectable using certain speakers, or a certain amplifier, or certain test samples, or even a certain sample population. (For example, since we all agree that the ability to hear high frequencies decreases with age, it seems to make sense to include at least some children in tests for the audibility of high frequencies; using 60 year old audiophiles is surely a badly chosen test population.)  HOWEVER, any legitimate positive is definitive..... because, if a legitimate difference exists, which can be demonstrated under ANY circumstances, then the two UUTs are "different". (So, for our purposes, if one ten year old child reliably hears a difference, it doesn't matter how many 50 year old audiophiles fail to do so.)

The real purpose of a simple test like that would be to serve as "a qualifying round". If we were TOTALLY UNABLE TO FIND A SINGLE SUBJECT WHO COULD RELIABLY NOTICE A DIFFERENCE, then we would have a null-result. And that null-result would preclude future more extensive testing as long as it remains authoritative. It means that you have not successfully completed the qualifying round. (Once you prove there are differences, there are all sorts of additional things you can test for. Until then, you have nothing to test... beyond perhaps looking for more sensitive tests that might detect the differences that have so far eluded you.)

All we have at the moment is a positive but unreliable result.
However, that is in fact rather different than a reliable negative result.



pinnahertz said:


> 1. Yes, of course.
> 2. That's "sighted", not partially sighted.
> 3. No, not at all.  What you have is called noise in your statistics.
> 4. A proper ABX test would provide you with actual scientific data.  What you have is more anecdotal and statistically noisy data.


----------



## castleofargh

KeithEmo said:


> This was not designed nor run as a scientific test (although I sort of wish we had taken the time to do so).
> Therefore, this is just a bit of anecdotal evidence.
> Note that everyone who posts "I don't hear any difference" is also simply adding one more piece of anecdotal evidence as well.
> 
> ...


you're overreaching with your conclusions. maybe this specific test was very clean in some instances, and maybe people weren't all pre-fed with marketing expectations? but the way you describe it, it's just a nice sighted test while presenting a product. did I misunderstand the protocol involved? 
if I did not, then no the existence of an audible difference hasn't been established. people "hear" differences when there is none to hear all the time when asked if they do. that's why we need a control to take those false positive out. maybe it's there and maybe it's enough for people to reach that conclusion based on sound alone. but as long as the test isn't about sound alone, it isn't conclusive about sound. 

because we seem to get into war those days and it's not my aim at all, I want to add that I'm not trying to suggest your DACs don't sound different. you know better than I do if that's the case or not. my remark is only about the protocol of the experience and the range of conclusions we can reach. and contrary to what it may start to look like(maybe it's irrelevant, maybe you care), I'm not on a crusade against you. I actually enjoy having you posting here, and I'm proud to say that I've learned several things from previous posts of yours. but I'm skeptical first, and fanboy second. ^_^
the wish to disprove everything is not personal, is what I'm trying to say poorly here. my default position, maybe I'm wrong about that and need a more open approach, is to go for the null hypothesis and move from there. so this always makes me look like I'm thinking everything is the same. I'm not. it's just my starting point and I wait for evidence to resist my hypothesis.


----------



## pinnahertz

KeithEmo said:


> re #3.
> 
> Err...... no....... because it's not a statistical test; it is a qualitative test.
> 
> ...


Qualitative testing and statistical analysis are not mutually exclusive, but rather, if any qualitative test is to have any meaning other than biased opinion, it must be statistically significant. One result is statistically too noisy, and in fact completely meaningless.


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## KeithEmo

Hmmmmmm..... maybe because the methodology involved in doing so is flawed.......

We have two "widely accepted claims" about the thresholds of human perception.
Both are based on very specific test conditions.
Yet you want to generalize them to _ALL_ conditions - without offering any sort of proof that such a generalization is valid.

Personally, I would NOT be willing to bet $1 million that someone cannot produce a single living human being who can hear 22 kHz.
(And, no, the fact that it hasn't happened yet does not _PROVE_ that it cannot or will not happen.)

At best you can state that nobody has demonstrated that any of the claimed benefits have been shown to be audible in a properly designed and executed test.
(And I'll cheerfully agree with you there.)

To answer your question.... I don't specifically expect to hear anything... but I can't entirely rule it out based on the currently available evidence either.
(And, even if I can't hear it, I can't rule out the possibility that someone else will. After all, I don't have "perfect pitch" either, yet some humans do.) 



bigshot said:


> Why not just do a little googling and find out what additional benefits higher bitrates and sampling rates offer, then compare that to the thresholds of perception and you'll have your answer.
> 
> higher bitrate = lower noise floor
> higher sampling rate = extended frequency response
> ...


----------



## bigshot (Nov 10, 2017)

KeithEmo said:


> Personally, I would NOT be willing to bet $1 million that someone cannot produce a single living human being who can hear 22 kHz.



I read somewhere that the record for highest frequency heard was 23kHz. But that really isn't all that much. 20kHz to 40kHz is one octave... 7 musical notes. 20kHz to 23kHz doesn't even get from "doe" to "rey". It's a drop in the bucket. That's the thing. You need to keep in mind relative importance. Frequencies above 17kHz are largely irrelevant when it comes to recorded music. A large proportion of the population can't hear them at all, the majority of musical instruments can't produce those frequencies, sound that high doesn't propagate through a room well at all, it's far beyond the point where we can perceive a musical note in sound, and even if those frequencies exist in recorded music they aren't audible because they're masked by lower frequencies. Super high frequencies just don't matter. If you focus on them, you'll be neglecting the things that really do matter.

Frequencies up that high are as useful as teats on a bull hog.


----------



## KeithEmo

I would like to be able to disagree with your initial statement - because I have tried to avoid claiming any conclusions whatsoever.
However, I agree entirely with your statement that "I didn't prove anything".
My only claim is that I failed to rule out the need for further testing... as has everyone else.
Actually, since people claimed to hear differences under certain conditions, it seems reasonable to suggest using those conditions as a starting point for a real test.

My only intent was to offer a counterbalance to various statements from people stating "I didn't hear any difference - so they sound the same".
I simply offered a significant quantity of anecdotal evidence that "a lot of people do seem to believe they hear a difference".
(Mostly to offset the perception that "nobody hears any differences".)
In terms of preference and bias, as with any situation, this one was quite complicated.
Some people would undoubtedly prefer to believe that the new product sounds better; while some, who own the current product, would no doubt prefer to believe they have no reason to upgrade.
I also note that, up until that point, we had been promoting the new product based on added features rather than improved sound..... 
(Of course, we have now adjusted our marketing strategy, based on those results..... we'd be dumb not to mention positive testimonials.)

In fact, I'm not claiming to have presented any evidence at all.... beyond some anecdotal evidence that suggests that actual testing MIGHT turn up something interesting.
(Incidentally, I'm also not at all convinced personally that high-res files do or do not sound audibly different... and neither I nor anyone else has done what I consider to be sufficient testing to prove the point either way.)

However, when the discussion shifts to the question of whether all DACs sound audibly identical, the whole discussion devolves into fantasy.
Clearly all DACs do _NOT_ sound the same, and all DACs do not measure the same, or even close.
(For example, one DAC I owned had a very dull sounding high-end, almost certainly related to the fact that its frequency response was -3 dB at 20 kHz.)
So the real question is whether DACs sound different _IN WAYS NOT OBVIOUSLY RELATED TO DIFFERENCES IN MEASURED PERFORMANCE_.
And, in order to test that question, we must predict what specific measured differences are audible.

We can't simply say "all DACs sound the same" - because obviously they don't.
We need to make a specific claim which can be tested - for example "all DACs which are flat within 0.2 dB from 20-20 kHz, and have a THD below 0.05%, and a S/N ratio above 95 dB, sound the same".
Once we have that specific claim, we can choose a bunch of DACs that meet the required criteria, and then we can see if people can tell them apart.

My experience, and that of many people I know, is that the vague claims being made by a lot of people don't seem to agree with our experience.
I've owned many DACs which had "very flat frequency response; very low noise; and very low distortion" and which still sounded distinctly different to me.
Therefore, either:
1) I (and they) must have been imagining the differences
2) the claimed specs were wrong, and real major measurable differences existed - even though the published specs said otherwise (unless you measure them yourself it's not really a good idea to accept the manufacturer's specs)
3) the claimed specs were right, but the differences were audible, and the assumption that the differences were "below the threshold of audibility" is wrong
4) there were easily measurable differences that simply don't fall under the specs we're currently measuring

I'm also going to offer an interesting example... in the form of a "thought experiment".
As with many thought experiments, this takes the idea to the absurd - but not the impossible.
I'm going to start with a perfectly clean tone burst - 200 milliseconds of a 440 Hz sine wave tone - sampled at 44.1 kHz.
I'm now going to create a really silly filter - a 440 Hz bandpass filter one billionth of a cycle wide, with really sharp skirts, centered at 440 Hz.
If I pass my my 200 millisecond tone through this filter, it will experience some (an absurd amount) of "pre and post ringing".
In fact, I may add several seconds of ringing to my original relatively short burst.
However, assuming I don't screw up, I will have altered the time/energy envelope, but  the frequency, THD, and noise floor, will remain unchanged. 
(I will have changed my 200 msec burst into a twenty second tone that gradually increases to full amplitude, then gradually declines to zero asymptotically, without adding THD or noise or shifting the frequency.)
I'm betting that the difference will be audible.
(Again, note that my filter has NOT altered the frequency response, THD, or noise level......)

To be totally honest here, I feel bad arguing against skepticism, when I personally believe that too many audiophiles are far too credulous.
Skepticism is always a good thing.
(However, that does extend to skepticism against_ ALL CLAIMS_... especially against claims of negative proof... because negatives are damn near impossible to prove.) 
I also agree that, in practical situations, absolutes are often not the best way to discuss things.
It's probably not a good idea for everyone to pay extra for something that 99% of the people can't hear..... unless they happen to be one of the 1%, or just prefer to buy "insurance" against the possibility.....



castleofargh said:


> you're overreaching with your conclusions. maybe this specific test was very clean in some instances, and maybe people weren't all pre-fed with marketing expectations? but the way you describe it, it's just a nice sighted test while presenting a product. did I misunderstand the protocol involved?
> if I did not, then no the existence of an audible difference hasn't been established. people "hear" differences when there is none to hear all the time when asked if they do. that's why we need a control to take those false positive out. maybe it's there and maybe it's enough for people to reach that conclusion based on sound alone. but as long as the test isn't about sound alone, it isn't conclusive about sound.
> 
> because we seem to get into war those days and it's not my aim at all, I want to add that I'm not trying to suggest your DACs don't sound different. you know better than I do if that's the case or not. my remark is only about the protocol of the experience and the range of conclusions we can reach. and contrary to what it may start to look like(maybe it's irrelevant, maybe you care), I'm not on a crusade against you. I actually enjoy having you posting here, and I'm proud to say that I've learned several things from previous posts of yours. but I'm skeptical first, and fanboy second. ^_^
> the wish to disprove everything is not personal, is what I'm trying to say poorly here. my default position, maybe I'm wrong about that and need a more open approach, is to go for the null hypothesis and move from there. so this always makes me look like I'm thinking everything is the same. I'm not. it's just my starting point and I wait for evidence to resist my hypothesis.


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## bigshot (Nov 10, 2017)

KeithEmo said:


> My only intent was to offer a counterbalance to various statements from people stating "I didn't hear any difference - so they sound the same".
> I simply offered a significant quantity of anecdotal evidence that "a lot of people do seem to believe they hear a difference".



And a lot of people believe that they can communicate through telepathy with life forms in an alternate universe.

You have generally accepted science. It is what it is. Then you have claims that contradict that. That's when you require a controlled test to prove that the generally accepted science is wrong. Those of us who have been in this forum for a while, have seen people come in here claiming all sorts of things that contradict established scientific principles. They'll talk and they'll talk, but they never go beyond the anecdotal. Most of them just end up getting mad and stomp off when one of the Sound Science regulars give them a "put up or shut up" challenge. A very tiny percentage are disingenuous and try to trick us. Very few go to the trouble to set up a test for themselves and try to prove something. At best they make a half hearted attempt and then gleefully indulge their bias, making deliberate compromises to undermine any value of the results. Why would they do that? Ego. They don't want to be wrong. Unless they can get past their own ego, they will always have problems with bias.

The truth is, if some generally accepted scientific principle of audibility is wrong, there are a lot of scientists and sound engineers who would like to know that, and they would be eager to hear what the truth is. Prove Nyquist wrong and you'll be famous. Your ego will be fed. But you better do it fair and square. It's easy to pump out a lot of smoke and raise a bunch of questions and blather on and on about things that may or not be true. High end audio salesmen do that all the time. They train audiophools to march into the Sound Science forum and do a parrot paraphrase version of their sales tear sheets. But that doesn't move actual knowledge an inch forward.

My approach is simple. I define what it is that I want... a sound system that plays music in my home and sounds great. Then I focus on understanding what makes a sound system in a home sound great. I don't worry about theoretical sound. I don't try to feed my ego. I only care about results. I focus on what I can hear. I start at the top of the priority list and work my way down. I set up a simple system for doing controlled listening tests and I verify that what I'm doing is working and nothing is sabotaging my efforts that I'm not aware of. When I reach audio nirvana, I'm happy and I listen to a whole lot of music. Sometimes I think I'm an anomaly. There aren't a lot of people out there approaching the problem of recorded sound in a practical and organized way like this. But I'm a producer and that's how my brain works. I have a goal, I break down the steps to reach that goal, I move forward efficiently, I check my work. When I reach the goal, I relax and enjoy my accomplishment. If everyone did that, we would all have great sounding systems, we wouldn't be churning through upgrade after meaningless upgrade, and we wouldn't be wasting a whole lot of money on transistors and circuits that would be better spent on music.

And every DAC and player I’ve ever come across all sound the same. I imagine there are some out there that might sound different, but if I bought one, I’d pack it back up and return it as defective. I require a digital player to be audibly transparent.


----------



## KeithEmo

There actually was a time when 19 kHz was a problem.

The way FM stereo works - and maintains compatibility with FM mono - is interesting. FM stereo is actually delivered as two matrixed channels; a main channel which is mono L+R, and a difference channel L-R.
A monaural tuner is able to simply play the main L+R channel without doing anything special, while a stereo FM tuner extracts the L-R channel, then does the math to derive separate L and R channels.
This way the same signal is compatible with both.
However, in the actual broadcast, the L-R channel is delivered as a signal modulated on a 19 kHz subcarrier, which is part of the main audio signal.
The tuner must demodulate and extract the L-R channel, use the result to recreate the stereo audio signal, and then filter the 19 kHz subcarrier out of the main channel.
However, FM mono tuners, built before the stereo standard existed, failed to filter out the 19 kHz subcarrier at all, and many early stereo tuners didn't do so especially well.
As a result, many vintage FM tuners included a significant amount of 19 kHz noise mixed in with the audio.
The 19 kHz noise wasn't very audible as noise..... but could often be noticed as a sort of pressure in the ear (if you walked too close to the tweeter it almost felt like a jet of air blowing in your ear).
Many tweeters in those days simply didn't reproduce it at all, while others delivered it in a very narrow beam, directly in front of the speaker, due to narrow dispersion at that frequency.....

I would agree that the value of such high frequencies as "musical content" is rather dubious.
However, good high frequency response does contribute to fidelity in some situations: for example, enabling the sound of recorded cymbals to sound "properly metallic", and helping the recorded sound of breaking glass to sound natural.
(A well recorded wire-brush cymbal should be audibly different than the hiss of steam escaping from a relief valve.)
So, if the high frequencies roll off too soon, or too sharply, it can adversely affect how natural a recording sounds.



bigshot said:


> I read somewhere that the record for highest frequency heard was 23kHz. But that really isn't all that much. 20kHz to 40kHz is one octave... 7 musical notes. 20kHz to 23kHz doesn't even get from "doe" to "rey". It's a drop in the bucket. That's the thing. You need to keep in mind relative importance. Frequencies above 17kHz are largely irrelevant when it comes to recorded music. A large proportion of the population can't hear them at all, the majority of musical instruments can't produce those frequencies, sound that high doesn't propagate through a room well at all, it's far beyond the point where we can perceive a musical note in sound, and even if those frequencies exist in recorded music they aren't audible because they're masked by lower frequencies. Super high frequencies just don't matter. If you focus on them, you'll be neglecting the things that really do matter.
> 
> Frequencies up that high are as useful as teats on a bull hog.


----------



## KeithEmo

I don't disagree at all...... however we do need to distinguish exactly what we mean by "generally accepted science" and "absolutely undisputed facts".
For example, several quite popular, and largely informative, videos would have us believe that "there are only three significant types of distortion: THD, IMD, and noise".
While this was true a few decades ago, when both amplifiers and recordings were analog, it is no longer true - because digital technology has introduced several new ways in which the audio signal can be unintentionally altered.
(For example, a faulty surround sound decoder could actually route the sound of a certain instrument in a recording to the wrong speaker; it's clearly wrong, but which category would you place the error in?) 

Generally accepted science does sometimes turn out to be wrong.... or incorrect in a new context.... or sometimes simply insufficient to fully understand a new context.
I've personally been around the audio industry for a long time..... and I've heard plenty of whoppers in both directions. 
Therefore, I'm inclined to do my best to avoid saying things like: "Don't bother to test it; it can't possibly be true."
(But, yes, being human, I do have certain areas where I allow my opinion to overrule that guideline.)
However, I would always agree that you shouldn't take anything "on faith" - especially if ti seems to conflict with established science.

I would also note (I wish I could claim credit for this one): 
Some audiophiles enjoy listening to music, while others seem to listen mostly to their equipment, and only some few seem to do both. 
(And I guess that omits the ones who prefer to _DISCUSS_ the subject rather than listen to it  )




bigshot said:


> And a lot of people believe that they can communicate through telepathy with life forms in an alternate universe.
> 
> You have generally accepted science. It is what it is. Then you have claims that contradict that. That's when you require a controlled test to prove that the generally accepted science is wrong. Those of us who have been in this forum for a while, have seen people come in here claiming all sorts of things that contradict established scientific principles. They'll talk and they'll talk, but they never go beyond the anecdotal. Most of them just end up getting mad and stomp off when one of the Sound Science regulars give them a "put up or shut up" challenge. A very tiny percentage are disingenuous and try to trick us. Very few go to the trouble to set up a test for themselves and try to prove something. At best they make a half hearted attempt and then gleefully indulge their bias, making deliberate compromises to undermine any value of the results. Why would they do that? Ego. They don't want to be wrong. Unless they can get past their own ego, they will always have problems with bias.
> 
> ...


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## KeithEmo

Video is actually a rather more interesting subject than audio.

As consumers we do NOT have access to uncompressed video. 
_ALL_ video we have access to, including the video on Blu-Ray discs and 4k UHD discs, is significantly compressed. 
And, if you do have the opportunity to compare actual uncompressed video to a Blu-Ray or 4k disc, you will see that there_ ARE_ visible differences.

Use your computer to generate a few seconds of true uncompressed video - especially containing sharp lines and fine print.
Take a bunch of perfectly sharp still frames and combine them into a video.
Compress it using any of the standard video compression profiles, and you will see that there are in fact significant visible differences, all of which relate to loss of detail or motion or color space compression.
You may find that the losses aren't especially noticeable, and you may be able to trade objectionable  artifacts for less objectionable ones, resulting in an "overall improvement".... but there will always be fidelity losses.

Video encoders are notorious for omitting random noise like film grain that was present in the original.
(They do this because true random noise is difficult to encode, and so uses up bandwidth that "could better be used for more important details".)
However, some producers, and some fans, complain that the "film look" of the video is destroyed when the film grain is removed.
(And, if your new digital audio recording omitted the tape noise that was present on the original master tape, would you describe it as an accurate reproduction or not?) 

(Incidentally, Darbee is not noise reduction..... it is a sort of dynamic color and detail enhancement process. And, no, I'm not fond of the look it delivers.)



bigshot said:


> A Lexus has better features. If you are buying a more expensive piece of equipment because it has additional features that you like, that is a good reason to spend more. I have an Oppo BDP103D player, and it has the exact same video and sound quality as my $100 Sony blu-ray player, but it plays DVD-A, SACD, MKV on thumb drives and it has Darbee video noise reduction. It has a whole slew of features that justify the price. When I bought it, they also made a BDP105D, which was the exact same player, except the specs were even further into audio overkill. I passed on that one. No point to throwing good money at sound I can't hear.
> 
> 
> 
> ...


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## bigshot (Nov 10, 2017)

Generally accepted science is the Nyquist Theory, the principles of acoustics developed at Bell Labs in the 20s, and the established thresholds of human perception.

Have you done any controlled listening tests yourself? If not, get to it. You’ll understand better what we’re talking about.

When you talk about compression artifacting, there is absolutely no point generalizing unless you're going to specify the codec and bitrate you're talking about. There's blatantly obvious artifacting and there's completely imperceptible compression. Darbee is the same. It has a continuous scale from absolutely no difference to horrible over sharpening. But if used properly, it is a great tool.

Absolutism gets you nowhere in digital audio and video. "Purity" theories were great in the analogue days where every layer of processing added noise and there was generation loss. It's a whole new world with digital. You can instantly see the effect of processing in a direct A/B comparison. You can process and compress and actually come out with something that looks and sounds better than the original. I love those tools and I know how to use them effectively. If I was afraid of them, I'd be stuck with what I'm handed. No thanks!


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## pinnahertz

KeithEmo said:


> There actually was a time when 19 kHz was a problem.
> 
> The way FM stereo works - and maintains compatibility with FM mono - is interesting. FM stereo is actually delivered as two matrixed channels; a main channel which is mono L+R, and a difference channel L-R.n
> A monaural tuner is able to simply play the main L+R channel without doing anything special, while a stereo FM tuner extracts the L-R channel, then does the math to derive separate L and R channels.
> ...


Your FM stereo description is a bit wrong.  The L-R channel is carried on a double-disband suppressed carrier channel centered at 38kHz, not 19kHz.  If a 19kHz subcarriers were used modulation side bands would end up in the audio band.  38kHz was chosen because the lowest modulation sideband lands at 23kHz, above the audio band. Double sideband supressrd carrier modulation was chosen for efficiency, no bandwidth was burned up be a carrier as it is in basic AM. The 19kHz pilot, which is required to regenerate the suppressed 38kHz carrier for demodulation, is injected 20dB below 100% modulation, then taken down another 20dB by de-emphasis (yes in mono radios too). So the most 19kHz you'd get out of any FM radio would be 40dB below 100% modulation. The better stereo tuners would have included a pilot null circuit that could be aligned for another 30 to 50dB of pilot reduction. The chances of  any audible 19 kHz coming out of the output is very minimal indeed,  certainly far too low to be filled as "pressure". In fact, the only 19 kHz leakage problems I'm aware of have to do with it beating against the bias frequency of a tape recorder.

In my youth, I was blessed with hearing up to 23kHz (which should partly address that issue in one of your other posts),  verified by the use of a sine wave oscillator in electronics lab. Horizontal fly back frequency  in television drove me nuts, and the ultrasonic burglar alarms used in retail stores of the time caused me actual pain ( they ran in the low 20 kHz range). However, I built a vacuum tube – based Heathkit stereo tuner with virtually no special handling of the pilot other than a null circuit, and I was never bothered by 19 kHz from that,  other than the bias beat in my tape recorder. 

 I disagree the 20 kHz plus is important  at all for  hi fidelity music reproduction in and of itself.  I do believe that systems that are severely limited in high frequency response can suffer from nonlinearities causing intermodulation products that can become audible, but high frequencies themselves above 20 kHz are not audible.  If good high frequency Linearity can be maintained such that Intermodulation products are kept very low, I see no need for greater than 20 kHz response.


----------



## pinnahertz

KeithEmo said:


> Hmmmmmm..... maybe because the methodology involved in doing so is flawed.......
> 
> We have two "widely accepted claims" about the thresholds of human perception.
> Both are based on very specific test conditions.
> Yet you want to generalize them to _ALL_ conditions - without offering any sort of proof that such a generalization is valid.


I trust you aren't addressing that comment to me.  I hate generalizations without qualification. 


KeithEmo said:


> Personally, I would NOT be willing to bet $1 million that someone cannot produce a single living human being who can hear 22 kHz.
> (And, no, the fact that it hasn't happened yet does not _PROVE_ that it cannot or will not happen.)


And as I addressed briefly in my previous post, you'd have been wise not to make that bet.  The fact that a high level, high frequency test tone was audible to me up to 23kHz at one point in my life means nothing in terms of that frequency and above contributing in any way to music preproduction.  You are probably aware that the high end of human hearing is highly variable and depends on many factors, the big ones being age and hearing damage.  The range of 15kHz detection, for example, is over 90dB across a wide population segment.  That's 90dB, not 9 or so.  90.  And believe it or not, that's partially correctable!  But that should tell you a bit about how human hearing averages out.  Young undamaged ears may detect high levels of sustained test tones above 20kHz, but over 20 years of age, that's pretty much gone already. 


KeithEmo said:


> At best you can state that nobody has demonstrated that any of the claimed benefits have been shown to be audible in a properly designed and executed test.
> (And I'll cheerfully agree with you there.)
> 
> To answer your question.... I don't specifically expect to hear anything... but I can't entirely rule it out based on the currently available evidence either.
> (And, even if I can't hear it, I can't rule out the possibility that someone else will. After all, I don't have "perfect pitch" either, yet some humans do.)


Please understand that hearing high frequencies is not a binary situation.  It's not "hear it or you don't", it's a question of level vs frequency vs threshold of detection.  Hearing response has quite a radical curve to it, even in ideal, young, undamaged ears.  But the curve gets very, very steep above 20kHz where, in fact, the threshold of hearing at the threshold of pain intersect somewhere around 140dB SPL.  Pretty much no point in designing for that condition, now is there?

Perfect pitch is completely outside of this discussion.  It can be developed, and then later lost.  Some come by it more readily, some almost innately, others not as much.  And trust me, it's not always a welcome gift.  But it has nothing to do with this discussion.


----------



## castleofargh

KeithEmo said:


> I would like to be able to disagree with your initial statement - because I have tried to avoid claiming any conclusions whatsoever.
> However, I agree entirely with your statement that "I didn't prove anything".
> My only claim is that I failed to rule out the need for further testing... as has everyone else.
> Actually, since people claimed to hear differences under certain conditions, it seems reasonable to suggest using those conditions as a starting point for a real test.
> ...


indeed skepticism should apply to all including ourselves, and to everything including negative results. I believe we don't hang around negative results much because most of us already know how little they prove. I don't believe many aside from the new born objectivists take failure to notice as a huge conclusive result.  
there is a little conflict with that idea when my default stand is to expect no difference, but what else can I do? the open mind approach where we seriously consider and investigate everything ever claimed, leads straight to the 9th dimension of sound that no instrument can measure but some old dude in a chair claims to hear. the need to draw a line between reality and delusion is just too important in this hobby IMO. so I picked the position of the jerk who doesn't believe what people say and goes "vid or it didn't happen". 

to me(and only me for me in my life) the all high res idea is very clear and not mysterious at all. I see what I can get in measurements from increasing it and changing variables. I see how different DACs deal with redbook with various approaches and degrees of success in being transparent. I can test a bunch of situations for my own gears and my own hearing, and have done it many times over the years. I'm also clear on how money is more important to me than very small doubts. something unrelated to sound but very significant to me. just like some other guy will make himself sick with insecurity, and if using highres can relax him, then obviously he should go high res. and for the guy with the crappy NOS DAC that some fool made with no or very bad filter, it's very possible that high res(or at least upsampling) will really improve things. and whatever other situation is fine.  whatever is significant in somebody's life for whatever reason, that's something worth dealing with for that person.

what makes the highres issue complicated is like you suggest, how everybody seems to be asking different questions about different situations while being different people. what's worst is when after that they assume that any idea/conclusion they reach is applicable to the rest of the world and every other situations. 
even within the noticeable, people tend to have very different perspectives. someone will go "I noticed that hat on that guy when we passed in the subway full of people". and even if the next day I go looking for that guy with the hat and I end up finding him, the meaning of noticeable takes a special sense. I could have walked there for 10 years without noticing that hat. or I could have seen it and not care "oh a hat, so what?". or I could have seen it and from that day on, could never not see it when I passed there. at some point I could become so obsessed that I would think I see the hat everywhere even when it's not there. or I would see another hat and assume it's THE hat. all those situations encompass "noticeable" but are night and day different when it comes to personal experience and the significance of it. so even the correlation between noticeable and significant is a delicate thing. and with highres, we're often at a point where just noticeable is hard to demonstrate. so I don't find it excessive to assume that for most people, highres is a trivial event as far as the sound of musical content is concerned. because they might still be of that opinion if noticeable was clearly ascertained. that's how much mental headroom I have on this ^_^.


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## Whazzzup

well thats that.


----------



## bigshot

castleofargh said:


> I'm also clear on how money is more important to me than very small doubts. something unrelated to sound but very significant to me. just like some other guy will make himself sick with insecurity, and if using highres can relax him, then obviously he should go high res.



Unfortunately, there's ample evidence that high data rate audio doesn't totally deal with _audiophilia nervosa._ It just makes the doubt move on to a different area... Instead of sampling rate, they start worrying about jitter in their DAC or how clean the caps are in their amp. They just bounce from variable to variable, spending money and never finding total relief. The problem with insecurity is insecurity. It doesn't have anything really to do with what the person is feeling insecure about.


----------



## frodeni

pinnahertz said:


> Qualitative testing and statistical analysis are not mutually exclusive, but rather, if any qualitative test is to have any meaning other than biased opinion, it must be statistically significant. One result is statistically too noisy, and in fact completely meaningless.



In interpretive qualitative research, there is no usage of bias. Bias is related to positivist research, and is not used in the tradition of interpretive research. Actually, objecting to how bias is traditionally used in the nature sciences, rejecting it completely, is so comon, that there is a large number of papers dealing with it. For really good reasons. Well founded, and well argument ed reasoning.

The notion that everything needs to be statistical significant is also a highly contested argument. Also, for very good reasons. There are plenty of papers, of highly regarded scientists, that argue against the need for being statistically significant.

As for meaningless, well, what do you mean by "in fact completely meaningless"? When did meaning become a fact? How do you factually prove meaning?



bigshot said:


> Unfortunately, there's ample evidence that high data rate audio doesn't totally deal with _audiophilia nervosa._ It just makes the doubt move on to a different area... Instead of sampling rate, they start worrying about jitter in their DAC or how clean the caps are in their amp. They just bounce from variable to variable, spending money and never finding total relief. The problem with insecurity is insecurity. It doesn't have anything really to do with what the person is feeling insecure about.



Same thing. How do you guys even know that anyone feel insecure? Care to prove how you arrived at that conclusion? How can you possibly know what other people feel? How do you know, what caused people to feel how they do?

Just as a warning, answer with care. You also need to apply your epistemology, at the topic of this tread. Good luck with that, give the position you are pushing.


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## bigshot (Nov 11, 2017)

frodeni said:


> How do you guys even know that anyone feel insecure? Care to prove how you arrived at that conclusion? How can you possibly know what other people feel? How do you know, what caused people to feel how they do?



I can do that very easily. I have a simple blind listening test that takes two very difficult passages of music to compress without artifacting and runs it though a variety of codecs and bit rates. It's a very easy way to determine where your own threshold of transparency lies. I can offer it to you to as a single lossless file containing ten different encodings, from lossless all the way down to MP3 192. All you have to do is listen to the samples and rank them from best to worst. Easy, right? If you'd like to take the test, all you have to do is tell me if you want FLAC or ALAC.

OK. Now it's time for your response. Will you agree to take the test? That depends on whether you really *want* to know if you can hear a difference. If you take the test and determine that you can't hear the difference with a particular codec and bitrate, will you stop ripping to a lossless format and use compressed audio? If not, why? Because you would feel more secure knowing that it's "lossless" regardless of whether it sounds identical or not.

My test doesn't just determine the level where compressed audio becomes audibly transparent. It also tests whether you will allow knowledge of the truth to override your insecurities about lossy codecs "throwing out something you might need". Most audiophiles just refuse to take the test because being wrong is preferable to them than knowing the truth. That shows that ego is the real force driving their decisions, not audio fidelity. I can test for transparency, insecurity and ego all with one simple test.


----------



## frodeni

bigshot said:


> I can do that very easily. I have a simple blind listening test that takes two very difficult passages of music to compress without artifacting and runs it though a variety of codecs and bit rates. I can offer it to you to as a single lossless file containing ten different encodings, from lossless all the way down to MP3 192. All you have to do is listen to the samples and rank them from best to worst. Easy, right? If you'd like to take the test, all you have to do is tell me if you want FLAC or ALAC.
> 
> OK. Now it's time for your response. Will you agree to take the test? That depends on whether you really *want* to know if you can hear a difference. If you take the test and determine that you can't hear the difference with a particular codec and bitrate, will you stop ripping to a lossless format and use compressed audio? If not, why? Because you would feel more secure knowing that it's "lossless" regardless of whether it sounds identical or not.
> 
> My test doesn't just determine the level where compressed audio becomes audibly transparent. It also tests whether you will allow knowledge of the truth to override your insecurities about lossy codecs "throwing out something you might need". Most audiophiles just refuse to take the test because being wrong is preferable to them than knowing the truth. That shows that ego is the real force driving their decisions, not audio fidelity. I can test for transparency, insecurity and ego all with one simple test.



Sure. If you dare to. Sure you want me to flaw your test? I am not scared of any testing, what so ever, rather I embrace it. I also embrace my current beliefs, assumptions, experience, and knowledge. Please, send me this test of yours, please explain it in detail, and the ethics of how you will use my results. Please describe the limits of usage, when the data will be destroyed, and the scheme you have to vet your usage of the data I supply to you? Also, describe what kind of research this is a part of, what tradition this is conducted within.

Also, please note that I hardly can tell any difference on sound quality using certain USB ports on my laptop, differences between uncompressed and lossy compressed. I just don't. Guess what my results will tell you. Give some thought.

As for you proving any insecurity, that is plain out false. You have not even established any correlation in any form with any insecurity. You simply have not even found any valid prove of any insecurity. None. It is all in your head. Which is kind of ironic.

If I do this test of yours, how do I know that you will publicly refrain from using it to mock me, making all sort of false claims of what I supposedly feel? Who will vet your claims? What makes them capable to vet you? In what tradition will you be vetted?

So great! Send me this file of yours. This is going to be great fun. Just not the fun you expect it to be.


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## bigshot (Nov 11, 2017)

The single ground rule is that you can only listen. No opening up the file and peeping at waveforms. Normal listening levels the same way you listen to music normally. If you cheat the test, you're only cheating yourself. (People have tried and it doesn't work. I will be able to tell.) You can listen on any equipment you want.

You rank the ten samples from best to worst. Discerning the degree of artifacting is as important as discerning lossless vs lossy. You send me your list. I send you back your results in PM. It won't be published or publicly discussed unless you want to. This isn't about ego. It's about you finding out the truth for yourself. Only you and I will know. If you discuss the test publicly, then I will discuss your results too. If you don't, I won't. PM me and I'll send you the file.


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## frodeni

bigshot said:


> The single ground rule is that you can only listen. No opening up the file and peeping at waveforms. Normal listening levels the same way you listen to music normally. If you cheat the test, you're only cheating yourself. (People have tried and it doesn't work. I will be able to tell.) You can listen on any equipment you want.
> 
> You rank the ten samples from best to worst. Discerning the degree of artifacting is as important as discerning lossless vs lossy. You send me your list. I send you back your results in PM. It won't be published or publicly discussed unless you want to. This isn't about ego. It's about you finding out the truth for yourself. Only you and I will know. If you discuss the test publicly, then I will discuss your results too. If you don't, I won't. PM me and I'll send you the file.



Oh boy. Sure. What ever.

Just answer the questions, particularly the one about how you know what other people feel.

If you cannot behave like a scientist, do not pretend to know how to.


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## Strangelove424

KeithEmo said:


> Video is actually a rather more interesting subject than audio.
> 
> As consumers we do NOT have access to uncompressed video.
> _ALL_ video we have access to, including the video on Blu-Ray discs and 4k UHD discs, is significantly compressed.
> ...



I have been lucky enough to work on uncompressed film masters in my career, with files large enough to saturate ethernet connections to the server. With a nice IPS monitor, it is a beautiful thing to behold, especially if the movie is an animated one. The gradients stand out particularly, perfectly smooth and organic. I never noticed such a startling difference with live action films, which have the resolving limitation of lenses and sensors to deal with. But with computer animated movies rendered uncompressed from software... wow.


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## bigshot (Nov 12, 2017)

frodeni said:


> Oh boy. Sure. What ever.
> 
> Just answer the questions, particularly the one about how you know what other people feel.
> 
> If you cannot behave like a scientist, do not pretend to know how to.



Didn't I already answer that? Do you want to take the test? Feel free to say no. If you don't want to know your thresholds of perception that's fine. I know where mine is, and I would guess it isn't that different from yours. Did me pointing out that it's strictly a listening test scare you off?


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## bigshot

Strangelove424 said:


> I have been lucky enough to work on uncompressed film masters in my career, with files large enough to saturate ethernet connections to the server.



I was inspired to create my home theater projection system when I was able to see The Incredibles at the Frank Wells Theater at Disney. It was very nice, but my own theater with a really good blu-ray is just as good. The only difference is the size. My theater holds about 12 people and we sit closer to the screen. The Disney theater holds about 50 and you sit further away. The end result is the same.


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## pinnahertz

frodeni said:


> In interpretive qualitative research, there is no usage of bias. Bias is related to positivist research, and is not used in the tradition of interpretive research. Actually, objecting to how bias is traditionally used in the nature sciences, rejecting it completely, is so comon, that there is a large number of papers dealing with it. For really good reasons. Well founded, and well argument ed reasoning.


If you've been reading the thread....

You might have discerned that the kind of testing we're talking about fits two basic forms: the statistical analysis of subjective opinion with regard to the basic question, "Is there an audible difference between A and B?", and if that analysis returns a statistically significant trend greater that random guessing, the next phase of testing, the statistical analysis of the subjective opinion with regard to the basic question, "Which is better, A or B?" can be performed.  The data structure is a simple binary response in each case and does not include the more broad spectrum of data that might be generated by interpretive qualitative analysis where data is comprised of description and opinion, which of course, is not directly or easily analyzed by simple statistics.  My error might be the use of the term "qualitative testing", which you then interpreted as "interpretive qualitative research", which I was not referring to.


frodeni said:


> The notion that everything needs to be statistical significant is also a highly contested argument. Also, for very good reasons. There are plenty of papers, of highly regarded scientists, that argue against the need for being statistically significant.


In this particular case, the first question, "Is there an audible difference?" can only be answered by statistical analysis of binary data.  If you don't apply statistical analysis of a significant quantity of data, then if you were testing "Can a person predict the future", and your test was the ability of a subject to "call" a coin flip in advance, your assumption could be that the correct guess of a single coin-flip indicates the subjects ability to predict the future.


frodeni said:


> As for meaningless, well, what do you mean by "in fact completely meaningless"? When did meaning become a fact? How do you factually prove meaning?


In the coin-flip example, you have a single subject and a single coin flip test with a single binary answer.  The resolution of that data is poor and includes  a very high degree of "noise" that is equal to the data itself; the correct result could be a random correct guess OR the subject could be predicting the future.  The resulting data has a noise level equivalent to the data "signal", and is therefore meaningless with regard to the question, "Can a person predict the future?"   From a single correct response, conclusion could only be "yes".  However, statistical analysis of a quantity of data from a number of tests increases the signal to noise ratio of the result by averaging the test responses, then returning a number rather than a binary result.  If we test many subjects many times, the degree to which the results are different than random noise (guessing) will indicate a probability that someone can predict the future.  In other words, the results of the test are a ratio of subject responses to random noise, which returns a probability figure, not a binary result.

This kind of testing is valid and used in scientific research all the time, most notably in drug efficacy testing.  The first question, "Does use of the drug return a result different than the placebo?" could be followed by "Does the drug result in improvement in patient condition?"  Both require statistical analysis of binary data.  That might then be followed by full interpretive qualitative analysis of patient impressions and side-effects.


----------



## gregorio

KeithEmo said:


> [1] Clearly all DACs do _NOT_ sound the same, and all DACs do not measure the same, or even close.
> [2] (For example, one DAC I owned had a very dull sounding high-end, almost certainly related to the fact that its frequency response was -3 dB at 20 kHz.)
> [2a] So the real question is whether DACs sound different _IN WAYS NOT OBVIOUSLY RELATED TO DIFFERENCES IN MEASURED PERFORMANCE_.
> [3] And, in order to test that question, we must predict what specific measured differences are audible.



1. No one is saying all DACs sound the same! We're saying all modern DACs, competently designed to achieve transparency, sound the same. Therefore, *obviously*, not all DACs sound the same! Some are not modern, some are not competently designed and some are deliberately designed not to achieve transparency (despite claiming to be high fidelity).

2. Just saying -3dB at 20kHz doesn't necessarily tell us much, where did that roll off start and how gradual was it? But, the "high-end" is not at 20kHz, the "high-end" is roughly from about 8-9kHz and extends to as much as about 16kHz. So if you really did hear the DAC you owned as having a "very dull sounding high-end" then you should look at what's happening in the high-end, rather than at what's happening significantly beyond the high-end where you can't hear anything at all!
2a. No, the real question is what is the actual measured performance, as opposed to what is just a published spec! What's disconcerting about your posts is your determination to validate or at least give some credence to audiophile nonsense claims: Misrepresent a spec as a "MEASURED PERFORMANCE", so you can imply measurements are invalid. State that anecdotal audiophile claims and sighted tests are flawed and so are all the published scientific studies and therefore scientific studies have effectively the same level of credence as audiophile claims. This is the same old typical audiophile nonsense just presented more intelligently and less absolutely. Of course, it's your choice if you wish to give equal credence but personally, I prefer to evaluate the flaws in the published science and other evidence and judge for myself how reliable it is, how much it's conclusions are affected by those flaws and how much credence to give it. Using this rational approach is largely why this sub-forum exists and it's clear there's virtually no credible evidence to support these audiophile claims, which is particularly telling given the very significant financial incentive in providing reliable evidence over the course of nigh on 20 years!

3. And we can! While in some cases we haven't laid to rest exact figures, even the most optimistic realistic figures are often thousands or even many tens of thousands greater than those actually achieved by modern, fairly cheap units.



KeithEmo said:


> As consumers we do NOT have access to uncompressed video.



What's your whole post about compressed video got to do with anything? We're not talking about compressed audio and even if we were, they're different beasts, we don't data compress raw audio by up to 1,000 times! It's hard to avoid the conclusion from your posts that you have an agenda, something to sell maybe? While you're not outright saying that measurements are useless and science has it all wrong, you are suggesting it might be wrong but for that suggestion to be anything more than just wishful thinking or marketing BS you've got to provide something substantially more than fallacies, misrepresentations and a bunch of audiophile impressions and beliefs!!

G


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## pinnahertz

gregorio said:


> It's hard to avoid the conclusion from your posts that you have an agenda, something to sell maybe?   While you're not outright saying that measurements are useless and science has it all wrong, you are suggesting it might be wrong but for that suggestion to be anything more than just wishful thinking or marketing BS you've got to provide something substantially more than fallacies, misrepresentations and a bunch of audiophile impressions and beliefs!!


I had a similar impression. In fact, my perception of a certain company has almost completely reversed as a result of this discussion.


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## castleofargh

pinnahertz said:


> If you've been reading the thread....
> 
> You might have discerned that the kind of testing we're talking about fits two basic forms: the statistical analysis of subjective opinion with regard to the basic question, "Is there an audible difference between A and B?", and if that analysis returns a statistically significant trend greater that random guessing, the next phase of testing, the statistical analysis of the subjective opinion with regard to the basic question, "Which is better, A or B?" can be performed.  The data structure is a simple binary response in each case and does not include the more broad spectrum of data that might be generated by interpretive qualitative analysis where data is comprised of description and opinion, which of course, is not directly or easily analyzed by simple statistics.  My error might be the use of the term "qualitative testing", which you then interpreted as "interpretive qualitative research", which I was not referring to.
> In this particular case, the first question, "Is there an audible difference?" can only be answered by statistical analysis of binary data.  If you don't apply statistical analysis of a significant quantity of data, then if you were testing "Can a person predict the future", and your test was the ability of a subject to "call" a coin flip in advance, your assumption could be that the correct guess of a single coin-flip indicates the subjects ability to predict the future.
> ...


don't get dragged into his controversy bait(that's usually my job to fall for it like an idiot). the best way to test something is obviously conditioned by the question someone is trying to answer at the time.


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## bigshot

I think he's gone anyway... When I challenged him to prove *he* could hear a difference, it took it away from theoretical arguments into real world proof and he bailed. This comment will probably irritate his ego and he'll be back trying to drag it back from the specific to the theoretical and general, but my test will still be waiting for him.

I'd like to set up a track with two more samples... 24/96 and 24/192. I don't have any music encoded that high though myself. If anyone would like to help me set one up, I'd like to do it. Imagine 12 samples from MP3 192 all the way up to 24/192 randomly shuffled into a single 24/192 file... haha! Sort that out.


----------



## JaeYoon

bigshot said:


> I think he's gone anyway... When I challenged him to prove *he* could hear a difference, it took it away from theoretical arguments into real world proof and he bailed. This comment will probably irritate his ego and he'll be back trying to drag it back from the specific to the theoretical and general, but my test will still be waiting for him.
> 
> I'd like to set up a track with two more samples... 24/96 and 24/192. I don't have any music encoded that high though myself. If anyone would like to help me set one up, I'd like to do it. Imagine 12 samples from MP3 192 all the way up to 24/192 randomly shuffled into a single 24/192 file... haha! Sort that out.


Let me know if he accepts your challenge. I did not like his attitude at the end of the debate.

I have LAME MP3 the latest by the way, 3.100 encoded from -V3 to -V0.


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## Strangelove424

bigshot said:


> I was inspired to create my home theater projection system when I was able to see The Incredibles at the Frank Wells Theater at Disney.



I can't speak to the Frank Wells showing of the Incredibles, but viewing uncompressed video requires a chain built around it... drives fast enough to not drop frames, and a display system capable of processing and displaying a wide gamut. Computers can be outfitted for that pretty easily, and work in high bit color space with monitors that reliably display the extra color info. There are consumer TVs available that show wide color gamuts and many receivers and BD players that support Deep Color too. I know my own PS3 and simple Denon receiver support it. Still not sure where uncompressed films would be sourced though, as Blu Ray compression typically uses a form of h.264. Though color depth may not be of supreme importance, I think it's atleast as important as resolution. Before the market rushes headlong to 4k, they might want to make use of those extra pixels with color definition too. If the extra file size needed to deliver 4k comes at the cost of color data, that's a bad compromise to make. I'd pick an equivalent bit rate 1080p any day of the week. But we'll see where that leads to I guess. 

Personally, for entertainment use, I have a PS3->DenonAVR->1080p 60hz Vizio. Nothing fancy, and I enjoy it very much. When it's time for popcorn and escapism, it certainly beats being huddled up to a hot IPS monitor. And the surround sound imo is the most important aspect of mimicing a theatre experience. The sound system brings the dimension.


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## KeithEmo

1) Unfortunately, the adjectives you've chosen are all subject to opinion and interpretation.... which is the problem. There are in fact many expensive DACs, being marketed by supposedly "reputable" companies, and highly thought of by a substantial number of reviewers and audiophiles, that clearly do NOT meet that criterion. And, while you and I might agree that this simply means they are poorly designed, that is obviously not sufficient to describe the difference to others. And, yes, there are clearly a number of people who believe that "any DAC made in the past ten years, and costing over $50, is going to sound audibly perfect" - and they are incorrect in that assumption. 

2) You're way overanalyzing that one. My sole point was to offer that particular DAC as an example of the many instances where measured flaws in performance do in fact correlate quite well with audible flaws in performance. (I didn't measure its frequency response; however, the point remains that the manufacturer didn't even spec it to have "an audibly flat frequency response", so it would be unreasonable to treat it as if it was claimed to have one.

3) Again, you take things out of context. When I'm talking about measured performance I'm simply talking about validating your test set. Claiming that "there's no audible difference with high-res files" is meaningless if your test setup includes speakers that are "rated to 30 kHz" unless you've actually measured them and demonstrated that they meet their spec. Likewise, failing to hear an audible difference between two DACs that are CLAIMED to be flat +/-0.1 dB isn't meaningful if you don't confirm that they both do in fact meet spec. (Otherwise, it's possible that you might hear a difference simply because one meets spec and the other does not.) 

In this context, I'm referring generally to the various tests claimed to determine "whether the difference between regular and high-res files is audible". In the majority of them, no measurements were taken to validate the claim that the listener was actually being presented with a MEASURABLE difference. (I don't care what someone claims. It's simple enough to pull out a test microphone and see if there's a measurable difference before going on to determine if it's audible. If there is in fact no measurable difference, then your test is invalid, and so are any results arising from it.)

4) My response about video was in response to another post about "how some types of video compression may look perfect - or may even sometimes make the video look better".

And, no, I'm not at all saying that measurements don't count.... simply that, if you do want to base your judgment on measurements, then you need to ensure that you make the proper measurements under proper conditions.
But, yes, if both are flawed, then neither can be considered to be absolutely reliable or definitive.... that's simple logic.
And, yes, a test that claims nobody can hear 25 kHz, conducted using a speaker that hasn't been tested and confirmed to be able to play 25 kHz, IS meaningless.



gregorio said:


> 1. No one is saying all DACs sound the same! We're saying all modern DACs, competently designed to achieve transparency, sound the same. Therefore, *obviously*, not all DACs sound the same! Some are not modern, some are not competently designed and some are deliberately designed not to achieve transparency (despite claiming to be high fidelity).
> 
> 2. Just saying -3dB at 20kHz doesn't necessarily tell us much, where did that roll off start and how gradual was it? But, the "high-end" is not at 20kHz, the "high-end" is roughly from about 8-9kHz and extends to as much as about 16kHz. So if you really did hear the DAC you owned as having a "very dull sounding high-end" then you should look at what's happening in the high-end, rather than at what's happening significantly beyond the high-end where you can't hear anything at all!
> 2a. No, the real question is what is the actual measured performance, as opposed to what is just a published spec! What's disconcerting about your posts is your determination to validate or at least give some credence to audiophile nonsense claims: Misrepresent a spec as a "MEASURED PERFORMANCE", so you can imply measurements are invalid. State that anecdotal audiophile claims and sighted tests are flawed and so are all the published scientific studies and therefore scientific studies have effectively the same level of credence as audiophile claims. This is the same old typical audiophile nonsense just presented more intelligently and less absolutely. Of course, it's your choice if you wish to give equal credence but personally, I prefer to evaluate the flaws in the published science and other evidence and judge for myself how reliable it is, how much it's conclusions are affected by those flaws and how much credence to give it. Using this rational approach is largely why this sub-forum exists and it's clear there's virtually no credible evidence to support these audiophile claims, which is particularly telling given the very significant financial incentive in providing reliable evidence over the course of nigh on 20 years!
> ...


----------



## KeithEmo

About the only place uncompressed live video can be sourced is as a direct feed from a high end camera. 
You can also find various samples of uncompressed computer rendered video (Google the publicly available renderings of the "Big Buck Bunny" cartoon).
Blu-Ray discs use h.264 compression; 4k UHD discs use h.265 (HEVC), which is more efficient, and is also often optimized to deliver a smoother looking picture, sometimes even at the expense of absolute sharpness. 
Even the CinePro format, used on the higher-end GoPro cameras, is compressed.
(A few industry experts have suggested that we would have been better off using moving to HDR without bothering with 4k..... but they were apparently in a minority.)



Strangelove424 said:


> I can't speak to the Frank Wells showing of the Incredibles, but viewing uncompressed video requires a chain built around it... drives fast enough to not drop frames, and a display system capable of processing and displaying a wide gamut. Computers can be outfitted for that pretty easily, and work in high bit color space with monitors that reliably display the extra color info. There are consumer TVs available that show wide color gamuts and many receivers and BD players that support Deep Color too. I know my own PS3 and simple Denon receiver support it. Still not sure where uncompressed films would be sourced though, as Blu Ray compression typically uses a form of h.264. Though color depth may not be of supreme importance, I think it's atleast as important as resolution. Before the market rushes headlong to 4k, they might want to make use of those extra pixels with color definition too. If the extra file size needed to deliver 4k comes at the cost of color data, that's a bad compromise to make. I'd pick an equivalent bit rate 1080p any day of the week. But we'll see where that leads to I guess.
> 
> Personally, for entertainment use, I have a PS3->DenonAVR->1080p 60hz Vizio. Nothing fancy, and I enjoy it very much. When it's time for popcorn and escapism, it certainly beats being huddled up to a hot IPS monitor. And the surround sound imo is the most important aspect of mimicing a theatre experience. The sound system brings the dimension.


----------



## KeithEmo

I disagree.....

If we ask: "Can humans hear 23 kHz?" then the answer is in fact a simple binary answer...... the answer is either "Yes" or "No".

You feel free to consider any number of other questions like:
"Can MOST humans hear the difference?"
"Is the difference SIGNIFICANT?"
"Does the presence or absence of 23 kHz affect the average person's appreciation of music?"
or even 
"What percentage of humans in each age group can hear 23 kHz?"
or
"Is it worthwhile designing equipment to be accurate to 23 kHz because some humans can hear it?"

However, none of them obviates the original - and quite binary - question.
(Of course, the reverse is NOT true. If the answer to the original question turns out to be "No" then the other questions become moot.)

As for perfect pitch, I simply offer it as an example of an ability that only a few humans have.
If a certain recording was off-speed, but only one in a million humans had perfect pitch accurate enough to notice the flaw in that recording, we would still have to report that "the flaw is audible to some humans".
(And, in the binary case of the question, "the flaw is audible to humans".)
The fact that we can find millions of people who cannot hear it does NOT change that binary result.)



pinnahertz said:


> I trust you aren't addressing that comment to me.  I hate generalizations without qualification.
> And as I addressed briefly in my previous post, you'd have been wise not to make that bet.  The fact that a high level, high frequency test tone was audible to me up to 23kHz at one point in my life means nothing in terms of that frequency and above contributing in any way to music preproduction.  You are probably aware that the high end of human hearing is highly variable and depends on many factors, the big ones being age and hearing damage.  The range of 15kHz detection, for example, is over 90dB across a wide population segment.  That's 90dB, not 9 or so.  90.  And believe it or not, that's partially correctable!  But that should tell you a bit about how human hearing averages out.  Young undamaged ears may detect high levels of sustained test tones above 20kHz, but over 20 years of age, that's pretty much gone already.
> 
> Please understand that hearing high frequencies is not a binary situation.  It's not "hear it or you don't", it's a question of level vs frequency vs threshold of detection.  Hearing response has quite a radical curve to it, even in ideal, young, undamaged ears.  But the curve gets very, very steep above 20kHz where, in fact, the threshold of hearing at the threshold of pain intersect somewhere around 140dB SPL.  Pretty much no point in designing for that condition, now is there?
> ...


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## RRod

It isn't a simple 'yes/no', Keith. It's a simple 'yes/no' when you reference some SPL in some environment. I can't hear 2kHz at 0.1dBSPL in my living room...


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## KeithEmo

I'd certainly be willing to take the test...... (but how about putting the file up on a like site like WeTransfer, and posting the link, so we can ALL take it)......
And, by all means, report the results.

I will, however request that you publish ALL the results.
I want to see the overall number of correct guesses - so we can see that percentage.
But I also want to see a list of how many correct guesses each participant made (they can be anonymous).
That way, if two or three people are consistently correct a statistically significant portion of the time, showing that at least a few people can consistently tell the difference, we will see that as well.
Since this is a binary question.... knowing that even one or two people can consistently hear the difference would be highly significant.

It would also be nice if you could include a few different pieces of music... just so we can each pick out something we're somewhat familiar with.
(And, of course, we need to know the provenance of the original, so we can confirm that, on its own, it is of sufficient quality to show differences in other factors.)

However, I will note right up front that I am NOT claiming that I will personally be able to choose correctly.
After all, I'm almost 60 years old, so it seems pretty unlikely that I have the best hearing of any human on the planet - or even close.
And, no, even if it turns out that I hear no difference on a specific sample, produced in a specific way, I will probably not generalize that to every piece of music ever made.
(So I will continue to store my music in whatever form preserves the amount of information present in the original - as I've received it.)

If I was very pressed for space, I might decide to evaluate each individual piece of music, and use lossy compression for those where I heard no difference.
However, since I am not at all pressed for space, I would be unlikely to bother (the easiest way to ensure that it will not be audibly altered is simply not to alter it at all).



bigshot said:


> I can do that very easily. I have a simple blind listening test that takes two very difficult passages of music to compress without artifacting and runs it though a variety of codecs and bit rates. It's a very easy way to determine where your own threshold of transparency lies. I can offer it to you to as a single lossless file containing ten different encodings, from lossless all the way down to MP3 192. All you have to do is listen to the samples and rank them from best to worst. Easy, right? If you'd like to take the test, all you have to do is tell me if you want FLAC or ALAC.
> 
> OK. Now it's time for your response. Will you agree to take the test? That depends on whether you really *want* to know if you can hear a difference. If you take the test and determine that you can't hear the difference with a particular codec and bitrate, will you stop ripping to a lossless format and use compressed audio? If not, why? Because you would feel more secure knowing that it's "lossless" regardless of whether it sounds identical or not.
> 
> My test doesn't just determine the level where compressed audio becomes audibly transparent. It also tests whether you will allow knowledge of the truth to override your insecurities about lossy codecs "throwing out something you might need". Most audiophiles just refuse to take the test because being wrong is preferable to them than knowing the truth. That shows that ego is the real force driving their decisions, not audio fidelity. I can test for transparency, insecurity and ego all with one simple test.


----------



## bigshot (Nov 13, 2017)

I have multiple copies of the test in different random orders. Plus I want people to give me their ranking before they find out their results. That’s why I administer the test individually in PM

It doesn’t matter really if a person can hear above 20kHz by the way, because those ultra high frequencies add nothing to perceived audio quality in music.


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## KeithEmo

We seem to be talking about different semantic interpretations here....

I tend to expect that any question I state will be considered USING THE CONDITIONS I INCLUDE AS PART OF THE QUESTION.
Therefore, if I ask: "What frequency range can a human hear at 80 dB SPL" then that's the information I want.
And, if I simply ask: "What frequency range can a human hear" then I'm opening that question to be considered under ALL possible conditions.

In other words, any question that fails to include specific conditions should be interpreted as a simple yes/no encompassing all possible conditions.



RRod said:


> It isn't a simple 'yes/no', Keith. It's a simple 'yes/no' when you reference some SPL in some environment. I can't hear 2kHz at 0.1dBSPL in my living room...


----------



## bigshot

At 80dB I would think a human could perceive super audible frequencies. They couldn’t hear them, but they certainly could feel the headache it would probably create


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## KeithEmo

That's fine.....

I just want to know, in the end, whether ANY INDIVIDUAL can tell the difference with an accuracy above statistical random.
(Because, if any individual can, then the binary form of the question - whether ANY HUMAN can hear the difference - will have been proven.)

As I've mentioned before, I personally have little interest in my own threshold..... (since I have no reason to believe that I represent the ultimate test subject).
And, since I'm not at all short of space or bandwidth, I have no reason whatsoever to expend the extra effort to use compression, especially with the added possibility that, at some point, it may do something audible.
(If I had space or bandwidth issues I might feel differently.)



bigshot said:


> I have multiple copies of the test in different random orders. Plus I want people to give me their ranking before they find out their results. That’s why I administer the test individually in PM.
> 
> This test is for individuals to determine their own personal thresholds. It’s not to generalize, but I will tell you that I’ve given this test a couple of dozen times and no one has been able to differentiate high bitrate lossy and lossless. Many can’t hear any real difference between any of the files.


----------



## KeithEmo

I'm pretty sure you're right..... and then we could discuss whether we were testing "hearing" or "perception".

As I think I mentioned before, I have a friend who works on ultrasonic plastic welding equipment. 
He informs me that, if you're foolish enough to operate one of the ultrasonic welders we works on with its shielding open, the totally inaudible ultrasonic frequencies actually make your eyeballs ache.
That would clearly qualify as "perceptible".
(We might also discuss whether we might prefer that our audio system were UNABLE to reproduce those frequencies accurately at full volume.)



bigshot said:


> At 80dB I would think a human could perceive super audible frequencies. They couldn’t hear them, but they certainly could feel the headache it would probably create


----------



## RRod

KeithEmo said:


> We seem to be talking about different semantic interpretations here....
> 
> I tend to expect that any question I state will be considered USING THE CONDITIONS I INCLUDE AS PART OF THE QUESTION.
> Therefore, if I ask: "What frequency range can a human hear at 80 dB SPL" then that's the information I want.
> ...



Too easy to lose such assumptions in oceans of text and tangential video discussion. Carry on then.


----------



## Strangelove424

KeithEmo said:


> About the only place uncompressed live video can be sourced is as a direct feed from a high end camera.
> You can also find various samples of uncompressed computer rendered video (Google the publicly available renderings of the "Big Buck Bunny" cartoon).
> Blu-Ray discs use h.264 compression; 4k UHD discs use h.265 (HEVC), which is more efficient, and is also often optimized to deliver a smoother looking picture, sometimes even at the expense of absolute sharpness.
> Even the CinePro format, used on the higher-end GoPro cameras, is compressed.
> (A few industry experts have suggested that we would have been better off using moving to HDR without bothering with 4k..... but they were apparently in a minority.)



So essentially no consumer content. Regarding 4k, the bandwidth required for atleast 4:2:2 easily exceeds most internet connections or even disc readers. Delivery technology would need to advance greatly on all fronts. There’s film cameras that can do 8k raw with HDR, but acquisition and delivery are two different beasts. Plenty of great cameras around, but still only so much space to deliver a movie on.


----------



## bigshot

[QUOTE="KeithEmo, post: 13847815, member: 403988"I just want to know, in the end, whether ANY INDIVIDUAL can tell the difference with an accuracy above statistical random.
(Because, if any individual can, then the binary form of the question - whether ANY HUMAN can hear the difference - will have been proven.)[/QUOTE]

Perception doesn't necessarily mean hearing. I'm sure high levels of ultra sonic sound could cause discomfort, but that isn't the same as *hearing* those frequencies. The general upper range of human hearing is 15kHz to 20kHz. with rare instances of very young people who can hear up to 23kHz. None of those frequencies contribute to audio quality in music however. Even if they exist in recorded music, you could do a roll off starting at 15kHz and I doubt anyone would be able to tell the difference. They just aren't important in recorded music. The balance of frequencies through the core two octaves- 2kHz to 8kHz is MUCH more important. If you want music to sound good, that is the area to focus on.


----------



## bigshot

frodeni said:


> Going on and on about methodology, sort of makes things just worse.



It is pointless. Remember I have that comparison listening test standing by for you if you want to find out your own personal thresholds of perception for compression artifacting. Just shoot me a PM letting me know whether you want ALAC or FLAC.


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## KeithEmo

Basically you are correct..... although I would also point out that what we're really talking about is more limitations on "value to the consumer" rather than limitations on the technology itself.

For example, video streaming services are essentially limited to about 1/5 the bandwidth of a 4k UHD Blu-Ray disc.
However, that is not an absolute limit on the bandwidth available to distribute content.
For one thing, you can still buy a disc, although it will cost more, and entail a two day delay.
For another, you don't HAVE to stream video in real-time.
A new service named Vidity proposes to make 4k movies available for rent at full 4k Blu-Ray quality.
With Vidity you purchase a single empty hard disk.
Then, when you want to rent a movie, you request it a day in advance, and it is downloaded to your hard drive overnight.
You then watch it the next day.
(Because Vidity hasn't "taken off", even with the backing of several major studios, we must assume that most viewers prefer to watch in lower quality today rather than in higher quality tomorrow.)

So it is really consumers who have decided that they will NOT wait 12 hours in order to see that movie in better quality.
The technology sets the threshold, and defines the options, but the consumers still make the choice.
We COULD be watching uncompressed movies if enough consumers were willing to pay a little more, and tolerate a little more inconvenience, in order to do so.

As with most products, if ENOUGH customers were willing to pay ENOUGH, someone would sell it.



Strangelove424 said:


> So essentially no consumer content. Regarding 4k, the bandwidth required for atleast 4:2:2 easily exceeds most internet connections or even disc readers. Delivery technology would need to advance greatly on all fronts. There’s film cameras that can do 8k raw with HDR, but acquisition and delivery are two different beasts. Plenty of great cameras around, but still only so much space to deliver a movie on.


----------



## pinnahertz

KeithEmo said:


> I disagree.....





KeithEmo said:


> If we ask: "Can humans hear 23 kHz?" then the answer is in fact a simple binary answer...... the answer is either "Yes" or "No".


No, it isn't and cannot be because you must also include the conditions in which a human can or cannot hear 23kHz.  The answer would be "Yes, under these conditions:" or "No, under these conditions:"  If you reduce everything to a binary answer you will miss the most important data completely, because you'll get a Yes, then, I assume, your designs and products will be adjusted to respond to a binary Yes. 


KeithEmo said:


> You feel free to consider any number of other questions like:
> "Can MOST humans hear the difference?"
> "Is the difference SIGNIFICANT?"
> "Does the presence or absence of 23 kHz affect the average person's appreciation of music?"


You're still trying for binary answers to multi-dimensional questions.  You've ignored conditions completely.


KeithEmo said:


> or even
> "What percentage of humans in each age group can hear 23 kHz?"


Getting closer, still ignoring other conditions.


KeithEmo said:


> or
> "Is it worthwhile designing equipment to be accurate to 23 kHz because some humans can hear it?"


Possibly the most disturbing binary conclusion of all.


KeithEmo said:


> However, none of them obviates the original - and quite binary - question.
> (Of course, the reverse is NOT true. If the answer to the original question turns out to be "No" then the other questions become moot.)


A binary answer to any of those will narrow the understanding of the problem to the extent that whatever solution is percieved as valid will effectively be the use of a pile driver to insert a thumb tack.  That's not science or engineering.


KeithEmo said:


> As for perfect pitch, I simply offer it as an example of an ability that only a few humans have.
> If a certain recording was off-speed, but only one in a million humans had perfect pitch accurate enough to notice the flaw in that recording, we would still have to report that "the flaw is audible to some humans".


No, actually the correct conclusion would be "the flaw is audible to one person in a million", then state the test conditions.


KeithEmo said:


> (And, in the binary case of the question, "the flaw is audible to humans".)
> The fact that we can find millions of people who cannot hear it does NOT change that binary result.)


But the binary result is not useful, it's just pedantic.  Good design cannot be arrived at using your binary method.


----------



## bigshot

KeithEmo said:


> For example, video streaming services are essentially limited to about 1/5 the bandwidth of a 4k UHD Blu-Ray disc.



It's hard to compare apples to apples with video because mastering differences can make a huge difference. But I found a direct comparison on Warner Archive Instant. I had the blu-ray of Billy Rose's Jumbo and Warner Archive had the exact same restoration on their streaming service. I set up a direct A/B switchable comparison between the blu-ray and streaming and couldn't tell the difference on my ten foot screen, even standing close. 4K really doesn't apply because most 4K movies are upscales anyway. Plus you would need a huge screen and sit very close to even gain the benefit of the added resolution.


----------



## KeithEmo

Science and engineering are different.
Science deals in facts while engineering, at least some of the time, deals in practicalities.
Engineering is about design; science is just about facts.

Therefore, as pure science, it is perfectly valid to question whether a thumb tack can be inserted with a pile driver.
(I suspect the answer is that, at least with some pile drivers, it is in fact quite possible.)
However, ENGINEERING might suggest that, even though it's possible, there are much more practical solutions.
Of course, with some subjects, there's a lot of overlap... and a lot of science doesn't get budgeted for research unless someone sees a practical aspect of the knowledge to be gained.

In this situation, the initial binary question serves as a sort of qualification for the more practical follow-on tests.
If we could prove that no human being could ever hear 23 kHz under any circumstances, then there would be no need to make more detailed tests.
We might even narrow that down and suggest that, if we could determine that no human could ever perceive 23 kHz under any conditions that weren't downright dangerous, we needn't test it in the context of listening to music.
However, if we find that even a few humans can hear it under some conditions which aren't totally excluded for safety reasons, then we move on to questions of practicality or usefulness.
You might be convinced that the experience of listening to music will never be improved by widening the frequency response of the recording... so let's find out for sure if you're always right.
However, I might be designing a new electronic toy for preschoolers, and be wondering if any of them might be annoyed by leakage of a small amount of 23 kHz power supply noise.
(Science makes no distinction; it simply provides pure information that each of us may then use for our own purposes.)

The correct answer to the initial question is simply: Yes, it is audible to some humans.
The next question is then: Which humans and under what conditions?
(Note that, if the answer to the first question was NO, then there would be no reason to ask the second question.)

And, yes, if you could determine, as a simple binary answer, that no human can ever hear 23 kHz, then we could quite usefully exclude the need for a frequency response extending to 23 kHz in our design specifications.
That would simplify our design goals, and probably save us a lot of money and effort.




pinnahertz said:


> ​No, it isn't and cannot be because you must also include the conditions in which a human can or cannot hear 23kHz.  The answer would be "Yes, under these conditions:" or "No, under these conditions:"  If you reduce everything to a binary answer you will miss the most important data completely, because you'll get a Yes, then, I assume, your designs and products will be adjusted to respond to a binary Yes.
> You're still trying for binary answers to multi-dimensional questions.  You've ignored conditions completely.
> Getting closer, still ignoring other conditions.
> 
> ...


----------



## pinnahertz

KeithEmo said:


> Science and engineering are different.
> Science deals in facts while engineering, at least some of the time, deals in practicalities.
> Engineering is about design; science is just about facts.
> 
> ...



1. But you cannot prove that no human could ever hear 23kHz under any circumstances unless you actually test every human being.  It's just impractical and illogical to look at it that way.  Even if you narrow down and limit your conditions the non-hazerdous, you still cannot prove the statement.  This gets to the "science cannot prove anything" line of logic, which is of course equally flawed. Science can prove things if you carefully structure what it is you're trying to prove.  The problem is the demand for a binary answer.  It's neither necessary nor desirable, and obscures the total picture of the situation just as if you processed a full image down to only black a white.  You'll clearly define black and white, but won't see anything between.  That is data reduction, and that should not really anyone's goal.

2. Now you're focussing on the conditions I was getting at, and will return more useful data.  That was my point: include the conditions, get more of a trend than a binary answer, and respond to that information.

3. Again, this is exactly my point: include the conditions.  You can never get a NO to the first question because you cannot practically test all humans.  That makes it the wrong question to ask, with easily misconstrued conclusions.

4. You can still get a useful answer even if it's not binary.  I'm sure your engineers would agree, and don't design for the extreme edges of the bell curve anyway.  At least I hope so (doubting everything now, though).   You could also bump the frequency up and come closer to a binary answer, but demanding a binary answer, well, that's hardly applied science.


----------



## KeithEmo

I agree entirely that mastering makes a major difference.
It's also well known that some differences are deliberate... for example, commercial videos often avoid certain colors, because the producers know that those colors tend not to come out well on consumer television sets.
Therefore, rather than risk their disc displaying inconsistently on different sets, the mastering engineer deliberately alters those colors to avoid causing problems on certain monitors.
(Look for scenes on commercial DVDs where there is any item colored bright saturated purple......  I can easily render that color on my computer screen - but TVs that follow the NTSC standard have trouble with it.)

If I were to set out to actually compare, I would start with an actual master original, encode it using several different CODECs, then compare each to the original.
(I'm assuming that the goal is not "to produce a copy that looks good" but "to produce a copy that cannot be distinguished from the original" or at least "a copy that is as close as I can get to the original".)

I've also found that, with video, and almost certainly because the processing that is applied in video compression tends to be quite complex, the errors seem to usually consist of specific types of artifacts.
One common artifact is that smooth color gradations on single-color objects, like a red marble, tend to become banded or posterized (banding is often present on copies and totally absent on the original).
Another common artifact is that gentle motion in dark areas tends to be lost. In one particular movie, in a particular scene, slow swirling in some dark clouds was clearly visible on an analog VHS copy, but totally missing from the DVD copy.
In many movies, if the original was made on actual film, there is a characteristic film grain in dark areas of the image, which is totally absent from some copies.
(The reason this occurs is well known. Random tape noise interferes with efficient compression, so filtering is applied to low level noise to remove it before the video is encoded. The filtering also sometimes removes low level random noise that belongs there.)
Also, even in high quality digital video, you often see blotching that follows square boundaries - where several pixels of similar color have been forced to the same color, creating an artificial seam where that zone touches one of a different color.
(It's painfully obvious because gradations in nature rarely follow areas with sharp vertical and horizontal boundaries. h.265 has improved this by introducing more variations outside of the large square boundaries used by h.264.)
These are all flaws which individually aren't usually "glaring" or "obvious" - but are easy to pick out if you look for them - and easy to notice by their absence on the original when you look there for comparison.

Note that the stated goal of most perceptual encoding is for the picture to "look natural" rather than specifically to be identical to the original..... 
If we were discussing audio using the same ideas, we would say that the goal is for it to sound good rather than to be accurate to the original.
(And we would probably say that the artifacts in h.265 are not only less than those in h.264, but they are more euphonic and less dissonant.)

And, yes, many current 4k movies are upscaled from an original that was filmed and/or processed at 2k....but not all.
If you want good samples to use for comparison.... check out some of the 4k demo videos from vendors like Sony and LG.
There are a few websites dedicated to demos (LG has one of a garden which is very impressive.)
Sadly, as usual, commercial movies rarely if ever live up to the capabilities of the standard - at least so far.



bigshot said:


> It's hard to compare apples to apples with video because mastering differences can make a huge difference. But I found a direct comparison on Warner Archive Instant. I had the blu-ray of Billy Rose's Jumbo and Warner Archive had the exact same restoration on their streaming service. I set up a direct A/B switchable comparison between the blu-ray and streaming and couldn't tell the difference on my ten foot screen, even standing close. 4K really doesn't apply because most 4K movies are upscales anyway. Plus you would need a huge screen and sit very close to even gain the benefit of the added resolution.


----------



## KeithEmo

Absolutely agreed......  but where we seem to disagree is simply in where I'm willing to accept something as "close enough not to matter".
I agree with you that very few humans can probably hear 23 khz.
However, given that possibility, and the current standards, I'd rather record a file at 96k and risk wasting a little space, rather than record it at 44.1k and risk causing some minor difference that a few people can hear.
(For the same reason that people go to museums to see great paintings instead of "very good copies".)
As I mentioned in another post, it also relaxes the design requirements on the DAC, which improves the chances that a DAC at a given price point will perform better with that copy. 

Another important factor is whether you consider your recordings as "something to listen to" or as "your own personal archival masters".
I consider mine to be the latter....... 

I'm going to give you an example where a recording made at 192k will definitely work as intended and one recorded at 44.1k definitely will not.
(I've avoided this so far because it is clearly a very specific situation - which does not apply to most listeners.)

Back in the days of vinyl, one of the problems that annoyed many of us was those ticks and pops you hear when the cartridge plays over a scratch.
Even with the best possible cleaning and care, it is often impossible to eliminate or avoid every single tick on a record.
Back when vinyl was a current technology, several companies devised electronic devices for removing ticks from records.
They operated on the principle that record ticks contain a lot of ultrasonic content.
They analyzed the spectrum of the record as it was playing, designated any short abrupt sound with excess ultrasonic content as a "tick", and muted it (or replaced it with a split second of delayed content from nearby).
Some of these devices worked pretty well (Garrard had one that was well known.)
Now, if you were to record a vinyl album, in perfect quality, at a 44k sample rate, and then apply one of these devices to your recording.... it wouldn't work - because your recording has failed to capture the ultrasonic information the device needs to operate.
However, if you applied it to a recording made at 192k, it would work as intended - because your recording would include that ultrasonic information.
(I'm guessing that the digital methods used by some editing software works much the same way.)
Would this ever matter to you or me?
Probably not.
But why take the chance and discard information that is present in the original and _MIGHT_ be useful later?
(You can always convert to a lower sample rate later, but you cannot get back information once you discard it.)

When I make a live recording, I keep a copy of the original, even if I edit it later, or make a "use copy" at a lower sample rate.
After all, I may choose to do a better edit later, or using software that allows me to do things I can't do now.
I tend to look at all music the same way.
If I have the opportunity, I prefer to have the best possible copy available, even if I don't expect to need it, or to hear the difference under the conditions I plan to use it.
(Note that we're talking about paying an extra $5 for a high-res download instead of a regular one..... not paying someone $5k to steal a copy of the studio master for my collection.)



pinnahertz said:


> 1. But you cannot prove that no human could ever hear 23kHz under any circumstances unless you actually test every human being.  It's just impractical and illogical to look at it that way.  Even if you narrow down and limit your conditions the non-hazerdous, you still cannot prove the statement.  This gets to the "science cannot prove anything" line of logic, which is of course equally flawed. Science can prove things if you carefully structure what it is you're trying to prove.  The problem is the demand for a binary answer.  It's neither necessary nor desirable, and obscures the total picture of the situation just as if you processed a full image down to only black a white.  You'll clearly define black and white, but won't see anything between.  That is data reduction, and that should not really anyone's goal.
> 
> 2. Now you're focussing on the conditions I was getting at, and will return more useful data.  That was my point: include the conditions, get more of a trend than a binary answer, and respond to that information.
> 
> ...


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## bigshot

KeithEmo said:


> Absolutely agreed......  but where we seem to disagree is simply in where I'm willing to accept something as "close enough not to matter".



You're free to accept or not accept whatever you want, but that doesn't mean that you could hear it, and it isn't terribly good advice for other people either. But if it makes you happy, go crazy!


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## pinnahertz

KeithEmo said:


> 1. Absolutely agreed......  but where we seem to disagree is simply in where I'm willing to accept something as "close enough not to matter".
> I agree with you that very few humans can probably hear 23 khz.
> However, given that possibility, and the current standards, I'd rather record a file at 96k and risk wasting a little space, rather than record it at 44.1k and risk causing some minor difference that a few people can hear.
> (For the same reason that people go to museums to see great paintings instead of "very good copies".)
> ...



1. But now you've drifted away from the question of can anyone hear 23kHz!  This is an entirely new question that must include: is there music-realated content above 23kHz, is it audible in the presence of music, can it be captured and passed unaltered through the entire signal chain to the listener's ears?  That last one appears to be a big NO, as reproducing ultrasonics with speakers in rooms plus propagation issues pretty much negates the possibility.  But those are all different questions that have nothing to do with the first. 

2. No recording represents the original event anyway, but how you consider your recordings is entirely psychological and has nothing to do with actual and real perception as it's pre-loaded with a payload of bias.

3. Nice example, and funny because I am actually intimately familiar with several of those devices, including the Burwen TNE-7000 and the SAE-5000.  But you clearly are not familiar with current software equivalent solutions for those problems, which _actually do work just fine at 44.1! _ The detection algorithms used in those early hardware devices had to be very basic, with relatively rudimentary ultrasonic filtering of an L-R signal.  Software is not limited in the same ways, and works fine even with the limits of 44.1kHz.  Additionally, several of the current declicker software options include many controls that vary the intensity of declick action offering the ability to tune the process to specific problems.  Add to all of that the process of capturing a "noise print" and applying noise reduction.  None of that was available on the analog devices of the past, and guess what? The new processes work much, much better!  And all without that extended ultrasonic response to deal with.  Though having ultrasonic content might help in this specific situation in some cases, 192kHz would be completely unnecessary given the actual transient speed of a click and the current detection methods.  Sorry, your example doesn't work.

4. Because we do know the actual bandwidth of the noise and signal of vinyl, and 192kHz is completely unnecessary for dealing with either.  Most vinyl was recorded on analog tape which has a remarkably sharp HF rolloff characteristic (yeah, only those of us who have maintained those machines realize that), and very little ultrasonic content other than distortion products.  No need to capture 96kHz bandwidth.

5. Anyone dealing with recording professionally, or even as an amatuer, keeps the original unaltered, either via back or nondestructive editing or both.  The error here is assuming high sampling rates = "better", when it more often means you've captured more noise but no more audio.


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## bigshot (Nov 13, 2017)

I read that the highest frequency a human has ever been able to hear is 23kHz. So yes, there is (or at least ways) someone who could hear that high. I’m sure it was very annoying to them, and the problem was rectified by the time they got out of their teens.


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## KeithEmo

First off, I disagree entirely on your first assumption. There are plenty of speakers that have a response out well past 25 kHz, including most of the models we currently sell at Emotiva (pretty well anything that uses the new AMT is good out past 25 kHz). And, since they measure flat past 25 kHz when we test them, with a microphone, in a room, those frequencies are obviously making it to the listening position. There are also plenty of microphones that can pick up and record those frequencies. Therefore, they certainly CAN be present in those high-res recordings. However, on to your second point, whether the recording engineer does their best to reproduce the original experience exactly, or simply uses it as a starting point on which to build their vision (which might be quite different), is always going to be up to the production folks. All I can do is to ensure that what I experience is as close as possible to what the production engineer intended. 

The original click-removal devices counted on the fact that ticks and pops contained ultrasonic content which was not present in normal musical content. Most of the new ones obviously work quite differently. The fact remains, however, that we may someday discover some other wondrous and amazingly useful process that requires the ultrasonic content - and, if we do, we may wish that our archival copies included it. I've never found any automated method to be entirely satisfactory - but I still prefer not to cut myself off from options. As I recall, CD-4 records required a cartridge with a frequency response out to about 50 kHz, so clearly vinyl _CAN_ contain frequencies far beyond 20 kHz - although a 96k sample rate should be sufficient even for CD-4.

I agree that assuming that high-res is _ALWAYS_ better would be an error. 
However, it has the _POTENTIAL_ to be better, and so that possibility cannot be ruled out.
Personally, I would simply prefer to have an actual copy of the master.... which is certainly possible with a digitally mastered recording.

I used to collect antiques..... and I tend to think of audio recordings like works of art.
If I were to walk into a shop, and see twelve identical looking vases on a shelf, but be told by the proprietor that:
"The shelf fell the other day, and eleven of them got damaged; our repair guy is really good, so we're sure you won't notice the repairs, and we'll sell you the repaired ones at a hefty discount"...... 
I would still choose to pay a little extra for the UNDAMAGED one.
(And that would still be true no matter how long I squinted into a magnifying glass and failed to find the repairs....)
I tend to view files with lossy compression, and in fact any file that's been reduced from the master version, the same way.
Even if I may not notice the difference, and may never have occasion for it to make any difference, I'd still prefer the one that has no damage, or the LEAST damage.
(And, yes, some high-res files are upsampled from CDs, and others may be mastered from tapes that really contain nothing useful above 20 kHz, in which case you've paid a little extra for insurance that has no value.)

I quite agree with you that.....
MANY high-res files are no better than the "normal" equivalents.....



pinnahertz said:


> 1. But now you've drifted away from the question of can anyone hear 23kHz!  This is an entirely new question that must include: is there music-realated content above 23kHz, is it audible in the presence of music, can it be captured and passed unaltered through the entire signal chain to the listener's ears?  That last one appears to be a big NO, as reproducing ultrasonics with speakers in rooms plus propagation issues pretty much negates the possibility.  But those are all different questions that have nothing to do with the first.
> 
> 2. No recording represents the original event anyway, but how you consider your recordings is entirely psychological and has nothing to do with actual and real perception as it's pre-loaded with a payload of bias.
> 
> ...


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## bigshot

I have a technique that I imagine a lot of people use... When people go into a mode where they're speaking solely for their own benefit and have stopped listening and interacting with others, I read only the first line or two and skip the rest. Your first couple of lines was about 25kHz, so I'll answer that...

Digital audio was designed to have a certain amount of overkill when it comes to frequency response. Just because it can capture ultrasonic frequencies, that doesn't mean that ultrasonic frequencies are present in music, or that it would make any difference if they were. (I also refer you back to my previous posts where I let you know that 23kHz is the highest any human ear could hear, and the one where I gave context to the scale of the frequency range you're talking about by noting how it relates to notes on a musical scale. Those two appear to have slipped by you a couple of times now.)

Now... back to your method of communicating... You keep going on and on and make no real point because you ignore anything anyone else says to you except an an opportunity to launch into another irrelevant sidetrack. You're talking in circles with self directed internal conversations. I hope you're enjoying your posts, because I doubt that anyone else is getting very much out of them. You might want to think about externalizing your thought processes a bit by considering your audience and what they are saying. The point of an internet forum is to talk *with* other people, not *at* them.


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## KeithEmo

I'm guessing you miss a lot of information that way.....

Generally I speak to inform - and I leave it up to the reader whether they want to inform, be informed, or simply argue.

Your statements didn't "slip by me". 
You have asserted that the ability to accurately reproduce ultrasonic frequencies serves no practical purpose because there is no meaningful musical content at those frequencies.
However, as it turns out, some instruments DO produce harmonics well up past 20 kHz (cymbals are one example).
And some humans can in fact hear frequencies above 23 kHz (although, admittedly, not many).
And some other audio processes, like certain click and pop removal hardware, also relies on the accurate reproduction of ultrasonic frequencies.
(Which demonstrates that those frequencies serve a potentially useful purpose in audio reproduction EVEN IF IT TURNED OUT HUMANS COULDN'T HEAR THEM.)
Therefore, it would seem to me that you have failed to prove your assertion.
(And the claim that "nothing important exists up there" is an opinion rather than a fact.)

Therefore, we are both simply "throwing out facts that tend to support our opinions".



bigshot said:


> I have a technique that I imagine a lot of people use... When people go into a mode where they're speaking solely for their own benefit and have stopped listening and interacting with others, I read only the first line or two and skip the rest. Your first couple of lines was about 25kHz, so I'll answer that...
> 
> Digital audio was designed to have a certain amount of overkill when it comes to frequency response. Just because it can capture ultrasonic frequencies, that doesn't mean that ultrasonic frequencies are present in music, or that it would make any difference if they were. (I also refer you back to my previous posts where I let you know that 23kHz is the highest any human ear could hear, and the one where I gave context to the scale of the frequency range you're talking about by noting how it relates to notes on a musical scale. Those two appear to have slipped by you a couple of times now.)
> 
> Now... back to your method of communicating... You keep going on and on and make no real point because you ignore anything anyone else says to you except an an opportunity to launch into another irrelevant sidetrack. You're talking in circles with self directed internal conversations. I hope you're enjoying your posts, because I doubt that anyone else is getting very much out of them. You might want to think about externalizing your thought processes a bit by considering your audience and what they are saying. The point of an internet forum is to talk *with* other people, not *at* them.





bigshot said:


> I have a technique that I imagine a lot of people use... When people go into a mode where they're speaking solely for their own benefit and have stopped listening and interacting with others, I read only the first line or two and skip the rest. Your first couple of lines was about 25kHz, so I'll answer that...
> 
> Digital audio was designed to have a certain amount of overkill when it comes to frequency response. Just because it can capture ultrasonic frequencies, that doesn't mean that ultrasonic frequencies are present in music, or that it would make any difference if they were. (I also refer you back to my previous posts where I let you know that 23kHz is the highest any human ear could hear, and the one where I gave context to the scale of the frequency range you're talking about by noting how it relates to notes on a musical scale. Those two appear to have slipped by you a couple of times now.)
> 
> Now... back to your method of communicating... You keep going on and on and make no real point because you ignore anything anyone else says to you except an an opportunity to launch into another irrelevant sidetrack. You're talking in circles with self directed internal conversations. I hope you're enjoying your posts, because I doubt that anyone else is getting very much out of them. You might want to think about externalizing your thought processes a bit by considering your audience and what they are saying. The point of an internet forum is to talk *with* other people, not *at* them.


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## bigshot (Nov 14, 2017)

Do you remember what I said about the ultrasonic frequencies in cymbals?

Who can hear sound above 23kHz, because I found 23 as the limit on a website on human hearing?

Noise reduction doesn’t have any relevance with digital audio for playback of music in the home, does it?


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## KeithEmo (Nov 14, 2017)

I believe you said that you think they don't count - in terms of audibility.
I found a few sites that list "the frequency spectrum of popular instruments" - and at least two of them list cymbals as "up to 23-24 kHz".
There was no footnote stating: "but they really don't matter because the level is so low".

In terms of the direct audibility of continous sine waves - 23 kHz as an "absolute top limit" seems to agree with most of the claims I've heard.

If I choose to apply noise reduction to the recordings I play, and I play them in my home, then I guess it is relevant.
(And, if you do not choose to, then I guess it isn't relevant to you.)
I even know one guy who still has a dBX noise reduction unit in his audio rack - and claims to still use it occasionally.
I also actually have at least one commercial CD that has what clearly sounds like record surface noise on it (clearly not put there intentionally).
(I'm guessing the only copy they had available was on vinyl.... and, yes, I do hope to get around to cleaning it up a bit someday.)

However, as I mentioned, I consider music I purchase as being added to my collection (which means that I do not rule out deciding to apply processing to it - or even editing it).
That makes music files I _BUY_ different than, for example, music I stream on my phone, used solely for live listening, and with no option to use it for anything else.
(And, yes, if I had a record I really liked that had some bad scratches, I would save it - just in case some future technology would remove those scratches better than the current technology.
Failing that option, I would digitize it using a sample rate I was sure would record every potentially useful piece of information on it - not just the ones I'm sure I need today. )



bigshot said:


> Do you remember what I said about the ultrasonic frequencies in cymbals?
> 
> Who can hear sound above 23kHz, because I found 23 as the limit on a website on human hearing?
> 
> Noise reduction doesn’t have any relevance with digital audio for playback of music in the home, does it?


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## bigshot (Nov 14, 2017)

There's a concept called "auditory masking"... It says that loud sounds at one particular frequency will block the ear from hearing frequencies an octave higher. A friend of mine did a startling demonstration for me once. He boosted the upper mids a little bit and boom! the treble disappeared and the sound got muffled. It's counter intuitive to think that boosting a frequency would make such a big difference in the octave above it, but it's true. A cymbal is a perfect example of masking. The crash in the upper mids / bottom treble pretty much makes the upper frequencies above them inaudible. And as the harmonics go upward, the volume level of the harmonics drop, so the masking continues up kind of like a ripple. There really isn't anything you can hear up around 18kHz in a cymbal crash. It's there in the recording, but your ears don't hear it. You can try an experiment and take a high sampling rate drum solo and roll off at different points. You'll find that you can roll off quite a bit off the top- as much as half an octave- and it won't affect the sound quality. Especially if you're over 50 and you can't hear those frequencies anyway!

I don't think anyone uses Burwen click filters on their CDs, and there are better digital filters available for that. In digital if you are gong to apply noise reduction, you'll do it in an audio editor and then save out the file and play that back. I don't know any real time filters that operate on ultrasonic frequencies that are used in playback. DBX has to be pre-encoded, and no CDs would have that. They would just decode the DBX and save it out. There really isn't any purpose for ultrasonic frequencies for playback. Only for mixing and mastering.


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## najo (Nov 14, 2017)

Wow!
These threads certainly made an interesting read!
Just wanted to put my 2cents as well i guess 


As far as reproducing the original audio with high fidelity goes, it depends entirely on the artist and production of the audio.

I've talked to my friend about this in the past, who is a recording artist with a record label and has a couple of albums with a few videos as well.

He told me that when they produce the audio to mix digitally the method of recording depends on the track.
Sometimes they will record with multiple instruments and other times individual instruments.

These form the inputs into the mixing console. However each of these inputs is then further eq'd.
The high cymbals and low bass for example.
They would take the bass and then use high/low pass filters on the eq to narrow the frequency range of each instrument.
So whilst the bass may bleed a little into 100++ they will effectively silence it above this frequency (or a chosen one. Im using arbitrary numbers)
With any high instrument in the high frequency band its the same. The instrument may cover a slight range but this is limited and the frequency spike is very sharp and narrow.
This helps to keep the separation between each instrument too as it provides a clean sound for each, and within a narrow range so less overlaps and muddy sounds.

He suggested that whilst there could be some content in that freq range on graphs it may have been introduced by other equipment/ripping/codecs etc or anything else if the artist has effectively cut the audio to a silence in that range. Even if the instruments used in the audio are typically though of as high or low they most certainly have a cut off.

This is also why 16bit 44.1 is more than enough to very accurately reproduce his audio as everything is trimmed and cleaned up.

I can imagine hearing a cymbal crash rise and trail for a seconds.... but he may well have cut it off...


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## old tech

That somehow having  a playback system that can reproduce frequencies greater than 20khz is beneficial is up there as one of the marketing scams designed to appeal to the gullible end of the spectrum of audiophiles.

It serves the interests of manufacturers who dabble in that quackery - product differentiation, higher sales margins

It serves the interests of audiophools who paid top dollar so they can "hear" these frequencies, assists with cognitive dissonance

It serves the interests of the less fidelity is more crowd (ie the vinylphiles) - records can have information past CD's 20khz so it must be better, never mind the roll-off from around 16khz, the noise floor limits, the masking effects or the fact that the other end of the frequency spectrum (ie subsonic) is far more important to the imagination effect as humans can at least feel the vibrations from low frequency energy.

Can you imagine videophiles being so gullible to want their video playback to display x-ray frequencies, as if that would make the picture any better?  Perhaps there are some videophiles that would be so gullible but nothing like the phoolery found in sections of the audio world.


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## pinnahertz (Nov 14, 2017)

KeithEmo said:


> First off, I disagree entirely on your first assumption. 1. There are plenty of speakers that have a response out well past 25 kHz, including most of the models we currently sell at Emotiva (pretty well anything that uses the new AMT is good out past 25 kHz).
> 
> 2. And, since they measure flat past 25 kHz when we test them, with a microphone, in a room, those frequencies are obviously making it to the listening position.
> 3. There are also plenty of microphones that can pick up and record those frequencies. Therefore, they certainly CAN be present in those high-res recordings. However, on to your second point, whether the recording engineer does their best to reproduce the original experience exactly, or simply uses it as a starting point on which to build their vision (which might be quite different), is always going to be up to the production folks. All I can do is to ensure that what I experience is as close as possible to what the production engineer intended.
> ...



1. So, you're going to apply that same warped logic here too?  If a tweeter has output past 23kHz, then it produces ultrasonic energy?  Under that definition EVERY tweeter produces ultrasonic energy!  That's true in the absolute, false in the practical.  We live in a practical world.

2. OK, you work for the manufacturer.  How about a set of polar plots of your "flat to past 25kHz" speakers, then?  Or link to manufacturer's data.  Any evidence at all would be great.  From what I've seen, and I've searched this oh just a bit, "flat past 25kHz" is a very, very tall order, especially when you look at dispersion and power handling too.  But I'll take those polars when you have them.  Then we can talk about off axis HF response of tweeters, microphones and oh yeah, ears.

3. Look again.  Very few are flat past 20kHz.  Very, very few.  And they'd need to be located in the ultrasonic field of some instrument producing ultrasonic energy to pick it up and transduce it.  No, the mic problem is not small, it's huge.  But, clearly, we're still at the same warped logic, "if one mic has response flat past 20kHz then a mic can transduce ultrasonic energy".  I own a calibrated measurement mic flat to 30kHz, but I'd never use it to record anything.  Microphones are part of the sonic palette, we don't select them because they are flat to 30kHz, we select them because their sonic signature compliments the application.

4. I already clearly outlined that old click removers needed extreme HF information, and that modern software works differently and much better.  You can worry about "someday", that's fine, but if there's no content at 92kHz, even from clicks (there are actual physical reasons for this), you're just blindly throwing technology at a non-problem.

5. Do not confuse the 45kHz response necessary for recovering the CD-4 carrier (FM, BTW), with the ability of vinyl to reproduce ultrasonic audio content. The two functions are completely and radically different.  The CD-4 carrier was injected at very low level, of physical necessity, and was frequency modulated.  When played the demodulator would "capture" the carrier with a phase locked FM demod, which effectively ignores the rather poor and ragged amplitude response of vinyl at 45kHz.  That's completely different than actually playing real ultrasonic content directly, and unrelated except that CD-4 stylus design pushed stylus shape development forward.  You do know the CD-4 groove had to be bigger, right? Less playing time, but it had to be just to get that ultrasonic carrier on and off the disc without damaging it. We don't use that size groove on any other kind of record.  Once again, a poor example.

6. But you ignore the fact that any high resolution file has also passed through a mastering phase where those in charge will make it sound differently if they intend it to be a perceived improvement.  And that has nothing whatever to do with resolution.  Potential for better sound from high-res has yet to be clearly proven, and cannot be proven by consumers buying files because there is absolutely no provenance.

7. Well...my other profession is an antique dealer. No, not kidding.  My wife has been a dealer for 34 years, and I joined her in the business about 15 years ago.   Now, isn't this fun?  I can tell you several things are amiss here.  There are virtually no antiques of any kind that are perfect.  Damage usually reduces value, but not always.  Sometimes certain damage increases value!  Repairs reduce value often, but not always.  Certain repairs increase value too.  The value of an antique is a complex function of rarity, desirability, condition and the market which is comprised of at least two or more buyers.  In  your example you seem to assume all the vases were identical in every aspect except for the damage and repair.  That's a condition that never occurs in the reality of antiques!  They would all be different, and you'll buy the one you like if you can accept the price and condition. You won't pay extra for the undamaged vase unless it happens to be the one you like, and at that point, damage is no longer a factor if it is minor or well repaired.

8. Why are you bringing up lossy compression???  Tangential at best.  We all prefer the least damaged audio file, but at some point the lossy codec becomes fully transparent, and it no longer matters.  Oh, sorry, if one person on or off the planet can detect the codec, then it's bad, right?

Thank you for mentioning that the bulk of high-res files are upsampled or analog masters, which fully disqualifies them for the classification of high-res, and casts doubt on the entire high-res file market for being scam artists.

9.  Yes, and those that are different could be made to sound the same different way at standard res.


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## KeithEmo (Nov 15, 2017)

1.
I'm sorry, but, yes. If the speaker has output past 23 kHz, and I can measure that output with a microphone at the listening position, then the tweeter is indeed "delivering ultrasonic energy to the listener" (assuming we're referring to 23 kHz as "ultrasonic").
In the practical world, if we can measure something, and the measurements show it's there, then it's there.
(I would agree that, if it's 20 dB down at 25 kHz, it probably won't be doing anything useful at that frequency.)

2.
Here you go.
Here's a link to our Emotiva Stealth 8 studio monitors.... (they're about $900 each - so cheap by audiophile standards):   https://emotiva.com/product/stealth-8/
They're rated at:
30 Hz to 23 kHz + / -1.75 dB
28 Hz to 32 kHz +0 / -6 dB
And here are the graphs:   http://cdn.shopify.com/s/files/1/0201/8878/t/2/assets/Stealth8_Plot_Graph.pdf
And, yes, their response drops off rather rapidly off axis at very high frequencies (but I usually sit in front of mine).
And, yes, those are real measurements.

3.
As for microphones..... I agree entirely...... and it's up to a whole bunch of people what microphones get used, and where the mixing engineer sets his (or her) filters and EQs.
My only goal is to make sure that MY equipment isn't the limiting factor - so I need to be able to handle anything that they MIGHT include.
(And, yes, in engineering, it's pretty standard to include SIGNIFICANT safety margins to eliminate the possibility of this happening.)

And, yes, if the mixing engineer decides to use a microphone that only hears up to 15 kHz, or even to apply a 10k low cut filter, that's his business - because he's PRODUCING music.
However, if it isn't there because MY equipment failed to reproduce it, then my equipment is broken - because it has failed to REPRODUCE what the engineer put there.

4.
As far as I know the "content" on vinyl _CAN_ extend to about 60 kHz (I'm defining "content" as "retreivable information").
I read that the pitch stability on some of the Grateful Dead albums reissued by HDTRacks was corrected by locking onto the residual 80 kHz carrier tone from the erase bias - which is still recoverable on tapes recorded on certain master tape machines.
I agree that I have no plans to either use a vintage click remover or restore any antique tapes.....
However, having extra information that I don't ever use is at worst useless, while not having information that I turn out to need later could be tragic.

5.
My point is simply that, if you recorded a CD-4 record directly off the stylus, and later decide to feed that signal to a decoder, it won't work if those frequencies weren't recorded.
This would obviously be a poor method for archiving four-channel content on CD-4 records.
And, again, if you want to claim that you have AN ACCURATE RECORDING of that Cd4 album, then it should be there.
(Or you must concede that you have saved a "processed" copy of the content rather than an accurate copy of the original.)
You have simply decided to set YOUR goal to exclude the requirement of saving all of the out-of-band content in its original form.

6.
I'm not forgetting that at all.
BY DEFINITION, whatever comes off that mastering console is "right".
BY DEFINITION, the job of a "high-fidelity reproduction system" is to reproduce whatever I play through it accurately and completely.
I absolutely agree that we have no provenance, and that it's a problem.
However, while I can't fix that problem, I can at least make very sure that I don't cause unnecessary damage or alteration.
I can only provide quality assurance on the parts which I control.

7.
If you're really trying to say that "damage is OK" then I guess we'll just have to agree to disagree.
In most situations, if a perfect example is available, it will be valued more highly than a damaged one.
That doesn't rule out the possibility that there may be no undamaged examples available, or that a certain damaged example might not actually be considered better than a perfect one for other reasons.
However, I don't think you'll find many dealers of legitimate antiques who would suggest deliberately breaking a vase, and then gluing it back together, as a way to increase its value.
Of course the price will be DIFFERENT if it's damaged.... and, of course, if you prefer to pay less for the damaged one, then that's your choice.
(Try putting up two ads on eBay - of the same vase - describe it in one as "perfect" and in the other as "broken - but with a perfectly invisible repair" and see which one goes for a higher closing bid.)

It occurs to me that this provides a perfect analogy to how *I* feel about both lossy compression and storing music in any format that is designed to carefully avoid storing any level of quality that isn't strictly and provably necessary.
To ME, that would be as if I had an undamaged vase which I needed ship, and someone were to suggest: "If you break it into pieces it will fit in a smaller box and be cheaper to ship. We can glue it back together at the other end and 
none of our customers will be able to tell the difference.... so what's the harm?"


8.
The reason I bring up lossy compression is the same reason that your favorite studio does NOT store their masters using it.
In order to "work", and deliver a "virtually indistinguishable" copy of the original, virtually all lossy compression relies on a whole bunch of assumptions and given conditions.
Feel free to suggest that, "in order to hear the noise floor on a CD I would have to turn my system up so loud that the loud parts would deafen me".
However, you seem to be ignoring the fact that I might decide to turn the volume way up on a quiet part to hear what one of the musicians mumbled under her breath.....
Are you suggesting that it is INVALID for me to do so?
Or are you simply suggesting that YOU wouldn't do it?
JPeG is a remarkably effective lossy compression for images; but NOBODY sane would use it as an archival format.
The reason is that, while JPG images may be "visually perfect copies of the original" under certain circumstances, that isn't true under ALL conditions.
Try boosting the contrast, or looking a little too closely at one tiny spot on the picture, and you often notice the flaws.
And, if you were to try and edit that picture, it's a virtual certainty that you'll "run into the limitations" caused by discarding all that data.

Of course, in real life, we always have to choose a set of requirements that suit US.
Someone who favors 24/96k as a recording format will be quick to point out that reducing the sample rate to 16/44k is obviously a form of lossy compression....
You are discarding information which cannot be gotten back.
I guess it also qualifies as "a perceptual lossy format" since you decided what was OK to throw away based on what you can hear - and what you expect me to be able to hear.

9.
As for ANALOG masters, since there's a format conversion involved, there is no direct comparison.
You may prefer that the tape hiss be omitted; someone else may insist that it be reproduced ACCURATELY.
(Just as, in video, some people prefer a smooth background, while others delight in figuring out what type of film was used by the original camera crew by examining the shape of the noise grains.)

Back in the analog days, it was IMPOSSIBLE to own a perfect copy of an album, because every copy process added a small amount of error.
Digital technology now makes it POSSIBLE for every customer to own an actual IDENTICAL copy of that original digital master.
To me, it seems like some sort of blasphemy to throw that opportunity away just to save a few bytes - or a few dollars.
It it seems as if I'm not overly concerned with whether they difference is "audible" or "significant"... then you're right.
I simply see no reason to forego "perfect" and go out of the way to look for an alternative which is "not perfect - but I'l never hear the difference".

I also do find it very sad that all of the attempts to claim to provide provenance have turned out to be either overreaching or just plain impractical.
I think it would be great if I COULD really buy a file knowing that it was a legitimate bit-for-bit copy of what came off the mastering engineer's console.
(Clearly the fact that so many companies promise it - even though they routinely fail to deliver on their promise - suggests that a lot of people agree.)

I'm also going to disagree with you on your definition of the word "practical".... simply because it is a word associated solely with personal opinion.
You seem to consider 24/96k to be "not practical".
I define 24/192k as quite practical - because almost all of the DACs I currently own support it (but I consider 32/768k to be a bit impractical with current equipment).
(Note that nothing was said about "necessary" - which is a different value judgment.)




pinnahertz said:


> 1. So, you're going to apply that same warped logic here too?  If a tweeter has output past 23kHz, then it produces ultrasonic energy?  Under that definition EVERY tweeter produces ultrasonic energy!  That's true in the absolute, false in the practical.  We live in a practical world.
> 
> 2. OK, you work for the manufacturer.  How about a set of polar plots of your "flat to past 25kHz" speakers, then?  Or link to manufacturer's data.  Any evidence at all would be great.  From what I've seen, and I've searched this oh just a bit, "flat past 25kHz" is a very, very tall order, especially when you look at dispersion and power handling too.  But I'll take those polars when you have them.  Then we can talk about off axis HF response of tweeters, microphones and oh yeah, ears.
> 
> ...


----------



## 71 dB

I believe this thread is about _consumer_ audio. In studios 24 bit / more than 44.1 kHz are _not_ worthless. Of course you digitize CD-4 vinyls with higher samplerate for archive reasons, but you can decode it, make for example matrixed Dolby Pro Logic stereo version and sell it to consumers as a 16 bit / 44.1 kHz downsampled version.


----------



## KeithEmo

But the thread title doesn't say anything about "consumer audio"......
And it doesn't say anything about higher sample rates only being beneficial under very limiter conditions.
It makes a blanket claim that "24-bit-audio-and-anything-over-48k-is-not-only-worthless-but-bad-for-music".

I would totally have no problem with the suggestion that "it usually doesn't matter"... and that "most people probably can;t hear the difference"... and that "with most content it probably doesn't matter anyway".
However, the thread, and the fellow who wrote the original article, reached much further.
He not only suggested that high sample rates were NEVER beneficial, but went on to claim that, because of the limitations in a lot of consumer equipment, would actually be audibly WORSE sometimes.
To me, that claim is about as silly as the claims of the vendors who sell high-res downloads are on the other.

The other thing I would like to point out is in the context of "practical" and "available" and "consumer".
As consumers, we are quite limited in what options we have.... and sometimes there is a distinct difference between "what could be" and "what is".

For example, a few years ago, HDTracks reissued the entire Grateful Dead studio album collection in high-res.
The albums were entirely remastered.
And, as it turns out, they sound very good - and far better than any previous versions (at least in my opinion and that of several others).
Do they sound better _BECAUSE_ they were remastered at 24/192k?
Or, at a minimum, will some of the improvement be lost if we down-sample them to 44k?
Does it really matter?
I bought the new remasters _BECAUSE THEY SOUND GOOD_.
It's moot to suggest that "they could have offered a version at 16/44k that sounded just as good" -  because they didn't.
(Well, in that case, I think they did offer a 16/44k version, but there wasn't much cost difference, since I still had to buy the whole set over again either way.)

So, yes, there is a big difference between theory and "what's good for most consumers".
However, as far as I can tell, this thread was started with a statement about the THEORY - that high-res files not only didn't sound audibly better but often sounded audibly worse.
As far as I'm concerned, in order to "prove" the original intent of the thread, you would have to prove that high-res files not only didn't sound better, but actually sounded worse at least some of the time.

What you seem to be talking about would be a thread titled: "Why high-resolution files are a waste of money for most listeners".
(And I would probably cheerrfully go along with THAT claim).



71 dB said:


> I believe this thread is about _consumer_ audio. In studios 24 bit / more than 44.1 kHz are _not_ worthless. Of course you digitize CD-4 vinyls with higher samplerate for archive reasons, but you can decode it, make for example matrixed Dolby Pro Logic stereo version and sell it to consumers as a 16 bit / 44.1 kHz downsampled version.


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## bigshot (Nov 15, 2017)

71 dB said:


> I believe this thread is about _consumer_ audio. In studios 24 bit / more than 44.1 kHz are _not_ worthless. Of course you digitize CD-4 vinyls with higher samplerate for archive reasons, but you can decode it, make for example matrixed Dolby Pro Logic stereo version and sell it to consumers as a 16 bit / 44.1 kHz downsampled version.



That's a given because we all know the benefits of high bitrates and sampling rates for professional recording studios. This is a forum for people who want to put together a home audio system. That's the context we're all participating in. But of course audiophiles rebel against context and try to spin things out into theoretical arguments so they can incrementally escalate the audiophoolery. That's a massive waste of time, because if you can't hear it and I can't hear it, it's totally irrelevant to the purposes of home audio. Inaudible is inaudible, and inaudible doesn't matter.

I sometimes think that audiophools should be philosophers or psychologists or priests. Those are areas where gray areas are celebrated and focused on. In Sound Science that kind of mental monkey knuckling is just boring.


----------



## 71 dB

KeithEmo said:


> For example, a few years ago, HDTracks reissued the entire Grateful Dead studio album collection in high-res.
> The albums were entirely remastered.
> And, as it turns out, they sound very good - and far better than any previous versions (at least in my opinion and that of several others).
> Do they sound better _BECAUSE_ they were remastered at 24/192k?
> ...



I'm not here to tell you how to spend your money. That's your own business. If great remasters are sold only at 24/192 kHz and you want them of course you buy them, but if they had them available at 16/44.1 kHz too, you probably would find them equally great. Maybe the idea was that it pays off to remaster if you can sell them at higher cost because it's 24/192 kHz?


----------



## bigshot

I've found that with the couple hundred SACDs and other "hires" formats I've bought, the odds of getting something that has better sound quality than the standard CD release is about 1 in 4. Half the time it sounds pretty much the same. 1 in 4 sound worse. Pretty much random. Multichannel has the potential for improving sound beyond just the quality of the mastering, but the quality of multichannel mixes vary too. I haven't found any correlation between hires and quality of sound at all.


----------



## KeithEmo

I'm inclined to agree with you..... some remasters sound better, some worse, and there seems to be little correlation with whether they happen to be high-res or not.
(Although, personally, while I like surround sound for movies, I tend to prefer stereo to surround sound for music.)
And, yes, I agree that the mixing and mastering make a FAR greater difference than the sample rate.



bigshot said:


> I've found that with the couple hundred SACDs and other "hires" formats I've bought, the odds of getting something that has better sound quality than the standard CD release is about 1 in 4. Half the time it sounds pretty much the same. 1 in 4 sound worse. Pretty much random. Multichannel has the potential for improving sound beyond just the quality of the mastering, but the quality of multichannel mixes vary too. I haven't found any correlation between hires and quality of sound at all.


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## KeithEmo

You're probably right. And, after hearing a few really well mastered CDs, it's very disappointing how bad the majority of the discs produced lately actually sound.
(Of course, they may sound exactly how the engineer wanted them to sound, or how he or she imagines the audience wants them to sound.)

It's also pretty obvious that re-masters have always been, and will continue to be, a viable way to sell more copies of albums that people already have. People have always been willing to buy "new and improved" copies of their favorite albums, and, if high-res, or MQA, provide one more selling point they can use to sell a few more, of course they're going to do so. The actual COST of storing a 24/192k copy instead of a 16/44k one is trivial for the vendor, as is the cost of the extra bandwidth it takes to download 1 gB of extra data these days, so, if it sells even a few more copies, then it's going to be worthwhile for them. 

(I would remind everyone, however, that remasters are nothing new... Dark Side of the Moon has been remastered somewhere around two dozen times.... and I assume people still buy the new ones. Studios have a huge investment in their music library - so they aren't going to stop trying to find new ways to make money from that investment.)

Another interesting thought is that, if a vendor sells copies of the same album at 16/44k and at 24/192k, it might actually be in their best interest to make sure that the 16/44k version sounds audibly worse - even if that entails using a different mix or using an inferior piece of software to create the "cheaper" version from the "premium" version. (I haven't heard anyone accused of doing so, but it might be interesting to see if, in specific cases, the 16/44k version someone sells is slightly different than what you get when you down-sample the 24/192k version they sell.)

I would also advise caution when assuming that all programs do an equally good job at converting between sample rates. 
There is a website that has published comparisons of how different programs perform when doing sample rate conversions.
It's interesting to note that, while many do an exemplary job, many do a rather poor job.
(Interestingly, for example, the CS6 version of Adobe Audition does an excellent job, while the CS6 version of Adobe Media Encoder does not.)
It's also informative to compare both sweep spectra and impulse response (note that most apodizing filters have a lot of aliasing).

I'm not interested in getting into a discussion about which flaws are audible.
(Or why, when the technology is obviously widely available, anyone would still do it badly.)
However, it makes sense to me to choose a program that technically does a good job, and avoid software that does a technically inferior job.
(And it makes you wonder which CDs were, perhaps, converted using one of the converters that performs very poorly.)

That website is:   http://src.infinitewave.ca/



71 dB said:


> I'm not here to tell you how to spend your money. That's your own business. If great remasters are sold only at 24/192 kHz and you want them of course you buy them, but if they had them available at 16/44.1 kHz too, you probably would find them equally great. Maybe the idea was that it pays off to remaster if you can sell them at higher cost because it's 24/192 kHz?


----------



## danadam

KeithEmo said:


> Another interesting thought is that, if a vendor sells copies of the same album at 16/44k and at 24/192k, it might actually be in their best interest to make sure that the 16/44k version sounds audibly worse - even if that entails using a different mix or using an inferior piece of software to create the "cheaper" version from the "premium" version. (I haven't heard anyone accused of doing so, ...


Well... there's this:


gregorio said:


> Not so much actually record but master differently, certainly. In fact, way back earlier in this very thread I remember an exchange I had with a representative of a distributor, I can't remember which distributor but I think it might have been HD Tracks or Linn, where they stated they do (or did at that point in time) routinely make their 16bit versions poorer quality! Of course, that's not exactly what they stated, they stated that they make their recordings as good as possible and that in the case of their 16bit versions they add a significant amount of compression. This is because, they say, many of their clients then lossy encode their 16bit version for use in portable devices where more audio compression would sound better. Now that is a reasonable/acceptable response but when I basically said fair enough but instead of distributing a more highly compressed 16bit version and a far more expensive "HiDef" version, why not just sell a second 16bit version but without the additional compression (which would be audibly the same as the HD version). No response, the silence was deafening!


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## bigshot

I have a Rolling Stones SACD where the redbook version isn't even the same mix as the SACD layer.


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## old tech

bigshot said:


> I have a Rolling Stones SACD where the redbook version isn't even the same mix as the SACD layer.


Interesting that it is a different mix, however it is not uncommon for the CD layer of a SACD to be more compressed.  The rationale of the labels is that the CD layer is better for everyday listening (eg in noisier environments) while the more dynamically mastered SACD layer is for quiet listening rooms.


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## pinnahertz

KeithEmo said:


> 1.
> I'm sorry, but, yes. If the speaker has output past 23 kHz, and I can measure that output with a microphone at the listening position, then the tweeter is indeed "delivering ultrasonic energy to the listener" (assuming we're referring to 23 kHz as "ultrasonic").
> In the practical world, if we can measure something, and the measurements show it's there, then it's there.
> (I would agree that, if it's 20 dB down at 25 kHz, it probably won't be doing anything useful at that frequency.)
> ...



1. That's exactly the kind of warped logic I'm talking about.  It's like defining "any"="all".

2.  Nice.  I'll admit to being impressed, except for the even slightly off-axis response, which means this works only for one seat.  And except you said "there are plenty of speakers", and you've cherry-picked an example.  You also said they measure flat past 25kHz.  I don't consider -10dB@30kHz "flat past 25kHz".  No matter, you've found an exception, not the norm or typical. 

3. NO relation to reality here.   There's no "production" going on above 20kHz.  How could there be?  Do you think all control room monitors are flat to 30kHz, or that engineers hear above 20kHz? The 10kHz high cut filter example is also ridiculous, completely over-reaching reality.  The actual question should be: how high is actual music content? That's content that is actual music, not distortion products caused by transducers. 

4a. What's the 3rd harmonic of 20kHz? How about the 3rd harmonic of 8kHz?  Are you aware that harmonic distortion in the vinyl system is HUGE?  See, this is the real problem, the assumption that 60kHz coming off vinyl or even 25kHz is somehow part of the original acoustic event.  Not very likely, especially considering they were mostly mastered from tape, which rolls off quite quickly above 20kHz.   The nonlinearities in the total system are teutonic at high frequencies.  The distortion characteristics jump up quite quickly as a function of level and frequency because of the physical limitations in the geometry of the groove, cutting stylus and reproducing stylus. For example, you can't have a cutting stylus velocity that exceeds a physical maximum without the stylus's rear facet scraping the groove wall just cut.  A similar physical limit exists in playback.  Given that, it's far, far more likely that what's going on is the result of distortion mechanisms...many of them.  What's wrong is, since you don't have the original acoustic event PLUS an ultrasonic-capable microphone system to evaluate it with, you make the assumption that it was original.  But that's just not true.

4b. Another completely unrelated point.  The bias signal used in analog tape is several orders of magnitude hotter than the actual audio signal.  It literally operates for the purpose of linearizing the recording capability of an otherwise extremely nonlinear medium.  Tape must have very high coercivity and retentivity to store information, which results in terrible linearity and very high distortion.  The bias signal enables acceptable distortion by carrying the audio signal way past the nonlinear portion of the magnetic characteristic of the tape, while still using a very magnetically stable medium. A bit of residual bias signal recorded on tape has been used by the Plangent process to effectively correct time-base errors, but that process involves special tape heads and extensive DSP to recover the bias signal and be able to derive a time-base signal.  N_one of that has anything to do with the actual HF response of magnetic tape machines!_  In fact, that same bias signal tends to erase HF audio as it is being recorded, and the level of bias has to be optimized for each tape type as a trade-off between minimum distortion and equalizable HF response. The optimum usually results in flat response to 20kHz, then a rather rapid roll-off above that as bias erasure rate exceeds the capabilities of record equalizers to compensate for it.   Once again, an inappropriate example.

4c. In some circles this is recognized as a manifestation of OCD.  In others, a full understanding of what those signals actually are will help make a reasonable capture decision.

5a. Yes, that would be a very poor method, and completely the wrong way to do it, as it would take post processing to recover the discrete 4 channels that doesn't exist. 
5b. You've taken the discussion over to archiving a CD4 record, and that has nothing to do with the HF response or content of an LP.   A CD4 record was a special case, full of flaws and limitations, and it was a commercial failure. The performance of the CD4 system was compromised at best.  I have no idea why anyone would want to archive the content of one of those records.  This is of course apart from the fragile nature of the HF carrier, which literally could be worn beyond usability with normal play.  If the recording is at all important it likely exists in some other more usable form, like the original analog tape masters.  Any attempt to even play a CD4 record would be so far inferior to the masters as to be pointless, especially from an archive point of view. The comparison of CD4 with the capability of vinyl to record and reproduce ultrasonic content is meaningless even if you cherry-pick a specific example.

6. Yes, and that quality assurance is admirable.  But what would be better is a full and complete understanding of the content, which isn't what's going on here.  It seems that you're saying "We don't understand what's really there so over-capture so we don't miss anything".  Well, we do understand what's there.  It's not even that hard.  And that understanding changes what we need to do.

7a. The problem is that a perfect example is often suspected of being a fake or reproduction.  A classic case is old metal and porcelain signs.  They are so hot right now that they've gone full steam into reproduction.  And the reproductions are very, very good!  However, they are also easily recognized as being far too perfect, so current reproductions include actual simulated damage.  Thats why the actual real old version, with all it's damage, is worth 10X that of a repro in much better condition.
7b. That's just silly, and I'm just slightly insulted at the insinuation.
7c.  No, back up one step to "try putting up two ads on eBay of the SAME VASE".  First challenge is right there.  Then of course if you do have that, and one is damaged and visibly repaired it will be worth much less.  That's not my point at all.  Have you failed to see my point, or do I need to go through this again?  Your example is not appropriate to this discussion at all.
7d. And hold it again.  Now you've equated all lossy compression with being by nature of lower quality than the original.  Sorry to use your own warped logic, but that cannot be a true statement at all.  If there is lossy compression that is indistinguishable from the original, then we must conclude that even a lossy codec exists that is capable of handling all material without audible loss.  It's just a question of how much bit rate reduction is applied and how.  No, the analogy still doesn't work.
7e. Completely wrong, yet again.  A vase cannot be broken into pieces and reassembled without any visual damage or repairs because the damage is random and not specifically chosen to be the type of damage that is invisible once reassembled because the cracks are masked somehow.  If you understand how lossy compression works, (and it's not a binary thing at all!), it's data reduction is based on psychoacoustic masking properties which are both variable by parameter setting, and variable dynamically based on content.  There is no parallel to breaking a vase.

8a.Yes, of course. Are you implying the assumptions are incorrect?
8b. You've moved over into creative writing now.  I don't believe I've mentioned your playback volume at any time, or anyone else's either.  I never said anything about playback conditions being valid or invalid.  Please don't make up things and attribute them to me.
8c. You've just labeled thousands of photo editors "insane".  JPEG image compression is, in fact, and has been a format that is extensively archived.  Not necessarily by desire or design, but if an image was originally shot that way, and in the early days of digital photography nearly every image was, then it could remain archived as the original file with no loss.  Yes, there are millions of archival jpg images in storage, just as there are millions of lossy film negatives in storage.  Here we go again into another inapplicable analogy....
8d.  JPG compression is completely variable, and the degree of loss is a choice.  What's far more important in image quality is bit depth and total resolution.  I have a professional DSLR that shoots JPG and RAW images, both at the same resolution, and the resulting file size (the degree of jpg compression) can be chosen to be nearly identical to the raw file.  If the exposure is correct, there's no visible difference.  RAW gets you more control of the extremes of exposure because it's essentially a copy of the sensor output, but that's not because of the lack of lossy compression, a RAW file is a completely different beast.  A TIFF version of the RAW image would be as limited in contrast editing as a jpg, yet the TIFF is also uncompressed and is not considered lossy.  However, Camera RAW is not a "release" file format, as each camera has a different profile and format that are not universally compatible.  There is no direct parallel in audio.  You need to leave off from analogies, they aren't working.
8e.Incorrect!  Lossy compression reduces data based on psychoacoustic masking.  Resampling doesn't.  If the total audio bandwidth and dynamic range of a given recording can be represented by 16/44, then  resampling at 24/96 recording to 16/44 is lossless!  No data that represents the original recorded signal is lost because no data actually uses the total dynamic or frequency range in the first place.  Resampling can be destructive if the resulting resampled version must exclude data that describes the original, but that's not always the case in audio. Resampling is not perceptual.  You are discarding information that may not be gotten back, but if that information only describes noise and not the original signal, it is not a loss to discard.

9.  I have no issue with capturing the original in as complete a representation as possible, and distributing that version.  I have big problems with the general assumption that high sampling rates and bit depths result in categorically better sound, and question strongly the value of ultrasonic content over 20kHz and the ability of the average or even high-end user to get that energy to his eardrums (much less to actually hear it).  I do think there are cases where ultrasonic energy, real or distortion products, can cause problems in devices not capable of handling them without distortion.  I do believe there are cases where the process of band-limiting audio by the use of certain devices results in intermodulation distortion that can be folded down into the audible region making wideband audio sound better for a reason other than the ultrasonic content.  I also believe that those cases are fewer today than say 25 or 30 years ago, and the real solution is to test for high frequency intermodulation and deal with the cause rather than to band-aid a solution either by passing a wider bandwidth or limiting ultrasonics by filtering.   Only a few decades back there were audio products that included ultrasonic filters to prevent those signals from wreaking havoc in other devices.  Hopefully today those devices are few, but I doubt they're completely gone.


----------



## KeithEmo

The problem is that you're making a straw-man argument there.

Yes, _IF I WERE TO SET MY VOLUME CONTROL AND NEVER MOVE IT_, then, in order to hear the noise floor on a CD, I would have to be playing it so loud that the loud parts would damage either my hearing or my playback equipment. 
However, like many people, I do occasionally turn up the volume on the quiet parts to hear something better.
And, sometimes, I even get up from my listening seat, and walk over near the speaker..... at which point that nice low noise floor suddenly becomes audible.
Now, I wouldn't claim that we all _NEED_ a 200 dB S/N ratio, just in case we crank it up 100 dB higher in the quiet spots.
However, that's one of those reasons why we include things like safety margins whenever practical; just in case we decide to.

You suggest that "we don't need" the stuff we won't normally hear...
I say that, while we may not _NEED_ it, I _PREFER_ not to hear background noise when I walk by the speaker, or to hear hiss with a less annoying spectrum.
Likewise, even though the highest speed limit I've seen is 70 mph, I prefer to have a car that can do 100 mph - just in case I need to pass someone who's speeding.
(It's also why people pay extra for sports cars that can go 150 mph, or can accelerate to 90 mph faster than mine - because, _TO THEM_, they're willing to pay extra for that extra safety margin.

I think it's always going to come down to what we consider "normal usage" - which is were we disagree.
I's also true that, in most situations (when you're looking at the picture at a "normal viewing distance" in a "normally lit room"), a JPG picture will look just as good as an uncompressed image.
However, I download a lot of pictures, and I run into situations quite often where I want to zoom in on a section of one, or turn up the contrast......
And, when I do, if all I have is a JPG, I often end up running into the limitations of the reduced quality, where my results would have been far better if I'd had a better quality non-compressed copy.
(I decide the picture is a little dark, so I turn up the brightness, and, suddenly, those "invisible" compression artifacts are very visible.....  )

Again, I may agree with you that I _USUALLY_ don't need it, and that it USUALLY won't make a difference....
But I'd rather pay a tiny bit extra for something I don't need 95% of the time than end up wishing I had it the other 5%.
And, yes, when I take pictures I_ ALWAYS_ use the lossless setting......
Because, like with audio, storage space in cameras is cheap these days, and I never know when I might need the extra quality or safety margin later.
I'd much rather throw away 50 tB of extra space on the pictures that turn out not to matter than find out that I screwed up one that _DOES_ matter.
(I _UNDERSTAND_ all the people who chose not to buy a camera that can take RAW pictures, or who prefer to save the space.... I just don't agree with their priorities.)

On that note.......
I've purchased quite a few albums from CDBaby (they offer "small volume productions" online).
Up until now, they have offered their content on physical CDs and/or downloads... and their downloads have always been available as lossless FLAC files or a variety of other formats.
(They charge a few bucks more - plus shipping - for the physical CD versions.)
They have recently announced that, while they will continue to offer actual CDs, they will soon only be offering downloads in lossy formats.
(They cite, as an excuse, that they can "ensure the provenance of a CD but not a file". Since they sell CD-R's, created from files anyway, I consider this to be a specious excuse.)
Obviously I won't be purchasing the download versions any more (just as I avoid purchasing music from iTunes)...
I guess we'll see how many other people agree (if we ever hear).



bigshot said:


> Why not just do a little googling and find out what additional benefits higher bitrates and sampling rates offer, then compare that to the thresholds of perception and you'll have your answer.
> 
> higher bitrate = lower noise floor
> higher sampling rate = extended frequency response
> ...


----------



## KeithEmo

That makes sense - sort of (except for the part where they neglect to mention it).
If they were honest, they could put a positive spin on it, and claim that "the CD layer is optimized for portable equipment" or some such similar claim.
I also suspect that sometimes it occurs for less planned out reasons (where the mixing engineer, knowing that the SACD master is for "the audiophile version", masters it differently according to whatever they believe "audiophiles" would want it to sound like.)
If people expect the DSD layer to sound different, in a certain way, and will interpret that as "better", it only makes sense to give them what they want...
Of course, sadly, it also fuels the claim the "SACDs sound better than CDs" since many people are under the mistaken assumption that they are making a fair comparison when they switch back and forth on a given disc.



old tech said:


> Interesting that it is a different mix, however it is not uncommon for the CD layer of a SACD to be more compressed.  The rationale of the labels is that the CD layer is better for everyday listening (eg in noisier environments) while the more dynamically mastered SACD layer is for quiet listening rooms.


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## bigshot (Nov 17, 2017)

KeithEmo said:


> However, like many people, I do occasionally turn up the volume on the quiet parts to hear something better.



That's the textbook definition of a bad mix. At normal listening volume, everything should be clear. You shouldn't have to reach for the volume knob when you listen to music. You should just listen to music without being interrupted like that. Imagine if I went to the Disney Concert Hall to hear the L A Philharmonic and I got up out of my seat and walked up on the stage because I wanted to hear the harp. They'd kick me out and rightly so. Stay in your chair! Don't touch that dial!

There's a proper way to appreciate music. You wouldn't go to an art museum and pull out a magnifying glass to examine the cracks in the impasto. The same goes for music. You should appreciate it for the entire composition, not go out of your way to look for flaws. We are imperfect humans living in an imperfect world. There's no pleasure in focusing on the imperfections to the exclusion of the beauty around us. In fact, that is an endless rabbit hole that will drag you down into audiophool places you don't want to go. Listen to music, don't diddle with your volume knob!


----------



## KeithEmo

I clipped the quote because it was getting terribly long......

2) 
I can't speak for other people's speakers or their capabilities.
I'm also not claiming that I choose my speakers based on their response above 20 kHz.
I'm simply disproving the claim that "speakers don't have useful response above 20 kHz". 
(The claim is wrong because, clearly, at least _SOME_ speakers do.)

3)
Offhand I have no idea who is filtering what where... nor what the response limitations are on the specific microphones they may have used (or their synthesizer's frequency response if it's electronic).
I simply prefer for my equipment to be able to reproduce the entire musical spectrum if it's there in the recording.
(I consider that to be the definition of "accurate reproduction".)

4a) - 5a)
I agree that most of us don't actually need to accurately reproduce CD-4 content....
Likewise, I agree that most of probably will never use the erase carrier tone leakage (and, yes, special tape heads and electronics were used)....
My point, which I stick by, is that we don't always know our future needs in advance....
(And, if someone says "if you turn the volume up at 14:27 you can hear the musician cough" or "a bat got into the studio and you can see his squeak on a spectrum display at 34:10" I would prefer to be able to try it.)
And, yes, I would prefer a full-spectrum copy of the tape rather than one that has been limited to "what I need" - based on someone else's ideas of what that comprises.

Again, I simply see the goal as an absolutely perfect and complete rendition of the original.
And I see _ANY AND ALL LIMITATIONS_ as a compromise.
And I prefer to avoid compromises unless absolutely necessary (or unless I agree to them).

Someone asked, jokingly, whether videophiles would prefer it if their TV could reproduce gamma ray frequencies so accurately they could be burned by the video of a nuclear explosion.
While the example is absurd, I would say that the answer is technically yes.
We would be better off if our video display could reproduce frequencies from DC to gamma rays - and then we or the producer of the video could _DECIDE_ which ones to limit or omit.
(I'm sure the government would cheerfully add some sort of safety standard to cover that aspect of things.)
However, in that case, both the safety and technical limitations _JUSTIFY_ setting a standard that falls short of that lofty goal.

As a counter-example I would offer the color purple.
Most NTSC-standard TV sets don't reproduce a deep yet bright saturated purple color very well at all.
Therefore, most SD videos rarely show anything in bright purple - like a shiny amethyst necklace.
We have a situation where "the equipment can't play the signal" and "there's no point in including the signal because nobody's TV will be able to play it".
It has reached a point where set designers avoid using certain colors because they know those colors will be poorly reproduced.
(And, when you play those few videos that ignore the limitation on a full-spectrum monitor, you see an immediate difference. Now, interestingly, delivering the full color gamut - or more of it - is a major selling point of 4k HDR.)

6)
I do have an easy question for you......
If "quality assurance is admirable" (your words), then why are you arguing against it?

7a)
I'm not concerned with fakes... or of how to distinguish them from legitimate copies.
In either case, a properly provenanced legitimate original that is undamaged will virtually always be worth more than one that is damaged.
Likewise, even an undamaged copy will be worth more than a damaged copy.
Notice that you said "VISIBLY" repaired..... while I didn't include it......
While visible repairs are surely worse than invisible ones.... damage is still damage.
If someone buys your expensive vase, and finds a repaired crack when they x-ray it, they probably _WILL_ sue you if you claimed it hadn't been repaired.
"Visible" and "nonexistent" are not the same thing at all......    (otherwise a perfect forgery of a Rembrandt would really be just as good as the original).

Yes, if you can find a lossy compression method whose output will be _INDISTINGUISHABLE_ from the original, using our ears, or any test we can device, then it will be perfect.
(Except, of course, by definition, it _WON'T_ be lossy at that point.)

The fact that some people find it easier to identify imperfect originals than perfect copies is irrelevant....

I am _WELL_ aware of how lossy compression works.......
And it _ALL_ amounts to "discarding content that someone else has decided I won't notice is missing".
(And I have very little faith in the choices made by other people - especially when those choices are often made based on "what 95% of people won't notice" rather than specifically on "what ***I*** won't notice".)

8)
You're entirely incorrect in one regard......
While the compression level used in JPG is variable - _THERE IS NO SETTING IN THE JPG STANDARD THAT EXACTLY REPRODUCES THE ORIGINAL_.
Even if you set a compression level that results in a file that is larger than the RAW file it is still lossy (you cannot retrieve the original pixels exactly). 
A RAW file contains all of the information that was available from the camera; which is why it gives you the most flexibility and retains the most information.
An "uncompressed TIFF file" contains most but not all of the information; but is a much more standard format - which justifies the slight loss.
A JPG file discards a significant amount of information.... because a lot of information is approximated or discarded outright.

I also suspect you haven't edited many image files.... especially JPGs.
The method of compression used by JPG is applied to square zones of the image (so each square of a certain size is processed separately).
The compression is applied with certain constraints that ensure that the seams between adjacent squares won't be visible.
However, those constraints are based on several assumptions, including the conditions under which the image will be displayed, and the characteristics of the monitor or printer that will be used.
As a result, even though a given JPG may look "visually perfect" under very certain conditions, their failings tend to become unpleasantly obvious when you change the conditions.
Specifically, when you adjust the brightness, contrast, or color saturation, the sharp discontinuities at block boundaries become visible, and you get that "JPG blockiness artifact" that so many people find annoying.
(I'm ignoring the fact that virtually every pixel has been changed from its original color even though the net overall difference may be "imperceptable" to most people.)

Of course, there is an "original loss" because, while many cameras may exceed the performance of the human eye in specific ways, none so far exceeds the human eye in _ALL_ regards simultaneously.

Take a picture of this posting, on a really sharp screen, and save it as RAW and as JPG.
Blow it up so you can see the individual letters.
I would be very surprised if you fail to see odd ghosts and echoes around the edges of the letters.
(Because JPG was optimized to compress "pictures with continuous tones", which works well for photographs, but does relatively poor job on sharp edges and narrow lines.)

And, yes, there is a parallel in audio.....

If the audio recording was originally recorded in MP3, then, yes, the most accurate rendition of the original would be a copy of that MP3.
And, if it was recorded at 24/96k, then that file would be the most accurate version.
(Of, assuming it was mixed, that would be the final output file that was sent to the production house.)

_BUT_, if you convert that MP3 file to 24/96k, you will be able to get back a very close approximation of the original MP3 file.
_HOWEVER_, if you convert that 24/96k file to an MP3 file, you will NOT be able to get back a close approximation of the original 24/96k file.

At all comes back to your original assertion that "none of the stuff we're throwing away really matters"...
I will agree that, for most people, on the equipment they'll be using, they probably won't notice the difference.....
But, if I was hoping to use the signature of that background noise to determine the brand of recorder the original was recorded on, then you have ruined my project if you discard it.
_YOU_ have decided that the background noise is_ NOT_ "a meaningful part of the recording" - but perhaps we don't all agree.
And, yes, if you're the recording engineer, then, by definition, any such decision you've made is "correct".

9)
We seem to be in perfect agreement on the final conclusion.
Personally, I want the unlimited version, and am willing to pay a little extra for it.
However, I do absolutely agree that many people may find no benefit to high-resolution files, and shouldn't buy them.

Incidentally, _as for your other comment...... ALL_ audio circuitry has bandwidth_ limitations_.....

They may be inherent in the circuit design or specifically imposed using an external filter - but they are necessary.
Essentially _ALL_ circuitry produces noise... and, in general, that noise has an infinite bandwidth.
So, even though your microphone may not pick up 30 kHz, the active devices in your microphone preamp are putting out _SOME LEVEL_ of 30 kHz noise (and some level of 10 mHz noise).
If you're planning to digitize that output, you must filter out any significant noise above the Nyquist frequency to prevent audible aliasing and other errors.
You also _MIGHT_ have issues, as some have suggested, with intermodulation distortion caused by high-frequency noise (or content).
And you really wouldn't want to die because your amplifier accurately amplifies the leaked noise it picks up from your microwave oven or cell phone.

There's also another technical issue that_ MOST_ (probably all) amplification circuits exhibit phase shift that increases as the frequency increases.
As a result, at some very high frequency, the phase shift is such that the circuit's negative feedback becomes positive feedback, and the circuit oscillates.
(And virtually all modern audio circuitry employs negative feedback.)
Because of this, audio circuits are designed so that their gain falls at high frequencies..... which is another way of describing "a bandwidth limiting filter".

With older equipment, this limitation was often more or less a random result of the overall design.... or simply a matter of luck.
However, with modern design, it is assumed that each piece of equipment has been designed to "protect itself from anything it would have a problem with".
Basically, with analog equipment, the circuit is designed so, as you look at higher and higher frequencies,  the gain reaches unity or lower before the phase shift reaches a dangerous point. 
(Most modern amplifiers have a frequency response that starts rolling off significantly around 80 kHz or so - as that is considered to be "well outside the audible frequency range".)



pinnahertz said:


> 9.  I have no issue with capturing the original in as complete a representation as possible, and distributing that version.  I have big problems with the general assumption that high sampling rates and bit depths result in categorically better sound, and question strongly the value of ultrasonic content over 20kHz and the ability of the average or even high-end user to get that energy to his eardrums (much less to actually hear it).  I do think there are cases where ultrasonic energy, real or distortion products, can cause problems in devices not capable of handling them without distortion.  I do believe there are cases where the process of band-limiting audio by the use of certain devices results in intermodulation distortion that can be folded down into the audible region making wideband audio sound better for a reason other than the ultrasonic content.  I also believe that those cases are fewer today than say 25 or 30 years ago, and the real solution is to test for high frequency intermodulation and deal with the cause rather than to band-aid a solution either by passing a wider bandwidth or limiting ultrasonics by filtering.   Only a few decades back there were audio products that included ultrasonic filters to prevent those signals from wreaking havoc in other devices.  Hopefully today those devices are few, but I doubt they're completely gone.


----------



## castleofargh

KeithEmo said:


> That makes sense - sort of (except for the part where they neglect to mention it).
> If they were honest, they could put a positive spin on it, and claim that "the CD layer is optimized for portable equipment" or some such similar claim.
> I also suspect that sometimes it occurs for less planned out reasons (where the mixing engineer, knowing that the SACD master is for "the audiophile version", masters it differently according to whatever they believe "audiophiles" would want it to sound like.)
> If people expect the DSD layer to sound different, in a certain way, and will interpret that as "better", it only makes sense to give them what they want...
> Of course, sadly, it also fuels the claim the "SACDs sound better than CDs" since many people are under the mistaken assumption that they are making a fair comparison when they switch back and forth on a given disc.


yup, and as is the case most of the time, instead of simply advocating for the good aspect of what they do, we end up with no information at all so that people can mistake mastering for format sound. or worst, we get the kind of BS marketing/propaganda we got for Daft Punk's Random Access Memory. where they went to interview the various guys involved and have the engineer say the dumbest most dishonest stuff about the reason why they made different masters for the different media.
the guy (forgot his name, the french dude, not the US guy doing the recording), implied that some stuff wouldn't "translate" on CD so they did it differently... I could imagine him talking to his friends in the industry in one of those "and then I told them" meme





 it gets even "funnier" when you actually listen to the versions. wow! yeah, now I get it, no way that massive low end boost could have worked on the CD version, CDs can't handle a bass boost, everybody knows that. it's just not dynamic enough...

it's infuriating because they take us for the dumbest milk cows ever. but what makes it worst is that they never needed to lie. they could have just said, "when you buy the SACD you get several versions of the song as an exclusive". or indeed argue that for listening in a noisy environment, the more compressed version would be more practical. there are numerous ways to sell it as a good thing. but at some point, marketing is just so used to lie about made up benefits of high res, that they forget how some products could actually sell themselves for what they are.


----------



## KeithEmo

I kind of agree.....

Except that you're mixing philosophy and technology.
Whether you _SHOULD_ diddle with that knob is a matter of _philosophy_; what happens when and if you do is a matter of _technology_.

And, saying that somebody shouldn't care about something is philosophy...
While saying that it isn't there is a matter of fact.
(There is a one-way philosophical overlap that says that worrying about something that is "meaningless".)

I should note that you seem to have highlighted my main issue with this thread.....
A lot of well meaning people have suggested that I shouldn't _CARE_ about certain things.
And, that being a philosophical argument, I happily concede that they may in fact be right.

However, rather than convince me that it isn't important, they seem determined to insist that "it really isn't there so I'm silly to even be talking about it".
I'm sure you're right about their not being pleased if you decide to go walkabout during a concert.
However, I wouldn't suggest you were wrong if you said you wanted to try different seats so you could decide which one you want to purchase next time.
And I especially wouldn't try to convince you that, "since it is a professionally designed concert hall, all of the seats are good, so it doesn't matter".

And, while I would also cheerfully agree that many "audiophile claims" really are simply untrue....
I don't think that's true in this case (more accurately, while I also concede that it _MIGHT_ be true, I don't think it's been _PROVEN_ beyond reasonable doubt).

If I, as a scientist, say "'there's no audible difference", and _ONE GUY_ shows up who can reliably hear one, then I've been proven both a liar and a fool (and, worse, a _bad scientist_).
I'd rather stick with "most" and "probably" - and get back to the _PHILOSOPHICAL_ discussion of whether a difference that less than 1% of us can hear is worth bothering with or not.



bigshot said:


> That's the textbook definition of a bad mix. At normal listening volume, everything should be clear. You shouldn't have to reach for the volume knob when you listen to music. You should just listen to music without being interrupted like that. Imagine if I went to the Disney Concert Hall to hear the L A Philharmonic and I got up out of my seat and walked up on the stage because I wanted to hear the harp. They'd kick me out and rightly so. Stay in your chair! Don't touch that dial!
> 
> There's a proper way to appreciate music. You wouldn't go to an art museum and pull out a magnifying glass to examine the cracks in the impasto. The same goes for music. You should appreciate it for the entire composition, not go out of your way to look for flaws. We are imperfect humans living in an imperfect world. There's no pleasure in focusing on the imperfections to the exclusion of the beauty around us. In fact, that is an endless rabbit hole that will drag you down into audiophool places you don't want to go. Listen to music, don't diddle with your volume knob!


----------



## KeithEmo

Perhaps they just want to force everyone to buy _ALL_ the versions .

I have a similar issue with certain symphonic metal bands I like.
They are very international, and each of their albums is published in a variety of different "country versions".
Annoyingly, each has one or two bonus tracks, and each national version has _DIFFERENT_ bonus tracks.
They insist that it's "to please different fans in different countries"....
But it seems clear to me that they simply want to force their serious fans, who don't want to miss any songs, to buy a copy of _EACH_ version to get all the bonus tracks.

(One band, as I recall, Edenbridge, actually had the courtesy to issue a separate album of the various bonus tracks so you could buy them all at once.)




castleofargh said:


> yup, and as is the case most of the time, instead of simply advocating for the good aspect of what they do, we end up with no information at all so that people can mistake mastering for format sound. or worst, we get the kind of BS marketing/propaganda we got for Daft Punk's Random Access Memory. where they went to interview the various guys involved and have the engineer say the dumbest most dishonest stuff about the reason why they made different masters for the different media.
> the guy (forgot his name, the french dude, not the US guy doing the recording), implied that some stuff wouldn't "translate" on CD so they did it differently... I could imagine him talking to his friends in the industry in one of those "and then I told them" meme
> 
> 
> ...


----------



## bigshot (Nov 17, 2017)

As I said to the other guy... My grandmother could hear the noise floor of 16 bit if you turn the volume up high enough. I'm sure you could hear the noise floor of an SACD too if you crank the volume. That isn't any surprise. If I grab a magnifying glass I can see pixels or film grain or halftone dots in photographs. If I look close enough I can see a single grain of sand on a beach. Heck, with a scanning electron microscope I can see an atom!

Turning up the volume of a CD so high that you can hear the noise floor is totally irrelevant to the purpose it was designed for. The purpose of a CD is to put it on your stereo and listen to music in your living room. Expecting 16/44.1 to wash your dishes for you or mow the lawn or split atoms is ludicrous. It was never designed to do that. Who cares? Why are we even talking about it? Beats me.

When an argument goes this far off the rails, you really don't have an argument any more. I see a lot of arguments around here lately that are seriously out of touch with reality. I think there is some sort of virus going around or something!

We're in a forum about home audio. For the purposes of listening to music in the home, 16/44.1 is all you need. Properly designed DACs (even cheap ones) are audibly transparent and do the job too. It's fine to talk about moonbeams and unicorns, but it's more entertaining to express those thoughts in small doses.


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## RRod

If you're turning up the quiet parts, you've become a human upward compressor. I think we all agree there are optimal ways to produce compressed material, but using a compressed version of a track as a justification for more bits for *delivery* seems a bit off.


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## 71 dB

I'm starting to feel this is pointless. Some of us think 16/44.1 is enough, others think we need more. Nobody is convincing anyone to change mind. I don't know how much more I am willing to keep on debating. Please continue if you want… …I guess 20 years from now it's about whether 24 bits is enough of if we actually need 32 bits to listen to atoms collading in our music. That's an era when us 16 bit advocates look like a bunch of cavemen learning to make fire.


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## castleofargh

well the loaded title of this topic isn't really helping. it makes anybody who doesn't disagree, look like they're claiming 16/44 sounds better. and although under some circumstances it could happen, it's really fighting a dishonest position with another dishonest position. 

I believe that 16/44 sounds just as good on most good devices(notice how that's an opinion), and that the potential improvement in resolution isn't really worth the extra money asked for high res files(opinion again, I'm on roll!). that's not much of a controversial position, and I don't think anybody would get mad at me for having it. at best the question of "how much money is too much money?", would lead to existential questions well outside of audio. but for sure it wouldn't have led to 177 pages of struggles saying the same stuff 50 times. ^_^

ultimately if somebody feels for any reason(including all the non audio ones, and even the unreal ones), that high res sounds better, we're not holding a knife on his neck to stop him from getting high res music. on both sides we're saying that people should try for themselves and decide what is worth it for them. the only sensitive area is about what should count as a valid listening test . but even that is relevant only for those who can't help but make claims, not to decide what we like most.


----------



## KeithEmo

Exactly.....

And, if we get it really right, we should be able to tell how many cylinders its engine has - and whether it's had a tuneup recently.



Hutnicks said:


> The vibrations caused by the Freightliner driving by the studio during the recording session?


----------



## KeithEmo

The problem, as I see it, is that not everybody here is participating in exactly the same discussion.

One discussion would be: "Are there audible differences?" - as "hard science".
Another would be: "Are there differences that most people would find significant?" - which has brought opinion into the matter (significance is a matter of opinion).
Yet another would be: "Are people being mislead about the differences - or lack thereof?"
And yet another variation would be: "Does it make SENSE - in some abstract sense - to worry about the difference?"

One person may argue that, if the difference between two DACs is less than the difference he hears when he moves his chair six inches, then it is meaningless.
Another may argue that, if he hears a difference between two DACs, _EVEN IF HE DOESN'T MOVE HIS CHAIR SIX INCHES_, then_ HE_ considers that difference to be significant.
Neither is right or wrong (because both acknowledge a difference - and whether that difference is "important" or not is a matter of opinion.
A third person may argue that the entire discussion is silly "because there really is no difference"........
However, because that is a scientific claim, it is something that _CAN_ be proven... or can fail to be proven (you can probably never prove the negative).

The original post on which this thread is based was a claim that "there is no audible benefit to sample rates above 48k - and, in fact, there is often a drawback".
This was stated as a scientific fact..... and the author did specifically claim that there was no detectable difference (other than occasional_ LOSS_ of quality due to distortion and interactions).
(He did NOT suggest that "the difference was minimal so we'd be silly to worry about it".

I consider this thread in particular to be a discussion of that particular claim - made in the article which was linked to in the Thread Starter post.
Heck.... 20 years from now the newest audiophile craze may be to have your hearing "cyber enhanced" so you _CAN_ hear up to 10 mHz...... using the latest neural implants.
(Anyone here a fan of "Ghost in the Shell"?)

Personally, I'd at least like enough safety margin that I have no doubt of failing to get 100.0% benefit out of even the current technology.





71 dB said:


> I'm starting to feel this is pointless. Some of us think 16/44.1 is enough, others think we need more. Nobody is convincing anyone to change mind. I don't know how much more I am willing to keep on debating. Please continue if you want… …I guess 20 years from now it's about whether 24 bits is enough of if we actually need 32 bits to listen to atoms collading in our music. That's an era when us 16 bit advocates look like a bunch of cavemen learning to make fire.


----------



## Strangelove424

71 dB said:


> I'm starting to feel this is pointless. Some of us think 16/44.1 is enough, others think we need more. Nobody is convincing anyone to change mind. I don't know how much more I am willing to keep on debating. Please continue if you want… …I guess 20 years from now it's about whether 24 bits is enough of if we actually need 32 bits to listen to atoms collading in our music. That's an era when us 16 bit advocates look like a bunch of cavemen learning to make fire.



I think a 32 bit dynamic range is theoretically 1680db. Perhaps the purists will finally be content once they are able to achieve the max SPL air molecules can sustain (194db), right before the resulting compression shock wave mashes their organs up instantaneously.


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## 71 dB

Strangelove424 said:


> I think a 32 bit dynamic range is theoretically 1680db.



194.4202 dB actually.


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## Strangelove424

71 dB said:


> 194.4202 dB actually.



32bit would theoretically have exponentially more value range than 24bit. I think you are referring to max SPL that can be attained, which is a mechanical limitation of air. Still yet, all of this is artificially limited by the SNR of integrated circuits. 

It's all pie in the sky.


----------



## castleofargh

take bits, multiply by 6 something. boom dB. lazy science power!


----------



## Strangelove424

I must have bad math for 32 bit. I do know that the compression limit for air is about 194db though. That is easily equivalent to a nuclear shockwave, and enough to kill you. I'm gonna go out on a ledge here, and say that somewhere south of that is ideal.


----------



## 71 dB

Strangelove424 said:


> 32bit would theoretically have exponentially more value range than 24bit. I think you are referring to max SPL that can be attained, which is a mechanical limitation of air. Still yet, all of this is artificially limited by the SNR of integrated circuits.
> 
> It's all pie in the sky.



Air pressure corresponds 194 dB, but of course you can have louder sounds than that. 100 bar explosion equals 234 dB peak SPL level for example.

It is a coincidence, that the theoretical dynamic range of 32 bit is ~194 dB too. The formula for dynamic range DR of n bit with normal quantization noise is:

*DR = 20*log10 (2^n * 1.5^0.5) dB = 6.0206 * n + 1.761 dB*​
Yes, the noise of electronic parts and circuits of course render 32 bit audio ridiculous, even 24 bit audio is for the most part.


----------



## RRod (Nov 17, 2017)

Strangelove424 said:


> I must have bad math for 32 bit. I do know that the compression limit for air is about 194db though. That is easily equivalent to a nuclear shockwave, and enough to kill you. I'm gonna go out on a ledge here, and say that somewhere south of that is ideal.



Your number is more around the dynamic range of 32-bit floating point.


----------



## Strangelove424

Ah, that would explain it. The number was in context to a DAW environment.


----------



## pinnahertz

KeithEmo said:


> I clipped the quote because it was getting terribly long......
> 
> 2)
> I can't speak for other people's speakers or their capabilities.
> ...


Your binary logic is noted.  It means nothing in terms of getting 20+kHz to the ears of the owners of hi-res audio recordings.


KeithEmo said:


> 3)
> Offhand I have no idea who is filtering what where... nor what the response limitations are on the specific microphones they may have used (or their synthesizer's frequency response if it's electronic).
> I simply prefer for my equipment to be able to reproduce the entire musical spectrum if it's there in the recording.
> (I consider that to be the definition of "accurate reproduction".)


That's your definition of accurate reproduction?  Ignoring everything else?  You've ignored most of the important factors, but locked onto possibly the least important factor in accurate reproduction, and with a death grip.  All my arguments here would reflect statistics and practical application, and that can't work with you, so I guess it ends here.  But I can't understand how a speaker or equipment manufacturer would focus on that and ignore every other factor.  Or, even weight ultrasonic response as a significant factor at all.


KeithEmo said:


> 4a) - 5a)
> I agree that most of us don't actually need to accurately reproduce CD-4 content....
> Likewise, I agree that most of probably will never use the erase carrier tone leakage (and, yes, special tape heads and electronics were used)....
> My point, which I stick by, is that we don't always know our future needs in advance….


OK... so you go capture all your tapes with the Plangent system (and don't use anything else!) and you can capture all your CD4 records using a CD4 stylus and 320kHz 32 bit sampling.  And you'll be The One Guy who does that.  I have no idea what you think you're capturing to preserve.


KeithEmo said:


> (And, if someone says "if you turn the volume up at 14:27 you can hear the musician cough" or "a bat got into the studio and you can see his squeak on a spectrum display at 34:10" I would prefer to be able to try it.)


You could capture the cough perfectly at 16/44, and unless the bat were included in the score, every musician involved would want, ask, or demand you filter it out.


KeithEmo said:


> And, yes, I would prefer a full-spectrum copy of the tape rather than one that has been limited to "what I need" - based on someone else's ideas of what that comprises.


A "full spectrum" copy of any analog tape could be accomplished without compromise at 16/48.  If you want to use the Plangent system, you'll need their entire system including custom hardware, but the purpose is flutter reduction, so if the masters are already low flutter, you wouldn't do that for economic reasons.  And, BTW, the bias signal is removed in the final result.  Be aware that there is actually a difference between signal and noise.


KeithEmo said:


> Again, I simply see the goal as an absolutely perfect and complete rendition of the original.
> And I see _ANY AND ALL LIMITATIONS_ as a compromise.
> And I prefer to avoid compromises unless absolutely necessary (or unless I agree to them).


Ok, but you don't seem to realize that even the highest possible capture introduces compromise that must be dealt with down the line.  Capturing ultrasonic noise or distortion products not only serves no useful purpose, but actually can cause problems later.


KeithEmo said:


> Someone asked, jokingly, whether videophiles would prefer it if their TV could reproduce gamma ray frequencies so accurately they could be burned by the video of a nuclear explosion.
> While the example is absurd, I would say that the answer is technically yes.
> We would be better off if our video display could reproduce frequencies from DC to gamma rays - and then we or the producer of the video could _DECIDE_ which ones to limit or omit.
> (I'm sure the government would cheerfully add some sort of safety standard to cover that aspect of things.)
> ...


...and another bad example!  4KHDR still falls far short of the full color gamut...like not even close!  Yet we can capture, fully and completely, the full audible spectrum...and have done so for many decades.


KeithEmo said:


> 6)
> I do have an easy question for you......
> If "quality assurance is admirable" (your words), then why are you arguing against it?


The binary thinker strikes again!  Not even going to try  to explain "gray" here.


KeithEmo said:


> 7a)
> I'm not concerned with fakes...


..but the market is, therefor the price you pay is impacted...so you actually are concerned even if you don't realize it.


KeithEmo said:


> or of how to distinguish them from legitimate copies.


That's where experience pays off.


KeithEmo said:


> In either case, a properly provenanced legitimate original that is undamaged will virtually always be worth more than one that is damaged.


But you'll never have both to compare!  Welcome to Antiques!  The example is just naive.


KeithEmo said:


> Likewise, even an undamaged copy will be worth more than a damaged copy.
> Notice that you said "VISIBLY" repaired..... while I didn't include it......
> While visible repairs are surely worse than invisible ones.... damage is still damage.
> If someone buys your expensive vase, and finds a repaired crack when they x-ray it, they probably _WILL_ sue you if you claimed it hadn't been repaired.
> "Visible" and "nonexistent" are not the same thing at all......    (otherwise a perfect forgery of a Rembrandt would really be just as good as the original).


None of this matters because you're talking about damage vs no damage of a physical object and trying...desperately...to link it somehow to audio. The comparison doesn't work.


KeithEmo said:


> Yes, if you can find a lossy compression method whose output will be _INDISTINGUISHABLE_ from the original, using our ears, or any test we can device, then it will be perfect.
> (Except, of course, by definition, it _WON'T_ be lossy at that point.)


This is a rather naive and inexperienced view.  It can be done, and has been for some time.  And yet it's still lossy.  Again..may I introduce you to the full spectrum of grays?



KeithEmo said:


> The fact that some people find it easier to identify imperfect originals than perfect copies is irrelevant....


No, it's part of the total picture of reality.

You say this:


KeithEmo said:


> I am _WELL_ aware of how lossy compression works.......


And yet you say this:


KeithEmo said:


> And it _ALL_ amounts to "discarding content that someone else has decided I won't notice is missing".


It's interesting what you choose to ignore.  Like in this case, ignoring massive psychoacoustic research and testing, yet taking lossy compression down to a personal level.


KeithEmo said:


> (And I have very little faith in the choices made by other people - especially when those choices are often made based on "what 95% of people won't notice" rather than specifically on "what ***I*** won't notice".)


The trap you find yourself stuck in here is, "I don't trust anyone to make decisions about what I don't understand", and therefore "I'll make decisions about what I don't understand myself" because "others decisions will be wrong but affect me."  I gotta say...I'll pass discussing this further.


KeithEmo said:


> 8)
> You're entirely incorrect in one regard......
> While the compression level used in JPG is variable - _THERE IS NO SETTING IN THE JPG STANDARD THAT EXACTLY REPRODUCES THE ORIGINAL_.
> Even if you set a compression level that results in a file that is larger than the RAW file it is still lossy (you cannot retrieve the original pixels exactly).
> ...


No, a RAW file contains MUCH MORE than any TIFF file.  In fact, there is no display available that can display the full content of a RAW file.  And a RAW file must be post-processed before the image in it can be used.  That post-processing, even if into a TIFF is vastly less data than included in the original.  Yet, at TIFF is uncompressed!


KeithEmo said:


> A JPG file discards a significant amount of information.... because a lot of information is approximated or discarded outright.


Again, an entirely binary view.  jpg compression can be varied in several ways by someone encoding the image.  You talk about it as if it's an on/off process, it's nothing of the kind.  As to reproducing the original exactly, you ignore (as is your way) the concept of "perceptual", and cling to the absolute binary definition.  I can't imaging living happily in your world.


KeithEmo said:


> I also suspect you haven't edited many image files.... especially JPGs.


Yeah, I'm probably a bit light on that compared with some others.  My experience with Photoshop is limited to version 2.0 (1991) through CS6, and my photo library is a paltry 100K+ images, so yeah, I'm just an inexperienced babe in those woods.


KeithEmo said:


> The method of compression used by JPG is applied to square zones .......<snip! We don't need all this here.>
> As a result, even though a given JPG may look "visually perfect" under very certain conditions, their failings tend to become unpleasantly obvious when you change the conditions.


Yes!  You define your process by the end usage.  So what?  Even an uncompressed original picture or audio recording can be shown/played in conditions that make it sound bad.  S what's the solution, keep striving for perfect capture?  You'll fail every time.  No, the key is to _understand your conditions!_


KeithEmo said:


> Of course, there is an "original loss" because, while many cameras may exceed the performance of the human eye in specific ways, none so far exceeds the human eye in _ALL_ regards simultaneously.


What IS your point? Perfection? Not achievable.  Anything else?


KeithEmo said:


> Take a picture of this posting, on a really sharp screen, and save it as RAW and as JPG.
> Blow it up so you can see the individual letters.
> I would be very surprised if you fail to see odd ghosts and echoes around the edges of the letters.
> (Because JPG was optimized to compress "pictures with continuous tones", which works well for photographs, but does relatively poor job on sharp edges and narrow lines.)


Sorry, this won't work.  The reason is simple: capture resolution is so very far in excess of any screen resolution you can't do your experiment and see what you hope to see.  You'd need a lens resolution target for that.  Yes, I did try it, but all I resolve is individual screen pixels, and that's of a 4K screen.  You've exited the real world, once again.


KeithEmo said:


> And, yes, there is a parallel in audio.....
> 
> If the audio recording was originally recorded in MP3, then, yes, the most accurate rendition of the original would be a copy of that MP3.
> And, if it was recorded at 24/96k, then that file would be the most accurate version.
> ...


Thank you for all of that, but unless you think of us all here as at least someone moronic, all you've done is state the obvious.  All true, all obvious, but not at all pertinent to this discussion.


KeithEmo said:


> At all comes back to your original assertion that "none of the stuff we're throwing away really matters"...
> I will agree that, for most people, on the equipment they'll be using, they probably won't notice the difference.....
> But, if I was hoping to use the signature of that background noise to determine the brand of recorder the original was recorded on, then you have ruined my project if you discard it.


Reaching now for the cherry at the top of the tree.


KeithEmo said:


> _YOU_ have decided that the background noise is_ NOT_ "a meaningful part of the recording" - but perhaps we don't all agree.
> And, yes, if you're the recording engineer, then, by definition, any such decision you've made is "correct".


OK, lets see now. My mic has self noise, my mic preamp has noise, and other devices contribute noise.  Is any of that part of the original acoustic event we are trying to capture? My tape recorder has a noise floor, and there's a vestige of bias too.  Is any of that part of the signal we are trying to reproduce?

Unless you actually like and prefer, and go out of your was to hear the added noise, the answer to both above must be NO.

Ok, now, if I capture all of that with a bandwidth up to 92kHz, what have I captured as compared to 23kHz?  More noise!  Yes, increasing bandwidth includes more noise.  Is there any of the original signal up there above 23kHz?  Possibly a very, very tiny bit (my analysis of 92kHz original acoustic recordings made with actual 30kHz capable mics shows it's there but more than 60dB down, typically 80 to 90dB down).  Now we look at human hearing, and we agree (don't we?) that the best ears on earth can detect the presence of strong 23kHz signals.  "Detect the presence" isn't the same as hearing, though, because it's not a sound that carries information, it's an anomaly, force fed to a test subject in unnatural means to prove a hearing limit.  And those that can hear 23kHz are very, very few very young people, likely not even able to purchase music yet.  Remember, that test signal is LOUD, not at some level 60dB below average human speech.

So, if we can't hear above 23kHz, and what's up there is not part of the original acoustic event, or so low as to be inaudible, then what are we capturing?

Bats and Bias (neither was in the original), random thermal noise (not in the original either), distortion products caused by nonlinear devices (unavoidable, but also not part of the original event).  That makes a 92kHz capture _*less perfect*_, relative to the original acoustic event, than a 48kHz capture!  *Because it includes more of what wasn't supposed to be there in the first place.*  You want perfection? Then don't capture more anomalous and spurious signals!


KeithEmo said:


> 9)
> We seem to be in perfect agreement on the final conclusion.
> Personally, I want the unlimited version, and am willing to pay a little extra for it.
> However, I do absolutely agree that many people may find no benefit to high-resolution files, and shouldn't buy them.


I'm fine with that up to the point where some manufacturer extolls the virtues of high-resolution audio without including the pit-falls.


KeithEmo said:


> Incidentally, _as for your other comment...... ALL_ audio circuitry has bandwidth_ limitations_.....


I never said it didn't.


KeithEmo said:


> They may be inherent in the circuit design or specifically imposed using an external filter - but they are necessary.
> Essentially _ALL_ circuitry produces noise... and, in general, that noise has an infinite bandwidth.
> So, even though your microphone may not pick up 30 kHz, the active devices in your microphone preamp are putting out _SOME LEVEL_ of 30 kHz noise (and some level of 10 mHz noise).
> If you're planning to digitize that output, you must filter out any significant noise above the Nyquist frequency to prevent audible aliasing and other errors.
> ...


Captain Obvious strikes again!  I have no idea why you felt the need to explain all of that, but I hope you feel better now.

You and I differ fundamentally on one basic area:  You feel that it is necessary to capture and distribute audio using a system that far exceeds human hearing because you think that some day in the future there might be a need that is answered by that information.  I feel that capturing signals that were not in the original is not only wasteful, but detrimental, as it doesn't represent the original as faithfully as eliminating them.  I recognize you are a perfectionist.  I admit to being a recovering perfectionist.  I've found that scaling my desire for perfection in some areas to more practical limits permits me to apply my desire for perfection in other areas that may benefit more from that attention.  I think we both desire perfection.  I get more joy out of letting go of perfection in some areas and doing the best possible at a composite of many areas.  That means some specifications won't meet your definition of perfection.

My tangent example would be high-res vs multi-channel.  Whatever effect high-res has, it's lost on most listeners for many reasons.  However, when compared to stereo, multichannel audio is recognized as different by nearly every listener regardless of the quality of his system.  There's even room for "different vs better" in that.  So which would you rather have? Two channels of high-res 24/96 or higher, or 5.1+ channels of 16/48?  My answer is 5.1+ of 16/48 or even 16/44.1 because it touches, moves and inspires me (and several friends) more than sitting alone in the sweet spot listening to stereo at any resolution.  I know, apples vs oranges, and why not have both high-res and multichannel?  Sure, but if you change one thing, and that thing is resolution, it makes very, very little difference, possibly none.  Switching 5.1 back to stereo, on the other hand, is like pulling the rug out.

And there's where I stand.  I'll capture at whatever rate makes sense (and there are commercial and financial reasons to go as high as 192kHz, just not audible ones), but the engineering half of my brain knows, without question, that capturing more of what's not supposed to be there in the first place makes no sense at all.


----------



## bigshot (Nov 18, 2017)

I think the second someone says they can't hear it at normal listening levels, only when they boost the volume to unlistenable levels, you have your answer. Some of us are talking in theory. Some of us are talking real world. That's the difference.

If the armageddon is televised I sure hope they record it in a sufficient bit rate to properly reproduce the shock wave that turns us all to slabs of meat.


----------



## pinnahertz

bigshot said:


> I think the second someone says they can't hear it at normal listening levels, only when they boost the volume to unlistenable levels, you have your answer. Some of us are talking in theory. Some of us are talking real world. That's the difference.
> 
> If the armageddon is televised I sure hope they record it in a sufficient bit rate to properly reproduce the shock wave that turns us all to slabs of meat.


Ha!  Well, the "clipping point" of air at sea level is around 194dB SPL, where the minimum pressure part of the wave is a vacuum.  Pretty sure even a tiny atomic blast does all of that and more in positive asymmetry, and last time I checked there wasn't a mic made ever that could do 194dB SPL.  At the point where the mic shakes to bits or melts into a blob I'm not sure bit rate matters too much.  And the meat, if any is left, would probably be well done.

I don't know if you heard this anecdote...and since I can't find the original reference I'll leave it a bit ambiguous...a certain well known film sound guy attempted to record a Space Shuttle launch by placing mics and a Nagra much closer than it was safe for a human to be, and letting it run then retreating to safety.  After the launch when he went back to retrieve the gear, the Nagra lay in pieces, shaken apart by raw SPL.  Now, I may have some of that wrong, and my reference isn't within reach right now, but that's about how I remember it.


----------



## bigshot

There is a great youtube video where a guy drops a GoPro camera into the vent hole for a volcanic flow.


----------



## Arpiben (Nov 18, 2017)

RRod said:


> Your number is more around the dynamic range of 32-bit floating point.



When dealing with *floating points* we have:
For n bit floating-point numbers, with n-m bits in the mantissa and m bits in the exponent:






n=64 bit floating points, m=11 n-m=53
n=32 bit floating points, m=8   n-m=24

Therefore an IEEE 754 half precision 16bits floating (->5bits = exponent) has an equivalent DR around 192 dB
32 bits floating -> DR around 1540 dB

Edited: reformulated


----------



## gregorio

Arpiben said:


> 32 bits floating -> DR around 1540 dB



Yep, that's the figure I've seen quoted. I bet it's not enough for some audiophiles though! Personally, I don't want to hear Clair De Lune at the same Sound Pressure Level as a supernova, but that's just me.

G


----------



## sonitus mirus

gregorio said:


> Yep, that's the figure I've seen quoted. I bet it's not enough for some audiophiles though! Personally, I don't want to hear Clair De Lune at the same Sound Pressure Level as a *supernova*, but that's just me.
> 
> G


 http://www.youredm.com/2015/10/13/a...create-a-black-hole-larger-than-the-universe/


----------



## Arpiben

Due to the very low density of interstellar gas molecules in space we may assume that sound waves don't propagate.
I am afraid I won't be able to listen to Au Clair de Lune whatever DR it may have if played from space or moon.
At least no more resolution talks over there (smile).


----------



## gregorio

sonitus mirus said:


> http://www.youredm.com/2015/10/13/a...create-a-black-hole-larger-than-the-universe/



Looks like I under-estimated. I'm presuming 1520dB would therefore be somewhere about the level of two universes starting simultaneously with big bangs? One thing's for sure, KeithEmo wouldn't be happy with the digital reproduction of that transient!

G


----------



## old tech

Arpiben said:


> Due to the very low density of interstellar gas molecules in space we may assume that sound waves don't propagate.
> I am afraid I won't be able to listen to Au Clair de Lune whatever DR it may have if played from space or moon.
> At least no more resolution talks over there (smile).


Yes but if we are talking about insane SPLs, it may generate and propagate through gravity waves...


----------



## Arpiben

old tech said:


> Yes but if we are talking about insane SPLs, it may generate and propagate through gravity waves...



Agreed in terms of insane pressure levels or power levels. Electromagnetic waves do propagate in space: light, radio waves, gravity waves etc....
But Sound or Sound Pressure barely exists.You may be wherever you want in space you will never listen to a supernova or black hole. 
Anyhow I took advantage of Debussy's "Clair De Lune" ( By the Light of the Moon) quoted by @gregorio. It is also a popular folk song kids learn in kindergarden where Pierrot is often represented seated in Moon....


----------



## old tech (Nov 19, 2017)

.


----------



## KeithEmo

I think you've actually summed up our differences in thinking very well......

I do my best to think and act in "virtual binary".......
By that I mean that, even though practical limitations may prevent me from achieving it, my_ GOAL_ is always perfection.
I often choose to avoid buying "the best" because I can't afford it, or even because it has real significant drawbacks, but never because "I just don't need it".....
When I buy a wrist watch, I don't carefully consider exactly how much accuracy I_ NEED_; instead, I'm quite pleased that I can buy a $20 watch that's a thousand times more accurate than I need,  so "I don't have to worry about it".

I absolutely agree that most of the arguments about why we _need_ high-resolution audio are ridiculous.
However, I find most of the arguments against it to be equally specious.

I figured it out...... 
At current prices, a typical song, stored at 24/192k, costs me between two and three _CENTS_ of disc space (a 6 tB USB hard drive costs $150)...
And that 6 tB doesn't put a scratch in my download allocation for a month...
And I didn't pay _EXTRA_ for a DAC that supports 192k (almost all of this year's models support 192k, and many of next year's models support 384k).
Therefore, the savings I would realize by _avoiding_ high-res files are minimal at best.
(I would save the $5 extra they charge me to buy it - but, of course, they could simply stop charging extra.)

Likewise, I'm sorry, but the claims of higher server costs are specious as well.
The majority of the costs incurred by streaming services have to do with licensing the music and operating their server equipment (mostly the licensing).
The extra cost of storing larger files, and the extra cost of the bandwidth people would use to download them, is really negligible (by percentage).

(There is a marginal cost of putting a bigger hard drive in a certain server.... but, when compared to the revenue that server generates, it really is negligible.
iTunes may save $20 million a year by using AAC instead of FLAC - which may be important to them - but the 0.1 cents a song that works out to really is negligible.
I'm reminded of a story about a guy who saved a certain food company $20k a year by deciding to put one less olive in each jar... and I wonder how many sales they lost to less satisfied customers.)

Likewise, I find the arguments against mastering at higher sample rates to also be rather specious.
Sure, if your studio has a great recorder that records at 44k, then they're not going to want to throw it away and replace it with one that records at 24/192k.
_HOWEVER_, if their equipment supports both, the cost of larger memory sticks, and the extra few seconds it takes to transfer a 24/192k file isn't significant. 
(But, yes, I agree that it is silly to re-master the same tape at 24/192k if the original didn't have a response extending past 20 kHz anyway.)

I disagree entirely about your claim vis-a-vis antiques.
If I start out with a 24/96k master recording and convert it to a 320k MP3 file, I have damaged it (I have thrown away part of what was there).
Feel free to argue that the damage is "invisible" - but it is still there.
(And, if you convert it back to a WAV file, any checksum utility will cheerfully tell you that you have FAILED to recover the original file without errors.)

I also disagree with your claim about image compression.....
The camera itself introduces one level of limitation - which is arguably unavoidable (there is no perfect camera).
After that, you either retain all of the information recorded by the camera or not...... it is a binary question.
The RAW file is "full quality" - because it contains the absolute most information available - and both the TIF and the JPG are "reduced quality".
So, given the choice, which I _AM_ given, I prefer to retain the full quality... 
The TIF is handy because everyone can read it - but I'm sure going to keep the RAW file as my archive copy.
And I know plenty of people who thought that JPG was "good enough" - until one day they needed to edit an important picture - and only had it in JPG...........
(The first time that happened to me was the day I switched my camera to "RAW + JPG" - and the switch hasn't moved since.)

I'm not personally a fan of multi-channel.... at least for music.

The real place where I find myself unable to agree with you is your _CERTAINTY_ that nothing above 20 kHz will_ EVER_ be useful.
(Or, for that matter, that the time smear introduced by a given ADC will be "absolutely inaudible".)
At one level - I simply don't agree with you.
And, at another level - I find myself very uncomfortable allowing someone else to make that judgment _FOR ME_.
Over the years I've had too many people telling me that "cassettes were good enough" and "open reel tape was good enough" and "JPG is good enough".
I've also had several of what I would have considered (at the time) to be "really good DACs" - until this or that flaw became obvious by comparison to another unit that had better performance.
And, to be honest, in my experience, my track record (in terms of results) with believing other people when they tell me that something is "good enough" hasn't worked out very well at all. 
I've ended up buying far too much stuff over again when "good enough" turned out to be "not really good enough". 

And, yeah, if that bat was in the studio, then _HE STAYS IN THE ORIGINAL_.
I'll be perfectly happy to remove him from the production copy if the customer requests it....
And I may even decide on my own that his presence is inappropriate....
But I sure don't want my equipment making that choice for me if I can avoid it.

And, if there's noise on that master tape, then it is part of the master....
I may choose to remove it, which I can do any number of ways, but I don't want my equipment making that decision for me either.

Obviously we disagree on this very basic point of philosophy...... which is fine (we are both entitled to our _OPINIONS_.)



pinnahertz said:


> Your binary logic is noted.  It means nothing in terms of getting 20+kHz to the ears of the owners of hi-res audio recordings.
> That's your definition of accurate reproduction?  Ignoring everything else?  You've ignored most of the important factors, but locked onto possibly the least important factor in accurate reproduction, and with a death grip.  All my arguments here would reflect statistics and practical application, and that can't work with you, so I guess it ends here.  But I can't understand how a speaker or equipment manufacturer would focus on that and ignore every other factor.  Or, even weight ultrasonic response as a significant factor at all.
> OK... so you go capture all your tapes with the Plangent system (and don't use anything else!) and you can capture all your CD4 records using a CD4 stylus and 320kHz 32 bit sampling.  And you'll be The One Guy who does that.  I have no idea what you think you're capturing to preserve.
> You could capture the cough perfectly at 16/44, and unless the bat were included in the score, every musician involved would want, ask, or demand you filter it out.
> ...


----------



## KeithEmo

And isn't he lucky that the designer didn't say something like: "there's no point in being able to record in 250 degree air because no human could survive it".



bigshot said:


> There is a great youtube video where a guy drops a GoPro camera into the vent hole for a volcanic flow.


----------



## Strangelove424

Throwing a go pro into a volcano and capturing inaudible frequencies are not equivalent to each other. An equivalent would be setting up a go pro in front of a radio antenna and expecting to_ see _the radio waves. Now _that_ would be dumb.


----------



## pinnahertz

KeithEmo said:


> I think you've actually summed up our differences in thinking very well......
> 
> I do my best to think and act in "virtual binary".......
> By that I mean that, even though practical limitations may prevent me from achieving it, my_ GOAL_ is always perfection.


The world we live in is not binary.  A binary thinker that tries to deal with that world will always be compromising his goals.  I choose to be happy with grays, because they might just be the exact grays the world has in it.


KeithEmo said:


> I disagree entirely about your claim vis-a-vis antiques.
> If I start out with a 24/96k master recording and convert it to a 320k MP3 file, I have damaged it (I have thrown away part of what was there).
> Feel free to argue that the damage is "invisible" - but it is still there.
> (And, if you convert it back to a WAV file, any checksum utility will cheerfully tell you that you have FAILED to recover the original file without errors.)


But you actually aren't looking at the real original!  That's an acoustic event.  Your "perfect" 24/96 copy isn't the original either, it's been through at least two transducers that have radically modified it, then it's been digitized and reconstructed.  You've lost far more audible information in all of that than you would with 320kbps AAC (sorry, mp3 is so yesterday).  Change a mic, change a speaker, change the room...radical and unmistakable difference.  Compare your 24/96 to the 320k AAC...not so much, in fact, none at all.  Pick your means of loss, obsess on the tiny ones if you like, the huge ones are far more in your life.


KeithEmo said:


> I also disagree with your claim about image compression.....
> The camera itself introduces one level of limitation - which is arguably unavoidable (there is no perfect camera).
> After that, you either retain all of the information recorded by the camera or not...... it is a binary question.


Not binary at all.  Camera sensors generate noise that's not part of the image.  They add something you really don't want.  There's no reason to keep what you don't want and wasn't part of the original.


KeithEmo said:


> The RAW file is "full quality" - because it contains the absolute most information available - and both the TIF and the JPG are "reduced quality".


Yes, but it's a question of degree, and end use.  Nobody that shoots film presents his negatives in a display because they are not usable that way.  They must be converted to positive images first, and that process discards information.  RAW files cannot be displayed, they must be converted first and that process discards information.  Always. It's just a question of what information you choose to discard. It's subjective. 


KeithEmo said:


> So, given the choice, which I _AM_ given, I prefer to retain the full quality...
> The TIF is handy because everyone can read it - but I'm sure going to keep the RAW file as my archive copy.


Totally agree, because I have gone back to RAW files and reprocessed them for alternate goals. No argument, but also, no parallel either.


KeithEmo said:


> And I know plenty of people who thought that JPG was "good enough" - until one day they needed to edit an important picture - and only had it in JPG...........
> (The first time that happened to me was the day I switched my camera to "RAW + JPG" - and the switch hasn't moved since.)


Oh sure, that's true.  But I don't shoot RAW+JPG.  I shoot RAW only, create the jpg if needed using my own judgement.


KeithEmo said:


> I'm not personally a fan of multi-channel.... at least for music.


You might give it a try.  Stereo is fatally flawed in many ways that mult-channel is not.  That's been known since the original Bell Labs experiments, but we got stuck with two channels for economic/practical reasons.  3 was the absolute minimum Bell arrived at. You might not like the "in the band" perspective, I usually don't.  But as far as immersing yourself in an event, give it a try.  Stereo is NOT more pure, it's loaded with compromise.


KeithEmo said:


> The real place where I find myself unable to agree with you is your _CERTAINTY_ that nothing above 20 kHz will_ EVER_ be useful.
> (Or, for that matter, that the time smear introduced by a given ADC will be "absolutely inaudible".)
> At one level - I simply don't agree with you.


Ok, that's fine.  But I said 23kHz.  and "time smear" audibility is unproven.  When it is proven, I'll be on that train.


KeithEmo said:


> And, at another level - I find myself very uncomfortable allowing someone else to make that judgment _FOR ME_.


I'm not, but if you buy music someone else has recorded, you're stuck with whatever they did, and if it's not using mics flat to over 30kHz and released unfiltered at 24/96 you won't be happy.  I differ.  I listen to their art for the enjoyment, and I'm happy with many different sample rates.


KeithEmo said:


> Over the years I've had too many people telling me that "cassettes were good enough" and "open reel tape was good enough" and "JPG is good enough".


I must take exception here.  As an audio professional for over 45 years, I NEVER considered cassettes "good enough".  They were the best most consumers could access.  I NEVER considered open reel tape "good enough", and I worked with some of the best pro reel machines ever made, and the best noise reduction systems, and the best mics, mixers...everything.  Tape was NEVER good enough, it was at one time the best we had.  JPG images when introduced were never "good enough", the too were the best digital camera images we had.  At that time 35mm film was much, much better...still not "good enough".  I saw an exhibit of Ansel Adams original prints...huge ones...and even they showed that 8x10 negatives were not quite "good enough".  So what?  I thoroughly enjoyed them, and images and recordings in all of those formats. 


KeithEmo said:


> And, yeah, if that bat was in the studio, then _HE STAYS IN THE ORIGINAL_.
> I'll be perfectly happy to remove him from the production copy if the customer requests it....
> And I may even decide on my own that his presence is inappropriate....
> But I sure don't want my equipment making that choice for me if I can avoid it.


But what if that bat is the only sound over 23kHz, the rest of that spectrum being random noise?  You still want to over-capture without understanding the original signal.


KeithEmo said:


> And, if there's noise on that master tape, then it is part of the master....
> I may choose to remove it, which I can do any number of ways, but I don't want my equipment making that decision for me either.


Part of the master, yes.  Part of the original, no.  And if it's part of the master in an area where none of the original information exists, it's a defect, a distortion.  The only way you're removing it is to limit bandwidth, there are not a "number of ways" to eliminate ultrasonic noise.


----------



## pinnahertz

Strangelove424 said:


> Throwing a go pro into a volcano and capturing inaudible frequencies are not equivalent to each other. An equivalent would be setting up a go pro in front of a radio antenna and expecting to_ see _the radio waves. Now _that_ would be dumb.


Depending on the antenna, frequency and signal strength, it might though!  I know I can capture cell phone signals with a microphone (not working just as a mic of course).


----------



## Strangelove424

I was speaking in regard to visually captured frequencies, I know that many kinds of waves can interfere with audio recording. Is it actually the mic's transducer picking up the radio waves or is it creating noise inside the circuitry? 

In regards to the visual stuff, the light sensors in some cameras can actually pick up quite a bit more than just visual light. I know of a few that are very sensitive into infrared, and with a few hacks to remove the IR filter over the sensor, and with a visible light filter over the lens (some will allow different nm lengths of light adjustable on a ring) can create beautiful photos such as this:


----------



## KeithEmo

Again, obviously we look at things differently......

For example..... yes, camera sensors, and lenses, impose their own limitations.
Therefore, the best possible version of the information I can get from my camera in digital form is that RAW file.
(I can obsess at another level how much I want to spend on my camera and lenses.)
Therefore, _MY_ only choice is to either keep_ EVERYTHING_ the camera lets me have, decide myself to discard some of it, or let someone or something else make that decision for me.
Given those choices, keeping everything seems like the safest choice.
(Deciding with absolute certainty what belongs there and what is an artifact of the camera is beyond my current capabilities.)

Likewise, if I hear hiss in a recording, I can't know _FOR SURE_ if it's hiss from the microphone preamp, a noisy steam radiator, of the air hissing in a pump-powered organ.
If it's microphone preamp hiss - then it's probably "an artifact".
But, if it's the hiss of the air in the organ pipes, then it actually belongs there - as part of the original experience.
And, if it's the radiator, then only the recording engineer knows whether he intended it to be there or not (maybe it's part of his "artistic vision").
Therefore, I'd rather keep it - at least in my master copy - than _ASSUME_ I know it is an artifact.

I tend to divide the world into "things I have control over" and "things I don't".
In the case of music, everything up until the output of the mixing console is "whatever the mixing engineer says it should be".
(So, if he didn't remove that hiss, then I guess maybe he wants it there.... or maybe not.... but I'll never know for sure.)
It's part of the _PRODUCTION_ rather than of the_ REPRODUCTION_.
But, once the music has been _PRODUCED_, then the goal is to _REPRODUCE_ it.
To me it seems obvious that part of the process should start with an exact copy of what the mixing engineer intended me to have.
I may _CHOOSE_ to discard parts of it, or alter parts of it, but I don't want equipment that limits my choices by being _UNABLE_ to reproduce all of it, or that eliminates parts of it without asking me.

I agree that stereo is a compromise....... but I guess I've just grown to those particular compromises.
(Surround sound also causes compromises of a different sort.)
We've also got the situation that the mix engineer has made that choice for me........ (either it was recorded and mixed in surround or not - so that's outside of my control).

I want to "over-capture" because it saves the most information.....
And, yes, sometimes there is a situation where the information "grows" because of our inability to retrieve it optimally.

To use the photo example......
I may take a photo at a certain resolution...
And then have that photo printed as a dot-lithograph... (let's assume we print it at 200 dpi).
And, by an unfortunate circumstance, that lithograph may be the only copy I have (someone's dog ate the negatives).
Now I'm stuck with the limitation of that lithograph.
So, what resolution do I scan it at to do the best possible job of making a "new digital negative" of my image?
Some people might suggest that 200 dpi is plenty...
But it's not.
Because the dots on my scanner won't line up perfectly with the litho dots...
In fact, the best chance I'll have for the best possible quality will be to scan it at WAY over 200 dpi...
I'll scan it at 2400 dpi..... so I can actually see the shape of the dots from the lithograph...
That way each dot can be represented by dozens of scanned pixels, and the color proportions will come out just right.

My point about tape (and even cassette) is that there were people who insisted that both were good enough.
I knew one person who had a very expensive Nakamichi cassette recorder... he swore that it could make cassettes that were "indistinguishable from the original".
My real point is that there is a long history of people claiming that something was "good enough" and being proven wrong. 
I'd rather spend a little extra time and money rather than RISK making that mistake again.
(again, within what's within my ability to control....... )

An, no, that doesn't mean that I don't enjoy a poor quality recording of a great performance....
It just means that I'd enjoy it MORE if the quality was better.
(And, yes, listening to a poor quality recording, and wondering if the other version I didn't buy sounds better, would make me unhappy.....) 



pinnahertz said:


> The world we live in is not binary.  A binary thinker that tries to deal with that world will always be compromising his goals.  I choose to be happy with grays, because they might just be the exact grays the world has in it.
> But you actually aren't looking at the real original!  That's an acoustic event.  Your "perfect" 24/96 copy isn't the original either, it's been through at least two transducers that have radically modified it, then it's been digitized and reconstructed.  You've lost far more audible information in all of that than you would with 320kbps AAC (sorry, mp3 is so yesterday).  Change a mic, change a speaker, change the room...radical and unmistakable difference.  Compare your 24/96 to the 320k AAC...not so much, in fact, none at all.  Pick your means of loss, obsess on the tiny ones if you like, the huge ones are far more in your life.
> Not binary at all.  Camera sensors generate noise that's not part of the image.  They add something you really don't want.  There's no reason to keep what you don't want and wasn't part of the original.
> Yes, but it's a question of degree, and end use.  Nobody that shoots film presents his negatives in a display because they are not usable that way.  They must be converted to positive images first, and that process discards information.  RAW files cannot be displayed, they must be converted first and that process discards information.  Always. It's just a question of what information you choose to discard. It's subjective.
> ...


----------



## KeithEmo (Nov 20, 2017)

As far as I know, while certain really obscure microphones can detect audio (vibration) up into the low RF frequencies, maybe as high as 2 mHz, they are few and far between, and absurdly expensive. What you normally encounter is that something in the preamp, usually a transistor junction, is "detecting" the radio signals (extracting the audio frequency from the carrier), the audio portion then leaks into the audio circuitry, and that's what you're hearing. If you have one of those old transistor radio crystal earphones, you can sometimes hear AM radio simply by touching the wire to a big piece of metal (try a chain link fence). The piezoelectric "crystal" element in the earpiece acts as both the detector and the speaker.

Near IR photos can be lovely.

Actually, as far as I know, all current camera sensors are fully sensitive to near-IR.
On the better cameras, there is a filter to block those frequencies because they interfere with the normal light image.
(If you add the image below to a full color version of the same image the IR information will make the trees look washed out and milky.)
Most cameras can have the filter removed - although it can be complicated - and you risk ruining an expensive camera (you need to add a piece of clear glass to avoid compromising the autofocus capabilities).
LifePixel is one company who sells commercially modified cameras - and do the modifications (I have a Nikon d40x they modified).
(they also have a big gallery of pictures on their website)

It would be really nice if the camera could actually record all the wavelengths - so you could choose the ones you want to use afterwards.
Unfortunately, cameras only "see" in three colors, so the IR is seen mostly by the red sensor.... and there's no way to separate them.
(You block the visible red with an IR filter - and so get a picture of just the IR....... but there's no way to photograph both, then later choose which one alone you want.)



Strangelove424 said:


> I was speaking in regard to visually captured frequencies, I know that many kinds of waves can interfere with audio recording. Is it actually the mic's transducer picking up the radio waves or is it creating noise inside the circuitry?
> 
> In regards to the visual stuff, the light sensors in some cameras can actually pick up quite a bit more than just visual light. I know of a few that are very sensitive into infrared, and with a few hacks to remove the IR filter over the sensor, and with a visible light filter over the lens (some will allow different nm lengths of light adjustable on a ring) can create beautiful photos such as this:


----------



## bigshot

I see no purpose to this end of the conversation.


----------



## pinnahertz (Nov 20, 2017)

KeithEmo said:


> Again, obviously we look at things differently......
> 
> For example..... yes, camera sensors, and lenses, impose their own limitations.
> Therefore, the best possible version of the information I can get from my camera in digital form is that RAW file.
> ...


Except that's not how this works.  Your RAW is not displayable at all.  Just to see it and make your judgement of what you think you want to discard you have to throw information out. 


KeithEmo said:


> Given those choices, keeping everything seems like the safest choice.


I've never once argued against keeping the RAW image.  It's what I do.


KeithEmo said:


> (Deciding with absolute certainty what belongs there and what is an artifact of the camera is beyond my current capabilities.)


But it's actually not.  You just need to learn what to look for.  What you're doing when you process a RAW file for display is making that choice, either you or your computer, but it's already being made, and quite intelligently.


KeithEmo said:


> Likewise, if I hear hiss in a recording, I can't know _FOR SURE_ if it's hiss from the microphone preamp, a noisy steam radiator, of the air hissing in a pump-powered organ.
> If it's microphone preamp hiss - then it's probably "an artifact".
> But, if it's the hiss of the air in the organ pipes, then it actually belongs there - as part of the original experience.
> And, if it's the radiator, then only the recording engineer knows whether he intended it to be there or not (maybe it's part of his "artistic vision").
> Therefore, I'd rather keep it - at least in my master copy - than _ASSUME_ I know it is an artifact.


You've once again missed the point entirely.  If the noise is from an instrument, it's probably not an artifact, it's made by the instrument.  If the noise is form electronics, it's not part of the acoustic event.  If the noise is audible, it's a flaw, a defect, but likely nothing we can do anything about.  If the noise is ultrasonic, it's not part of the original event AND something we can eliminate. 


KeithEmo said:


> I tend to divide the world into "things I have control over" and "things I don't".
> In the case of music, everything up until the output of the mixing console is "whatever the mixing engineer says it should be".
> (So, if he didn't remove that hiss, then I guess maybe he wants it there.... or maybe not.... but I'll never know for sure.)
> It's part of the _PRODUCTION_ rather than of the_ REPRODUCTION_.
> ...


That's fine.  As when I have my engineer hat on, I make those decisions all the time, and you already know what I'm going to do. 


KeithEmo said:


> I agree that stereo is a compromise....... but I guess I've just grown to those particular compromises.
> (Surround sound also causes compromises of a different sort.)
> We've also got the situation that the mix engineer has made that choice for me........ (either it was recorded and mixed in surround or not - so that's outside of my control).


You need to study this a bit more.  You're implying that an engineer's surround mix is somehow wrong, and that's incorrect.  Surround sound includes a rather well standardized speaker layout that even you can follow.  And standard calibration.  And even known play levels.  With those tools alone you can stand a far, far better chance of hearing what the engineer heard, and that is, in fact, the goal of some higher end home AV systems.  That goal does not, and cannot exist in two channel stereo.  And we're ignoring all the rather important issues of phantom imaging (or the lack of).  I'm surprised that you don't understand all of this given your connection with your employer.


KeithEmo said:


> I want to "over-capture" because it saves the most information.....
> And, yes, sometimes there is a situation where the information "grows" because of our inability to retrieve it optimally.


And I want to over capture too, to preserve as much of the information *of the original even*t as possible.  I don't want to capture more of what wasn't there in the first place.


KeithEmo said:


> To use the photo example......
> I may take a photo at a certain resolution...
> And then have that photo printed as a dot-lithograph... (let's assume we print it at 200 dpi).
> And, by an unfortunate circumstance, that lithograph may be the only copy I have (someone's dog ate the negatives).
> ...


You might consider not doing any more imaging analogies, you don't seem to understand printing or image processing.  I get what you're trying to say, but your analogy is horrendously out of touch with reality.


KeithEmo said:


> My point about tape (and even cassette) is that there were people who insisted that both were good enough.
> I knew one person who had a very expensive Nakamichi cassette recorder... he swore that it could make cassettes that were "indistinguishable from the original".
> My real point is that there is a long history of people claiming that something was "good enough" and being proven wrong.
> I'd rather spend a little extra time and money rather than RISK making that mistake again.
> (again, within what's within my ability to control....... )


Here's where we differ a lot.  You don't seem to feel comfortable with your understanding of what is and is not audible.  In the tape days it was easily provable at any point in time that the system was not perfect or even adequate for capturing the original.  Anyone making the claim above was deluded (though clearly very happy with his gear).  The medium was vastly worse than its input signal.  Today we can still easily prove what's audible and what's not.  Our digital medium is also measurable, provable, and verifiable as to its efficacy and deficiencies.   You seem to think that knowledge  doesn't exist.  So you slap more resolution on it as a solution, when it solves nothing, and creates further potential problems. 

You admit that the production process is out of your control.  So, then, why are we even arguing?  You'll take what we give you and like it...or hate it (I'm sure there's no neutral). 


KeithEmo said:


> An, no, that doesn't mean that I don't enjoy a poor quality recording of a great performance....
> It just means that I'd enjoy it MORE if the quality was better.
> (And, yes, listening to a poor quality recording, and wondering if the other version I didn't buy sounds better, would make me unhappy.....)


Pretty sure we've both bought multiple version of material hoping for something better or different.  That goes back the the tube/vinyl days, nothing new there.  One of my favorite pieces of music was recorded in the 1950s, and the original RCA pressings were only fair. I bought several pressings, none were good.  Decades later I got the CD.  Guess what? The same distortion I didn't like on vinyl was perfectly preserved in bits!  They pushed the record level into tape saturation.

Ultimately you have two choices: take it or leave it.  Hey! That's a binary decision!  You should have no trouble with it.


----------



## Strangelove424

KeithEmo said:


> Again, obviously we look at things differently......
> 
> For example..... yes, camera sensors, and lenses, impose their own limitations.
> Therefore, the best possible version of the information I can get from my camera in digital form is that RAW file.
> ...



You are conflating the demands of an acquisition or editing format to that of a delivery format. In terms of this ongoing (and often irrelevant) comparison to film, imagine going down to Best Buy, picking up a copy of your favorite film, and finding that they delivered the movie to you on a reel of undeveloped negatives. All of the data is right there! Completely intact! Not a single missed color, not a compression artifact in sight… but completely useless to an end consumer. 

The need to preserve data through generational loss (which really isn’t a problem with digital audio anyway) isn’t a consumer issue, it’s a production issue. Retaining data to make editing decisions later. Consumers don’t need to make those decisions, don’t need to second guess the colorist, don’t need to wonder if a scene should have a lower gamma level. For a delivery codec, you find out what the perceptual limits are for 99.9% of people, you find out what your storage or delivery limit is, and you hit that target. 

You are constantly trying to make your point via allegory, and those allegories are false and misleading. Like when they tell you the protective coating is an invisible bubble of protection around your vehicles paint. It's not a bubble. It's not protective. It's not even there. It's story telling.


----------



## pinnahertz

KeithEmo said:


> As far as I know, while certain really obscure microphones can detect audio (vibration) up into the low RF frequencies, maybe as high as 2 mHz, they are few and far between, and absurdly expensive. What you normally encounter is that something in the preamp, usually a transistor junction, is "detecting" the radio signals (extracting the audio frequency from the carrier), the audio portion then leaks into the audio circuitry, and that's what you're hearing. If you have one of those old transistor radio crystal earphones, you can sometimes hear AM radio simply by touching the wire to a big piece of metal (try a chain link fence). The piezoelectric "crystal" element in the earpiece acts as both the detector and the speaker.


No point here.  We have transducers that exceed human capabilities.  Have had for a very long time.  We don't use them to record music.


KeithEmo said:


> Near IR photos can be lovely.


They are artistic, using different tools.


KeithEmo said:


> Actually, as far as I know, all current camera sensors are fully sensitive to near-IR.
> On the better cameras, there is a filter to block those frequencies because they interfere with the normal light image.


No, ALL color cameras must have IR filters. 


KeithEmo said:


> (If you add the image below to a full color version of the same image the IR information will make the trees look washed out and milky.)
> Most cameras can have the filter removed - although it can be complicated - and you risk ruining an expensive camera (you need to add a piece of clear glass to avoid compromising the autofocus capabilities).
> LifePixel is one company who sells commercially modified cameras - and do the modifications (I have a Nikon d40x they modified).
> (they also have a big gallery of pictures on their website)


That's art, not accurate reproduction.  You certainly must know the difference.


KeithEmo said:


> It would be really nice if the camera could actually record all the wavelengths - so you could choose the ones you want to use afterwards.


A camera that records all the wavelengths would't reproduce a visible picture.  What you actually want is a camera sensor that behaves like a perfect retina.  Any more is just art.


KeithEmo said:


> Unfortunately, cameras only "see" in three colors, so the IR is seen mostly by the red sensor.... and there's no way to separate them.
> (You block the visible red with an IR filter - and so get a picture of just the IR....... but there's no way to photograph both, then later choose which one alone you want.)


A camera with it's IR filter removed and no visible block filter photographs both visible and IR.  

 Each RGB color channel can be easily separated in software.


----------



## KeithEmo

1) 
Actually, no, in the past, many cameras omitted the filters.
Up around when commercial hand-held cameras passed the 1 mP point most of them started including the filter.
Up until last year, the cameras in many phones omitted them (that's why you can test your IR remote control by pointing it at the camera on your phone).
In the last two or three years most of the better phones include the filter.
(Many security cameras specifically are still intended to work in B&W with an IR light source at night.)

2)
IR images are art... obviously.

3)
Your retina is able to "see" all visible frequencies by being sensitive to three specific bands.
It does not detect three specific frequencies, but detects three rather wide bands of frequencies, each CENTERED around red, green, and blue respectively.... and your brain applies a sort of image analysis to figure out what's going on.
With cameras, it was determined that, by using sensors that detect those same three frequency ranges, and phosphors that emit them, a facsimile could be reproduced that would fool our eyes.
However, the image itself is far from accurate.

If we had both a camera that stored all visible frequencies accurately, and a monitor that reproduced them, then it would work perfectly - and would look exactly like the original.
You would also get the same results as from the original if you used IR or UV filters.
And, since the image would contain all frequencies there in the original:
a) it would look exactly like the original when viewed by a human retina (or any other sensor)
b) unlike current photographs, you would also be able to decide whether to view red, green, blue, infrared, or any combination of light colors in the resulting image
c) you would also get a proper rainbow when you passed white from it through a prism (when you pass white light from a current image through a prism you get three stripes - one in each primary color - which is INCORRECT)

The current tri-stimulus system is a very inaccurate compromise - but it was designed to work very well when used ONLY with the human retina as a sensor.
(Arguably it is a very good example of perceptual encoding.)

You can separate the R, G, and B ......   but you cannot separate the near IR - because it's using the red sensor.
(In a "full spectrum system" you would be able to filter any color or range of colors you wanted to.)



pinnahertz said:


> No point here.  We have transducers that exceed human capabilities.  Have had for a very long time.  We don't use them to record music.
> They are artistic, using different tools.
> No, ALL color cameras must have IR filters.
> That's art, not accurate reproduction.  You certainly must know the difference.
> ...


----------



## pinnahertz

KeithEmo said:


> 1)
> Actually, no, in the past, many cameras omitted the filters.
> Up around when commercial hand-held cameras passed the 1 mP point most of them started including the filter.
> Up until last year, the cameras in many phones omitted them (that's why you can test your IR remote control by pointing it at the camera on your phone).
> ...


I am, and I suspect just about everyone else is, now lost on what point you are trying to make.  None of the above has anything to do with the topic, _"Why 24 bit audio and anything over 48k is not only worthless, but bad for music."_

I've attempted to clarify the principles of engineering, science and application. You have been, and and keep on posting inapplicable analogy after inapplicable analogy, with wild concepts of pseudoscientific theory.   

I must honestly thank you for the entirely new (and rather opposite of my former) view I now have of the company you represent, Emotiva.  I formerly thought of it as a company grounded in practical applications here on Earth.  I admit to being disappointed, but oh well, live and learn to recognize futility.  I believe I've done both.


----------



## bigshot

I really can't believe how irrelevant this thread has gotten. What do cameras and gamma rays and all this stuff have to do with 24 bit audio being overkill? I'm getting to the point where there is nothing in here that is worth reading any more. As for the subject title, there's a link in my sig file that covers that completely.


----------



## KeithEmo

I'm not sure exactly what you mean. I have several image editing programs that will happily display the RAW images from all of my cameras.... as will most of my image viewers.
And, while my Nikons have a pretty good dynamic range, a good modern TV can match it (DSLRs don't tend to have an excessively wide dynamic range - although some of the new video cameras are much better).
But, yes, I always convert them to something else "for distribution".... because a lot of displays won't.... and, having the part of "the recording engineer", it is my decision to make.
(And there are also a few situations where no current camera can handle the dynamic range - then we use HDR.) 

And, again, we're back to my original point....
How do you _KNOW_ which noise was made by the instrument and which was an artifact?
As a recording engineer, _MAKING_ music, you can easily decide if you _LIKE_ a particular sound or not.
However, as someone arranging to _REPRODUCE_ it, I (or the customer) should not.

I have one recording that, in one part, has an odd little noise that sounds very much like a midrange with a jangled voice coil.
However, it plays the same on multiple speakers, so it's really part of the recording.
In fact, when you turn it up, and listen carefully enough, you can tell it's something vibrating on the drum kit.
I'm not sure why the recording engineer left it there - but I'm glad that it didn't mysteriously disappear when it passed through my equipment - because then my reproduction of that recording would be incorrect.
I may _DECIDE _to make a copy with that noise edited out - but I want to make that decision consciously if at all.

I never implied that any given surround sound mix is wrong.
In fact, by definition, _WHATEVER_ the engineer did is "right" (which doesn't mean that I have to like it).
However, strictly speaking, other than possibly binaural, both stereo and surround sound fall short of being an accurate reproduction of the original.
The original was a bunch of sounds, produced by a variety of different instruments, each of which interacts differently with the room.
Neither a stereo pair of speakers nor a set of surround sound speakers can reproduce the complex radiation and reflection patterns accurately.
Therefore, both are a compromise, where we do our best to do what we can accurately, correct for or null out the obvious discrepancies, and hope for the best.
But, no, I have never heard a system, at any price, or with any level of detailed setup, where I could honestly say:
"If I dragged the conductor of that performance into my listening room he would not be able to tell whether this was a recording or a live performance."
I would consider both the stereo and surround sound mixes to be "artistic renditions of the original" - so the best we can do is to reproduce the mixing engineer's intent there.

We seem to be agreeing that no recording is perfect...
Therefore, it would seem logically obvious that it must follow that "there's room for improvement"...
In fact, I suspect that I more or less agree with your priorities (except that I prefer stereo to surround).
However, I'm also not prepared to declare that any area is "so perfect there is no possible room for improvement".

I think our biggest point of disagreement may be in your "faith" about modern equipment.
I've owned a lot of DACs..... many of them sound very similar... and many sound very obviously different.
In some cases, the differences in sound can be clearly traced to specific obvious differences in specifications, but in others it seems less so.
For example, _MOST_ of the units I've owned that used the Sabre DAC chip have had a distinctive sound....
(Since the company who designed the chip originally claimed that they'd chosen their filter characteristics based on "what people liked in focus groups rather than what was the most numerically accurate", I see no surprise there.)
My problem is that, of the two dozen or so DACs I've owned, about half of them had a distinctive sound signature of one sort or another.
And, while I think it might be interesting to do the research to figure out where those differences come from, I'm too lazy to do so.
However, from my experience, it would be untrue to say that "all modern DACs sound the same" or even "_MOST_ modern DACs sound the same".
I'm much more comfortable saying that "some modern DACs sound the same, and many sound quite similar, but you shouldn't ASSUME that a given one does without finding out for yourself".
(Sadly, I'm forced to suggest that people find out for themself, even though we all know people have all sorts of biases and odd notions, because reviewers seem even more prone to odd opinions, and the commonly available specs don't seem to tell us quite enough - at least not yet.) 



pinnahertz said:


> Except that's not how this works.  Your RAW is not displayable at all.  Just to see it and make your judgement of what you think you want to discard you have to throw information out.
> I've never once argued against keeping the RAW image.  It's what I do.
> But it's actually not.  You just need to learn what to look for.  What you're doing when you process a RAW file for display is making that choice, either you or your computer, but it's already being made, and quite intelligently.
> You've once again missed the point entirely.  If the noise is from an instrument, it's probably not an artifact, it's made by the instrument.  If the noise is form electronics, it's not part of the acoustic event.  If the noise is audible, it's a flaw, a defect, but likely nothing we can do anything about.  If the noise is ultrasonic, it's not part of the original event AND something we can eliminate.
> ...


----------



## reginalb

KeithEmo said:


> I'm not sure exactly what you mean. I have several image editing programs that will happily display the RAW images from all of my cameras.... as will most of my image viewers.
> And, while my Nikons have a pretty good dynamic range, a good modern TV can match it (DSLRs don't tend to have an excessively wide dynamic range - although some of the new video cameras are much better).
> But, yes, I always convert them to something else "for distribution".... because a lot of displays won't.... and, having the part of "the recording engineer", it is my decision to make.
> (And there are also a few situations where no current camera can handle the dynamic range - then we use HDR.)



What he is trying to explain is that there is a background process that is converting your RAWs for display. Those image editing programs are displaying a conversion of the image that they've generated when you imported them. At no point can you see a RAW file, because all monitors on cameras, image editors, and even the drivers you can get to just view them in Windows, are performing a conversion and showing you a limited amount of the data contained in the RAW image. 

With regard to Nikon, what are you even talking about? No, a TV can not match a D850, D750, D810, D500, D7500, D7200, or even a D3300 level of dynamic range. Frankly, you wouldn't want them to. The contrast level would be too low, the color grading and contrast decisions that go in to image and video processing make the video a lot more appealing to actually look at. 

In regards to video cameras, the Red Helium 8k has 15.2 stops of dynamic range at base ISO, which does best any current Nikon (by less than half a stop of dynamic range at base ISO - 14.8 for the D850), but that is basically the king of dynamic range. The Red Epic ties the D850, does no better. I don't think anything from any company has better DR than the Helium 8k - though there are other things that make some cameras perhaps more desirable to some, the Helium is a monster for pure image quality. I mean, once you get all the necessary stuff to use it, you're north of $100,000, but you gotta pay to play!

But eyes and ears aren't comparable, and analogies between them always have a tendency to confuse the issue, not clarify it, when it comes to hifi.


----------



## bigshot (Nov 21, 2017)

Digital audio is a lot like ventriloquism...

Big long irrelevant spiel about whether lighter composition dummies are better than heavier ones carved of wood "is it the same for headphone construction?", throwing your voice and perceived directionality of sound, moving your lips as an analogy for excursion of transducers, the philosophy of illusion and how it relates to subjective perception of sound and the placebo effect, audibility of artifacts and substituted consonants like B, P and V and F, speed and pacing of patter and voice switching and how it compares to micro timing in sampling rates, open and closed mouths are they like open and closed headphones? etc.

Someone replies explaining some misconception about ventriloquism without any reference to audio.

More irrelevent analogies and sidetracks into arcane ventriloquilia.

More explaining

Rinse and repeat 5X.

See! I can do it too!



reginalb said:


> But eyes and ears aren't comparable



They are both cute!


----------



## KeithEmo

I'm sorry, but my point there was simply to point out that your assertion that "all cameras have filters" was in fact simply incorrect.

However, since you bring it up, I will take this opportunity to clarify Emotiva's "company line" on this subject.
(I should also mention that, since we really don't have a company line on the subject, what I've been posting here are mainly my opinions on the subject.)

We here at Emotiva don't produce or sell music... we sell hardware that reproduces music.
Therefore, we're just as happy if you buy the latest high-res remasters, or keep playing your old CDs, or subscribe to Tidal instead.
And we design the hardware we sell based on _BOTH_ sound engineering principles_ AND_ the needs and desires of our customers.
Therefore, since many of our customers want to play high-res files and downloads, we design our equipment to do so...... and do it very well.
We're not going to try very hard to convince our customers that they will or will not hear a difference; whatever files they choose to listen to will sound good on our DACs.
You may even get a different opinion depending on who you talk to..... but we all agree that our goal is to make _WHATEVER_ music you decide to play sound as good as it possibly can.
And you will find plenty of specs to back up the claim that the performance of our hardware is very good.

Individually, we have a wide variety of tastes and preferences, and that extends to music formats.
Some of us find high-res files to sound better often enough that we buy them; others are satisfied to stream music using Spotify; and yet others still like physical CDs; (a few even still like vinyl).
(So we're not really here to _convince_ anyone either way.)

However, I do want to stress one point....... which seems to get lost in this discussion.
Our customers are _NOT_ "paying extra for something they don't need".
You may pay extra when you purchase a high-res download.... but you're _NOT_ paying extra to own a DAC that can play it.
All of the current high-quality DAC chips support 24/192k; and that includes the ones we use in our current products (and now most of the latest chips support 384k).
However, you're not paying extra for that high-res capability... it's simply "standard" on this year's good DAC chips.
(There are some really low-end DAC chips that don't support over 48k, but most of them are undesirable for other, and far more compelling, reasons.... and the cost difference is on the order of $5.)



pinnahertz said:


> I am, and I suspect just about everyone else is, now lost on what point you are trying to make.  None of the above has anything to do with the topic, _"Why 24 bit audio and anything over 48k is not only worthless, but bad for music."_
> 
> I've attempted to clarify the principles of engineering, science and application. You have been, and and keep on posting inapplicable analogy after inapplicable analogy, with wild concepts of pseudoscientific theory.
> 
> I must honestly thank you for the entirely new (and rather opposite of my former) view I now have of the company you represent, Emotiva.  I formerly thought of it as a company grounded in practical applications here on Earth.  I admit to being disappointed, but oh well, live and learn to recognize futility.  I believe I've done both.


----------



## castleofargh

reginalb said:


> What he is trying to explain is that there is a background process that is converting your RAWs for display. Those image editing programs are displaying a conversion of the image that they've generated when you imported them. At no point can you see a RAW file, because all monitors on cameras, image editors, and even the drivers you can get to just view them in Windows, are performing a conversion and showing you a limited amount of the data contained in the RAW image.
> 
> With regard to Nikon, what are you even talking about? No, a TV can not match a D850, D750, D810, D500, D7500, D7200, or even a D3300 level of dynamic range. Frankly, you wouldn't want them to. The contrast level would be too low, the color grading and contrast decisions that go in to image and video processing make the video a lot more appealing to actually look at.
> 
> ...


that's also my understanding. cameras have come challenging the eye's dynamic range ability for a few years(also with a linearity films couldn't reach), while TVs are still struggling to have a dark black and a white that doesn't bleed over 10pixels. 
also true about RAW, that it cannot be displayed without entering a color profile beforehand. usually what we see is what the camera would pick for a JPG version of it(or whatever setting we have in our app). there is a good deal more data available for manipulation and that's the greatness of RAW images, but the display is what it is.


----------



## bigshot

KeithEmo said:


> We here at Emotiva don't produce or sell music... we sell hardware that reproduces music.
> Therefore....



Oh the commercial just came on. I'm going to take a bathroom break. Let me know when it's over.


----------



## KeithEmo

Well, obviously, since my brain can't read digital file formats, there's always going to be a conversion involved.
My point was simply that, regardless of what goes on inside the camera, the RAW file is the most accurate version of the information which the camera makes available _TO ME_.
And, yes, most of the software I use is able to deal with that directly, so I have no compelling need to sacrifice using the best version of that data available because of limitations on my part.
If you put the camera in the position of the mixing engineer, then the RAW file is analogous to the master file that the engineer saves from his (or her) console.
(I see that as the dividing line between "production" and "reproduction".)

That helium 8k sounds awfully nice (but I don't really do video).

My point, in the context of this discussion..... which seems to have gotten lost in the smoke.....
Was simply that I prefer to get the most complete and accurate information available - and then make my own decisions.
In the case of an image, I want the RAW file (which is as close as I can get to "the data coming off the sensor").
And, in the case of audio, it would be the highest resolution version or the original content available (excluding up-sampled versions).
In both cases, I prefer _NOT_ to trust someone or something to decide _FOR ME_ what information I need and what information I can do without.

Honestly, people seem to get far too excited about the whole "high-res question".
From a _PURELY_ practical aspect, it doesn't matter why they're being sold at all.... all that matters is whether a given file sounds better or not. 
Of course the vendors want to sell more downloads; and, of course, we prefer to only buy them if they're better; it's Economics 101 (and there's nothing sinister about it). 
Most of my current DACs support 24/192k.
But I didn't agonize over the decision.
Its just that sometimes I get 24/192k files, and It's easier to be able to play them without having to convert them.... and I don't have to wonder if something was lost in the translation.
(For the same reason, if I were to get a RAW file of an important picture, I would view it in a RAW viewer..... rather than convert it to a TIF, then wonder if something important had been omitted.)

My personal "offensiveness scale" seems to be the exact opposite of many others here.....
I hear many people who seem to be downright _outraged_ that the music industry might be trying to convince them to buy something that has no practical value (high-res files).
Personally I was outraged when Apple decided to only offer lossy compressed files in their iTunes store.... because I find it offensive that they've taken it on themselves to decide_ FOR ME_ that I don't need the lossless copy I asked for.
(Please be clear that the point is not whether I can hear a difference or not, but simply that they've taken it on themselves to deprive me of the choice, which seems to me as if they're trying to force me to agree with their opinion on the subject.)



reginalb said:


> What he is trying to explain is that there is a background process that is converting your RAWs for display. Those image editing programs are displaying a conversion of the image that they've generated when you imported them. At no point can you see a RAW file, because all monitors on cameras, image editors, and even the drivers you can get to just view them in Windows, are performing a conversion and showing you a limited amount of the data contained in the RAW image.
> 
> With regard to Nikon, what are you even talking about? No, a TV can not match a D850, D750, D810, D500, D7500, D7200, or even a D3300 level of dynamic range. Frankly, you wouldn't want them to. The contrast level would be too low, the color grading and contrast decisions that go in to image and video processing make the video a lot more appealing to actually look at.
> 
> ...


----------



## reginalb (Nov 21, 2017)

KeithEmo said:


> Well, obviously, since my brain can't read digital file formats, there's always going to be a conversion involved.
> My point was simply that, regardless of what goes on inside the camera, the RAW file is the most accurate version of the information which the camera makes available _TO ME_.
> And, yes, most of the software I use is able to deal with that directly, so I have no compelling need to sacrifice using the best version of that data available because of limitations on my part.
> If you put the camera in the position of the mixing engineer, then the RAW file is analogous to the master file that the engineer saves from his (or her) console.
> ...



But it's not just any old conversion to make the 0's and 1's in to an image. It's a lossy conversion to a format akin to a JPEG. Now, it has enough data in the background that you can work the color and change it - change the white balance, bring up the shadows, etc, but what you're displaying on the screen at any given time is still a lossy conversion - those sliders are just changing how to display that lossy conversion.

So yes, the RAW is the master file saved at the recording studio. But it's more akin to the one that you never get (unless you're the recording engineer) because there are decisions and manipulations to be made so that the file is ready to be listened to.

If you hire a photographer, they will probably deliver results to you in JPEG, and that's OK, part of what you're hiring is someone who is skilled at converting that RAW file in to something that is pleasant to the actual human eye. Not something that's flat with little color or contrast. I don't know your skill level with professional photo editing, or color grading software. But I'd be willing to wager you'd find you're better off with a JPEG that a pro converted from RAW, than anything you'd get out of your RAW editor of choice.

The RAW is NOT like a higher bit-rate or bit-depth file, rather it's like the unmastered file that a mastering engineer works with. I would get a better file from a professional mastering engineer (even if it's an MP3) than an FLAC or WAV or DSD or whatever, that I converted myself. So maybe it's not a bad analogy at all. You're just proving the wrong point.


----------



## pinnahertz

KeithEmo said:


> Well, obviously, since my brain can't read digital file formats, there's always going to be a conversion involved.
> My point was simply that, regardless of what goes on inside the camera, the RAW file is the most accurate version of the information which the camera makes available _TO ME_.
> And, yes, most of the software I use is able to deal with that directly, so I have no compelling need to sacrifice using the best version of that data available because of limitations on my part.
> If you put the camera in the position of the mixing engineer, then the RAW file is analogous to the master file that the engineer saves from his (or her) console.
> (I see that as the dividing line between "production" and "reproduction".)


Absolutely wrong, your analogy is inapplicable. 

Here's why:
*
24/96 "master" file vs Camera RAW
*
24/96: Linear representation of its input voltage, which can be directly linearly converted to an output voltage
CR: is relatively unprocessed sensor data. In some cases not even directly RGB (i.e. Sony), not directly linear conversion to usable image

24/96: Playable easily as recorded
CR: must always be processed to be displayed or printed

24/96: Workflow into a DAW - input original directly, then internally processed with 64 bit float, then down-sampled for export in 24/96 (or anything else).
CR: Adjustments are made_ before conversion_ to display image with known CR profiles, monitor profiles, and printer profiles.  Pre processing before conversion is typically 16 bits/channel.  Once the display image, or export image is created, _CR data is no longer available._   Export images are typically 8 bit/channel, any format.  Everything after CR is lossy.  Everything after 24/96 can be lossless.

24/96: all file types are interchangeable without loss (aiff > wav, etc.)
CR: Each manufacturer has its own CR file, incompatible with others.  The only open source raw standard file (DNG) has not been universally adopted.

24/96: contains only linear quantized information
CR: data must be processed to be displayed or printed by doing (the last 6 items are often done by processing in the camera (from Wiki):

decoding – image data of raw files are typically encoded for compression purpose, but also often for obfuscation purpose (e.g. raw files from Canon[34] or Nikon cameras).[35]
demosaicing – interpolating the partial raw data received from the color-filtered image sensor into a matrix of colored pixels.
defective pixel removal – replacing data in known bad locations with interpolations from nearby locations
white balancing – accounting for color temperature of the light that was used to take the photograph
noise reduction – trading off detail for smoothness by removing small fluctuations
color translation – converting from the camera native color space defined by the spectral sensitivities of the image sensor to an output color space (typically sRGB for JPEG)
tone reproduction[36][37] – the scene luminance captured by the camera sensors and stored in the raw file (with a dynamic range of typically 10 or more bits) needs to be rendered for pleasing effect and correct viewing on low-dynamic-range monitors or prints; the tone-reproduction rendering often includes separate tone mapping and gamma compression steps.
compression – for example JPEG compression
removal of systematic noise – bias frame subtraction and flat-field correction
dark frame subtraction
optical correction – lens distortion, vignetting, chromatic aberration and color fringing correction
contrast manipulation
increasing visual acuity by unsharp masking
dynamic range compression – lighten shadow regions without blowing out highlight regions
Until the above is done, you don't have a usable picture.

24/96: Files are universally playable directly on computers and most portable devices (possibly with additional software)
CR: New RAW formats are created all the time, they are incompatible with existing RAW software without updates, some require proprietary software (Sony's RAW files can't even be handled in Photoshop!  They must be converted to DNG files first).  None are directly and universally viewable on any device.

24/96: A DAC produces a signal that, with only simple amplification, can drive any speaker, though the acoustic environment severely limit the reproduced dynamic range, it is still playable.
CR: Cannot even be displayed on any monitor without lossy processing first.  You never see the RAW data represented completely on any display. 

If that doesn't give at least some idea of how inapplicable the analogy is, then I must just give up. 


KeithEmo said:


> My point, in the context of this discussion..... which seems to have gotten lost in the smoke.....
> Was simply that I prefer to get the most complete and accurate information available - and then make my own decisions.
> In the case of an image, I want the RAW file (which is as close as I can get to "the data coming off the sensor").


I agree with that, but realize you never actually see all that data.  What you do see is a choice, (possible NOT yours!), of what part of the RAW data is important to display and how.  When you manually adjust the RAW converter, you are doing so through the mask of your monitor and light environment.  Unless you precision calibrate your monitor to a standard like D65, your decisions will be skewed to compensate.  Even then, your decisions will not translate to a print viewed under 2700K light.  _Your conversions will always reflect your own imperfect preference._


KeithEmo said:


> And, in the case of audio, it would be the highest resolution version or the original content available (excluding up-sampled versions).
> In both cases, I prefer _NOT_ to trust someone or something to decide _FOR ME_ what information I need and what information I can do without.


I read this as an incredibly arrogant stance.  You admit your lack of ability, but don't trust anyone else.  There is apparently only one expert in your world, and you trust no one else, even those with far more experience and knowledge.  If I'm reading that wrong, please correct.

No matter. You can desire all the data, but you won't ever get it.  The original content is never released.  There are judgements made along the line.  You'll get what you are allowed to get, and it's the best you can get.  It might be related to the original 24/96 ADC output, but likely not even close.  It will be resampled, processed, mixed, mastered, and released.  You're obsessed with the original, but you can't ever have it.  So, which would you rather have?  The original ADC file at 24/96?  OR  the produced version at 16/44?  It doesn't matter, you still have two choices: take it or leave it.


KeithEmo said:


> Honestly, people seem to get far too excited about the whole "high-res question".
> From a _PURELY_ practical aspect, it doesn't matter why they're being sold at all.... all that matters is whether a given file sounds better or not.
> Of course the vendors want to sell more downloads; and, of course, we prefer to only buy them if they're better; it's Economics 101 (and there's nothing sinister about it).
> Most of my current DACs support 24/192k.
> ...


Yes, it's always rough to lose control.  But Apple's decision doesn't cause loss of your control, not at all.  You still have control: don't buy from Apple, buy the CD and rip it.  Others will choose to have control by impulse-purchasing from iTunes and getting that song for $1.29 right now.   Both are correct, both are valid, and both possible.  If you think in grays, you'll see that there are benefits to both ways.  I know that's a tall order.


----------



## KeithEmo

If *I* hire a photographer to take pictures for me personally he'll be delivering the RAW files to me - otherwise I won't be hiring him.
(If it's a commercial contract, and I'm hiring him to deliver finished images, then I may well expect him to do the processing so I don't have to.)
And, no, I'm not going to be "trusting" a photo editing program to make the decisions for me either.
I am quite well versed in how to adjust color - so I'll be doing it the way I want it.

However, I would say that the RAW file qualifies as both "raw footage" and "the archival master of the original content".
On the one hand, just like a "raw track", it may well be improved by skilled editing and adjustment.
However, on the other hand, most RAW files are stored at 12 bits, which really is a higher bit depth than you'll get with most other formats.
(And, if I have a proper profile for my camera, I can even backtrack and make it better by compensating for flaws in the camera - to a point.)
Another benefit is that, if I buy a new monitor next year, and it allows me to notice flaws I didn't notice today, I can re-process that RAW file - starting with all the information I had to begin with.

Remember that, while what's displayed on the screen is lossy, it is NON-DESTRUCTIVE.
Moving the slider and changing what's on the screen is not the same as performing a format conversion where information is actually discarded.
(I can put the slider back, or in a different spot, and my information is still there.)





castleofargh said:


> that's also my understanding. cameras have come challenging the eye's dynamic range ability for a few years(also with a linearity films couldn't reach), while TVs are still struggling to have a dark black and a white that doesn't bleed over 10pixels.
> also true about RAW, that it cannot be displayed without entering a color profile beforehand. usually what we see is what the camera would pick for a JPG version of it(or whatever setting we have in our app). there is a good deal more data available for manipulation and that's the greatness of RAW images, but the display is what it is.





reginalb said:


> But it's not just any old conversion to make the 0's and 1's in to an image. It's a lossy conversion to a format akin to a JPEG. Now, it has enough data in the background that you can work the color and change it - change the white balance, bring up the shadows, etc, but what you're displaying on the screen at any given time is still a lossy conversion - those sliders are just changing how to display that lossy conversion.
> 
> So yes, the RAW is the master file saved at the recording studio. But it's more akin to the one that you never get (unless you're the recording engineer) because there are decisions and manipulations to be made so that the file is ready to be listened to.
> 
> ...


----------



## bigshot

I found a great place to discuss photography that I bet no one here knew about!
https://www.head-fi.org/forums/gear-fi-non-audio-gear-and-gadgets.90/

I'm going to start talking about music and posting pictures of my dogs here again soon.


----------



## pinnahertz

KeithEmo said:


> If *I* hire a photographer to take pictures for me personally he'll be delivering the RAW files to me - otherwise I won't be hiring him.
> (If it's a commercial contract, and I'm hiring him to deliver finished images, then I may well expect him to do the processing so I don't have to.)
> And, no, I'm not going to be "trusting" a photo editing program to make the decisions for me either.
> I am quite well versed in how to adjust color - so I'll be doing it the way I want it.


Having worked as a photographer, I can tell you we'll never work together.  I NEVER release my RAW files.  Ok, perhaps not never, but you would pay a lot to get them.   And that means we'd never work.


KeithEmo said:


> However, I would say that the RAW file qualifies as both "raw footage" and "the archival master of the original content".
> On the one hand, just like a "raw track", it may well be improved by skilled editing and adjustment.


No, the RAW file is the camera negative, full of information but not directly usable for _anything other than storage_ without conversion.  A "raw track" is directly usable without conversion, though it may or may not be improved on later.


KeithEmo said:


> However, on the other hand, most RAW files are stored at 12 bits, which really is a higher bit depth than you'll get with most other formats.


Current cameras from Canon (for one) are 14 bit RAW, even the entry-level DSLRs. I'd be surprised in Nikon wasn't there too.  TIF and PSD (and others) can be formatted to 16 bits/channel.  Actually, that's recommended for advanced editing, with up-sampling from RAW to 16bit done as part of the initial conversion before display/editing/exporting. 


KeithEmo said:


> (And, if I have a proper profile for my camera, I can even backtrack and make it better by compensating for flaws in the camera - to a point.)


Yes, camera flaws, lens flaws, etc., etc....all of which breaks your analogy to audio.


KeithEmo said:


> Another benefit is that, if I buy a new monitor next year, and it allows me to notice flaws I didn't notice today, I can re-process that RAW file - starting with all the information I had to begin with.
> Remember that, while what's displayed on the screen is lossy, it is NON-DESTRUCTIVE.
> Moving the slider and changing what's on the screen is not the same as performing a format conversion where information is actually discarded.
> (I can put the slider back, or in a different spot, and my information is still there.)


Basic non-destructive editing has existed for images, audio, video, text...pretty much everything for a very long time.   Heck, it's been that way in iPhoto (free with a Mac) pretty much since the beginning, including cropping!

What's your point? Again??


----------



## Whazzzup

here you go big shot


----------



## reginalb

KeithEmo said:


> If *I* hire a photographer to take pictures for me personally he'll be delivering the RAW files to me - otherwise I won't be hiring him...



Then you're unlikely to find a very good one to hire.


----------



## bigshot

Whazzzup said:


> here you go big shot



Those aren't yours! They have a logo in the corner!


----------



## Strangelove424

Whazzzup said:


> here you go big shot



awwwwwwww. Even if you stole it.   

Keith will now link adorable fluffy wuffy puppy play to the need for hi res audio in 3...2...1....

To be fair, some of us are fascinated by certain things, especially those things related to their hobbies, and subject matters wander around here when we share those hobbies. It's the allegorical connection of these shared interests to the justification for high audio sample rates that I personally find to be manipulative and contrived. If that perspective had substance, it would have establish itself literally not metaphorically. We are, after all, talking about scientific concepts of bit rate and perception, not 'if the banana is really a banana'. 

To that extent, the justification is always looking for a new analogy to grapple onto. I have a feeling I could have mentioned any of the Sesame Street characters instead of IR, and certain company would find a way to connect it back to high res audio. Just like volcanoes were somehow.


----------



## bigshot

The Treachery of Analogies!






This is not a pipe.


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## Strangelove424

LOL

I pray that an analogy will not be made of this analogy of analogies. We could here for a while.


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## Whazzzup

that would be an enigma wrapped in a conundrum


----------



## castleofargh

reginalb said:


> Then you're unlikely to find a very good one to hire.


 I guess it can depend on the country's laws, and on the kind of job. but again I agree it doesn't seem like a thing pros would do willingly. if there is enough money, then of course anything is possible ^_^.  

@bigshot not long ago when I pretended to give the angry look to people going off topic, I believe you were the one telling me that when a topic has reached its purpose, it's fine to go off topic. now karma is making you pay.


----------



## bigshot

castleofargh said:


> @bigshot not long ago when I pretended to give the angry look to people going off topic, I believe you were the one telling me that when a topic has reached its purpose, it's fine to go off topic. now karma is making you pay.



When I go off topic I keep it brief. I don't wallpaper the place with big long diatribes and enter into arguments with people about the off topic subject. I just share a concise juicy tidbit... What we have here is a big sloppy mess!

I'm chasing my dogs around the house to shoot a glamour shot of them as we speak.


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## Whazzzup




----------



## bigshot




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## reginalb

Geez, here come the topic police. I finally had something I could speak more authoritatively on!


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## bigshot (Nov 21, 2017)

WHO LET THE DOGS OUT? WOOF WOOF WOOFWOOF!


----------



## RollsDownWindowsManually (Nov 22, 2017)

I have nothing significant to add compared to the rest of you. However, I have been re-ripping my cd collection because I finally have the hard drive space to store it in FLAC (yeah yeah, I know most people had that 10 years ago, I'm a laptop user though and spent all my storage money on headphones).  Along with that, I discovered I have a few 24/96 albums and the Schiit Jotunheim on my nightstand has a 24/96 capable DAC. So.... Into the Foobar2000 ABX I went to compare the 24/96 files to the same songs encoded LAME V2 Mp3 ~ 190 kpbs VBR, which is what I've been using for the past 15 years or so.  First try 2 out of 8.  Second try 5 out of 8.  I think that'll be enough for me.  The two files sound the same on jotunheim - HD800.  I tried it on my Home Theater too, which has phase technology 3 way floor standers in  treated room with an SVC PC-13 subwoofer.  Similar results.

I will admit there is a sense of satisfaction in knowing I'm listening to higher quality files, but I can't detect an audible difference despite my best efforts.  Trust me, I wanted to be able to do so.

Also here is the noise floor for my room as measured with the UMIK-1 at the main listening position:


----------



## bigshot

I bet if you tried a third and fourth time your odds would continue to fluctuate. I use AAC 256 VBR and I know for a fact that it's transparent. I spent the better part of two weeks encoding a variety of music into multiple codecs and data rates and whenever I could tell a difference, I dismissed that encode from consideration. After a while I was left with AAC 256. I added VBR just to feel secure and to optimize the files. No more re-encoding. I'm happy for good.


----------



## gregorio

KeithEmo said:


> If *I* hire a photographer to take pictures for me personally he'll be delivering the RAW files to me - otherwise I won't be hiring him.



And if you hire me, you can have it at whatever sample rate/bit depth you want, up to and including 192/32 float. You can even have a hard disk full of just all the raw takes, with no editing, mixing or mastering but of course that's not a piece of music. If you do want an actual piece of music (mixed) then I can't give it to you in that mix format, as I cannot export a 64bit file and I also can't guarantee it will sound any good played back anywhere other than my studio unless you have it mastered.



KeithEmo said:


> I prefer to get the most complete and accurate information available - and then make my own decisions. ..  in the case of audio, it would be the highest resolution version or the original content available (excluding up-sampled versions). ... I prefer _NOT_ to trust someone or something to decide _FOR ME_ what information I need and what information I can do without.



Of course if you're talking about someone else hiring me and some other artist, then you take the artistic decisions made by them, in the format they decide to release, or you don't take it at all. You seem to think (along with many other audiophiles) that, for example, Coldplay went into a studio, recorded a perfect performance of a song, then the engineers came along, spent a considerable amount of time and effort butchering it and only this butchered version is available, while you want that original perfect performance. That original perfect performance never existed, that's not how modern music is recorded/made, it's not how music has been made since about the 1950s! Coldplay do not leave after they've recorded the perfect performance, they records some parts, play around with those parts in the studio with the engineers, process them, go to another studio record some more, edit, mix and process some more, throw some of it away and record some different things, process and mix some more and the song gradually takes shape. Song creation is an evolution, what I've described is pretty much how everyone has done it for decades. It's a collaborative process between Coldplay and the engineers, with Coldplay always having the final decision, same when the mix is finished and it goes off for mastering. You don't like or trust the decisions made by Coldplay, then don't buy Coldplay. If you do buy Coldplay, it's because you like the decisions they make! But no, you can't have the raw takes, you can't have half a mix and you can't have it without the mastering Coldplay have approved!

G


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## gregorio (Nov 22, 2017)

RollsDownWindowsManually said:


> Also here is the noise floor for my room as measured with the UMIK-1 at the main listening position:



I might be wrong but AFAIK, that probably is NOT the noise floor of your listening environment! Those cheap measurement mics are actually very good at measuring room response to a signal, provided you've got a professionally calibrated correct file for it. However, they are ONLY very good in response to a signal (say pink noise or a sweep tone) and, only if that signal is played pretty loud, say over 70dBSPL or so, because they have pretty lousy self noise. So what you've measured is probably as much the self noise of your mic as it is your listening environment noise floor! They are the wrong tool for the job of measuring noise floor, you need something quite different and significantly more expensive! BTW, I'm not knocking cheap measurement mics, I own and use one myself (though not that particular one) but I don't use it for measuring noise floor.

G


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## KeithEmo

Obviously it depends on "what circles you travel in".

Some DACs cannot play 24/96k files - but most can (so I wouldn't say that "everyone can play them").
Likewise, every image editing program I use can accept and process Nikon RAW files (but some low end consumer viewers can only show JPGs).
Some image editing programs can only handle TIFs, or even only JPGs..... and some music players can only play MP3's..... or can;t handle FLACs..... there are always a range of capabilities.

Wikipedia provided what looks like a more or less exhaustive list of all the steps that go into processing a typical image (although they left out a few of the more interesting ones, and grouped a few differently than I would).
However, the fact remains that:
- _SOME_ of those steps (like de-mosaicing) are virtually_ ALWAYS_ performed in the camera hardware
(so I have no option to get the sensor data before that and use my own de-mosaicing software... a least not with most cameras)
- _SOME_, like rendering to a JPG image, _MAY_ be done by the camera, but can usually be done better by a human being with some skill in image editing
- and _SOME_, like adjusting the white balance, fall somewhere in the middle 
(If I'm shooting a bunch of snaps I'll probably let my favorite plugin handle the white balance; if it's important stuff, I'll do it by hand, or let the plugin have a go at it, then tweak it by hand; if it's _REALLY_ important I'll use a ColorChecker).
- and with some, like optical lens correction, there are benefits and drawbacks to both methods (so the "best" choice depends on the circumstances)

And, yes, I DO calibrate all my monitors....
And, if I'm being really critical, I will snap a frame of the ColorChecker so I can calibrate the conversion of a particular set of images to both the camera and the lighting....
Of course, I can only wish that I had that much control over the audio tracks I purchase (or the photos).

As for Apple......
Unfortunately, on at least three or four albums that I know of, they were _ONLY_ released on iTunes... and only in a lossy format.
With two of them I was able to contact someone closer to production and get a real copy.
With the other two I was stuck with the AAC copy (which I consider to be a significant failure on their part to consider my desires - as a customer).

And, yes, it seems as if we're rarely if even going to get the "real original master content".
However, the fact that so many companies seem to _PROMISE_ to give it to us sure suggests that many people would like to have it.
Pono's promised to give us "versions approved by the artist" and MQA promises music with an actual provenance..... 
And, yes, I can understand why the studios would prefer _NOT_ to do so.....    
(However, I also can't shake this nasty feeling that Apple hopes someday to be able to charge their customers _AGAIN_ for the music they already paid for - to get the lossless copy.)

And, to answer your question, I would personally like to have that choice.
I may well decide I like the produced 16/44k version better than the RAW files (or I may prefer to get the 24/96k PRODUCED copy - which probably does exist).

HOWEVER.... let's let this thread die since it's gotten pretty far off topic.

To get back to my point.....
I'm not determined to always get that 24/96k console copy (although I would cheerfully pay extra for it with _SOME_ albums I especially like).
(And, if it turns out that it sounds exactly like the CD, then I will have bought myself the assurance that I have the best copy available.... so I can avoid wondering.)
But, to put it _VERY_ bluntly, years ago some guy told me that 128k MP3 files were "plenty good enough".
He lied (or he was right that those 128k MP3 files really were good enough - but only for 95% of his customers - and not for me).
And I'm sure glad I didn't buy all my albums in MP3, or on the iTunes store, instead of CD - to save a few bucks - or bytes.
And a lot of people seemed to think that DVD video was "plenty good enough" - especially when compared to the previously available VHS and Beta tape formats.
So I'm simply not especially comfortable when "some other guy" insists that, this time, 16/44k files are "good enough".
I've simply experienced too many situations where technology that seemed to be "good enough" looked different in hind-sight... after a new and better technology became available.



pinnahertz said:


> Absolutely wrong, your analogy is inapplicable.
> 
> Here's why:
> *
> ...


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## pinnahertz

This thread seem to have gone to the dogs.


KeithEmo said:


> Obviously it depends on "what circles you travel in".
> 
> Some DACs cannot play 24/96k files - but most can (so I wouldn't say that "everyone can play them").
> Likewise, every image editing program I use can accept and process Nikon RAW files (but some low end consumer viewers can only show JPGs).
> ...


Your image analogies are just plain broken.  I'm not replying to them any further.  Your only remaining points in that discussion is to argue with me.

I think your "point" above is ridiculous.  Anyone looking at new tech compared to old tech only, and thinking it's good enough is comparing the wrong thing.  Hind sight is often lack of knowledge not just lack of foresight.  The very first time I encountered 128 mp3 and listened to it I rejected it as not good enough. That was easy!  But I took it for it's intended use as adequate.   The very first time I saw a DVD I recognized the improvement, better than some of what we had, but never thought "good enough".   However, comparing 16/44 with with actual live analog console out...very different story.  That was a complete fooler!  We had a live mix in studio, and an early digital recorder.  The monitor was accidentally set to monitor the ADC/DAC output (very short time delay), and we monitored and mixed that way!  Didn't catch the mistake for qutie some time.  When the monitor was switched back to the 2-mix, there was no audible difference!  And that's in the early 80s with what we'd consider a really bad ADC/DAC now.  Did the same thing when DAT came along and we could do 16/48.  

But look, I'm really done here.  Have fun.


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## gregorio

KeithEmo said:


> [1] And, yes, it seems as if we're rarely if even going to get the "real original master content".
> [2] But, to put it _VERY_ bluntly, years ago some guy told me that 128k MP3 files were "plenty good enough".
> He lied (or he was right that those 128k MP3 files really were good enough - but only for 95% of his customers - and not for me).
> And a lot of people seemed to think that DVD video was "plenty good enough" - especially when compared to the previously available VHS and Beta tape formats.



1. I covered all this in my last post to you!
2. Yes, he lied. He and many companies do that in order to sell their products. That's why we need to separate the marketing from the facts! Also, DVD obviously wasn't good enough, neither was HDTV and 4K HDR is nearly there but I don't think it's quite up to the limit of the human eye. I'm not sure what the limit of the human eye is, but let's say it's 8K Ultra-HDR, what happens if someone comes out with 16K and says it looks better? What happens when 16K is pretty standard and someone brings out 32K, what about 64K and 128K? This is what's ALREADY happening in audio, we're so far beyond what the ear can hear it's ridiculous, 192kHz is already 4 times beyond what we can hear and now we've got 384kHz and even 768kHz!!

G


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## bigshot

KeithEmo said:


> But, to put it _VERY_ bluntly, years ago some guy told me that 128k MP3 files were "plenty good enough".
> He lied



How are you finding the lossy tracks above 128 in the listening test I sent you? Have you determined your threshold for the three codecs there yet? When you're done, I'll let you know what other people have found. There is a range but it's narrower than you would expect.


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## RollsDownWindowsManually

gregorio said:


> I might be wrong but AFAIK, that probably is NOT the noise floor of your listening environment! Those cheap measurement mics are actually very good at measuring room response to a signal, provided you've got a professionally calibrated correct file for it. However, they are ONLY very good in response to a signal (say pink noise or a sweep tone) and, only if that signal is played pretty loud, say over 70dBSPL or so, because they have pretty lousy self noise. So what you've measured is probably as much the self noise of your mic as it is your listening environment noise floor! They are the wrong tool for the job of measuring noise floor, you need something quite different and significantly more expensive! BTW, I'm not knocking cheap measurement mics, I own and use one myself (though not that particular one) but I don't use it for measuring noise floor.
> 
> G



So you're saying that the noise floor of my room is probably even lower than that measurement?  If so, I guess it would be a good environment for evaluating 24/96 then. 

And yeah, UMIK-1 comes with 90 and 0 degree calibrations files.


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## castleofargh

gregorio said:


> I might be wrong but AFAIK, that probably is NOT the noise floor of your listening environment! Those cheap measurement mics are actually very good at measuring room response to a signal, provided you've got a professionally calibrated correct file for it. However, they are ONLY very good in response to a signal (say pink noise or a sweep tone) and, only if that signal is played pretty loud, say over 70dBSPL or so, because they have pretty lousy self noise. So what you've measured is probably as much the self noise of your mic as it is your listening environment noise floor! They are the wrong tool for the job of measuring noise floor, you need something quite different and significantly more expensive! BTW, I'm not knocking cheap measurement mics, I own and use one myself (though not that particular one) but I don't use it for measuring noise floor.
> 
> G


agreed, going from cheap mics to cheap mics, and various power sources and cheap ADCs along the way, it has become painfully clear that measuring variations at 80dB or 90dB is a piece of cake for them all(within a limited frequency range), but the noise floor is rarely the room's or at least not only the room. although it's easy enough to play around trying to isolate the mic from outside noises with some materials, and see if the noise floor measured goes significantly lower or not. at least in the mid/high freqs where isolating is easy.
 behold all the headroom I have... :'(  I can detect that my laptop is turned ON, and that's close to the limit of what I can measure with my rig.


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## bigshot

I would think that the sound measuring stuff by MiniDSP would be at least in the ballpark of correct for noise floor. I can't see the image though. What was the reading?


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## Strangelove424

I find that computers are really a troublesome source of ambient noise, and we are all using them now for media consumption. I struggled to build a desktop computer that was inaudible from listening range at low/medium work loads. The loudest component is the PSU, and it's not fan noise, just hum. I'm very happy with what I accomplished, partially by using fans that shift the noise to a lower, less detectable frequency. But even though I dropped the computer a few db, I still have a minimum ambient of 31db in my house because of the damn roads nearby. In rush hour it gets to 35db, but when there is little traffic I can occasionally see noise floors just below 30db. That's not with an expensive mic though, so the numbers might be off.    

Anyway, if anybody wants any help with computer cooling, feel free to PM me, I love the subject matter, and am always happy to give people some pointers on lowering computer ambient noise.


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## KeithEmo

Honestly, I want to sit down and do it right, and I haven't had a chance yet.
I promise I will do so over the long weekend and report my results.

I'll also state publicly something that I already said in one of our PMs......
I am NOT especially confident that I'll be able to distinguish major differences, or tell for sure which is which, among samples of music I'm not familiar with.

I'm going to use one last analogy to image processing here.....
If you were to show me two different images, on two different monitors, I may not necessarily be able to tell which one is correct if they look different.
The reason I calibrate my monitors is so that I KNOW which one is correct.
So, even if I can't tell which of two monitors is "right" or "better" I still prefer taking steps to ensure that the monitor I'm using is correct - rather than not - and I don't have to worry whether it is or not.
(I do not accept the logic that "if I can't tell which is which then it doesn't matter".)



bigshot said:


> How are you finding the lossy tracks above 128 in the listening test I sent you? Have you determined your threshold for the three codecs there yet? When you're done, I'll let you know what other people have found. There is a range but it's narrower than you would expect.


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## KeithEmo

I do have a quick suggestion there......

The computer I currently use to play music is a Raspberry Pi.... running Volumio.
It's about three times the size of a deck of cards.... and totally silent (no fans or other moving parts).
I have a USB hard drive connected directly to it; and it's connected to my pre/pro via USB.
(Volumio plays FLAC and all the other popular formats..... )

Volumio is free, and the Raspberry Pi (including a case and power supply) is about $60 total.

For a minimalist, and totally silent, solution it really is hard to beat.



Strangelove424 said:


> I find that computers are really a troublesome source of ambient noise, and we are all using them now for media consumption. I struggled to build a desktop computer that was inaudible from listening range at low/medium work loads. The loudest component is the PSU, and it's not fan noise, just hum. I'm very happy with what I accomplished, partially by using fans that shift the noise to a lower, less detectable frequency. But even though I dropped the computer a few db, I still have a minimum ambient of 31db in my house because of the damn roads nearby. In rush hour it gets to 35db, but when there is little traffic I can occasionally see noise floors just below 30db. That's not with an expensive mic though, so the numbers might be off.
> 
> Anyway, if anybody wants any help with computer cooling, feel free to PM me, I love the subject matter, and am always happy to give people some pointers on lowering computer ambient noise.


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## bigshot (Nov 22, 2017)

There is a broad range of rates and codecs there. The point is to find your level of transparency. Artifacts are artifacts. They are either there or they aren’t. The differences between the samples aren’t a graduated scale like frequency imbalances or distortion levels. We’re talking here about either/or... either there are artifacts, or there is sufficient bandwidth to reproduce the sound transparently. You will be able to tell if it doesn’t matter or not.

When the test is done, you can pull the file apart and A/B them against the known lossless file using foobar, but if there are no artifacts, the results will be the same.


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## Strangelove424

KeithEmo said:


> I do have a quick suggestion there......
> 
> The computer I currently use to play music is a Raspberry Pi.... running Volumio.
> It's about three times the size of a deck of cards.... and totally silent (no fans or other moving parts).
> ...



For a dedicated media player, the Raspberry Pis are great. Low power consumption, easy to mount, wireless/HDMI capable. I have one too, complete with LEGO case! I use my media computer for a few different things though, including my sim racing rig, so it’s a bit of a beast.


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## Whazzzup

dedicated external music server running roon core via ssd is the best component i purchased in absolute sq enjoyment.


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## castleofargh

I have a fanless weak crap I bought second hand for almost nothing, I got it just to play music. but in practice I almost always end up using the other computer anyway, for all the important stuff like telling that somebody is wrong on the web and other vital stuff like playing games(siriouz biznezz). so the fanless craputer has about as much use as my CD player nowadays, almost none. 

the glass is half full: good thing about the fans making noise is that I never get to complain about my speakers hissing even so slightly(I need to get close to notice).


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## castleofargh

bigshot said:


> I would think that the sound measuring stuff by MiniDSP would be at least in the ballpark of correct for noise floor. I can't see the image though. What was the reading?


it just suggests that the self generated noise is really close to the noise in the room despite measuring close to the computer, so not a "quiet" room.  I don't get much improvement from a physical seal and isolation of the mic(of all the mic not only the opening for the capsule). I'm using a Behringer EMC8000 and someone reported it to have 32dBA of noise floor. which is weirdly in line with what I just got(mine isn't weighted).
I've come to a point of self skepticism where getting the expected result makes me even more suspicious ^_^.


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## KeithEmo

Obviously we simply look at the same thing from opposite directions (or we interpret the semantics differently).

To me, when you're talking about lossy compression, there are _ALWAYS_ artifacts (by definition_ ANY_ detectable or measurable variation from the original is an artifact).
I can convert a WAV file to a FLAC, then back to a WAV, and there will be _NO_ differences between the "before" and "after" file - thus no artifacts.
However, if I do the same with MP3, or AAC, I will be unable to get back my original WAV file (when I compare the before and after there will be differences; those differences are artifacts; binary statement there).
(I don't need to do any complicated tests - a simple bit-compare will tell me with absolute certainty whether there are artifacts or not.... )
Therefore, the only question is whether I am able to _NOTICE_ the artifacts/errors under a specific situation or not (or perhaps, assuming I notice them, whether I consider them significant or not).

If I am totally unable to pick out the artifacts, I will have proven is that, with that particular sample, one particular human (myself), was unable to notice the artifacts.
Of course, since I'm probably not the human with the best hearing on the planet, that won't prove that nobody else can hear them.
And, even if nobody else can hear them, that still won't prove that they don't exist - just that nobody can hear them.

The biggest difference between you and I seems to be the way we interpret the data.
To me, there is absolute data - for example, a copy of a file either is or is not bit perfect.
And there are an infinite number of ways in which it can be _IMPERFECT_, some of them more obvious than others, but only one "100% right".
And, while we can debate whether a given flaw is or is not noticeable, there is no doubt that "perfect is perfect" - which is why, to me, it's just easier to stick with perfect.
Note that, for the sake of this discussion, I'm assuming that whatever version of music I acquire is "my local master", so my only goal is to maintain that copy without degrading it.

However, the term "transparent" is more a flexible term, based on some sort of opinion......
An optics expert will tell you that there is no such thing as a transparent sheet of glass - because all glass absorbs somewhat more than 0.000% of the light passing through it.
Therefore, what you're really defining is "transparent enough" or "so transparent I personally cannot see the flaws".



bigshot said:


> There is a broad range of rates and codecs there. The point is to find your level of transparency. Artifacts are artifacts. They are either there or they aren’t. The differences between the samples aren’t a graduated scale like frequency imbalances or distortion levels. We’re talking here about either/or... either there are artifacts, or there is sufficient bandwidth to reproduce the sound transparently. You will be able to tell if it doesn’t matter or not.
> 
> When the test is done, you can pull the file apart and A/B them against the known lossless file using foobar, but if there are no artifacts, the results will be the same.


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## bigshot (Nov 22, 2017)

Audibly transparent means that the quality of the sound exceeds your ability to hear it. Beyond that point, better quality doesn't matter because it can't be heard. If you can't hear something, it's irrelevant to your enjoyment of recorded music. No one wants to pay more or suffer inconvenience because of stuff that doesn't matter.

You see, there's a difference here... I've spent a great deal of time to try to carefully document my perception. I spent two weeks straining to hear differences between a wide range of rates and codecs and a wide range of different kinds of music. I've also invested a great deal of time sharing this test with a wide range of people and I know the results of those tests. You just have a concept of sound purity that you have faith in, but you haven't made much effort to see if it's a valid concept. I understand how you can't be sure where the line of transparency lies. And I'm sure you can understand how I can be pretty sure.

To know something, you have to want to know. At least half the time, I hand out this test and I never hear back from the people. They aren't really interested enough to put their ears on the line like this. Ignorance is more comfortable to them because they can remain the same and their thinking won't have to change. Changing your mind can hurt. Other people will fudge the text reports from Foobar or cherry pick their tests to make it look the way they want it to look or engage in logical fallacies to prop up their weak theories. That goes beyond ignorance into disingenuousness. But we're in Sound Science here. We enjoy finding out. We don't want to prop ourselves up and pat ourselves on the back. We just want to know so we can apply that knowledge to get better sound out of our audio systems.

If you make the effort to find out the truth, you don't need to depend on faith. Knowing what matters and what doesn't is the highest level of understanding.


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## RollsDownWindowsManually (Nov 22, 2017)

ABX testing is not fun to me, especially when you're close to the threshold of audibility and really have to strain to hear any difference. I can see why some people wouldn't like to do it. For me that starts to happen at 128 kbps LAME MP3; at that point I can hear the difference clearly on some songs (hard rock) but not at all on others (rap).  At 96 I can hear the difference clearly on almost anything without even listening carefully.  I think it's mostly a function of the high frequency cutoff - I can only hear up to about 18.5 khz and 96 kbps MP3 starts to roll off a little below that I think whereas 128 goes right up to about 18.5 iirc.


Did 16 more trials of 24/96 vs V2 VBR MP3 and and got 3/8 followed by 5/8, so I'm batting right at about .500 - same as I would be just guessing I reckon.


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## bigshot (Nov 22, 2017)

RollsDownWindowsManually said:


> ABX testing is not fun to me, especially when you're close to the threshold of audibility and really have to strain to hear any difference.



If you have to strain to hear a difference, you're in the neighborhood of your line of audible transparency. You don't have to strain. The goal is to determine what matters and what doesn't, not to force yourself to hear the unhearable. Call that your line, bump up your encoding one notch so you're sure you're on the other side of the line, add VBR so your files are efficient and go with it. No reason to look back.

By the way, your findings are pretty typical. For some people (and with some kinds of music) the line might be a notch higher than yours, but you are in the majority. If you go two notches up, no one can hear a difference with any kind of music.

Congrats! There are a lot of people around here who are much less self aware of their hearing limits than you are.


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## pinnahertz

bigshot said:


> I would think that the sound measuring stuff by MiniDSP would be at least in the ballpark of correct for noise floor. I can't see the image though. What was the reading?


It's not.  All of those mics, and even really good measurement mics like the Earthworks M30 ($700), are too noisy for good room noise measurements.  It's the 1/4" capsules.  The good low noise measurement mics, like the B&K 4179 capsule, are all 1" capsules and very expensive.  But then you get -5.5dBA.  Otherwise all the 1/2" stuff is 14-18dBA, and the 1/4" stuff is in the 20+ range.


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## bigshot

You're going to have a higher noise floor than 20dB in your living room anyway most likely. 20dB is like a recording booth. My $100 SPL meter came up with between 35 and 40dB in my room, and that sounds totally in line with what I would expect. There's a place for "close enough for government work" measurements.


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## KeithEmo (Nov 22, 2017)

There's something here that seems obvious to me, and probably to you, but I would like to clarify it for everyone else........

I agree entirely that "audibly transparent means that the quality of the sound exceeds your ability to hear it" - and, by that definition, I agree with virtually everything you've ever said.
Where we seem to disagree in in terms of usage.
You seem quite content to declare that "if I can't hear the slightest flaw when I listen to this file today then it's good enough".
Personally, I am far less certain than that...... mostly because I suspect that my needs may change later.
If I were to determine with absolute certainty that a given "level of accuracy" was "absolutely audibly perfect" to me today.....
When I buy my next file, I would still buy the one that's "200% as good as I need" - just to provide myself a margin or error... in case something changes...
(Like I _DO_ decide to turn the volume up a bit on the quiet parts; or I buy a new pair of speakers and, on them, certain things become more obvious).
I can honestly say that, contrary to the alarmist title applied to this thread, I have rarely been disappointed later to find out that I bought something _BETTER THAN_ what I needed.
(I wouldn't pay $100 for a 24/96k copy which didn't sound better to me..... but I'd cheerfully pay an extra $5 to get 200% of the quality I really need instead of just exactly 100.0000000%...... I like safety margins.)

JUST TO BE PERFECTLY CLEAR........ since I may appear to be "taking the other side on this issue".......... I'm not.
I agree entirely that it's extremely useful to have ALL the information.
After all, nobody can make an informed decision without all the information.
I don't personally buy high-resolution files because I'm convinced they sound better.....
And, since that's not the reason I buy them, I doubt anything we find out here will convince me not to.....
However, I am still interested to know whether the differences are really easily audible or not.....
(And, yes, it's also interesting to know whether they're "downright obvious" or "only maybe a tiny bit audible with music I'm really familiar with" or "not audible at all".)
And, for others who _MAY_ be buying high-res downloads based solely on claims of superiority which they may not find to be true, this information will be even more valuable.
There's no such thing as "bad information" - as long as it's accurate and clearly stated.

I might mention one other (small) area where I disagree with you.
I personally enjoy certainty... and I find uncertainty... disquieting.
Therefore, I really do enjoy listening to something more when I am absolutely certain that it is the best copy I have access to... and being unsure of that does reduce my enjoyment - at least a little.
And, when I look at images on my calibrated monitor, I do enjoy them a little more knowing that they're right because my calibration is current and up to date.
I may enjoy them quite a bit on an uncalibrated monitor - but I enjoy them just a tiny bit _more _with that last little nagging doubt removed.
(And I suspect that lots of "audiophiles" feel this same way about the music they listen to.)




bigshot said:


> Audibly transparent means that the quality of the sound exceeds your ability to hear it. Beyond that point, better quality doesn't matter because it can't be heard. If you can't hear something, it's irrelevant to your enjoyment of recorded music. No one wants to pay more or suffer inconvenience because of stuff that doesn't matter.
> 
> You see, there's a difference here... I've spent a great deal of time to try to carefully document my perception. I spent two weeks straining to hear differences between a wide range of rates and codecs and a wide range of different kinds of music. I've also invested a great deal of time sharing this test with a wide range of people and I know the results of those tests. You just have a concept of sound purity that you have faith in, but you haven't made much effort to see if it's a valid concept. I understand how you can't be sure where the line of transparency lies. And I'm sure you can understand how I can be pretty sure.
> 
> ...


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## danadam

bigshot said:


> My $100 SPL meter came up with between 35 and 40dB in my room


Isn't 35 dB actually the limit for cheap (or even not so cheap) SPL meters?


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## bigshot

I guess so. 35dB is also as low a noise floor as most living rooms have too.


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## KeithEmo

Pretty much....

There's also a context here.

For environmental noise measurements, you want a microphone that measures down to very quiet levels.... 
but you may not be especially concerned with really good frequency response accuracy.
For room and speaker calibration, you want a microphone that is exceptionally flat (or well calibrated).... 
but, since you're going to run test tones well above the noise floor to ensure accurate measurements, you probably don't care so much about superior noise performance or superior flatness at very low SPLs.

A cheap "general purpose" meter will probably not be very good for either.... but will more or less get the job done.
However, you're going to pay a lot more for a meter that's really good for EITHER purpose...... 
And you're going to pay a much bigger premium for one that's very good for both.... 



danadam said:


> Isn't 35 dB actually the limit for cheap (or even not so cheap) SPL meters?


----------



## bigshot (Nov 22, 2017)

KeithEmo said:


> You seem quite content to declare that "if I can't hear the slightest flaw when I listen to this file today then it's good enough".
> Personally, I am far less certain than that...... mostly because I suspect that my needs may change later.



Think about it logically... If you are ripping a file from a CD and the compressed version sounds completely identical to the original CD, then for the purposes of listening to it, it's perfect. You won't need better sounding than identical in the future. It can't sound any better than the CD it's ripped from, and your ears aren't going to hear any better 20 years from now. It's a good idea to bump the bitrate up a notch just to cover some particularly difficult to compress sort of sound. But I can tell you that I have ripped over 10,000 CDs to AAC 256 VBR and I have never run across an artifact in any of my rips.

I've experimented a lot with compression codecs and there's something most people don't know about how the way they work. If you compress a song and it throws out inaudible information to make the file smaller; if you run it through the compression again, it has already thrown out all the inaudible information, so it makes no change. You can compress a file over and over and it doesn't degrade. Once it's been compressed, it won't compress any further unless you change the data rate. This means, if the file is audibly transparent and you need to transcode it to some other new format, as long as the new format is also audibly transparent, there will be no degradation in the sound quality.

If you have recorded and mixed the song yourself, it makes sense to archive your mix in the full quality without compression. You might want to go back into the track and remix it and higher bitrates will be useful to you. But for the purposes of listening to music in the home, there is no functional difference between 24/96 or a FLAC file or a high bitrate lossy file that has reached the level of audible transparency. You can pack hard drives with big fat WAV files or high bitrate/high sampling rate files, or lossless files if you want. But there is no audible advantage to it. All that extra file size might as well be excelsior in a shipping box. It's just bulk with no purpose.

Now I can't speak to how you *feel* about having hard drives full of big fat files. If you sleep better at night knowing you are storing a bunch of inaudible bits for posterity, that's fine. But that has nothing to do with sound quality, it has nothing to do with compatibility in the future, and it has nothing to do with logic. It's just plain OCD- the digital audio version of excessive hand washing. Your hands are clean. Your files are audibly perfect. You don't need to wash them again. You don't need to store a bunch of bits and bytes that you can't hear anyway.

The key issue here is that it doesn't make any practical difference if a file is lossless or lossy as long as it is audibly transparent.


----------



## danadam

bigshot said:


> You can compress a file over and over and it doesn't degrade. Once it's been compressed, it won't compress any further unless you change the data rate. This means, if the file is audibly transparent and you need to transcode it to some other new format, as long as the new format is also audibly transparent, there will be no degradation in the sound quality.


In my experience that's not true. It may still be audibly transparent after a few transcodings, but at some point it won't be any more. Here's an example how it sounds after 100 transcodings wav->X->wav for mp3 (preset insane) and vorbis (q10) respectively: SymphonyNo5.zip


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## bigshot (Nov 22, 2017)

I did 10 transcodings of AAC 256 VBR, which is my standard and it sounded fine. I figured since I've been encoding for 15 years or so and I haven't had to re encode yet, 150 years worth should be safe. How likely is it that you're going to do 100 transcodings in your lifetime? Also, perhaps it's different with an older Frauenhofer MP3. AAC is a much more sophisticated codec.

The only caveat is that iTunes tends to boost the volume a hair each time, so you can't be normalized all the way up to zero or it will clip.


----------



## pinnahertz

bigshot said:


> I did 10 transcodings of AAC 256 VBR, which is my standard and it sounded fine. I figured since I've been encoding for 15 years or so and I haven't had to re encode yet, 150 years worth should be safe. How likely is it that you're going to do 100 transcodings in your lifetime? Also, perhaps it's different with an older Frauenhofer MP3. AAC is a much more sophisticated codec.
> 
> The only caveat is that iTunes tends to boost the volume a hair each time, so you can't be normalized all the way up to zero or it will clip.


With the exact same codec settings each recode, the damage should be minimal.  However, if you change codec settings each time, or even one time, or change to a different codec (mp3 > aac), you'll take a pretty obvious hit as perceptual coding is re-done with different rules.


----------



## bigshot

That could be. But if AAC 256 VBR is audibly transparent and the codec you're transcoding to is audibly transparent, then the results should be audibly transparent. A new codec from the future might be able to throw out more info, but it would be inaudible info if both are transparent.


----------



## pinnahertz

bigshot said:


> That could be. But if AAC 256 VBR is audibly transparent and the codec you're transcoding to is audibly transparent, then the results should be audibly transparent. A new codec from the future might be able to throw out more info, but it would be inaudible info if both are transparent.


Yes, but each codec arrives at "audibly transparent" differently, and when the algorithms don't match recoding causes more data loss.  Cascading dissimilar codecs definitely results in audible degradation more quickly and with fewer recodes. 

A number of years ago I evaluated a broadcast digital STL made by Dolby using AC3.  The radio station also used an on-air computer audio system that stored files in MP3 in a wav container. I found the combination of MP3 and AC3 easily audible, where each was transparent by itself. This became a non-issue when both the on-air system and the STL went to uncompressed, but the situation did reveal that recoding twice with very different codecs was audible.

None of this should be an issue now of course.


----------



## bigshot

That's interesting. I should try my transcode test with 320 LAME alternating with 320 AAC for ten generations and see what happens.


----------



## 71 dB

Pinnahertz is totally right here. You can't cascade independently transparent codecs and assume transparent result. It should be "illegal" to do further lossy coding, because the result can be surprisingly bad depending on how the codecs work. You do lossy coding once and that's it!


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## RRod (Nov 24, 2017)

Going between codecs will necessitate an intermediate WAV file, so you need to watch out for quantization errors building up as well.


----------



## KeithEmo

I can't speak for ALL lossy CODECs, but I can tell you that you're wrong about MP3. When you initially compress a file, it does indeed do its best to "throw away unnecessary information". However, the process is not as simple as identifying what doesn't matter and deleting it, and re-encoding something that has already been encoded WILL produce "generational degradation". Basically, the encoder does NOT "just throw away the information you won't miss". What it does is to divide the audio signal into a bunch of frequency bands, each for a short block of time, decide how much "important" information is contained in each, and then divide its "quality/priority" depending on how important the information is that's contained in each. It may discard some information entirely, while other information is simply encoded at lower quality. Each "section" of the information is encoded at the least quality for which "you won't notice the difference" - and the decision of what that will be depends on psychoacoustic properties like masking. Therefore, the majority of information in an MP3 encoded file is neither full quality, nor minimum quality, but somewhere in-between - encoded at "just high enough quality" that you won't notice the loss.

HOWEVER, in no part of this process is there any sort of specific identification of how each individual sound was treated, and so no way to ensure that the process won't be applied repeatedly to a given section. Therefore, if a given frequency/time slice has been encoded with a lot of quantization error (because it was deemed to contain "unimportant content"), and you re-encode it, it will AGAIN be encoded with a lot of quantization error - and those errors will compound. If you take a file that's been encoded at 128k VBR MP3 and re-encode it at the same settings, either as is or after converting it back into a WAV file, you will probably not lose much ADDITIONAL quality (because pretty much the same decisions are being made), however the encoder will NOT "simply leave it as is" either. It will be re-encoded, AGAIN with encoding that introduces further quantization errors, so the total sum of the errors will increase. (The result is that areas which are considered unimportant will get significantly worse when you re-encode them, because they will have been encoded at poor quality twice instead of once. Areas which are deemed more important will suffer less degradation, because they will have been encoded twice, but both at a higher quality setting, which causes less loss of quality. You may argue that, since those areas were unimportant to begin with, the additional loss of quality won't matter - but it is there - and the overall quality will decrease with repeated generations.)

With lossy compression, the analogy of a photocopier is quite valid, and illustrates the situation quite well. If you make a copy of a good quality original on a good quality photocopier there will be little loss. And, if you copy that copy again on that same photocopier, there will again be little loss. However, if you make a copy on a poor quality photocopier, a lot of quality will be lost; and more quality will be lost if you make a copy of that copy at the same poor quality. In this analogy, the way MP3 encoding works is that, for each frequency slice, for each windows interval, the encoder "decides how important the content is and how good a copy it needs to make to avoid audibly noticeable degradation". However, at least some quality is lost in each, and that loss of quality WILL COMPOUND. (The file won't get smaller because some of the data being encoded the second time will be quantization errors caused by the first encode process...... so the amount of "useful data" won't change much, but the "useless data" will be quite different, and most of that (but not all of it) will be discarded when you encode it the second time.



bigshot said:


> Think about it logically... If you are ripping a file from a CD and the compressed version sounds completely identical to the original CD, then for the purposes of listening to it, it's perfect. You won't need better sounding than identical in the future. It can't sound any better than the CD it's ripped from, and your ears aren't going to hear any better 20 years from now. It's a good idea to bump the bitrate up a notch just to cover some particularly difficult to compress sort of sound. But I can tell you that I have ripped over 10,000 CDs to AAC 256 VBR and I have never run across an artifact in any of my rips.
> 
> I've experimented a lot with compression codecs and there's something most people don't know about how the way they work. If you compress a song and it throws out inaudible information to make the file smaller; if you run it through the compression again, it has already thrown out all the inaudible information, so it makes no change. You can compress a file over and over and it doesn't degrade. Once it's been compressed, it won't compress any further unless you change the data rate. This means, if the file is audibly transparent and you need to transcode it to some other new format, as long as the new format is also audibly transparent, there will be no degradation in the sound quality.
> 
> ...


----------



## bigshot (Nov 27, 2017)

There's more than one MP3. Frauenhofer MP3 is pretty primitive compared to LAME, and MP4 like AAC is a step beyond both of them. I'm mainly speaking about AAC. I haven't done generation tests with the older codecs, only AAC. Generation loss is minimal to the point of insignificance with AAC 320 VBR with a normal number of generations (under 10). For the purposes of listening to music in the home, high bitrate AAC is pretty much the same as lossless. You'd have to come up with a pretty extreme circumstance to find a difference between them.

But unless I'm mistaken, you're speaking entirely in theory about this. You haven't actually tried it yourself. It's easy to find out for yourself. Just take a great sounding CD. Rip it to WAV and normalize it down to 85%. Convert it to AAC 320 VBR then back to WAV. Rinse and repeat to 10 generations, then compare your original WAV to the 10th generation AAC. Then you'll know for sure what the difference is. I've done this. I know. The results surprised me because I was expecting significant degradation like you are.

The great thing about lossy is that it's so easy to do listening tests to determine exactly what the thresholds are and how it works, but I guess most people don't bother. They go with what seems logical to them or a gut feeling instead of making the minimal effort to find out for themselves. It isn't like a xerox. A xerox throws out stuff you can clearly see. High bitrate lossy is audibly transparent. It throws out things you can't hear. Perhaps after 100 generations it would be audible. But most people will never transcode more than 2 or 3 times in their lifetimes. For the purposes it was designed for it works better than your theories say it does.


----------



## pinnahertz

You say this:


KeithEmo said:


> I can't speak for ALL lossy CODECs, but I can tell you that you're wrong about MP3.


How can he be "wrong" when he's speaking about his observations?  And how can anything you've said about mp3 apply to his being "wrong" when he didn't use that codec (he used AAC)?

Then you say:


KeithEmo said:


> When you initially compress a file, it does indeed do its best to "throw away unnecessary information". However, the process is not as simple as identifying what doesn't matter and deleting it, and re-encoding something that has already been encoded WILL produce "generational degradation". Basically, the encoder does NOT "just throw away the information you won't miss". What it does is to divide the audio signal into a bunch of frequency bands, each for a short block of time, decide how much "important" information is contained in each, and then divide its "quality/priority" depending on how important the information is that's contained in each. It may discard some information entirely, while other information is simply encoded at lower quality. Each "section" of the information is encoded at the least quality for which "you won't notice the difference" - and the decision of what that will be depends on psychoacoustic properties like masking. Therefore, the majority of information in an MP3 encoded file is neither full quality, nor minimum quality, but somewhere in-between - encoded at "just high enough quality" that you won't notice the loss.


...which is, if I'm allowed to be as binary as you, an inaccurate explanation of how MPEG coding works. 


KeithEmo said:


> HOWEVER, in no part of this process is there any sort of specific identification of how each individual sound was treated, and so no way to ensure that the process won't be applied repeatedly to a given section. Therefore, if a given frequency/time slice has been encoded with a lot of quantization error (because it was deemed to contain "unimportant content"), and you re-encode it, it will AGAIN be encoded with a lot of quantization error - and those errors will compound. If you take a file that's been encoded at 128k VBR MP3 and re-encode it at the same settings, either as is or after converting it back into a WAV file, you will *probably* not lose much ADDITIONAL quality (because pretty much the same decisions are being made), however the encoder will NOT "simply leave it as is" either. It will be re-encoded, AGAIN with encoding that introduces further quantization errors, so the total sum of the errors will increase. (The result is that areas which are considered unimportant will get significantly worse when you re-encode them, because they will have been encoded at poor quality twice instead of once. Areas which are deemed more important will suffer less degradation, because they will have been encoded twice, but both at a higher quality setting, which causes less loss of quality. You may argue that, since those areas were unimportant to begin with, the additional loss of quality won't matter - but it is there - and the overall quality will decrease with repeated generations.)


Note the emphasized word above.

OK, Mr. Black and White, now please explain (with minimal verbosity...if possible) how Bigshot was able to repeatedly recode without observable quality loss. 


KeithEmo said:


> With lossy compression, the analogy of a photocopier is quite valid,...


NO, it's not.  NO visual analogies...again...please!!!  Again, if I'm permitted to be as binary as you are, your visual analogies...all of them...are wrong.  Copiers don't use perceptual coding!!!! They are just lossy.  PLEASE let's not spend time trying to defend your pointless analogies.


----------



## KeithEmo

OK.... blunt statements.... blunt answers.

He claimed that:
"I've experimented a lot with compression codecs and there's something most people don't know about how the way they work. If you compress a song and it throws out inaudible information to make the file smaller; if you run it through the compression again, it has already thrown out all the inaudible information, so it makes no change. You can compress a file over and over and it doesn't degrade. Once it's been compressed, it won't compress any further unless you change the data rate."

I see specific claims there:
1) "it makes no change"
2) "You can compress a file over and over and it doesn't degrade"

To me, the intent of the statement is crystal clear...... He is claiming that, if you run an encoded file through the encoder again, the file will remain the same, and the encoder DOESN'T CHANGE ANYTHING. (The presumption being that, since the file has already been encoded, and any necessary changes have already been performed, there's nothing else to change.)

This is absolutely NOT true for MP3 - because, EVERY time you run an MP3 encoder, it deconstructs the audio file, then re-constructs it, after determining the maximum amount of quantization error that will be "audibly undetectable". And, each time this process is repeated, with no knowledge of the previous encoding, and no option to avoid encoding altogether because the file is already "fully optimized", the quantization errors will accumulate.... resulting in more errors overall. The file will not get smaller, but it will change (unless the encoder is smart enough to discard the newly encoded file when it discovers it is not smaller.... which some may be.)

I don't know the details of how AAC works - but I have seen it claimed that it works "similarly to MP3".....

You can find the details about how MP3 works (without the heavy math) here:
https://arstechnica.com/features/2007/10/the-audiofile-understanding-mp3-compression/

You will note that there is no part of the process where processing is avoided because it won't deliver an improvement.



pinnahertz said:


> You say this:
> How can he be "wrong" when he's speaking about his observations?  And how can anything you've said about mp3 apply to his being "wrong" when he didn't use that codec (he used AAC)?
> 
> Then you say:
> ...





pinnahertz said:


> You say this:
> How can he be "wrong" when he's speaking about his observations?  And how can anything you've said about mp3 apply to his being "wrong" when he didn't use that codec (he used AAC)?
> 
> Then you say:
> ...


----------



## pinnahertz

KeithEmo said:


> OK.... blunt statements.... blunt answers.
> 
> He claimed that:
> "I've experimented a lot with compression codecs and there's something most people don't know about how the way they work. If you compress a song and it throws out inaudible information to make the file smaller; if you run it through the compression again, it has already thrown out all the inaudible information, so it makes no change. You can compress a file over and over and it doesn't degrade. Once it's been compressed, it won't compress any further unless you change the data rate."
> ...


1. "audibly"
2. "audibly"


KeithEmo said:


> To me, the intent of the statement is crystal clear...... He is claiming that, if you run an encoded file through the encoder again, the file will remain the same, and the encoder DOESN'T CHANGE ANYTHING. (The presumption being that, since the file has already been encoded, and any necessary changes have already been performed, there's nothing else to change.)


No, you've applied binary filtering again.  That's NOT what he's saying, that's what you're reading.


KeithEmo said:


> This is absolutely NOT true for MP3 - because, EVERY time you run an MP3 encoder, it deconstructs the audio file, then re-constructs it, after determining the maximum amount of quantization error that will be "audibly undetectable". And, each time this process is repeated, with no knowledge of the previous encoding, and no option to avoid encoding altogether because the file is already "fully optimized", the quantization errors will accumulate.... resulting in more errors overall.


Your argument is theoretical.  You don't supply any actual proof of anything.  And your explanations are inaccurate anyway.


KeithEmo said:


> The file will not get smaller, but it will change (unless the encoder is smart enough to discard the newly encoded file when it discovers it is not smaller.... which some may be.)


Again, theoretical.  Have you tried this?  IF it doesn't get smaller, what's being changed?  I'm not saying you're wrong, I'm saying you have supplied no proof.  Regardless, that's not what Bigshot is getting at anyway.  He's saying it's not audibly changed by successive recodes...up to a point.  He's making a claim based on personal observation that includes a gray area...that, I'm not surprised, misses you entirely. 


KeithEmo said:


> I don't know the details of how AAC works - but I have seen it claimed that it works "similarly to MP3".....


Yeah, well if you don't know how AAC works but state, definitively, that someone making observations about AAC is wrong, guess who's actually wrong?


KeithEmo said:


> You can find the details about how MP3 works (without the heavy math) here:


Don't care, we're not discussing mp3 (only or even specifically) anyway...YOU are...we aren't.


----------



## bigshot

Just give it a try and recompress a maxed out AAC file. I think you'll be surprised at how well it recompresses. Trying it is better than arguing about it!


----------



## KeithEmo

I've read those sentences pretty carefully..... and I can't seem to find the word "audibly" in either claim.
I have no problem with saying that "it won't make an audible difference.... up to a point".
However, saying that "it makes no difference" is incorrect - since it does in fact make a difference, which is cumulative, and eventually becomes audible.
(And the truth makes it a bad idea to multiply encode audio, even though, in some instances, it might not produce audible issues..... because in other instances it might.)

The idea that "nothing is being changed because the file isn't getting smaller" is simply logically invalid.
A significant reduction is size is one external indicator of a successful compression.......
But lack of a change in size suggests nothing at all - besides the obvious.

Incidentally, I know the single most important piece of information about ALL LOSSY ENCODERS....... they're LOSSY.
A LOSSLESS CODEC must retain all of the original content without alteration.
This is a relatively simple task, and one which is easily proven to be true (we can do a bit compare between the copy and the original).
A LOSSY CODEC admits that it is going to alter the information, but asks us to believe that "the change will be small enough that we'll never notice".
Clearly, claiming that something will be damaged, but "we won't notice the damage" is the exceptional claim requiring the exceptional proof.
Furthermore, while I recall many instructions and manufacturer's recommendations that lossy compression should be applied as the last step before distribution....
I don't recall EVER reading a claim that "it's OK to repeatedly apply lossy compression to a file".
In fact, most SPECIFICALLY suggest that, if you wish to re-encode a file with different settings, you start with a fresh LOSSLESS ORIGINAL.

I don't doubt that certain lossy CODECs, when applied to certain content, with certain settings, produce minimal additional changes if applied a second time.
However, even though you may luck out and suffer no ill effects, applying multiple iterations of lossy compression is surely about as far from "best practices" as you can get.



pinnahertz said:


> 1. "audibly"
> 2. "audibly"
> No, you've applied binary filtering again.  That's NOT what he's saying, that's what you're reading.
> Your argument is theoretical.  You don't supply any actual proof of anything.  And your explanations are inaccurate anyway.
> ...





pinnahertz said:


> 1. "audibly"
> 2. "audibly"
> No, you've applied binary filtering again.  That's NOT what he's saying, that's what you're reading.
> Your argument is theoretical.  You don't supply any actual proof of anything.  And your explanations are inaccurate anyway.
> ...


----------



## KeithEmo

I don't doubt it..... 

However, I would expect a slow drift down in quality over multiple iterations.

At each iteration the compression algorithm is going to decide how much "quality priority" to give each frequency and time slice.
Those with lowest priority will simply be discarded - which takes them out of the picture.
Those with highest priority will be encoded well, and so should suffer little change.
However, I expect those in the middle, which are allocated just enough priority that they suffer significant quantization errors that fall just below the threshold of audibility, will gradually deteriorate.
(I would expect each iteration to be audibly identical to the previous ones...... but for drift between between later iterations and the original to become more noticeable as the iteration count increases.)



bigshot said:


> Just give it a try and recompress a maxed out AAC file. I think you'll be surprised at how well it recompresses. Trying it is better than arguing about it!


----------



## pinnahertz

KeithEmo said:


> I've read those sentences pretty carefully..... and I can't seem to find the word "audibly" in either claim.


You are far to literal.  It's right there between the lines where most of us can see it.  If not, Bigshot will correct me. 


KeithEmo said:


> I have no problem with saying that "it won't make an audible difference.... up to a point".
> However, saying that "it makes no difference" is incorrect - since it does in fact make a difference, which is cumulative, and eventually becomes audible.
> (And the truth makes it a bad idea to multiply encode audio, even though, in some instances, it might not produce audible issues..... because in other instances it might.)


There is nothing anyone could post that covers gray areas that will ever satisfy you, though. 


KeithEmo said:


> The idea that "nothing is being changed because the file isn't getting smaller" is simply logically invalid.


What logic is invalid?  If a codec's purpose by design is to result in less data, and after a second pass through it the amount of data didn't change, then what did?


KeithEmo said:


> A significant reduction is size is one external indicator of a successful compression.......


Well lets see now: if reduction in size is the goal...make that the ONLY goal...of a lossy codec, then what is the other indicator of successful bit-rate reduction?


KeithEmo said:


> But lack of a change in size suggests nothing at all - besides the obvious.


Now THATs what I would call logically invalid!


KeithEmo said:


> Incidentally, I know the single most important piece of information about ALL LOSSY ENCODERS....... they're LOSSY.
> A LOSSLESS CODEC must retain all of the original content without alteration.
> This is a relatively simple task, and one which is easily proven to be true (we can do a bit compare between the copy and the original).
> A LOSSY CODEC admits that it is going to alter the information, but asks us to believe that "the change will be small enough that we'll never notice".
> ...


But we are not talking about lossless codecs now, are we?  In fact, Bigshot didn't mention just any lossy codec, he was specific: AAC.  You can expound on lossless codecs all you want, that is a tangential discussion.


KeithEmo said:


> I don't doubt that certain lossy CODECs, when applied to certain content, with certain settings, produce minimal additional changes if applied a second time.
> However, even though you may luck out and suffer no ill effects, applying multiple iterations of lossy compression is surely about as far from "best practices" as you can get.


Obviously!  And nobody has argued against that either.  Your binary filter is working full tilt again.  The example was specific: multiple passes through a high-rate AAC codec.  Your binary filter has blocked the fact that codecs are adjustable as to the target bit rate, and that codecs are not all the same, and that the number of recodes through a single specific codec may actually not result in audible changes, or indeed, any changes.  Codecs are not binary in their action, they are not just on or off.  That means results will vary too. 

Your example is polarized, and silly: an additional pass produces minimal changes?  Really?  How's that codec set?  I can show you a second pass that is totally destructive, and one that is transparent...oh, and the grays in between them.  

"...surely about as far from "best practices" as you can get."???? Why are you even mentioning this?  How insulting! Do you actually think everyone here are so stupid and unaware that we'd consider a dozen or so recodes part of "best practices"?  Get real, and try to have a little respect.  Some of us are audio professionals, even!


----------



## pinnahertz

KeithEmo said:


> I don't doubt it.....
> 
> However, I would expect a slow drift down in quality over multiple iterations.
> 
> ...


So what you're doing here is a flat-out refusal to even try it, instead expounding on your personal opinions of what will happen.  Is that correct?  And all of this while still admitting to not be familiar with the codec in question.  

Wow!


----------



## bigshot (Nov 27, 2017)

KeithEmo said:


> I've read those sentences pretty carefully..... and I can't seem to find the word "audibly" in either claim.



"Audibly" is a given when talking about lossy. Everyone knows that lossy is different than lossless. There just isn't an audible difference if the bit rate is high enough and if it hasn't been transcoded too many times. Personally, "audibly" is what I worry about. I'm generally a happy and satisfied person. I don't feel anything lacking in my feelings of security that I have to fill with bitrate. As long as it sounds the same, for my purposes it *is* the same.

I would have assumed that re-encoding AAC 320 VBR over and over would create generation loss. But I tried it and found out that you have to re-encode a whole lot of times to create any audible difference. I was surprised to find that out. There's no point arguing about something that is simple to try for yourself. AAC is a damn good codec. For my purposes, it's interchangeable with lossless. But it's a lot smaller. I use it exclusively.


----------



## KeithEmo (Nov 28, 2017)

I'm not sure that I agree with your primary assertion: "That 'everyone knows the difference between lossy and lossless'". From my experience, a lot of people seem NOT to know the difference. 
That's why I tend to argue against statements which might conceivable mislead people who actually don't know into thinking that they are the same.

I think we're sort of discussing two different things here.

As far as I'm concerned, in terms of the technology, there's nothing to try.
Whenever I listen to a piece of music I've never heard before, I don't know what it sounds like, so I'm relying on my system to let me find out - by playing it accurately.
We all KNOW that lossy CODECs alter the information; they "say so right on the package"; so there's nothing to question.
To me the choice between lossless and lossy would be like the choice between buying a GPS that at least claims that it will take me to the exact right address.....
And one that is advertised as: "It never takes you to the exact right place; in fact it specifically avoids taking you to the exact right place; but it will get you close enough that you won't mind".
Personally, rather than wonder how big the error is, I'd rather just buy the one that takes me to the right place.
(And, in order to convince me to deliberately take the inaccurate version, they're going to have to offer a pretty compelling reason....... and, to me, smaller file size just isn't a compelling reason.)
That's why I personally am never going to try or use lossy compression..... because, to me, it has at least potential serious drawbacks, and no significant benefits.

However, as far as my statement about cumulative errors summing..... well, that's just math.
If you were to ask me "what does 2 + 2 = " I wouldn't go out and buy a bunch of marbles, put two in my left hand, two in my right hand, then put them together and confirm that I now have four on the table.
I would use math and logic to figure out what to expect..... based on how the process works.
Now, on every lossy audio CODEC I've ever read the description of, there is a design intent to ensure that the first generation copy will be "audibly identical to the original" - at least as much as possible.
However, I've never seen any that claim that there is any mechanism included that will prevent iterative changes from summing to a value greater than a single change.

If you were to tell me that you were going to walk for one block in a random direction from your home.... we can both agree that you will end up one block from home.
However, if you were to tell me that you're going to walk for one block in a random direction, then, starting from there, walk for one block in a random direction, and repeat the process five times.....
MATH tells me that, at least some of the time, you will end up more than one block from home.
(Remember that we've included nothing in the process to ensure that this doesn't happen.)

However, I don't dispute that running a lossy CODEC multiple times on the same content MAY, IN SOME CASES, still result in a final copy that is audibly indistinguishable from the original.
And neither do I dispute that, in a specific situation, and with a specific CODEC, a certian person may have had that experience.
However, I do oppose making it as a general statement, when the science suggests that we're looking at the exception and not the general case.

(And, yes, if someone were to suggest that "a cup of Drano is a great cure for a stomach ache" I would probably argue against that too...... WITHOUT trying it.)



pinnahertz said:


> So what you're doing here is a flat-out refusal to even try it, instead expounding on your personal opinions of what will happen.  Is that correct?  And all of this while still admitting to not be familiar with the codec in question.
> 
> Wow!


----------



## KeithEmo

1)
Some things are grey - and some are black and white - but, in many cases, which applies to something depends on your point of view.
For example...... in terms of DATA, the question is clear black and white, either data is retained accurately or not.
I can do a bit compare - and the result will be a simple black and white pass or fail.
(Personally, I like black and white, I can run a checksum on my music library and KNOW, with absolute certainty, that it's exactly the same as yesterday..... and nothing has been changed.)

In terms of technology, lossy CODECs aren't "a grey area" at all.
Lossy CODECs discard information.... this is a given.
Likewise, either the result is or is not audibly identical to the original.... and that's also black and white.
The only grey area I see would be with lower quality CODECs.... where we concede that the losses are audible, but there is a question of opinion about whether the loss is justified.
(The "grey" arises because it's a matter of opinion whether the loss is significant or not.... and whether we consider the cost to be justified by the benefits.) 
There is also an area of UNCERTAINTY..... it may turn out that, on 95% of all files processed, the result is perfect, but on 5% it is not......
(If so, we may still claim - in black and white - that "on 95% of a random selection of processed files nobody can hear the difference".)

2)
My problem with your "size assertion" is simply that it isn't true.
Your assumption that "if the file is the same size then it contains the same amount of data" is entirely incorrect.
It is in fact quite simple to make a file larger or smaller without changing the amount of data it contains; or to add or remove data without changing the size of the file.

When we initially run the CODEC, we can assume that, if the file got smaller, then information was discarded.......but that is a VERY special case.
There are several unstated assumptions on which that assertion is based...... and lots of exceptions.
For example, I can compress a file using FLAC, and the file will get smaller, but NO information will have been discarded.
Likewise, any process that makes the information LESS CORRECT, but does so in a way that doesn't result in LESS information, may leave the file the same size or even make it larger.

Your base assertion that "reduction in size is the goal...make that the ONLY goal...of a lossy codec" is incorrect.
The goal of a lossy CODEC is to reduce the size of the file while avoiding altering the contents in an audible way.... 
And the indicator of success would be that the file has gotten smaller but remains audibly the same.

However, there are several possible "indicators of failure"........
- if the file got larger that would be a definite fail
- if the file sounded audibly different that would be a definite fail
- if the file sounded the same and remained the same size that would be a sort of null result (a waste of time but no harm done).

There is also the potential for "generational failure"...... which is a concept that is applied deliberately in certain copy protection schemes (including the original CD-R music protection scheme).
In "generational failure" the copy is functionally the same as the original - but only in CERTAIN regards - while being very different in other ways.
In one such copy protection scheme, the user was allowed to make a copy of an "original".
However, even though the copy was AUDIBLY identical to the original, the user was unable to make a copy of that copy.
Therefore, while the copy was AUDIBLY identical, it was inferior in OTHER WAYS.
(For a user whose goal was strictly to listen, the copy was 100% perfect; for a user who wished to copy it, it was "broken".)



pinnahertz said:


> You are far to literal.  It's right there between the lines where most of us can see it.  If not, Bigshot will correct me.
> There is nothing anyone could post that covers gray areas that will ever satisfy you, though.
> What logic is invalid?  If a codec's purpose by design is to result in less data, and after a second pass through it the amount of data didn't change, then what did?
> Well lets see now: if reduction in size is the goal...make that the ONLY goal...of a lossy codec, then what is the other indicator of successful bit-rate reduction?
> ...


----------



## KeithEmo (Nov 28, 2017)

That is useful to know...... and I suspect part of that may be due to the fact that AAC is a proprietary standard.
As a result it is more tightly controlled and far more consistent.

One of the strengths and weaknesses of MP3 has always been that the encoding process is entirely open.
All MP3 decoders are supposed to follow a given standard - so, if you play the same MP3 file on different players, you should get the same exact result.
However, the MP3 ENCODER is "open"..... the only real requirement is that the file it outputs will play in a standard decoder (which does set lots of practical constraints).
On the plus side, this encourages programmers to think up new and better encoding methods...... and new and better ways to apply perceptual coding.
However, on the downside, it means that two MP3 encoders may produce different results, from the same input file, even if the same exact settings are applied.
So it is literally true that a given MP3 encoder may produce better results with certain content and poorer results with others.
The upside of this is that it encourages competition and product improvements; the downside is that you never know exactly what you're getting when you receive an "MP3" file.
(I recall one product from years ago that actually encoded each file tree times, using three different encoders, then prompted the user to "pick the version that sounded better" - individually for each song.)

I also do apologize for seeming to be so "pedantic" about the subject of lossy encoders.
However, far from what some folks seem to think, it has been my experience that many people DO NOT understand the difference between lossless and lossy encoders.
For example, I frequently encounter "audio CDs" made from iTunes..... whose owners seem to have no idea that, even though the CD itself is lossless, since their SOURCE was a lossy AAC file, the CD they have is NOT identical to a real purchased CD copy.
I haven't looked lately, but iTunes used to offer a very misleading "make a CD" option in the menu - with no warning that the CD you make will be different than a CD you might purchase.
I see this as a dangerous side effect of "too many grey areas" - which is why I tend to view such grey areas as a significant problem.
I have no problem whatsoever when people use AAC or MP3...... but, when someone presents me with a CD, which he says is "just a CD", but then I find out it was sourced from an MP3 or AAC file, THEN I have a major problem.
Unfortunately, from my experience, an awful lot of people do NOT know enough to make that distinction.

Many non-technical people I talk to seem to understand that there are different ways of encoding a file - but don't understand the lossy/lossless distinction....
Likewise, many people apparently don't understand that JPG is a lossy image CODEC, and that the encoding used on Blu-Ray discs is also lossy (just less so that the one used by their favorite streaming service).

It's also worth noting that, because each MP3 encoder is slightly different, generational copy issues will vary depending on which encoder is used.
(For example, encoding a file a second time using the same encoder as was originally used is likely to produce less loss of quality - especially if the same settings were used. While encoding a file a second time on a different encoder,
or using different settings, is more likely to result in different information being discarded each time, with an overall higher loss of accuracy.)
It makes sense that this would be less important with AAC - because the encoders themselves, being proprietary, should be consistent.



bigshot said:


> "Audibly" is a given when talking about lossy. Everyone knows that lossy is different than lossless. There just isn't an audible difference if the bit rate is high enough and if it hasn't been transcoded too many times. Personally, "audibly" is what I worry about. I'm generally a happy and satisfied person. I don't feel anything lacking in my feelings of security that I have to fill with bitrate. As long as it sounds the same, for my purposes it *is* the same.
> 
> I would have assumed that re-encoding AAC 320 VBR over and over would create generation loss. But I tried it and found out that you have to re-encode a whole lot of times to create any audible difference. I was surprised to find that out. There's no point arguing about something that is simple to try for yourself. AAC is a damn good codec. For my purposes, it's interchangeable with lossless. But it's a lot smaller. I use it exclusively.


----------



## bigshot (Nov 28, 2017)

Some of your info is out of date there... AAC isn't proprietary. It's been an open standard for years now. Open doesn't mean that people can tinker with the encoding process and make their own version of AAC. Every current AAC encoder works exactly the same. The encoding and decoding is performed by stock cut and paste burned right into the chips of the DAC. There's no difference in quality. It's a standard, even if it is an open standard. And it doesn't work the same as an MP3... it's an MP4 which is a totally different and more advanced compression scheme. AAC is audibly transparent, which means to human ears, it's identical to the original. No loss in fidelity. And generation loss is also transparent for more generations than anyone would be likely to need to re-encode. From a practical standpoint it's all positives and no drawbacks.

I think you're projecting a bit on other people about lossy. Everyone knows that it throws out data. It says so right there in the name "lossy". They just don't care because it's inaudible information. I don't care about things I can't hear. I never have. I focus on improving things I *can* hear. That gets me a lot further when it comes to sound quality, because the best sounding systems sound the best because of the way they present the core audible frequencies. What you hear is what matters. A truly great system will sound just as good with high rate lossy as they do with lossless. The only argument I've heard in favor of lossless is that it assuages people's OCD. I can totally understand that. If I had anxiety over bitrates, I'd want to make sure my file sizes were portly too I suppose.

Lossy... lossless... none of that matters. What matters is how the music sounds. Audio reproduction has advanced to the point where bitrates don't matter. They're nothing more than advertising points, especially in blu-ray where absurd bitrates are touted as being "necessary". The truth is that redbook is plenty and high bitrate AAC sounds exactly like lossless. So if it doesn't make you neurotic to be throwing out unnecessary bits, then it makes sense to use it. My whole music library is AAC 256 VBR. I've ripped tens of thousands of CDs to the music server and boxed the discs up in the garage. It all fits on one disc drive. It's sorted automatically by iTunes. And it sounds perfect on my best equipment. For me, that's all a win with no loss.

The reason I share my listening test with people is so they can find out for themselves where their line of transparency lies. That is very important. If you know that, then you don't have to worry about lossy throwing out something important. You know that above a certain rate, it's identical to lossless. That should be a comforting thought.


----------



## 71 dB

KeithEmo said:


> However, far from what some folks seem to think, it has been my experience that many people DO NOT understand the difference between lossless and lossy encoders.



How could they? I have an education that makes the difference trivial to me, but most people have other kind of education or even lack higher education altogether. Lossy encoders remove perceptual redundancy while lossless encoders remove information redundancy. One needs to understand these concepts in order to understand the difference. This is how you can explain them in layman's terms to someone who doesn't understand them:

If you have five 10 dollar notes and one 1 dollar note in your wallet, instead of telling you have $10, $10, $10, $10, $10 and $1 you use lossless coding and say you have 5 * $10 + $1 because that's a much shorter expression and no information is lost. You can also use lossy coding: You give $1 to a hobo on the street and tell you have $50. That's very short, but you lost $1 and also the information about what kind of notes you have.


----------



## bigshot

What does the hobo have? Is he stuck with ear buds or something?


----------



## bigshot

I think a better analogy is...

You tell your best friend that you're going to cut school and go fishing. He is afraid he's going to get in trouble, so he goes to school while you head for the creek. You fish all day and catch some beautiful trout. You wrap them up like they came from the market, put them on your front doorstep and ring and run. Your mom finds them and thinks they're a delivery from the market. That night neither you nor your friend who went to school get in trouble because your mom never found out. But you get pan fried brook trout for dinner, and he gets leftover tuna casserole.

Aesop says: What mom doesn't know won't hurt you... and it might actually be better than doing it by the book!


----------



## 71 dB

bigshot said:


> My whole music library is AAC 256 VBR. I've ripped tens of thousands of CDs to the music server and boxed the discs up in the garage. It all fits on one disc drive. It's sorted automatically by iTunes. And it sounds perfect on my best equipment. For me, that's all a win with no loss.



If it takes 6 minutes to rip one CD, you have spend thousands of hours on that hobby.


----------



## KeithEmo

Perhaps "proprietary" isn't the correct word - although many people I know would consider it to be correct in this context (someone else owns it and you aren't allowed to change it).
As you say, "people (can't) tinker with the encoding process and make their own version of AAC" - whereas you _CAN_ do exactly that with MP3.
AAC currently offers lots of options - but I believe that the actual processing used by each is spelled out in the standard.....
There is essentially an MP3 _DECODER_ standard.... but there is no MP3 _ENCODER_ standard per-se (or you may prefer to view it as having a huge number of possible variations). 
Your MP3 encoder can discard whatever it likes, using whatever version of perceptual encoding you like; as long as it plays on an MP3 decoder, it is "a valid MP3 file".

Obviously we each deal with very different segments of the population.
The majority of people I've spoken to about the subject don't even know that MP3 and AAC are lossy CODECs (and many don't even notice the type of file they're playing).
This applies to a significant proportion of the customers I speak to officially at Emotiva as well as the majority of my (non-audiophile) personal friends.
For example, most of the people I speak to who use ITunes don't even know that it used AAC; they simply "RIP their CDs with iTunes" and have no idea what it's set to.
(Unfortunately, these people are also quite unlikely to run audibility tests either..... they simply leave everything set to the defaults - or blindly follow the instructions of someone they trust.)



bigshot said:


> Some of your info is out of date there... AAC isn't proprietary. It's been an open standard for years now. Open doesn't mean that people can tinker with the encoding process and make their own version of AAC. Every current AAC encoder works exactly the same. The encoding and decoding is performed by stock cut and paste burned right into the chips of the DAC. There's no difference in quality. It's a standard, even if it is an open standard. And it doesn't work the same as an MP3... it's an MP4 which is a totally different and more advanced compression scheme. AAC is audibly transparent, which means to human ears, it's identical to the original. No loss in fidelity. And generation loss is also transparent for more generations than anyone would be likely to need to re-encode. From a practical standpoint it's all positives and no drawbacks.
> 
> I think you're projecting a bit on other people about lossy. Everyone knows that it throws out data. It says so right there in the name "lossy". They just don't care because it's inaudible information. I don't care about things I can't hear. I never have. I focus on improving things I *can* hear. That gets me a lot further when it comes to sound quality, because the best sounding systems sound the best because of the way they present the core audible frequencies. What you hear is what matters. A truly great system will sound just as good with high rate lossy as they do with lossless. The only argument I've heard in favor of lossless is that it assuages people's OCD. I can totally understand that. If I had anxiety over bitrates, I'd want to make sure my file sizes were portly too I suppose.
> 
> ...


----------



## 71 dB

bigshot said:


> What does the hobo have?


Maybe the $1 gets him a hamburger to eat?


----------



## bigshot

71 dB said:


> If it takes 6 minutes to rip one CD, you have spend thousands of hours on that hobby.



I've fed in CDs as I work at my computer. It rips in the background.



KeithEmo said:


> As you say, "people (can't) tinker with the encoding process and make their own version of AAC" - whereas you _CAN_ do exactly that with MP3.



I don't manufacture equipment, but my understanding is that proprietary means that you can only include the technology if you get permission from the owner. An open standard means that you can use the technology without permission as long as you pay a mechanical license. There really are just two types of MP3- Frauenhofer and LAME. LAME was a separate standard designed to optimize the MP3 standard. It has the same MP3 suffix but it's a different codec. In the old days there were DACs that implemented MP3 decoding poorly, but that was more of a design error than it was intentional.



KeithEmo said:


> most of the people I speak to who use ITunes don't even know that it used AAC; they simply "RIP their CDs with iTunes" and have no idea what it's set to.
> (Unfortunately, these people are also quite unlikely to run audibility tests either..... they simply leave everything set to the defaults - or blindly follow the instructions of someone they trust.)



You might find it surprising but a lot of people in this forum talk about the audibility of lossy artifacting without ever doing an audibility test too! They just assume that because the name says "lossy" it must sound inferior to "lossless". Those people are just as misinformed as the ones who don't know how iTunes works. However, the default encoding in iTunes and the iTunes store is AAC 256 VBR which is audibly transparent for just about everyone, so Apple has dummy proofed the process for them. The people who use lossless without understanding just get stuck not having as much music to choose from on their phone or DAP. No one is dummy proofing for them.


----------



## JaeYoon

bigshot said:


> I've fed in CDs as I work at my computer. It rips in the background.
> 
> 
> 
> ...


Yeah 256 VBR is a very good idea for me.
I don't want to buy a 400 GB sd card for $249.

I got a 256 GB sd card. But like, my entire ripped library is almost that size.
I also have another library that is bought off music stores that is around an extra 80 gbs. That won't fit together so ripping to lossy for my SD card is a perfect choice to make it all fit.


----------



## 71 dB

bigshot said:


> I've fed in CDs as I work at my computer. It rips in the background.



You work from home? That's convenient…


----------



## bigshot

I work all the time. I have a job at a studio during the day and weekends and nights I operate a non-profit digital archive out of my home.


----------



## pinnahertz

KeithEmo said:


> I'm not sure that I agree with your primary assertion: "That 'everyone knows the difference between lossy and lossless'". From my experience, a lot of people seem NOT to know the difference.


I don't agree with that assertion either, because that is not MY assertion.  If you going to quote me, please do so exactly, and stop making things up! 


KeithEmo said:


> That's why I tend to argue against statements which might conceivable mislead people who actually don't know into thinking that they are the same.


That's a fine goal...but you actually do the opposite quite frequently.


KeithEmo said:


> I think we're sort of discussing two different things here.
> 
> As far as I'm concerned, in terms of the technology, there's nothing to try.
> Whenever I listen to a piece of music I've never heard before, I don't know what it sounds like, so I'm relying on my system to let me find out - by playing it accurately.
> *We all KNOW that lossy CODECs alter the information; they "say so right on the package"; so there's nothing to question.*


As I said above, I disagree with that assertion!  "WE" do NOT know that.


KeithEmo said:


> To me the choice between lossless and lossy would be like the choice between buying a GPS that at least claims that it will take me to the exact right address.....
> And one that is advertised as: "It never takes you to the exact right place; in fact it specifically avoids taking you to the exact right place; but it will get you close enough that you won't mind".
> Personally, rather than wonder how big the error is, I'd rather just buy the one that takes me to the right place.
> (And, in order to convince me to deliberately take the inaccurate version, they're going to have to offer a pretty compelling reason....... and, to me, smaller file size just isn't a compelling reason.)
> That's why I personally am never going to try or use lossy compression..... because, to me, it has at least potential serious drawbacks, and no significant benefits.


OK, fine...but you already do use lossy compression, like it, know it, or not.  It has a place, it has an application.  Please realize that something is not categorically "bad" just because it can be misapplied.
The GPS example fails: that's not the same as a lossy codec based on perceptual coding.


KeithEmo said:


> However, as far as my statement about cumulative errors summing..... well, that's just math.
> If you were to ask me "what does 2 + 2 = " I wouldn't go out and buy a bunch of marbles, put two in my left hand, two in my right hand, then put them together and confirm that I now have four on the table.
> I would use math and logic to figure out what to expect..... based on how the process works.


Again, your analogy fails.  That's not the same as a lossy codec based on perceptual coding, that's just brute-force loss.


KeithEmo said:


> Now, on every lossy audio CODEC I've ever read the description of, there is a design intent to ensure that the first generation copy will be "audibly identical to the original" - at least as much as possible.
> However, I've never seen any that claim that there is any mechanism included that will prevent iterative changes from summing to a value greater than a single change.


Yes!  What you're saying is that lossy codecs can achieve their goal of transparency and not use all the original data.  Mission accomplished. 


KeithEmo said:


> If you were to tell me that you were going to walk for one block in a random direction from your home.... we can both agree that you will end up one block from home.
> However, if you were to tell me that you're going to walk for one block in a random direction, then, starting from there, walk for one block in a random direction, and repeat the process five times.....
> MATH tells me that, at least some of the time, you will end up more than one block from home.
> (Remember that we've included nothing in the process to ensure that this doesn't happen.)


Wrong analogy again!  That's just information loss, and that is absolutely NOT the same as a lossy codec based on perceptual coding!


KeithEmo said:


> However, I don't dispute that running a lossy CODEC multiple times on the same content MAY, IN SOME CASES, still result in a final copy that is audibly indistinguishable from the original.
> And neither do I dispute that, in a specific situation, and with a specific CODEC, a certian person may have had that experience.


So we did change your mind, then.


KeithEmo said:


> However, I do oppose making it as a general statement, when the science suggests that we're looking at the exception and not the general case.


It wasn't made as a general statement, the conditions were specific.  Go read Bigshot's post again.  And you are misapplying science.  Again.


KeithEmo said:


> (And, yes, if someone were to suggest that "a cup of Drano is a great cure for a stomach ache" I would probably argue against that too...... WITHOUT trying it.)


I cannot possibly imagine what THAT analogy is all about.  Stupidity?


----------



## pinnahertz

KeithEmo said:


> 1)
> Some things are grey - and some are black and white - but, in many cases, which applies to something depends on your point of view.
> For example...... in terms of DATA, the question is clear black and white, either data is retained accurately or not.
> I can do a bit compare - and the result will be a simple black and white pass or fail.
> ...


I understand that in your binary view of the world if a codec's impact is audible in one test performed by one person out of 8 billion on earth, then it's audible.  That's unrealistic, and not how codecs are designed or used.  That's your binary view of the world only.


KeithEmo said:


> The only grey area I see would be with lower quality CODECs.... where we concede that the losses are audible, but there is a question of opinion about whether the loss is justified.
> (The "grey" arises because it's a matter of opinion whether the loss is significant or not.... and whether we consider the cost to be justified by the benefits.)
> There is also an area of UNCERTAINTY..... it may turn out that, on 95% of all files processed, the result is perfect, but on 5% it is not......
> (If so, we may still claim - in black and white - that "on 95% of a random selection of processed files nobody can hear the difference".)


That's a very important gray area though.  And it applies to all codecs, not must lower quality ones.


KeithEmo said:


> 2)
> My problem with your "size assertion" is simply that it isn't true.
> Your assumption that "if the file is the same size then it contains the same amount of data" is entirely incorrect.
> It is in fact quite simple to make a file larger or smaller without changing the amount of data it contains; or to add or remove data without changing the size of the file.
> ...


Actually file size is the definition of how much data it contains.  That data may not be audio data, or usable data, or necessary data, but it IS data.  FLAC files are losslessly compressed, and the original audio data can be perfectly recovered, but that's because the actual FLAC file contains LESS DATA using data of a different type to represent actual sample data.
Here's a link that clearly defines what file size means:
https://en.wikipedia.org/wiki/File_size

And FLAC is described here: https://en.wikipedia.org/wiki/FLAC

"FLAC uses linear prediction to convert the audio samples. There are two steps, the predictor and the error coding. The predictor can be one of four types (Zero, Verbatim, Fixed Linear and FIR Linear). The difference between the predictor and the actual sample data is calculated and is known as the residual. The residual is stored efficiently using Golomb-Rice coding. It also uses run-length encoding for blocks of identical samples, such as silent passages."
See?  Different data results in perfect storage of audio information...but using LESS DATA.  In fact LESS DATA is the entire goal of FLAC.


KeithEmo said:


> Your base assertion that "reduction in size is the goal...make that the ONLY goal...of a lossy codec" is incorrect.
> The goal of a lossy CODEC is to reduce the size of the file while avoiding altering the contents in an audible way....
> And the indicator of success would be that the file has gotten smaller but remains audibly the same.


Yes, of course.  Thank you for being so literal.  I thought that much was understood.


KeithEmo said:


> However, there are several possible "indicators of failure"........
> - if the file got larger that would be a definite fail
> - if the file sounded audibly different that would be a definite fail
> - if the file sounded the same and remained the same size that would be a sort of null result (a waste of time but no harm done).
> ...


Fine. No disagreement.  What does any of that have to do with anything we are discussing now?  Or the title of this poor corrupted thread?


----------



## KeithEmo

You seem to have a knack to "deciding" when things should and should not be 'taken literally".......

When discussing DATA there is no ambiguity about "exactly the same" or "different".
You do a bit compare; if the bits are identical - it passes; if a single bit is different - it fails.
There is no ambiguity and the definition is quite well established.

The easiest way that "WE" know that LOSSY CODECs are lossy is that they are described that way (if they didn't alter the data then they would be LOSSLESS CODECs).
FLAC is lossless because, if I take a WAV file, convert it to FLAC, convert it back again, and compare the new copy to the original, they will be IDENTICAL.
All of the bits will be the same........  therefore, when we convert out original file to FLAC, data was rearranged, but NO DATA WAS LOST OR PERMANENTLY ALTERED.
There is a topological difference between discarding or altering data and simply changing its format.
The encoding used by FLAC DOES NOT DISCARD ANY DATA - it simply stores it temporarily in a different format.

Your statement is incorrect - FLAC does NOT store less data - it stores 100% of the original data in a more compact format.
Not only won't you hear a difference, but no test known to man will be able to detect one.... which is the difference between "no audible difference" and simply "no difference".
If I were to convert a WAV file to AAC, then convert the AAC file back to a WAV, then do a bit compare, the result will tell me if the process was lossless or not... a simple binary fact.
When we look at the digital data we will find that it is NOT the same.......
(So, when we use AAC, the original data CANNOT be recovered exactly; but, when we use FLAC, it can.)

The title of "this poor corrupted thread" is based on a claim that is far overreaching...... which is why I dispute it.
The thread (and the article) don't say that "most audiophiles would be silly to digitize audio at a sample rate over 48k" or that "for most people 48k is more than good enough" (I would probably agree with those claims).
It makes a blanket assertion that using higher sample rates has no value (or even negative value) - and NEVER has any positive value.
(And nowhere does the original article suggest that lossy compression is "good enough" either - therefore all of this discussion about lossy compression is far afield from the original thread.)

And, just for the record, I make no apology for "taking reality literally".

If I were to say "most swans are white", I believe I would be statistically correct.
(I would also have correctly described the experience of most occupants of North America.)
If I were to say that "ALL swans are white" I would be wrong.
The fact that most people might not realize that I'm wrong doesn't make me right - it just makes that most people don't KNOW I'm wrong (and I will have contributed to their incorrect "knowledge" by providing them with incorrect information).
If I wanted to avoid saying something that was untrue, I might say that "it is statistically very unlikely that you'll ever see a swan that's any color except white in North America".
(There is a species of BLACK swan that lives mostly in Australia - although there are a few in the UK, and New Zealand - and there are probably a few in North American zoos.)
There is an excellent book on the subject (named "The Black Swan") which discusses the pitfalls of propagating errors and inaccurate generalizations - among other things.



pinnahertz said:


> I understand that in your binary view of the world if a codec's impact is audible in one test performed by one person out of 8 billion on earth, then it's audible.  That's unrealistic, and not how codecs are designed or used.  That's your binary view of the world only.
> That's a very important gray area though.  And it applies to all codecs, not must lower quality ones.
> Actually file size is the definition of how much data it contains.  That data may not be audio data, or usable data, or necessary data, but it IS data.  FLAC files are losslessly compressed, and the original audio data can be perfectly recovered, but that's because the actual FLAC file contains LESS DATA using data of a different type to represent actual sample data.
> Here's a link that clearly defines what file size means:
> ...


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## KeithEmo (Nov 29, 2017)

I NEVER said that "lossy compression was categorically bad" or that "lossy compression should never be used"; in fact, I do occasionally use it myself.
However, since I place a high priority on intellectual certainty, and a low priority on storage efficiency, I use it very rarely.
(I would personally prefer to pay a little extra to KNOW that I've got all the data, or, at the very worst, that I haven't contributed more errors to those already present that I can't avoid.)
I think lossy compression is an excellent solution to the engineering goal of "how can I store music in a lot less space in such a way that the majority of people will think it sounds good".

I do have a nasty habit of generalizing and treating both the GPS location system and the street mapping system used in most street navigators as a single process - which they are not.
The actual mechanism whereby the GPS system figures out your location is a form of triangulation, based on satellite broadcasts, and is simply limited by the accuracy of the process itself.
(GPS location data used to be deliberately corrupted, and provided at reduced accuracy, as a way to prevent terrorists and foreign governments from using it, but that was discontinued some time ago.)
HOWEVER, the way that information is correlated with features like street addresses is somewhat closer to a form of lossy perceptual compression.
The GPS system uses longitude and latitude..... but it is a database in your "street navigator" that maps that information to local features like building address numbers.
And that database includes a lot of "perceptual approximations" (for example, it knows that the house at one end of the block is #100, and the house at the other is #120, so it ASSUMES that #110 is half-way between them).
The database OMITS the specific information about the longitude and latitude of each individual home based on the assumption that it isn't critical - which is why it sometimes puts you in front of the wrong building.
At some point, it was decided that, in the interest of minimizing database size, some points would be stored exactly, while approximating others was "good enough".
I could list several specific situations where my navigator is able to return me to a physical location I've told it to store within a few feet, yet, when I ask it to take me to that address by map, it is off by as much as a hundred yards.

Since it's up to you and me to decide whether being put in front of the house next door to the one we entered is "a critical error" or "good enough" I would consider that to be a "decision based on perception".
I would perceive it as a problem if a missile that was aimed at my next door neighbor were to hit my house by mistake; or an assassin were to show up at my door by mistake; but people who end up next door due to a GPS error usually seem to find the Emotiva office pretty quickly.
(Even though the GPS system itself is accurate to a few yards, the street map shared by most popular systems seems to have the location of our main office incorrect by about thirty yards.)
I don't know what percentage of errors that occur with a typical street navigator occur due to measurement inaccuracy, what percentage are due to actual data errors in the map, and what percentage are due to "lossy approximations"... but there are certainly some of each.
(And, yes, those errors occur often enough that I DO check the address on the door before assuming that the system has brought me to the correct building - because it frequently does not.)

I'm guessing you wouldn't be happy if, when you checked your checkbook, the answer you got was: "Your balance is about $1100; it may be a few cents one way or the other, but the difference is trivial, so don't worry about it."
(And that's how I feel about my digital audio files.)

You seem to prefer the logic path: "If we decide to discard information, then we need to determine if the loss is audible, and, if it is, whether the benefits outweigh the costs".
I prefer the simpler: "If we ensure that NO information is lost, then those other questions are clearly moot."
(And, with digital audio files, thanks to a lot of effort spent figuring out how to store and transmit computer files accurately, it happens to be very simple to confirm - or fail to confirm - that no change has occurred.)
Note that the logic is "one way".
If the data has not changed, then I know with certainty that it will be audibly the same.
But, if the data HAS changed, then I have to either test it to see if the change was inaudible THIS TIME, or ABSOLUTELY TRUST whatever has changed it to have done so inaudibly.



pinnahertz said:


> I don't agree with that assertion either, because that is not MY assertion.  If you going to quote me, please do so exactly, and stop making things up!
> That's a fine goal...but you actually do the opposite quite frequently.
> As I said above, I disagree with that assertion!  "WE" do NOT know that.
> OK, fine...but you already do use lossy compression, like it, know it, or not.  It has a place, it has an application.  Please realize that something is not categorically "bad" just because it can be misapplied.
> ...


----------



## MrIEM

bigshot said:


> Just as a bit of interesting trivia... 44.1K covers the full spectrum of frequencies that humans can hear- 20Hz to 20kHz, with a bit to spare. Higher sampling rates extend the frequency response higher, far beyond our ability to hear, but the core frequencies below 20kHz are rendered exactly the same at 44.1 as they are at 192. So whatever it is that you seem to think is clearly audible isn't audible with human ears. Perhaps a bat!
> 
> However, it is possible that your equipment isn't designed to deal with super high frequencies and is adding distortion down in the audible range. So if you are positive you are hearing a difference, it is almost certainly noise, not music.



There is a valid reason to use his bitrates and depths at the recording stage. Additive noise. If you are recording voice, instruments and digital sources it's hard to avoid having to re-route and process sounds multiple times. That means you're doubling and tripling noise. With 16 bits it can easily reach a point where the noise is a distraction. With higher depths it's not. 

As for sample rate, the same logic applies. Re-processing and re-routing sounds can knock the edges off. If the bitrate is excessive to begin with when finalized at 16 bits it will still sound perfect. If you start where you want to end-up though you limit your ability to play with the sound before artifacts become a problem.


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## bigshot (Nov 29, 2017)

Is intellectual certainty a cure for OCD? You probably don't need intellectual certainty if you simply do a controlled listening test to determine your perceptual thresholds and then go with it and not worry any more. That's what I did. I don't care about theoretical sound I've proven to myself that I can't hear. I guess that's a form of intellectual certainty too!


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## KeithEmo

Intellectual certainty is a cure for _ANY_ sort of uncertainty 

Obviously a lot of all this depends on what you're hoping for (or expecting).
Personally, when I listen to music, my goal is to hear exactly what the artist and mixing engineer intended.
And any doubt that this isn't the case takes away from my enjoyment.
(I may choose to alter a file if I detect what I consider to be a flaw, but I absolutely do not want anyone or anything making that decision for me.)

As I mentioned before, when I look at a picture on my monitor, I may not know what the original looked like - so I calibrate my monitor; that way I can trust that it is showing me what the original looked like.
I don't want to use my judgment to decide whether what I'm seeing is as close to the original as I can tell.... that's extra work.... I'd rather just have a guarantee I can trust that it is.
I look at my audio system much the same way; I may not know for sure what the original sounded like, so I rely on my system to let me hear what's there without alteration.
(Of course I cannot know that the music file I have is accurate to the true original; but at least I can ensure that I don't change it any further.)

Back when MP3 was the norm, I recall various claims that "MP3 files sounded just like the original"; however, when I listened to them, I occasionally heard artifacts in certain recordings. (I don't recall the settings involved.)
And, when I investigated, I found that I _WAS_ able to make up special test files which _NO_ current MP3 encoder was able to encode without artifacts.
When you dig into the encoders, and the assumptions they make, it's often not difficult to figure out how to "trick" them (I used to test computer network products for a living).
And, with today's modern sampling synthesizers, any waveform I can concoct in a test file _MIGHT_ turn up in electronic music.

A similar situation occurs when a video tape master is converted to a DVD...... 
There is an algorithm which sets a filter level which is used to remove tape noise (which is necessary if you want to achieve a reasonable level of compression).
However, in specific instances, a particular visual feature (in this case dark swirling clouds or smoke), sometimes ends up "tricking the intelligence", and being _INCORRECTLY_ removed by the filter.
(The encoder usually does a very accurate job - but, in some small percentage of instances, it gets it significantly wrong.)

I haven't tried this with AAC, so it's possible that it_ NEVER_ makes mistakes and removes something that might have been audible....
However, because of the complexity of the algorithms involved, I would have to run an awful lot of tests to claim that I was 100.0% sure it would _NEVER_ happen.
Alternately, I would have to listen carefully to _EVERY_ file I encoded to ensure that "it wasn't the one where the process failed".

Now, to a lot of people, reducing the size of their music library by 75%, even at the risk that one of their 10,000 songs might contain a single audible artifact, might seem like an excellent tradeoff.
However, since I have no issue whatsoever with storage size, and I'll admit to being a bit OCD when it comes to knowing what's going on, I'd still rather take the sure thing.
It also boils down to a matter of process.
If I wanted to fit a bunch of files on an iPod, I could encode them, then carefully compare each to the original to confirm that it was audibly perfect.
However, since I'm not going to discard the original copy anyway, that's an awful lot of extra work (the time it would take is worth more to me than the cost of buying a bigger SD card).
(My "library drive" is simply one of the three copies of each file I retain - one live copy plus two backups - so, if I were to create another AAC encoded copy, it would simply be another copy to keep track of.)

Also, to be honest, I tend to compartmentalize how I prioritize my music.
I could probably cheerfully get rid of about 90% of the CDs I currently own - and simply listen to whatever version of those tunes I can punch up on Tidal.
However, for the small percentage of my collection that I very much care about, a single bit out of place counts as a "fatal flaw".

And, much as everyone on this thread likes to disparage such claims.....
I absolutely have encountered situations where certain details or flaws became audible on a new piece of equipment (that were totally inaudible on my previous equipment).
Therefore, I am very disinclined to believe with absolute certainty that I cannot possibly discover differences tomorrow that really were inaudible today.
(And, again, the _EASIEST_ way to ensure against that is simply to keep the original file intact.)

And, yes, I _DID_ have quite a few MP3 copies of albums "back in the day"....
And, yes, it cost me a lot of money to go out and buy the CDs for all of them when I realized there was an audible difference.



bigshot said:


> Is intellectual certainty a cure for OCD? You probably don't need intellectual certainty if you simply do a controlled listening test to determine your perceptual thresholds and then go with it and not worry any more. That's what I did. I don't care about theoretical sound I've proven to myself that I can't hear. I guess that's a form of intellectual certainty too!


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## bigshot (Nov 29, 2017)

I won’t comment on whether you’re hearing exactly what the artists and engineers heard, because it will just upset you and make you go out and buy more audio equipment! But I will say lossless wouldn’t be the reason you aren’t achieving that.


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## reginalb

bigshot said:


> I won’t comment on whether you’re hearing exactly what the artists and engineers heard, because it will just upset you and make you go out and buy more audio equipment! But I will say lossless wouldn’t be the reason you aren’t achieving that.



Yeah, but MQA will solve it, right? I mean, the mastering quality is authenticated!


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## BaTou069

Am I reviving and old war on audio claims? Did in the meantime other threads pop up citing the xiph article that @bigshot has in his sig?
I went down that rabbit hole after discovering the 2nd xiph video and article

I kind of understand but then again, I don't

There are several claims kinda mixed in but related, then again, correct me if I'm wrong

1) Playing HiRes files makes no sense (per xiph) since the resulting analog signal wouldn't be measurable different 
2) Playing HiRes files, assuming we play a real mastering that was recorded lets say at 24/192, doesn't make sense since we don't try to reproduce the ADC but the original sound (connecting the dots)
3) I read how HiRes extends the bandwith to the inaudible spectrum, and thus or maybe not thus but also lowers the noise floor

If 24 bit or higher sampling rate doesn't matter, how come so many users here and on other forums use HQPlayer that upsamples, or use equipment as the chord m-scaler to upsample? If playing in the same bit depth and sample rate as the source makes no sense, why would upsampling make sense?

I remember when we learned about waves and did fourier transform during my physics studies, that each frequency or sound is basically a combination of an infinite series of waves with different wavelengths and amplitudes, including of course those inaudible frequencies. 
From playing around with fourier (try for yourself google fourier simulator and just add more and more elements/n) I know that those inaudible frequencies make no big difference, but they have an effect on how the audible frequencies look or behave. So I could imagine while the 16bit frequency response  covers the full spectrum of human hearingt, hat having those inaudible frequencies in the file could still make a difference since they affect those audible frequencies (finetune maybe), but then maybe it's not big enough to be audible?

I'd be happy to hear some thoughts


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## sander99

BaTou069 said:


> 1) Playing HiRes files makes no sense (per xiph) since the resulting analog signal wouldn't be measurable different


Eh, no.
a. It makes no sense because it wouldn't be audibly different
b. IF the same bandlimited signal is recorded and played back with a higher sampling rate than required and the same bit depth is used THEN the resulting analog signal also wouldn't be measurable different.
c. If you use a higher bit depth then the signal to noise ratio will be different. But not audibly different (under normal listening conditions, if you record a very low level signal and amplify it like crazy then it would be audible).


BaTou069 said:


> 2) Playing HiRes files, assuming we play a real mastering that was recorded lets say at 24/192, doesn't make sense since we don't try to reproduce the ADC but the original sound (connecting the dots)


Eh, what does "reproducing the ADC" mean?


BaTou069 said:


> If 24 bit or higher sampling rate doesn't matter, how come so many users here and on other forums use HQPlayer that upsamples, or use equipment as the chord m-scaler to upsample? If playing in the same bit depth and sample rate as the source makes no sense, why would upsampling make sense?


Upsampling and digital filtering can be helpful steps in the signal reconstruction process. (First calculating a few additional dots before connecting them).


BaTou069 said:


> From playing around with fourier (try for yourself google fourier simulator and just add more and more elements/n) I know that those inaudible frequencies make no big difference, but they have an effect on how the audible frequencies look or behave.


Every waveform can be decomposed into (is a summation of) a finite or infinite number of sine shaped waves. If we talk about frequency content in a signal we refer to those sine shaped waves. So audible frequencies don't look different due to inaudible frequencies being present, all frequencies look the same: sine shaped.
If a certain sound corresponds to a complicated waveform composed of many sine shaped waves then we recognise that sound from the audible overtones that accompany the fundamental frequency. The inaudible overtones don't matter. Somewhere in the inner ear - I don't know the exact English terminology - are hairs all tuned to certain small frequency bands. Let's look for example at a note being played with an acoustic bass. This is a complicated wave form that can be decomposed in a number of sine shaped waves, a number of different frequencies. Groups of hair cells correspomding to the different frequencies are triggered. The brain recognizes the combination of frequencies as the sound of the bass. There are no hair cells tuned to inaudible frequencies. Inaudible frequencies play no role in the process.


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## BaTou069

sander99 said:


> If a certain sound corresponds to a complicated waveform composed of many sine shaped waves then we recognise that sound from the audible overtones that accompany the fundamental frequency. The inaudible overtones don't matter. Somewhere in the inner ear - I don't know the exact English terminology - are hairs all tuned to certain small frequency bands. Let's look for example at a note being played with an acoustic bass. This is a complicated wave form that can be decomposed in a number of sine shaped waves, a number of different frequencies. Groups of hair cells correspomding to the different frequencies are triggered. The brain recognizes the combination of frequencies as the sound of the bass. There are no hair cells tuned to inaudible frequencies. Inaudible frequencies play no role in the process.


Exactly my point, but maybe I misunderstood. If we take that acoustic bass note for example, that is basically a composition/mix made out of an infinite series of sine waves of different amplitudes and frequencies (audible and inaudible frequencies), wouldn't recording in 24/32 bit enable those inaudible frequencies to be kept in the file, if the mic can pic those up...
high frequency sine waves do have an influence on the low frequency sine waves, I don't remember exactly the term but if you for example create a square wave out of sine waves, you'd need quite a lot of HF waves to get the corners rights.
Just saying I can imagine why not removing inaudible frequencies could be of importance if the effect those frequencies have on the audible ones is .... audible


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## sander99

BaTou069 said:


> if you for example create a square wave out of sine waves, you'd need quite a lot of HF waves to get the corners right


1. You only need the ones up to the limit of hearing, 20 kHz.
2. You wouldn't want a perfect square wave going into your amp because no amp can change it's output voltage infinitely fast and no loudspeaker driver or headphone driver can move infinitely fast and hence distortion would be unavoidable if you tried!
3. A perfect square wave probably doesn't exist in the real world.


BaTou069 said:


> Exactly my point, but maybe I misunderstood.


I fear you did misunderstand. The shape itself of the waveform is not of direct importance. The audible subset of frequencies - all sine shaped waves - that together form the waveform is. The individual hair cells don't know the difference between a sine wave or another wave, they just react to the - sine shaped - frequencies they are tuned to. If for example you would play a square wave - assuming for a moment that is possible without audible distortion - with a fundamental frequency of 10 kHz then the first harmonic would be at 30 kHz. You would not be able to tell the difference between that square wave and a 10 kHz sine wave because you would only have hair cells reacting to 10 kHz and none reacting to any of the harmonics. You probably would be able to hear the difference between a square wave of 3 kHz and a sine wave of 3 kHz because you do have hairs reacting to 9 kHz (the first harmonic above the fundamental of the 3kHz square wave).


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## 111MilesToGo

Let me try to give some background on human hearing capabilities. There are two ”domains“ involved, frequency range and timing perception.

Frequency range: It‘s so well known that humans can capture the range 20 Hz to 20 kHz - ideally, depending on age, on what destruction happened during life etc. So Nyquist and Shannon‘s theorem justifies the basic range of sampling frequencies chosen in audio technology, 44.1 kHz for CD, 48 kHz for DAT. Higher sampling frequencies allow for better representation of the continuous audio waves by the discontinuous sequence of samples (”dots“ on a timeline).

But don‘t forget the aspect of timing. Humans perceive timing differences in the arrival of wavefronts between left and right ear down to a few microseconds (like 8 to 5 microseconds or even less). That is one aspect of perceiving the direction of a source of sound - you know, in evolution, the famous lion out there that is going to eat you … So, in order to represent such small timing differences in sampled representations of sound, it takes sampling frequencies of more than 100 and up to 200 kHz or more. (More precisely, 1 microsecond corresponds to 1 MHz, 5 microseconds to 200 kHz, 10 microseconds to 100 kHz.)

I think that is an easy explanation of what sampling rates are needed for a good representation of sounds by discontinuous samples.


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## old tech

BaTou069 said:


> Exactly my point, but maybe I misunderstood. If we take that acoustic bass note for example, that is basically a composition/mix made out of an infinite series of sine waves of different amplitudes and frequencies (audible and inaudible frequencies), wouldn't recording in 24/32 bit enable those inaudible frequencies to be kept in the file, if the mic can pic those up...
> high frequency sine waves do have an influence on the low frequency sine waves, I don't remember exactly the term but if you for example create a square wave out of sine waves, you'd need quite a lot of HF waves to get the corners rights.
> Just saying I can imagine why not removing inaudible frequencies could be of importance if the effect those frequencies have on the audible ones is .... audible


As a thought experiment, let's assume you are correct that these inaudible frequencies affect those that are audible.  It would only be relevant to a live acoustic event, not a recording of it.  The recording would have captured the sound as we would have heard it, inclusive of any effects the inaudible frequencies may have had on the audible.


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## sander99

111MilesToGo said:


> But don‘t forget the aspect of timing. Humans perceive timing differences in the arrival of wavefronts between left and right ear down to a few microseconds (like 8 to 5 microseconds or even less). That is one aspect of perceiving the direction of a source of sound - you know, in evolution, the famous lion out there that is going to eat you … So, in order to represent such small timing differences in sampled representations of sound, it takes sampling frequencies of more than 100 and up to 200 kHz or more. (More precisely, 1 microsecond corresponds to 1 MHz, 5 microseconds to 200 kHz, 10 microseconds to 100 kHz.)
> 
> I think that is an easy explanation of what sampling rates are needed for a good representation of sounds by discontinuous samples.


That is what one could intuitively think. However the sampling theorem - a proven mathematical theorem - states that the original analog signal can be reconstructed completely with the original timing and phase preserved.


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## 111MilesToGo

sander99 said:


> That is what one could intuitively think. However the sampling theorem - a proven mathematical theorem - states that the original analog signal can be reconstructed completely with the original timing and phase preserved.


Yes, exactly. So the one requirement of representing 20 kHz leads to 40 kHz in sampling. But the other requirement of representing say 5 microseconds time difference between the L/R ears leads to 200 kHz.


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## old tech (Jan 25, 2022)

111MilesToGo said:


> Yes, exactly. So the one requirement of representing 20 kHz leads to 40 kHz in sampling. But the other requirement of representing say 5 microseconds time difference between the L/R ears leads to 200 kHz.


IIRC, the timing error with 44.1 sampling is measured in the nano seconds.  I doubt that Superman could hear that.

The other thing to consider is that timing errors of 44.1 sampling are in extreme order of magnitude lower than for any well engineered analog playback device (ie tape or record players) yet there are few complaints.

Edit: Nano seconds, not pico - still way beyond any human...


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## sander99

111MilesToGo said:


> But the other requirement of representing say 5 microseconds time difference between the L/R ears leads to 200 kHz.


No. I put a link to the part in one of the xiph videos where timing is covered, just watch 2 minutes from this point:




And here some more detailed information about the timing limits (there are limits, but they are far smaller than the inter sample time):


SoundAndMotion said:


> IIRC, you are some kind of engineer. You may have had some training in sampled-data systems (aka discrete-time systems). The ability to resolve temporal differences between 2 sets of sampled data depends on the nature of the signal (the highest slope therein) and the number of possible values (resolution or bit depth). The steeper the max slope and the larger the number of possible values (states), the better the temporal resolution. For sine waves the calculation is rather straightforward:
> ​where ∆t_res is the temporal resolution, f_sig is the frequency of the sine wave (the signal) and N_st is the number of states of the values.
> The best case sine wave for 44.1/16 is: f_sig=22050Hz and N_st=65536, then ∆t_res=110ps. I imagine @gregorio mistakenly used the sampling frequency rather than the Nyquist frequency.
> Perhaps more "typical" would be f_sig=2205Hz at -20dBFS (N_st=6554), with ∆t_res=11ns
> ...


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## 111MilesToGo (Jan 26, 2022)

old tech said:


> IIRC, the timing error with 44.1 sampling is measured in the nano seconds.  I doubt that Superman could hear that.
> 
> The other thing to consider is that timing errors of 44.1 sampling are in extreme order of magnitude lower than for any well engineered analog playback device (ie tape or record players) yet there are few complaints.
> 
> Edit: Nano seconds, not pico - still way beyond any human...


You are talking about electronic timing errors like jitter, e.g. within a DAC. But I am talking about discretization timing errors, in the ADC. I.e. about having something in sample number n or n+1.

Edit: At 44.1 kHz sampling, time discretization errors are of the order 22.7 microseconds (1/44,100 s).


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## SoundAndMotion

Good morning @111MilesToGo,

You are mistaken. Although it is easy to understand your intuitive sense, it turns out it is not true.
What @sander99  and @old tech have written is correct. If you would like to understand why, I (or others here) would be happy to explain. Would you prefer a more mathematical or more conceptual explanation? Or perhaps, a simple example that shows much greater temporal resolution than the 22.7 µs resolution you claim.

Schönen Tag noch...


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## gregorio (Jan 26, 2022)

To clarify a few points:


BaTou069 said:


> 3) I read how HiRes extends the bandwith to the inaudible spectrum, and thus or maybe not thus but also lowers the noise floor


This is true in theory but not really in practice, we have to be careful to specify which noise floor. More bits, EG 24bits rather than 16bits, lowers the DIGITAL noise floor, from inaudible to even more inaudible BUT in practise doesn't really change the noise floor of recordings. The noise floor of 16bit is about -96dBFS and of 24bit is about -144dBFS. However, the vast majority of recordings have a noise floor of about -60dBFS or higher, due to the noise floor of the recording venue, mic self-noise, etc. More bits doesn't affect the acoustic or analogue noise floor, only the digital noise floor. So, it doesn't matter whether we distribute these recordings in 16 or 24 bit, the noise floor will still be -60dBFS or higher.


BaTou069 said:


> If 24 bit or higher sampling rate doesn't matter, how come so many users here and on other forums use HQPlayer that upsamples, or use equipment as the chord m-scaler to upsample? If playing in the same bit depth and sample rate as the source makes no sense, why would upsampling make sense?


Oversampling makes sense and almost all DACs (and all ADCs) do this internally. Upsampling before the DAC therefore doesn't make sense and "_so many users here and on other forums_" have been duped by marketing nonsense!


111MilesToGo said:


> So, in order to represent such small timing differences in sampled representations of sound, it takes sampling frequencies of more than 100 and up to 200 kHz or more. (More precisely, 1 microsecond corresponds to 1 MHz, 5 microseconds to 200 kHz, 10 microseconds to 100 kHz.)
> 
> I think that is an easy explanation of what sampling rates are needed for a good representation of sounds by discontinuous samples.


It is "_an easy explanation_", unfortunately though it's also an INCORRECT explanation! Because ...


111MilesToGo said:


> You are talking about electronic timing errors like jitter, e.g. within a DAC. But I am talking about discretization timing errors, in the ADC. I.e. about having something in sample number n or n+1.
> 
> Edit: At 44.1 kHz sampling, time discretization errors are of the order 22.7 microseconds (1/44,100 s).


Due to marketing misinformation, it's a common audiophile failing to consider digital audio as an analogue signal, which of course it isn't. 22.7 microsecs is the sampling interval of 44.1kHz, it is NOT the "time discretization error" of 44.1kHz. The sample points are ONLY ordinates which allow the reconstruction of a continuous waveform, ALL information between the sample points is preserved (below the Nyquist freq)!

Please watch the Ziph video already posted. If you don't understand it, it's easy to test yourself with an audio editor/DAW: Upsample a 44.1kHz recording to 96kHz, move the waveform 1 sample (11.3 microsecs), convert it back to 44.1kHz and back to 96kHz again. According to your explanation we should loose this 96kHz 1 sample movement because it's only half a 44.1kHz sample period. However, as you will see, it's not lost, it's preserved. If you're still not convinced, try upsampling to 192kHz and move by 1 sample period (~5.7 microsecs). Same again, it's preserved after converting to 44.1kHz!

G


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## danadam (Jan 26, 2022)

111MilesToGo said:


> But don‘t forget the aspect of timing. Humans perceive timing differences in the arrival of wavefronts between left and right ear down to a few microseconds (like 8 to 5 microseconds or even less). That is one aspect of perceiving the direction of a source of sound - you know, in evolution, the famous lion out there that is going to eat you … So, in order to represent such small timing differences in sampled representations of sound, it takes sampling frequencies of more than 100 and up to 200 kHz or more. (More precisely, 1 microsecond corresponds to 1 MHz, 5 microseconds to 200 kHz, 10 microseconds to 100 kHz.)


From https://imgur.com/a/KVFOJU1


> A 16 bit, 44.1 kHz file with 33 impulses. Impulses in the right channel (bottom) are exactly 0.5 second apart, while the distance between impulses in the left channel (top) increases by 1.4 microsecond.
> 
> The animation skips to each impulse as evidenced by the time bar at the top. The grey waveform is this 16/44 file upsampled 16x. The highlighted area in the middle is 2-samples wide and centers on zero-crossing of the right, "stationary", channel.


(see attachment below for the file with impulses)


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## 71 dB (Jan 27, 2022)

Time resolution of 44.1 kHz 16 bit digital audio is at least 1000 times bigger than needed. It is all about how band-limited signals work.

If the time resolution was only 22.7 µs, applying minimum phase filters would be interesting as the phase shift would be quantised to junks corresponding to temporal delays of multiples of 22.7 µs. Fortunately digital audio doesn't work like that. 

Digital audio is deceptively easy to misunderstand and surprisingly demanding to understand correctly.


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## 111MilesToGo

Thanks to all who provided me with all educational stuff, good reads, good links to the theory behind digital audio and video.

And besides, contenance and netiquette are there for a good reason …


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## 71 dB

111MilesToGo said:


> Higher sampling frequencies allow for better representation of the continuous audio waves by the discontinuous sequence of samples (”dots“ on a timeline).


No, they don't if we are talking about band-limited signals and we should be, because sampling theorem _requires_ that signals are bandlimited according to the Nyquist frequency. By bandlimiting the signal, we know 100 % what the signal does between the sample points. There is mathematically only one way it can connect the dots. In order to behave differently the signal would need frequencies above Nyquist, but band-limited signals do not have those!

Also, how audio waves look to our eyes is different from how our ear hear them. For our eye the visual shape of the wave is easy to see, because that's what eyes are for: seeing shapes. Our ears however work differently and are much more into analysing the frequency content of the waveform. Since different _looking_ waveforms can have the exact same frequency content, ears are not interested much about the exact shape of the waveform. In fact, rooms acoustics with all the reflections and reverberation render the original waveform pretty much unrecognisable to the eye, but that doesn't matter much, because for the ear the frequency content is more or less intact (depending on how good the acoustics are). Even speakers/headphones alone change the waveform drastically. 



111MilesToGo said:


> But don‘t forget the aspect of timing. Humans perceive timing differences in the arrival of wavefronts between left and right ear down to a few microseconds (like 8 to 5 microseconds or even less). That is one aspect of perceiving the direction of a source of sound - you know, in evolution, the famous lion out there that is going to eat you … So, in order to represent such small timing differences in sampled representations of sound, it takes sampling frequencies of more than 100 and up to 200 kHz or more. (More precisely, 1 microsecond corresponds to 1 MHz, 5 microseconds to 200 kHz, 10 microseconds to 100 kHz.)
> 
> I think that is an easy explanation of what sampling rates are needed for a good representation of sounds by discontinuous samples.


Again, the core idea of the sampling theorem is to represent (band-limited) _continuous_ signals fully by taking samples often enough. 

Higher sampling frequency allows sampling of higher frequences. That's all.
Higher bit depth allows lower noise floor. That's all.

For _consumer_ audio 44.1 KHz sampling frequency and 16 bit quantisation allow enough bandwidth (up to 20 kHz) and dynamic range (technically about 96 dB and perceptually up to 120 dB depending on what kind of dithering is used). In music _production_ 24 bit is useful/beneficial for practical reasons and also higher sampling rates may have a place in the production, for example when recording ultrasonic sounds and using them at lower samplerates as sound effects.


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## 71 dB

111MilesToGo said:


> Edit: At 44.1 kHz sampling, time discretization errors are of the order 22.7 microseconds (1/44,100 s).


That would be true, I believe, if the bit depth was 1, but it is 16 or more.


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## MooMilk

As a delivery format title stays fundamentally true,
For recording - higher sample rates may proof useful for some VST effects (esp. for compression\limiting), and a bit lower latency


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## gregorio

MooMilk said:


> For recording - higher sample rates may proof useful for some VST effects (esp. for compression\limiting)


That was true, roughly 20 or more years ago but not today. It’s true that some plug-ins perform better at higher sample rates, certain modelled plugins such as compressors/limiters, certain soft synths and some other plugins, but these days those plugins would over-sample internally, so a higher sample rate file format is unnecessary.

G


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## Whitigir

Have anyone even considered about physics beside mathematics? In reality, there are tolerances parameters that play a huge role, and why over sampling.  Higher sampling will give better accuracy, because resistors and all other physical components have precision tolerances.  For band limited with Nyquist sampling, the tolerances of resistors needed are not existed yet, let alone temperatures drift parameters and so on.


----------



## 71 dB

Whitigir said:


> Have anyone even considered about physics beside mathematics? In reality, there are tolerances parameters that play a huge role, and why over sampling.  Higher sampling will give better accuracy, because resistors and all other physical components have precision tolerances.  For band limited with Nyquist sampling, the tolerances of resistors needed are not existed yet, let alone temperatures drift parameters and so on.


Higher sampling rate can reduce the accuracy when the circuits with their time constants are not "fast enough". That's why this "physics" argument doesn't hold water.


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## gregorio

Whitigir said:


> Have anyone even considered about physics beside mathematics?


What physics besides mathematics? There is no “physics besides mathematics” because all physics is defined by mathematics. So of course no one has ever considered physics besides mathematics. 


Whitigir said:


> In reality, there are tolerances parameters that play a huge role, and why over sampling.


What do you think those “_tolerances parameters_” are defined by, if not mathematics? Oversampling is used because it provides the most efficient method of overcoming certain issues. For example the cost/complexity of analogue anti-alias/reconstruction filters or of say emulating ultrasonic freq induced IMD. 


Whitigir said:


> Higher sampling will give better accuracy, because resistors and all other physical components have precision tolerances.


Higher sampling does NOT give better accuracy, it gives the same accuracy but over a wider frequency spectrum. 


Whitigir said:


> For band limited with Nyquist sampling, the tolerances of resistors needed are not existed yet,


True but you’re contradicting yourself! If the needed tolerances of resistors do not exist yet for 24/48 how will increasing the sample rate “give better accuracy”?


Whitigir said:


> let alone temperatures drift parameters and so on.


A fundamental limitation with resistors is the thermal noise. First measured by Johnson, quantified by Nyquist and published in 1928, this equation defines the amount of thermal noise produced by resistors depending on bandwidth and temperature:





G


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## Whitigir

71 dB said:


> Higher sampling rate can reduce the accuracy when the circuits with their time constants are not "fast enough". That's why this "physics" argument doesn't hold water.


Then why most of the current DAC are all at the least 8X OverSampling ?


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## gregorio

Whitigir said:


> Then why most of the current DAC are all at the least 8X OverSampling ?


Because:


gregorio said:


> Oversampling is used because it provides the most efficient method of overcoming certain issues. For example the cost/complexity of analogue anti-alias/reconstruction filters or of say emulating ultrasonic freq induced IMD.


However, @71 dB is correct, oversampling *can* reduce the accuracy. But the word “can” does not mean “always will”, it depends on how it’s implemented. Just increasing the sample rate, say from 48kFS/s to 192kFS/s, obviously requires processing 4x the amount of data in the same period of time. This can be achieved with no loss of accuracy provided that 4x the processing power is available. There have been implementations which solved the problem by using decimation or reconstruction filters that did not adequately attenuate at the Nyquist point (say 96kHz) and therefore reduced accuracy. Most DACs these days oversample by factors of more than 64x but reduce the processing demand by reducing the bit depth to just a handful.

G


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## 71 dB

Whitigir said:


> Then why most of the current DAC are all at the least 8X OverSampling ?


Because the benefits of oversampling surpass the negatives.


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## Whitigir

71 dB said:


> Because the benefits of oversampling surpass the negatives.


Thank you! This answers the thread topic itself


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## 71 dB

Whitigir said:


> Thank you! This answers the thread topic itself


I don't think it answers the thread topic itself, but I'm glad you are pleased.


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## castleofargh

Whitigir said:


> Thank you! This answers the thread topic itself


It does not end the conversation because oversampling and hires files aren't entirely the same thing, and for oversampling, where it happens does matter for some of the tricks a DAC can have in its bag. 
So we still need to consider various scenarios.  

BTW, when I say that oversampling and hires aren't the same, in my mind the hires file can cause more issues because of the occasional high ultrasonic content. Most audiophiles think the other way around but to me, more content I don't hear that has the potential to negatively affect the audible range I can hear, that's not good news. 
On the other hand, oversampling is super useful, no denying that.


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## Whitigir

castleofargh said:


> It does not end the conversation because oversampling and hires files aren't entirely the same thing, and for oversampling, where it happens does matter for some of the tricks a DAC can have in its bag.
> So we still need to consider various scenarios.
> 
> BTW, when I say that oversampling and hires aren't the same, in my mind the hires file can cause more issues because of the occasional high ultrasonic content. Most audiophiles think the other way around but to me, more content I don't hear that has the potential to negatively affect the audible range I can hear, that's not good news.
> On the other hand, oversampling is super useful, no denying that.


Yes, and I agreed on all of those points.  Especially high res issues with ultra frequencies as you have pointed out.  People is just swallowed up by marketing most of the time.  I also would take high res of 24/48Khz though


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## castleofargh

48 is everywhere and it helps some of the really bad reconstruction filters a little bit, I think it's a done deal now. 
24bits IDK. IMO it's too big of a file size change not to become a matter of personal priority and compromises.


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## 71 dB

I think for consumer audio *58.8/12* could have been an interesting alternative. It would have had exactly same bitrate as 44.1/16. The sample rate would be 4/3 times bigger and the bith depth 3/4 times smaller. The larger samplerate would allow the use of more "relaxed" anti-alias and reconstruction filter. Aggressive shaped dithering concentrating the dither noise energy above 20 kHz would allow _perceptual_ dynamic range up to maybe 100 dB.


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## bigshot

At this point oversampling is built in, so it isn’t a problem. I imagine when oversampling was introduced, they knew DACs would soon be supporting higher data rates out of the box, so they didn’t need to limit their data resources.

But you’re absolutely right, there’s headroom built into 16/44.1. It’s a little over the line into overkill.


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## AndreRitter

https://www.hifinews.com/content/philips-cd100-vintage
One of the most musical cdp's I owned and its 14bit.
Folk get hung up on bitrates etc and forget about the music.


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## gregorio

71 dB said:


> I think for consumer audio *58.8/12* could have been an interesting alternative…. Aggressive shaped dithering concentrating the dither noise energy above 20 kHz would allow _perceptual_ dynamic range up to maybe 100 dB.


It’s possibly an interesting alternative as just a contemporary thought experiment but it wouldn’t have been an alternative at the time. As far as I’m aware, the DSP power required to implement noise-shaped dither wasn’t possible at the time but more importantly, the Noise-Shaping Theorem (Gerzon, Craven) wasn’t published until 1989, a decade after the Redbook specs were being decided. 


AndreRitter said:


> https://www.hifinews.com/content/philips-cd100-vintage
> One of the most musical cdp's I owned and its 14bit.
> Folk get hung up on bitrates etc and forget about the music.


Some of that article is more than a little dubious, particularly the subjective claims.

Although I agree that the audible difference between the first consumer CD players and modern digital audio players is a lot smaller than the vast majority of audiophiles would imagine and that they “get hung up” on bit/sample rates. That’s hardly surprising though, because of all the audiophile marketing for the last 20+ years “bigging up” higher bit/sample rates to justify the purchase of new audiophile products.

G


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## 71 dB

gregorio said:


> It’s possibly an interesting alternative as just a contemporary thought experiment but it wouldn’t have been an alternative at the time. As far as I’m aware, the DSP power required to implement noise-shaped dither wasn’t possible at the time but more importantly, the Noise-Shaping Theorem (Gerzon, Craven) wasn’t published until 1989, a decade after the Redbook specs were being decided.
> 
> G


Of course, but maybe in parallel universe the Noise-Shaping Theorem was invented in 1964 by a jew who Nazis killed in our universe in WWII and DSP power was enough...


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## Joe Bloggs (Oct 29, 2022)

Whitigir said:


> Have anyone even considered about physics beside mathematics? In reality, there are tolerances parameters that play a huge role, and why over sampling.  Higher sampling will give better accuracy, because resistors and all other physical components have precision tolerances.  For band limited with Nyquist sampling, the tolerances of resistors needed are not existed yet, let alone temperatures drift parameters and so on.


Certainly it pays to oversample before DAC conversion, to avoid the issue with physical components you mentioned.  But that does not mean the music storage format needs to be oversampled.  Oversampling is totally a thing internal to a DAC.  For example, it is done even when the indicated sampling rate is 44.1kHz, on any HiBy R player and even RS players in OS modes.


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## Davesrose

gregorio said:


> It’s possibly an interesting alternative as just a contemporary thought experiment but it wouldn’t have been an alternative at the time. As far as I’m aware, the DSP power required to implement noise-shaped dither wasn’t possible at the time but more importantly, the Noise-Shaping Theorem (Gerzon, Craven) wasn’t published until 1989, a decade after the Redbook specs were being decided.
> 
> Some of that article is more than a little dubious, particularly the subjective claims.
> 
> ...



It's interesting for me growing up in the 1980s, and first having "hi-fi" with a Sony monitor headphone and various Discmans....I did find the best euphonic sound was a DSP setting with various Sony Discmans: it was Sony's "surround" DSP.  My dad had a D-555 (what audiophiles consider a holy grail for best Discman). All sorts of digital EQ bands, all metal, digital volume. I had a D-824K (less premium that was all plastic).  The D-555 had a lot of steps for this surround DSP, but the few steps the "surround" button had for my D-824K sounded more natural to me (also with marketing, the D-555 said 8 times oversampling, while the D-824K said 1 bit sampling).  These surround DSPs had an interesting processing to recess vocals and boost percussion frequencies (to make it sound like a live concert).  All to say for me it's evident that people will react to inherent EQ/DSP that they may not be aware of or sound level (for example, in my home theater system, I'm amazed how low a sound level could be for a movie with Dolby Atmos streaming being much lower vs same title Dolby Atmos on disc being much higher).

When it comes to high-res distribution in the home market, I do have SACD titles.  Probably the most unique which I'll hold on to was a quadrophonic recording of Bach toccatas and fugues in a cathedral with 4 organs.  I have one "movie" title that is 96khz surround: Baraka: which is also a cinematography odyssey (even though it still hasn't been remastered to 4K).  Perhaps due to mastering with its DTS track, it has some better separation with subwoofer and surround speakers.  A lot of my concert blu-rays have hi-res DTS surround tracks in 96khz.  They sound great, but also should carry for 44.1.  Maybe it's just the standard of that media: at least operas sound great as well as genres like live blues (that can have heavy percussion making my subwoofer active).


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## gregorio

Davesrose said:


> The D-555 had a lot of steps for this surround DSP, but the few steps the "surround" button had for my D-824K sounded more natural to me (also with marketing, the D-555 said 8 times oversampling, while the D-824K said 1 bit sampling).


As far as I’m aware, the D-555 was released in 1988 while the D-824K was released in 1994. A significant difference because available DSP power increased significantly during that period. Not only did that allow for more sophisticated/complex programming but also the programming routines became more efficient. Also, around 1990 most of the DAC chip manufacturers moved to a 1bit, 64x oversampling topology, although in both cases the “surround” DSP would be applied prior to the oversampling step. 


Davesrose said:


> All to say for me it's evident that people will react to inherent EQ/DSP that they may not be aware of or sound level (for example, in my home theater system, I'm amazed how low a sound level could be for a movie with Dolby Atmos streaming being much lower vs same title Dolby Atmos on disc being much higher).


Loudness/Sound Level control all got a bit complicated. Partially due to the fact that Dolby was ahead of the rest of the industry when it came to loudness control and when the rest of industry did catch-up, it used a largely different approach. So, we started with Dolby’s “Dialogue Normalisation” (Dialnorm), then they introduced “Dolby Volume” and then we had the rest of the industry’s “Loudness Normalisation”, which isn’t standardised across media types. So, not only do we have different implementations within the hardware/software of devices but we also have different delivery specifications for the audio content for different distributors. For example, TV broadcast is mandated (by law in some countries) at about -23LUFS integrated loudness, while for example Netflix specifies -27LUFS which isn’t “integrated loudness” but integrated Dialnorm loudness. The result could easily be that Netflix is 8LUFS lower loudness. Apple Music on the other hand, requires Atmos mixes at -18LUFS. BluRay is often also -27LUFS but that could be 4-5LUFS higher than Netflix’s -27LUFS dialogue anchored loudness and it’s not uncommon for BluRay disks to have -24LUFS integrated loudness. 


Davesrose said:


> Perhaps due to mastering with its DTS track, it has some better separation with subwoofer and surround speakers.


This could easily be a Dialnorm issue, which will usually make a Dolby soundtrack sound 4dB quieter than a DTS soundtrack, while “Dolby Volume” could change the spectral content depending on your volume level.

G


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## Davesrose (Nov 5, 2022)

gregorio said:


> As far as I’m aware, the D-555 was released in 1988 while the D-824K was released in 1994. A significant difference because available DSP power increased significantly during that period. Not only did that allow for more sophisticated/complex programming but also the programming routines became more efficient. Also, around 1990 most of the DAC chip manufacturers moved to a 1bit, 64x oversampling topology, although in both cases the “surround” DSP would be applied prior to the oversampling step.
> 
> Loudness/Sound Level control all got a bit complicated. Partially due to the fact that Dolby was ahead of the rest of the industry when it came to loudness control and when the rest of industry did catch-up, it used a largely different approach. So, we started with Dolby’s “Dialogue Normalisation” (Dialnorm), then they introduced “Dolby Volume” and then we had the rest of the industry’s “Loudness Normalisation”, which isn’t standardised across media types. So, not only do we have different implementations within the hardware/software of devices but we also have different delivery specifications for the audio content for different distributors. For example, TV broadcast is mandated (by law in some countries) at about -23LUFS integrated loudness, while for example Netflix specifies -27LUFS which isn’t “integrated loudness” but integrated Dialnorm loudness. The result could easily be that Netflix is 8LUFS lower loudness. Apple Music on the other hand, requires Atmos mixes at -18LUFS. BluRay is often also -27LUFS but that could be 4-5LUFS higher than Netflix’s -27LUFS dialogue anchored loudness and it’s not uncommon for BluRay disks to have -24LUFS integrated loudness.
> 
> ...



I did have a D-555 and D-828K series more recently to keep comparing.  It becomes more a point of diminishing returns as their capacitors age and tracking becomes an issue.  More power to the audiophiles still trying to listen to these earlier digital components. But I do also have some nostalgia of listening to my favorite music having more recession in vocals and more pronounced instruments in max "surround" vs what I can hear in headphones now with my local storage devices.  Then again, overall quality now seems better as I listen through better headphones that provide a sense of diffusion and greater dynamics due to harmonized FR curves.

We've previously had our differences about studio mastering and home distribution when it comes to audio and video.  I'm more the expert with VFX and video: what I do think amazing right now is that home media is just as good if not better with many cinemas when it comes to visuals.  Dynamic range of current flat panel TVs surpasses most projectors, and Dolby Vision is a great standard for grading.  My audio system for home theater is also more advanced than other multiplexes in my area: kind of surprised that there's a limited number that are "Dolby Cinema" who do have true 3D audio...most being the regular 2D surround I had with blu-ray.  From what I'm gathering with movie titles I compare, I don't really think there is a true standard for home media.  Even though the processing itself for Dolby Atmos and DTS:X are the same for cinema and home now....think most differences are sound levels.  So one recent example is Top Gun:Maverick, not a great movie, but awesome cinematography.  I recently received it on disc.  The sound is extremely dynamic and subwoofer bass full at my amp being 60% (my amp now going from 0-100 in scale).  If I try watching its digital equivalent, at least 65% for overall volume, but still not as much subwoofer trim and I don't think as much treble.  These are my subjective perceptions, but I know everyone would agree that the soundtrack of the disc vs the stream is different (and how much of it is just sound levels with a sound mix with that master).  Most the 4K movies I have on disc do sound optimal at "60" on my amp be it 3D audio or other blu-ray lossless.  But Disney has thrown a wrench with 4K discs: with their 4K Star Wars movies, they did set their own standard and lowered the sound level.  Theirs is more like what I've found with streaming Atmos for other movies.

RE: DTS-MA with concerts.  I was refering to it being the 5.1 surround mix vs PCM stereo mix.  So the one concert I have in blu-ray that may arguably be better in PCM stereo is a Police concert.  It does have better dynamics than the 5.1 track.  Otherwise, I've found many blu-ray concerts have better mixing on my surround system with the DTS-MA 5.1 track (be it vocals on center channel, instruments left and right, good bass with subwoofer, and subtle ambience in surround with crowd).


----------



## bigshot

I've got a Prince concert on blu-ray that has a couple of advanced audio formats... specifically, Atmos and Auro3D. But the best sounding mix is the regular stereo run through a Dolby Pro Logic DSP. I think that is how the film was released in theaters and it still sounds the best. If they don't mess with the stereo track, the Dolby Stereo encoding still works.


----------



## Davesrose

bigshot said:


> I've got a Prince concert on blu-ray that has a couple of advanced audio formats... specifically, Atmos and Auro3D. But the best sounding mix is the regular stereo run through a Dolby Pro Logic DSP. I think that is how the film was released in theaters and it still sounds the best. If they don't mess with the stereo track, the Dolby Stereo encoding still works.


I remember you asked about Dolby Atmos....do you have a Dolby Atmos system now bigshot?  That's cool that your disc has Atmos and Auro3D.  My system actually supports Auro....but since I have no source that supports that, I've just used it for matrixed surround with various titles (find mono and stereo especially sound good with forced Auro).  There are some titles in Europe that have Auro, but it never made it to the US when it comes to media.  So I do have one remastered concert in BD that's 4K Dolby Atmos: INXS.  My system does just default it to Atmos, and it sounds fine.  Can't say there's any difference than the other concerts I have in BD that's DTS-MA 5.1.  They all have good surround sound that pans around you: and nothing that's above you.  To date, the only main musical where I've heard a height channel is Bohemian Rhapsody (where they have an orbit shot of Live Aid.


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## 71 dB (Nov 5, 2022)

bigshot said:


> I've got a Prince concert on blu-ray that has a couple of advanced audio formats... specifically, Atmos and Auro3D. But the best sounding mix is the regular stereo run through a Dolby Pro Logic DSP. I think that is how the film was released in theaters and it still sounds the best. If they don't mess with the stereo track, the Dolby Stereo encoding still works.


How are those advanced audio formats produced? I am not a fan of multichannel sound made from stereo track using whatever phase tricks myself. These things have to be done carefully from the start.


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## Joe Bloggs

71 dB said:


> How are those advanced audio formats produced? I am not a fan of multichannel sound made from stereo track using whatever phase tricks myself. These things have to be done carefully from the start.


Me on the other hand have been experimenting with every phase trick under the sun to create multichannel sound from stereo tracks programmatically.  Wanna try?


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## 71 dB (Nov 6, 2022)

Joe Bloggs said:


> Me on the other hand have been experimenting with every phase trick under the sun to create multichannel sound from stereo tracks programmatically.  Wanna try?


Yes, I wanna try.


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## Whitigir

Joe Bloggs said:


> Me on the other hand have been experimenting with every phase trick under the sun to create multichannel sound from stereo tracks programmatically.  Wanna try?


Isn’t this a part of 360 virtual sounds by Sony ?


----------



## Joe Bloggs

71 dB said:


> Don't ask my money. I am poor.


Don't ask me to monetize my work, I'm too dumb and tired to try 🙃
I can't even give this stuff away for free if they're not serious about trying it and they have a suitable PC platform, hence the question.


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## 71 dB (Nov 6, 2022)

Joe Bloggs said:


> Don't ask me to monetize my work, I'm too dumb and tired to try 🙃
> I can't even give this stuff away for free if they're not serious about trying it and they have a suitable PC platform, hence the question.


I am a Mac mini user if that answers your question.


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## bigshot

Davesrose said:


> I remember you asked about Dolby Atmos....do you have a Dolby Atmos system now bigshot?


No, but these disks don’t have a 5.1 track, so the Atmos fold down is the one I can use. But it seems that they didn’t have multitrack masters to mix from, so they EQed the elements out to separate them. The surround on the Dolby Stereo sounded better.


----------



## Davesrose

bigshot said:


> No, but these disks don’t have a 5.1 track, so the Atmos fold down is the one I can use. But it seems that they didn’t have multitrack masters to mix from, so they EQed the elements out to separate them. The surround on the Dolby Stereo sounded better.


Well if it has an Atmos track, then that's the one that's also multitrack mix.  The standards now are that Atmos on streaming is Dolby Digital+ 5.1 and the Atmos positional metadata (so older HDMI receivers will not recognize the Atmos stream and see DD+).  There are more UHD blu-ray discs with Dolby Atmos tracks, but there are also some HD blu-rays with Atmos.  With them, their core is usually a core TrueHD 7.1 channel setup with Atmos metadata.


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## bigshot

Yes, I was pointing out this Prince concert was like your Police one.


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## VNandor

71 dB said:


> Sorry, but your "Member of the Trade" card indicates you are after our money.
> I don't think I have a suitable PC platform because I am a Mac mini user...


You're quick to write him off for his member of the trade card. Instead of getting caught up in cables/dacs/amps dont make a difference kind of posts for the nth time, he makes actually useful posts such as this or this. Here's some more evidence of his DSP wizardry. Anyways, he's definitely not posting here to sell you DAPs.


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## Davesrose

bigshot said:


> Yes, I was pointing out this Prince concert was like your Police one.


Got it now...it's good to have those options.  Have found that besides the Police blu-ray I have, other concerts surround tracks sound better.  Since you mention the Prince disc also has Auro3D, is it Sign O The Times?  I might try seeing if I can pick it up just to see what an encode of it sounds like.  I'm now usually defaulting to DTS:Neural as my 3D matrix format when playing 5.1 surround sources (but will sometimes switch to stereo with some old mono movies).  It and Dolby's matrix 3D will make use of all 7.1.4 channels I have.  If I do make it a point to try cycling between Dolby, DTS, and Auro3D with sources, sometimes the Auro3D sounds better and might have more heights (but it also goes to 5.1.4).


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## Davesrose

71 dB said:


> How are those advanced audio formats produced? I am not a fan of multichannel sound made from stereo track using whatever phase tricks myself. These things have to be done carefully from the start.


I would think it safe to assume most every disc advertising multichannel sound is a track that was remixed in surround.  But the matrixing for a stereo track to multichannel on a home receiver has gotten pretty convincing.  Or also getting height effects off a 5.1 track if you have a 3D receiver (I noticed that when I upgraded my speaker system and heard helicopter chopper blades on top of me from a 5.1 track).  With a stereo track, I do get some ambience around me (probably not as much distinct sounds in back of me as a surround track would have).  Most surround receivers now have the Dolby Surround matrix technology, and DTS: Neural.  Some also have Auro (which is a European company that has had some more sources there...there's not much of any in Auro tracks in the US).


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## bigshot (Nov 6, 2022)

I think the problem with Sign O The Times is that the multitrack masters didn't exist. They EQed out certain elements and then mixed them back in over the top to place them in a different spot. It sounds really gimmicky. I don't think it represents the format at all.

You might check to see if the stereo of your Police concert is Dolby Stereo. If so, try turning on Dolby Pro Logic. If it's encoded that way, you'll get an additional center and rear channel, which is better than just plain stereo. I've found that a lot of movies and TV shows are encoded in Dolby Stereo even if the logo doesn't appear on the cover.


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## 71 dB

VNandor said:


> You're quick to write him off for his member of the trade card. Instead of getting caught up in cables/dacs/amps dont make a difference kind of posts for the nth time, he makes actually useful posts such as this or this. Here's some more evidence of his DSP wizardry. Anyways, he's definitely not posting here to sell you DAPs.


My mistake. I apologise.


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## Davesrose (Nov 6, 2022)

bigshot said:


> I think the problem with Sign O The Times is that the multitrack masters didn't exist. They EQed out certain elements and then mixed them back in over the top to place them in a different spot. It sounds really gimmicky. I don't think it represents the format at all.
> 
> You might check to see if the stereo of your Police concert is Dolby Stereo. If so, try turning on Dolby Pro Logic. If it's encoded that way, you'll get an additional center and rear channel, which is better than just plain stereo. I've found that a lot of movies and TV shows are encoded in Dolby Stereo even if the logo doesn't appear on the cover.


New A/V receivers don't have Dolby Pro Logic (II).  To make things more confusing, "Dolby Surround" is the new surround matrix. (that will matrix to height channels as well).  DTS:Neural is the DTS equivalent.  When I watch that Police concert, I will matrix to DTS:Neural.  My receiver's display is pretty nice for checking source: it will display Multi-Chanel, TrueHD, DD+. DTS-MA, etc + DTS:Neural:X or Dolby Surround depending on what surround mode I've got set (if it's a Dolby Atmos track, then it will just decode that and just show that....also DTS:X).  If the source is stereo, then it just displays DTS:Neural:X or Dolby Surround.  I can also see the complete number of channels and source format if I use its app on my phone.

I did see that Sign O The Times doesn't have Auro3D in the US release, but does for the European version.  I've ordered it on Amazon UK.  Will be interesting to see how the Auro track sounds if it's natively decoded.


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## bigshot

Ack! No Dolby Pro Logic?! There's an ocean of content out there encoded in Dolby Stereo. That is nuts.


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## Davesrose (Nov 6, 2022)

bigshot said:


> Ack! No Dolby Pro Logic?! There's an ocean of content out there encoded in Dolby Stereo. That is nuts.


Dolby Surround is the successor to Dolby Pro Logic II.  Like Pro Logic being able to matrix surround with regular stereo or stereo with Dolby Stereo.  Pro Logic II matrixed stereo to 5.1.  Dolby Surround will matrix to 7.1.4 on my setup (whether source is stereo or 5.1).


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## bigshot (Nov 6, 2022)

Can you make Dolby Stereo decode to the way it was encoded to sound? That would be three (later 5) front channels and one rear channel split mono between the two rear speakers. That is the way that the Sign O The Times sounds best, probably because that is the way it was mixed to sound. Splitting it off to more channels doesn't make it better. I can run it through the more modern DSPs and it doesn't sound as organized as through Dolby Pro Logic.

For years and years that 4 channel Dolby Stereo system was standard in theaters. I think the first one was "A Star is Born" in 1976 or so, by 1979 (largely because of theaters gearing up to run "Star Wars") it became a standard. Just about all regular movies were encoded that way until digital projection pushed out 35mm around 2000. That's almost a quarter century of movies that were encoded in that system. DSPs to take it further are fine, but an AVR really should be able to decode a film to sound the way it was mixed to sound too.

My Yamaha AVR is 7.1 and it has DSPs for Dolby Stereo, Dolby Pro Logic II and Dolby 5.1 (regular and Master Audio decode automatically).


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## Davesrose

bigshot said:


> Can you make Dolby Stereo decode to the way it was encoded to sound? That would be three (later 5) front channels and one rear channel split mono between the two rear speakers. That is the way that the Sign O The Times sounds best, probably because that is the way it was mixed to sound. Splitting it off to more channels doesn't make it better. I can run it through the more modern DSPs and it doesn't sound as organized as through Dolby Pro Logic.
> 
> For years and years that 4 channel Dolby Stereo system was standard in theaters. I think the first one was "A Star is Born" in 1976 or so, by 1979 (largely because of theaters gearing up to run "Star Wars") it became a standard. Just about all regular movies were encoded that way until digital projection pushed out 35mm around 2000. That's almost a quarter century of movies that were encoded in that system. DSPs to take it further are fine, but an AVR really should be able to decode a film to sound the way it was mixed to sound too.
> 
> My Yamaha AVR is 7.1 and it has DSPs for Dolby Stereo, Dolby Pro Logic II and Dolby 5.1 (regular and Master Audio decode automatically).


Dolby Digital was actually introduced earlier than that: first movie with it was Batman Returns.  the digital source was read in the optical print.  It’s been ages since I’ve had a receiver that does original Dolby Pro Logic (having 4 channels): it was most popular with prerecorded VHS tapes.  By the time I invested in a DVD system, my reciever supported Dolby Digital and Pro Logic II (which came out in 2000 and matrixed to 5.1 channels).  More of my movie titles were Dolby Digital then as stereo.  Then when I got a plasma HDTV, I got a lossless HDMI Harman Kardon receiver.  It’s matrix surround was either Harman’s own processor or Dolby Pro Logic II.  The current Dolby implementation is known as Dolby Surround and was first introduced in 2014.  Just as I couldn’t set number of speakers output with Pro Logic II, I can’t with Dolby Surround (unless I actually disable them I suppose).  But it really doesn’t make sense anyway: if the source is stereo, Surround doesn’t do crazy things to move sound around you.  With old mono movies, though, I’ll normally set them to stereo since speech might not be centered in Surround.


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## bigshot (Nov 6, 2022)

Pro Logic II automatically decodes the old Dolby Stereo into the four channels and Pro Logic II into 5.1. It doesn't add anything else to the old Dolby Stereo. I have both decoders on my Yamaha, but there isn't any difference playing Dolby Stereo through both. I'm curious if Dolby Surround just intelligently applies the decoding depending on the signal, knowing whether it should be 4 channels or more. It would be fine if it just spread the same four channels over more speakers, but it wouldn't be good if it is performing additional processing, like splitting off bits of channels for the additional speakers or adding phase and reverb tricks that aren't part of the original mix. It's easy to check for that by just playing a Dolby Stereo encoded track with Dolby Surround activated and stand in the rear of the room. All the sound you hear coming from the rear should be the same, not divided into left or right or Atmos elevation.

I suspect that they just put it all in one button and it figures out what the intended codec to use based on the signal. It probably still decodes Dolby Stereo, Pro Logic and Pro Logic II, it just doesn't tell you which one it's using.

I've gotten into the habit of checking the old Dolby formats with movies from 1975 to 2000. Often the Dolby Stereo encoding is intact and is better mixed than the modern 5.1 mixes. Maybe selecting the stereo track and engaging Dolby Surround will give you the same result.

One other thing I've noticed is that some music albums, beginning in the mid to late 80s, seem to be mixed for Dolby Stereo. Engaging the DSP puts the vocals in the middle and adds effects in the rear channel. It makes no difference for other albums. There were albums that were specifically released as being in Dolby Surround (Tomita in particular) that were re-released on CD without the Dolby logo, but the master is clearly encoded in Dolby Surround. These are encoded identically to the Dolby Stereo movie format. http://www.surrounddiscography.com/dolby/dolby.htm

By the way, I highly recommend the Charles Gerhart / National Philharmonic classic film scores CDs. They sound great in surround.


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## castleofargh

bigshot said:


> Can you make Dolby Stereo decode to the way it was encoded to sound? That would be three (later 5) front channels and one rear channel split mono between the two rear speakers. That is the way that the Sign O The Times sounds best, probably because that is the way it was mixed to sound. Splitting it off to more channels doesn't make it better. I can run it through the more modern DSPs and it doesn't sound as organized as through Dolby Pro Logic.
> 
> For years and years that 4 channel Dolby Stereo system was standard in theaters. I think the first one was "A Star is Born" in 1976 or so, by 1979 (largely because of theaters gearing up to run "Star Wars") it became a standard. Just about all regular movies were encoded that way until digital projection pushed out 35mm around 2000. That's almost a quarter century of movies that were encoded in that system. DSPs to take it further are fine, but an AVR really should be able to decode a film to sound the way it was mixed to sound too.
> 
> My Yamaha AVR is 7.1 and it has DSPs for Dolby Stereo, Dolby Pro Logic II and Dolby 5.1 (regular and Master Audio decode automatically).


I think the idea now is to output to whatever it is you have as a setup. It's the principle behind all modern DTS and Atmos stuff, where the "sound objects" are anywhere they want and a device decides how to feed the channels you have for the subjective placement to match the "object's" location as best as possible. 
Then you get some variations where the sound objects are already a speaker setup(5.1 or more). In which case, each channel goes to each actual audio channel and we're done. I get the feeling that most stuff coming as atmos on TV or from older movies, tends to be good old 5.1 or 7.1 with a fancy atmos logo. But that's fine too. 5.1 well done can be glorious. 
In such a case, if you have a setup with more speakers, there might be some option to leave it at 5.1 or to populate all the speakers with sound. But the modern concepts clearly is to turn anything into as many channels you have.



P.S. If at some point you decide to discuss multichannel stuff in a thread that has anything to do with it, I won't complain.


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## bigshot (Nov 6, 2022)

Dolby Stereo doesn't have sound objects to place since it's only 2 channels. It has specific channel information encoded with different phase. The only way to decode it is with phase. And just like you've noticed that a lot of stuff branded Atmos is actually 5.1 or 7.1, a lot of stuff labelled stereo is actually encoded with Dolby Stereo with a center and surround channel.

If you happen to have the first blu-ray release of "The Changeling" with George C. Scott, try listening to the stereo track with Dolby Surround on. That disk was released with a messed up 5.1 track but an intact Dolby Stereo surround track. They later released a corrected disk that had a proper 5.1 track, but the stereo track no longer decoded properly. I kept the defective disk because the stereo surround was MUCH better than the corrected 5.1 surround. There was a release of Argento's "Suspiria" that unintentionally had a brilliant Dolby Stereo surround track too. I would bet that earlier releases of Star Wars would include the Dolby Stereo soundtrack on that. I remember being blown away by it when it first came out in theaters. You can do a lot with Dolby Stereo.

I think blu-ray Atmos is essentially 10 channels.


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## Davesrose

bigshot said:


> Pro Logic II automatically decodes the old Dolby Stereo into the four channels and Pro Logic II into 5.1. It doesn't add anything else to the old Dolby Stereo. I have both decoders on my Yamaha, but there isn't any difference playing Dolby Stereo through both. I'm curious if Dolby Surround just intelligently applies the decoding depending on the signal, knowing whether it should be 4 channels or more. It would be fine if it just spread the same four channels over more speakers, but it wouldn't be good if it is performing additional processing, like splitting off bits of channels for the additional speakers or adding phase and reverb tricks that aren't part of the original mix. It's easy to check for that by just playing a Dolby Stereo encoded track with Dolby Surround activated and stand in the rear of the room. All the sound you hear coming from the rear should be the same, not divided into left or right or Atmos elevation.
> 
> I suspect that they just put it all in one button and it figures out what the intended codec to use based on the signal. It probably still decodes Dolby Stereo, Pro Logic and Pro Logic II, it just doesn't tell you which one it's using.
> 
> ...


No, Pro Logic II matrixes stereo to 5.1 sound (whether it has a Dolby stereo encode or is a stereo source) Dolby Pro Logic II .  AFAIK, some vinyl records were quadrophonic stereo…which would be a different format that 4 channel Dolby Stereo (that was left, center, right, surround).

I think at this point, movies that are 4.0 Dolby Stereo is pretty academic since most all titles were converted to 5.1 by DVD and then blu-ray.  More movies in the 70s and 80s advertising Dolby Stereo were also mastered in 6 channel (6 channels were present in a movie’s 70mm blow up).  Such was the case for movies like Star Wars or Amadeus. So watching Amadeus on BD with its TrueHD 5.1 track is pretty much as the author intended.  Since there was also confusion about Dolby Pro Logic surround receivers being able to matrix regular stereo and Dolby Stereo to surround (old analog surround receivers outputting 4.0, digital Pro Logic II ones outputting 5.1), I don’t think you can say that home users were getting presentations as intended (well also, plenty of theater goers would go to an older theater to see a movie in stereo and not get the 6 channel master of a 70mm projector with Dolby processor).

And again, the processing of a stereo source on Dolby Surround or DTS:Neural is pretty minimal.  You’re not getting unusually loud sounds on rear left or right.  Sometimes I hear some ambience…but nothing drastic.


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## Davesrose

bigshot said:


> Dolby Stereo doesn't have sound objects to place since it's only 2 channels. It has specific channel information encoded with different phase. The only way to decode it is with phase. And just like you've noticed that a lot of stuff branded Atmos is actually 5.1 or 7.1, a lot of stuff labelled stereo is actually encoded with Dolby Stereo with a center and surround channel.
> 
> If you happen to have the first blu-ray release of "The Changeling" with George C. Scott, try listening to the stereo track with Dolby Surround on. That disk was released with a messed up 5.1 track but an intact Dolby Stereo surround track. They later released a corrected disk that had a proper 5.1 track, but the stereo track no longer decoded properly. I kept the defective disk because the stereo surround was MUCH better than the corrected 5.1 surround. There was a release of Argento's "Suspiria" that unintentionally had a brilliant Dolby Stereo surround track too. I would bet that earlier releases of Star Wars would include the Dolby Stereo soundtrack on that. I remember being blown away by it when it first came out in theaters. You can do a lot with Dolby Stereo.
> 
> I think blu-ray Atmos is essentially 10 channels.


Atmos is usually positional metadata on top of 5.1 channels for streaming DD and 7.1 TrueHD for discs


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## bigshot (Nov 6, 2022)

Nope. They didn’t translate to 5.1. The 5.1 mixes are usually entirely different and are created by a totally different team than the original soundtrack.  If I have to upgrade my AVR, I’ll definitely look for a way to decode Dolby Stereo properly. I’m sure it’s possible. There are too many films from this period that sound better in the original sound than the remixed surround.

Dolby Surround CDs aren’t quad. They have L, R, Center, and a mono Rear. You shouldn’t be hearing any difference between the two rears. The rear is mono in this configuration. Maybe you’re misunderstanding what I’m talking about. It’s a stereo signal that decides to three front channels and one rear. Pro Logic II added split rears left and right and a sub channel to simulate 5.1. That came later and wasn’t as widely implemented as four channel Dolby Stereo.

The only way to benefit from this kind of surround is to properly decode it using phase. You can do something close with a mono center and a Haffler Doss Sum and Difference Matrix System wiring of the speakers, but that isn’t practical for a system that needs to decode other surround types. Adding Atmos simulation over the top would be the same as regular stereo with Atmos upscaling.

I guess if what you’re saying is true about how your Dolby Surround button works, you’ll be unable to hear Sign O The Times with Dolby Stereo decoding. That’s a shame because the Auro and Atmos mixes suck on that one.


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## Davesrose (Nov 6, 2022)

bigshot said:


> Nope. They didn’t translate to 5.1. The 5.1 mixes are usually entirely different and are created by a totally different team than the original soundtrack.  If I have to upgrade my AVR, I’ll definitely look for a way to decode Dolby Stereo properly. I’m sure it’s possible.
> 
> Dolby Surround CDs aren’t quad. They have L, R, Center, and a mono Rear.
> 
> The only way to benefit from this kind of surround is to properly decode it using phase. You can do something close with a mono center and a Haffler Doss Sum and Difference Matrix System wiring of the speakers, but that isn’t practical for a system that needs to decode other surround types.



My point is that 6 channel mastering and 5.1 surround schemes were mastered for movies before the 2000s.  The studio did have to have different mixes for 35mm prints, and theaters that were mono, stereo, Dolby Stereo, or Dolby Stereo 6 track.  If the movie was already 6 channel, then it wasn't a great effort to go 5.1 for home distribution.

RE never upgrading your receiver.  Again, all my receivers meant for DVD on to BD on to UHD have been Pro Logic II, which has put a discrete channel on left surround and right surround.  When I switch between Dolby Surround or DTS:Neural, main differences I hear are in their EQ (with DTS weighting to bass a bit more).  And again: with the processing, the surround is very subtle....I really don't think if you could compare Dolby Pro Logic I and Pro Logic II or Surround, you'd be hearing much difference between surround left and surround right.  Especially a music BD.  For a disc like the Police, I'll watch the stereo track in stereo.  Or if I switch to DTS:Neural or Dolby Surround, there might just be some amorphous crowd noises (no distinct sound surround left or right).

I'll be interested more about how the Auro track sounds for Sign O The Times.  It does sometimes have a nice effect if matrixing a stereo, Dolby, or DTS track...but I think it would also sound wonky on a system that doesn't have Auro decoding (as their marketing says their system is very different than Dolby or DTS).


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## bigshot (Nov 6, 2022)

Just to clarify, Dolby Stereo is a two channel track which when decoded throws all of the info common to both channels to the center speaker, subtracting it from the left and right. Then it adds a rear channel which is inverse phase to the front. On decoding, all out of phase sound goes to the rear two speakers. Almost all movies during that time were released this way. Only the blockbusters had six track mixes. So with Indiana Jones, it might not make a difference, but with The Changeling which was never in six track, it makes a big difference.

I don’t think a lot of these were archived as four tracks. And the stems survive sometimes, so they can create a totally new surround mix. That can either be better or worse depending on the quality of the engineering. But if you want to hear the original surround mix, the only place it survives often is in the two channel Dolby Stereo track. It’s like The Beatles… there are new mixes, but if you want to hear the ones The Beatles themselves did, it’s the mono box.


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## Davesrose (Nov 6, 2022)

bigshot said:


> Just to clarify, Dolby Stereo is a two channel track which when decoded throws all of the info common to both channels to the center speaker, subtracting it from the left and right. Then it adds a rear channel which is inverse phase to the front. On decoding, all out of phase sound goes to the rear two speakers.


I know that Dolby Stereo and Dolby Pro Logic I were for analog systems where stereo was matrixed to L,C,R,surround.  I've been pointing out that Pro Logic also matrixes standard stereo.  Dolby Pro Logic II upgraded that to matrix to L,C,R, SL, SR, and subwoofer.

To get back to your premise that movies should be presented in 4.0 Dolby Stereo…you mentioned Star is Born 1976.  If you look at its DVD title, it was 5.1 Dolby Digital, and the BD is DTS-MA 5.1.  Even with it, there’s no way to try getting 4.0 on a receiver that’s digital (be it Dolby Digital vintage or HDMI DTS-MA/TrueHD vintage).  Or you were wondering about Star Wars.  That existed in Dolby Stereo with VHS tapes.  Even later editions of it on laserdisc were Dolby Digital.  By the time they made it to DVD, it was the special editions: which had film restorations and a new sound mix in 5.1.  Now they’ve done another sound mix with Dolby Atmos for UHD.  A VHS image certainly sucks on new large image displays.  And Atmos improves the original presentation.  Now with the begining scene of the star destroyer above you, you actually hear sounds above you…your not relying on visual queues.


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## Elegiac

This seems to make sense to me.


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## 71 dB

Elegiac said:


> This seems to make sense to me.



Hans makes more sense here than earlier. He kind of confesses that hi res makes no difference, but still tries to build a case for it by saying ONLY the most expensive DACs can perform oversampling well enough to overcome the problem of brickwall-reconstruction filtering with 44.1 kHz. I don't buy that claim. Sorry.


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## Elegiac

71 dB said:


> Hans makes more sense here than earlier. He kind of confesses that hi res makes no difference, but still tries to build a case for it by saying ONLY the most expensive DACs can perform oversampling well enough to overcome the problem of brickwall-reconstruction filtering with 44.1 kHz. I don't buy that claim. Sorry.


Why not? I have no dog in this race, except my amendable opinion.

I have an album that sounds better to me in 24/44 than in 16/44. The problem is that it's with different sources. 
I'll be getting a new DAC any day now, and then I'll run the two different files through the same DAC, Amp and headphones to better compare.


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## 71 dB

Elegiac said:


> Why not? I have no dog in this race, except my amendable opinion.


Why not what? My only dog in this race is the truth.



Elegiac said:


> I have an album that sounds better to me in 24/44 than in 16/44. The problem is that it's with different sources.


So it is the _source_, not the bit depth. If that 24/44.1 version is converted properly (using proper dithering) to 16 bit it will sound just as good. The accuracy of the sound isn't affected one bit (pun intended), only the background noise gets higher, but is still well below audibility in any reasonable listening scenario. Hans himself admits 14 bits (or even 12!) is enough. I say 13 bits is enough for consumer audio. 24 bit is for music production where dynamic headroom is beneficial.



Elegiac said:


> I'll be getting a new DAC any day now, and then I'll run the two different files through the same DAC, Amp and headphones to better compare.


Okay, but if they are different mixes all you will hear is different mixes...


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## Joe Bloggs (Nov 7, 2022)

bigshot said:


> Just to clarify, Dolby Stereo is a two channel track which when decoded throws all of the info common to both channels to the center speaker, subtracting it from the left and right. Then it adds a rear channel which is inverse phase to the front. On decoding, all out of phase sound goes to the rear two speakers. Almost all movies during that time were released this way. Only the blockbusters had six track mixes. So with Indiana Jones, it might not make a difference, but with The Changeling which was never in six track, it makes a big difference.
> 
> I don’t think a lot of these were archived as four tracks. And the stems survive sometimes, so they can create a totally new surround mix. That can either be better or worse depending on the quality of the engineering. But if you want to hear the original surround mix, the only place it survives often is in the two channel Dolby Stereo track. It’s like The Beatles… there are new mixes, but if you want to hear the ones The Beatles themselves did, it’s the mono box.


As far as I can tell Dolby Stereo is just plain old stereo mastered with a view to being surround upmixed by a particular algorithm of its time.  I would be interested to hear what you think of it upmixed through the algorithm I've settled on for the past few years . Firstly, which is the particular version of this album where this original Dolby Stereo exists?


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## bigshot

Again, I’m not saying it’s important to have Dolby Stereo decoding for big pictures like Star is Born or Star Wars. I’m saying it’s an important alternative for the thousands of smaller movies released in that format where the elements might not survive to make a proper 5.1 remix. It’s clear that the stems no longer exist for Sign o The Times and the 5.1 remix on The Changeling is vastly inferior to the original. These are just two examples, but there are a lot more films where the Dolby Stereo four channel surround mix done for the original theatrical release is better than the recent 5.1 remix for home video. The threads on the home video forums have lots more examples. There are also examples of TV shows released in Dolby Stereo that haven’t been released in multichannel on home video. But if you engage the Dolby pro Logic DSP, you can hear them in surround.

I think it’s important to be able to decode that original format, not just slap processing over the top without properly decoding it first. They had a lot more time and budget to mix films for first release than home video companies do for Blu-ray release. It shouldn’t be surprising that sometimes the original mix is better than the new one.


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## Joe Bloggs

bigshot said:


> Again, I’m not saying it’s important to have Dolby Stereo decoding for big pictures like Star is Born or Star Wars. I’m saying it’s an important alternative for the thousands of smaller movies released in that format where the elements might not survive to make a proper 5.1 remix. It’s clear that the stems no longer exist for Sign o The Times and the 5.1 remix on The Changeling is vastly inferior to the original. These are just two examples, but there are a lot more films where the Dolby Stereo four channel surround mix done for the original theatrical release is better than the recent 5.1 remix for home video. The threads on the home video forums have lots more examples. There are also examples of TV shows released in Dolby Stereo that haven’t been released in multichannel on home video. But if you engage the Dolby pro Logic DSP, you can hear them in surround.
> 
> I think it’s important to be able to decode that original format, not just slap processing over the top without properly decoding it first. They had a lot more time and budget to mix films for first release than home video companies do for Blu-ray release. It shouldn’t be surprising that sometimes the original mix is better than the new one.


Do you listen to stereo music through surround or stereo?


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## bigshot

Joe Bloggs said:


> which is the particular version of this album where this original Dolby Stereo exists?


I posted a link to CDs that are encoded in Dolby Surround above. Most are out of print, but you can easily find used copies for cheap. Pick up a few of those. They’ll give you plenty to experiment with.


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## bigshot (Nov 7, 2022)

Joe Bloggs said:


> Do you listen to stereo music through surround or stereo?


That depends on the recording. Some mixes are better in the new 5.1 remixes, some are better with the original release stereo mix, a few that were encoded for it are better in matrixed surround, and some are better in my AVR’s DSP that upmixes stereo to 5.1. There are even some really dry recordings that I apply a hall ambience DSP to give them a little depth that the recording is lacking. There are even some things that are better in mono. No one size fits all.


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## Joe Bloggs

bigshot said:


> Again, I’m not saying it’s important to have Dolby Stereo decoding for big pictures like Star is Born or Star Wars. I’m saying it’s an important alternative for the thousands of smaller movies released in that format where the elements might not survive to make a proper 5.1 remix. It’s clear that the stems no longer exist for Sign o The Times and the 5.1 remix on The Changeling is vastly inferior to the original. These are just two examples, but there are a lot more films where the Dolby Stereo four channel surround mix done for the original theatrical release is better than the recent 5.1 remix for home video. The threads on the home video forums have lots more examples. There are also examples of TV shows released in Dolby Stereo that haven’t been released in multichannel on home video. But if you engage the Dolby pro Logic DSP, you can hear them in surround.
> 
> I think it’s important to be able to decode that original format, not just slap processing over the top without properly decoding it first. They had a lot more time and budget to mix films for first release than home video companies do for Blu-ray release. It shouldn’t be surprising that sometimes the original mix is better than the new one.


what I mean is that from what I understand of Dolby Stereo, you can't really "encode" or "decode" for it in the sense that it is usually understood today.  As noted by others, engaging the Pro Logic decode "decodes" everything, not just Dolby Stereo "encoded" material.  There's no way for Dolby Stereo to identify itself as something special for the "decoder" to focus on, and that's because it isn't:  the decoder does what it does to all comers, and Dolby Stereo probably just stands for mixes where the engineers had actually listened to the upmix and found it good and/or applied particular tweaks to make it sound better through that upmixer at the time.


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## Joe Bloggs

bigshot said:


> Pro Logic II automatically decodes the old Dolby Stereo into the four channels and Pro Logic II into 5.1. It doesn't add anything else to the old Dolby Stereo. I have both decoders on my Yamaha, but there isn't any difference playing Dolby Stereo through both. I'm curious if Dolby Surround just intelligently applies the decoding depending on the signal, knowing whether it should be 4 channels or more. It would be fine if it just spread the same four channels over more speakers, but it wouldn't be good if it is performing additional processing, like splitting off bits of channels for the additional speakers or adding phase and reverb tricks that aren't part of the original mix. It's easy to check for that by just playing a Dolby Stereo encoded track with Dolby Surround activated and stand in the rear of the room. All the sound you hear coming from the rear should be the same, not divided into left or right or Atmos elevation.
> 
> I suspect that they just put it all in one button and it figures out what the intended codec to use based on the signal. It probably still decodes Dolby Stereo, Pro Logic and Pro Logic II, it just doesn't tell you which one it's using.
> 
> ...


I don't see Sign O the Times listed here?


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## bigshot

Pro Logic is the decoder being used back then. It’s the first iteration of Dolby Surround. Pro Logic II adds left and right to the rears and a sub channel for things encoded in Pro Logic II. I think Pro Logic II is backwards compatible. From what Daverose says, it appears that the new Dolby Surround DSP isn’t. And yes, you’re right, there’s no flag to indicate the type of encoding.


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## bigshot (Nov 7, 2022)

Joe Bloggs said:


> I don't see Sign O the Times listed here?


That is a movie. Almost all movies during the 80s and 90s had Dolby Surround audio. Big films had more elaborate multichannel tracks as well. Dolby Stereo was the bread and butter surround format that just about any theater could decode. Larger theaters had six track sound, which is more like 5.1.

Back in the day, the poster would have a little Dolby Stereo logo if it was being shown with matrixed surround. If the soundtrack hasn’t been remastered too much, the stereo track on home video release will still decode to matrixed surround.

In a nutshell, Pro Logic uses phase cancellation to isolate the material in the mix that is pure mono and sends it to the center channel. This is always dialogue/vocals. Then it throws all the out of phase content to the rear channels. I’ve never been able to figure out if it subtracts the mono from the front left and right. Maybe Gregorio knows that. It’s a pretty basic encoder/decoder, but it was used for over twenty years and it works pretty well, albeit with considerable spill over from channel to channel.


----------



## Elegiac

71 dB said:


> Why not what? My only dog in this race is the truth.
> 
> 
> So it is the _source_, not the bit depth. If that 24/44.1 version is converted properly (using proper dithering) to 16 bit it will sound just as good. The accuracy of the sound isn't affected one bit (pun intended), only the background noise gets higher, but is still well below audibility in any reasonable listening scenario. Hans himself admits 14 bits (or even 12!) is enough. I say 13 bits is enough for consumer audio. 24 bit is for music production where dynamic headroom is beneficial.
> ...


Why not the DAC thing. About only higher-end DAC's and the oversampling.

Hmm. I'm not sure about different mixes. They _should_ be the same mix. One on CD, and the other a FLAC file.
I won't be surprised if they sound the same, or different. Either way. Curious to see whether the CD _is_ more condensed, as I hear it.


----------



## Joe Bloggs

Elegiac said:


> Why not the DAC thing. About only higher-end DAC's and the oversampling.
> 
> Hmm. I'm not sure about different mixes. They _should_ be the same mix. One on CD, and the other a FLAC file.
> I won't be surprised if they sound the same, or different. Either way. Curious to see whether the CD _is_ more condensed, as I hear it.


Because the DAC-on-a-chip solutions that are advertised by TI, ESS, AKM etc. (and advertised as being on board any number of budget devices) already feature exemplary oversampling capabilities that would put flagship CD players of yesteryear to shame.


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## bigshot

Every format is remastered for that format. You can’t compare a CD to a download. Two different things. The way to compare is to take a high data rate file and bounce it down to a lower data rate.


----------



## Elegiac

Ah. Interesting.


----------



## 71 dB

Elegiac said:


> Why not the DAC thing. About only higher-end DAC's and the oversampling.


Because oversampling isn't "rocket science." There is not reason why only a few high-end DACs could do it well.



Elegiac said:


> Hmm. I'm not sure about different mixes. They _should_ be the same mix. One on CD, and the other a FLAC file.


Perhaps they _should_ be the same mix, but the notion of them sounding different to you indicates they are not. Having a different master for hi-res format is pretty common (otherwise people would notice no dfference).



Elegiac said:


> I won't be surprised if they sound the same, or different. Either way. Curious to see whether the CD _is_ more condensed, as I hear it.


Condensed as in dynamically more compressed?


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## 71 dB

bigshot said:


> Every format is remastered for that format. You can’t compare a CD to a download. Two different things. *The way to compare is to take a high data rate file and bounce it down to a lower data rate.*


Exactly!


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## Davesrose (Nov 7, 2022)

bigshot said:


> Pro Logic is the decoder being used back then. It’s the first iteration of Dolby Surround. Pro Logic II adds left and right to the rears and a sub channel for things encoded in Pro Logic II. I think Pro Logic II is backwards compatible. From what Daverose says, it appears that the new Dolby Surround DSP isn’t. And yes, you’re right, there’s no flag to indicate the type of encoding.



It is backwards compatible: Dolby considers the new Dolby Surround badge to be the successor to Dolby Pro Logic IIz (which took Pro Logic from processing standard stereo to 7.1.2).  And again, the Pro Logic decoder can process standard stereo to surround.  I really don't know if there are any blu-rays that have an old stereo track that was encoded in the "35mm Dolby Stereo" format of 4.0.  Movies that were surround were mixed to 5.1 (like Star is Born or Star Wars), and lower budget movies that had been standard stereo stayed stereo.  I'm looking at the specs of Sign O the Times: it says that the BD is PCM stereo-so it's not Dolby Stereo.  If you set your receiver to Dolby Pro Logic, then it's taking that stereo source and is processing to 5.1 (or more depending on your speaker setup).

The main addition with Dolby Surround I've heard is how it processes 5.1 sound to 7.1.4 surround.  I have heard scenes in which the camera is inside a helicoptor, and you hear the blades above you.  Or Master and Commander, where there are scenes in which you can hear sounds happening above deck.  I find that to be pretty impressive.  Yet it also doesn't go over the top with a source that's stereo (the sound field isn't much different than when I had a 7.1 setup using Harman Kardon surround or Dolby Pro Logic).

You said it was too bad that I can't listen to my Police BD on the old Dolby Stereo format, but it never had a Dolby Stereo track.  It's a reunion tour concert: where they could have recorded in stereo and/or multichannel.  The stereo track is PCM 2.0 (like all other concerts I have on blu-ray: PCM 2.0, or DTS-MA 5.1).  If on your system, you switch away from stereo mode and go to pro logic, it's processing that 2.0 track to at least 5.1.

What sparked our dialogue was me indicating that I liked that discs stereo track over the multi-channel (when for other BD concerts, I've enjoyed the multi-channel).  It's just the way that the stereo track was mixed to sound more dynamic and have use of bass.


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## Elegiac

71 dB said:


> Condensed as in dynamically more compressed?


Perhaps. But mostly I notice less detail. On the CD the guitar takes up less space.. it's spread-out less. It's more a solid mass.
On the 24bit files I can pick over the entire texture, peer into it. Get inside of it.
It's like the difference between looking at something from a distance, and looking at it up close.


----------



## 71 dB

Elegiac said:


> Perhaps. But mostly I notice less detail. On the CD the guitar takes up less space.. it's spread-out less. It's more a solid mass.
> On the 24bit files I can pick over the entire texture, peer into it. Get inside of it.
> It's like the difference between looking at something from a distance, and looking at it up close.


If you can, make a 16 bit version of the 24 bit file and listen if it still has these fine qualities.


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## ubs28

In music production, 24-bits is used as the minimum. So it not bad for music, it is how music is even created.

/close thread


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## 71 dB

ubs28 said:


> In music production, 24-bits is used as the minimum. So it not bad for music, it is how music is even created.
> 
> /close thread


24 bit is not "minimum" in music production. It is possible to produce good sounding music with only 16 bit (but one has to be careful about how dynamic range is used). It is a practical amount of bits in music production. It is however overkill for consumer audio considering 13 bits would be enough and we have 16 bit digital audio in common use.


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## bigshot (Nov 7, 2022)

ubs28​Elegiac​Please see the article in my sig file titled "CD sound is all you need". It clearly and thoroughly explains why 24 bit and high sampling rates are not beneficial to listening to music in the home. If that is too technical, the link to Mark Waldrop might be easier to parse. The original post in this thread does a good job of explaining it too. Based on your comments, I'm betting you haven't read that before replying to this thread.

If you have any interest in the subject, and want to understand the replies that you are receiving, take a moment and make an effort to understand. It's a waste of everyone's time for you to comment the same thing over and over without making any effort to process the replies you receive. A discussion is give and take. We are listening to you and replying to what you say. You need to do the same for us.


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## old tech (Nov 7, 2022)

Elegiac said:


> Perhaps. But mostly I notice less detail. On the CD the guitar takes up less space.. it's spread-out less. It's more a solid mass.
> On the 24bit files I can pick over the entire texture, peer into it. Get inside of it.
> It's like the difference between looking at something from a distance, and looking at it up close.


I am willing to bet it is all in your head. There is no difference in resolution, only the noise floor.

Watch this video for a practical demonstration, seriously, and then consider if the slight difference in noise (there is nothing else) between dithered 8bits and 16bits, is it possible for any human to pick a difference between 16 and 24 bits in a blind test?

https://productionadvice.co.uk/bit-depth-and-resolution/


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## old tech

ubs28 said:


> In music production, 24-bits is used as the minimum. So it not bad for music, it is how music is even created.
> 
> /close thread


As far as I am aware 24bits has not (generally) been used for music production for at least a couple decades now, mostly 32 bit float.

In any event headroom for music production is not the same requirement for music reproduction. 16 bits is already overkill for playback.


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## bigshot (Nov 7, 2022)

I’ve never heard of anyone who could discern any difference between 16 and 24 bit at normal listening levels. In order to hear a difference in noise floors, you would have to boost the volume level of 16 bit up to ear shattering volumes. Likewise, to tell the difference between 44.1 and 96, you would have to be able to hear frequencies humans can’t hear. Within the range covered by 16/44.1, the sound quality is identical to the same range in 24/96. There is no difference in detail, nor resolution between them audibly.

I don’t doubt you perceive something. It’s just that what you hear is due to comparing apples to oranges or perceptual error/ expectation bias.


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## Elegiac

old tech said:


> I am willing to bet it is all in your head. There is no difference in resolution, only the noise floor.
> 
> Watch this video for a practical demonstration, seriously, and then consider if the slight difference in noise (there is nothing else) between dithered 8bits and 16bits, is it possible for any human to pick a difference between 16 and 24 bits in a blind test?
> 
> https://productionadvice.co.uk/bit-depth-and-resolution/



It may all be in my head. I'm going to test it. Trust me, I have low expectations in these matters, and if anything, I'm biased towards CD. My ears are as skeptical as my brain  ...so I want to figure it out, not just assign some arbitrary reason for my perception.



bigshot said:


> ubs28​Elegiac​Please see the article in my sig file titled "CD sound is all you need". It clearly and thoroughly explains why 24 bit and high sampling rates are not beneficial to listening to music in the home. If that is too technical, the link to Mark Waldrop might be easier to parse. The original post in this thread does a good job of explaining it too. Based on your comments, I'm betting you haven't read that before replying to this thread.
> 
> If you have any interest in the subject, and want to understand the replies that you are receiving, take a moment and make an effort to understand. It's a waste of everyone's time for you to comment the same thing over and over without making any effort to process the replies you receive. A discussion is give and take. We are listening to you and replying to what you say. You need to do the same for us.


Lol, I'm just replying to replies, and making every effort to understand, which is why I'll view your links. Easy, tiger.

For example, I'd thought that what we had decided I was hearing differently was the difference between CD and digital download- because, as you said, the music would be remastered for CD- not the difference between 16 and 24 bit per se.


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## bigshot (Nov 7, 2022)

When I was puzzling stuff out, I found it useful to read about the basics of how digital audio works... Nyquist Theory, the difference between bit rate and sampling rate... basic stuff. Then you can understand what high data rate files are adding to the sound, and what is or isn't missing on a CD. It's also good to understand how to apply controls to listening comparisons to minimize the effects of bias and perceptual error, so you can focus on the sound itself, not the way your ears and brain are interpreting it.


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## old tech

Elegiac said:


> It may all be in my head. I'm going to test it. Trust me, I have low expectations in these matters, and if anything, I'm biased towards CD. My ears are as skeptical as my brain  ...so I want to figure it out, not just assign some arbitrary reason for my perception.


Make sure when you do test it that you are using the same mastering and level matched. Perhaps the easiest way of doing this is to download Foobar and the DBT plugin. Then all you need to do is use your 24bit file and the software will do the rest for you, i.e. convert a 16bit version, play random sections which you pick which you think is the 16bt version. At the end of the test it will provide you with a score, including the probability of just guessing.


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## Elegiac (Nov 8, 2022)

Alright, I get it. As much as I'm going to get without some serious study of audio science. Which I feel no urgency to pursue. I'm a student of the humanities... most specifically writing and history, but also cultural/philosophical ideas in general. What I mean to say is... while I may not understand down to the deepest level of this, I have a good nose for the truth, and an eye for what an unbiased set of ideas looks like. I don't always get it right, but where my instincts fail, my inquiring mind picks up the slack.
Anyway, nothing seems suspicious in what I've been presented with here.

Pretty much gels with what experience has taught me. There are crap-quality productions in 24 bit, and excellent-quality productions in 16 bit. The production means more than the sample rate and bit depth.


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## 71 dB

old tech said:


> As far as I am aware 24bits has not (generally) been used for music production for at least a couple decades now, mostly 32 bit float.


32 bit floating point is effectively 24 bit just like 314*10^-2 is pi given at 3 digit accuracy.


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## 71 dB

Elegiac said:


> Alright, I get it. As much as I'm going to get without some serious study of audio science. Which I feel no urgency to pursue. I'm a student of the humanities... most specifically writing and history, but also cultural/philosophical ideas in general. What I mean to say is... while I may not understand down to the deepest level of this, I have a good nose for the truth, and an eye for what an unbiased set of ideas looks like. I don't always get it right, but where my instincts fail, my inquiring mind picks up the slack.
> Anyway, nothing seems suspicious in what I've been presented with here.


You don't need to understand audio science, but the more you do the easier it is to see when an audio-related claim is bs (often for marketing/selling something people don't really need). Even us "math-heads" with scientific background have had to "convince" ourselves that there are limits beyond which sample rates and bitrates don't matter. That's because we are human beings too. Bigger numbers just "feel" better. It doesn't help that digital audio can be quite unintuitive. People who don't understand it well are easily mislead by intuitive thinking (such as "Surely 24 bit must mean more accurate sound than 16 bit!").



Elegiac said:


> Pretty much gels with what experience has taught me. There are crap-quality productions in 24 bit, and excellent-quality productions in 16 bit. The production means more than the sample rate and bit depth.


Yes. Sound quality comes pretty much entirely from music production. It is possible to create great sounding music using only 16 bit, but it requires that the dynamic range is handled carefully at every step which is cumbersome. That's why 24 bit is great in music production, but when the music has been mixed and mastered, consumers need only about 13 bits meaning 16 bit "CD quality" is more than enough. You can't make music sound better by increasing bit depth and sample rate beyond 16/44.1. You have to write/compose/perform/record/produce/mix/master your music better.


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## bigshot

One of the best sounding albums ever made was recorded, mixed and mastered entirely in 16/44.1... Donald Fagan's "The Nightfly".


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## BobG55

bigshot said:


> One of the best sounding albums ever made was recorded, mixed and mastered entirely in 16/44.1... Donald Fagan's "The Nightfly".


An independent station
WJAZ
With jazz and conversation
From the foot of Mt. Belzoni
Sweet music
Tonight the night is mine
Late line
Till the sun comes through the skylight


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## Davesrose

71 dB said:


> 32 bit floating point is effectively 24 bit just like 314*10^-2 is pi given at 3 digit accuracy.


Conceptually they're different.  Different number expressions and potential for many more values.  Question is if there's much need for it in audio.  This article brings up the potential application of utilizing its inherent flexibility.

https://www.wired.com/story/32-bit-float-audio-explained/

One application for 32bit float files that is pretty much a given is with 3D rendering.  As an image format, it has enough dynamic range to have realistic light simulation in any environment.  Digital cameras are getting up to 16bit exposure: so at least 3 bracketed exposures are merged to 32bit to create a lightmap: which is then used in the environment of a 3D scene.


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## 71 dB

Davesrose said:


> Conceptually they're different.


Yes. Floating point part is like having a measuring "stick" that has 1 km, 500 m, 200 m and 100 m modes. In 1 km mode you can measure distances up to 1 km, but only at 1 m accuracy. In 500 m mode you can measure up to 500 m, but with 50 cm accuracy and so on so that in 100 m you can measure 100 m distances with 10 cm accuracy. With sound this scaling property has less importance, because digital full scale is arbitrary anyway. It can mean 50 dB or 100 dB or whatever.


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## DaveStarWalker (Nov 12, 2022)

Personally, I don't know :

https://www.sciencedirect.com/science/article/abs/pii/S0531513101004940

https://en.m.wikipedia.org/wiki/Hypersonic_effect

Etc.

Perfectly happy with my 16/44.1 wave  disc and demat files... So I don't know... 🤔


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## bigshot (Nov 11, 2022)

The frequencies they are talking about there aren't particularly present in most commercially recorded music anyway.

Ultrasonic frequencies are used to disperse riots. https://en.wikipedia.org/wiki/Sonic_weapon


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## Killcomic

Wow, I’m gone for a couple of years and people here are still trying to convince others that they have dog hearing.


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## bigshot

We thought my dog was deaf because you can call her name and she doesn't react. But then we figured out that she was just ignoring us.


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## IanB52 (Dec 3, 2022)

Just want to point out that it is flawed reasoning to strictly equate digital bit depth with dynamic range in analog systems and especially against musical dynamics. In order for things to sound accurately real you need to be able to record not only the musical peak, but the transient peak, and _simultaneously_ the most subtle room acoustics and fine details at the very bottom of the scale, and perhaps even lower so as to provide a coherent rendering of the fine details.

That is the difference between bare minimum digital capture for music, and having good audio quality.

The other aspect to think about is whether your bandwidth and dynamic range are sufficient to avoid artifacts, and to allow less drastic implementation of filters and dither. The latter gets argued to death, but no designer is going to desire having zero margin for designing things like filters.

I would add one other thing, it is also flawed to look at audio from a pure frequency response point of view. Sounds also have amplitude envelope, which is not itself and audible frequency. From my understanding the human ear is not totally analogous to an FFT, and can detected changes in the amp envelope that are shorter in duration than a 20khz wave cycle. These are leading edges, transients etc, and the reason why you start to see squiggly lines when you filter a waveform. The conventional implication is that these are an optical illusion, but again that is thinking like an FFT. The ear also responds to very short pulses. This is the same problem with Monty using an oscilloscope to view steady state tones as opposed to complex recorded music, and also brushing off Gibbs ringing on square waves etc as irrelevant. The ringing itself may not be audible, but the effect of shifting that transient energy and distortion waveform edges should be.


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## bigshot (Dec 3, 2022)

Most commercial music has a dynamic range of around 35dB. The most dynamic commercial recordings don’t exceed 55dB and those can be uncomfortable to listen to. The noise floor of 16/44.1 is overkill when it comes to bit depth. I doubt even picky audiophiles really need much more than 12 bits in normal listening.

An impulse shorter than a 20kHz wave would be a higher frequency than 20kHz wouldn’t it? 20,000 cycles per second is a VERY short sliver of time. None of the transients of musical instruments are anywhere near that. Square waves do, but that is an illegal signal in digital audio and nothing like that exists in the real world.

Are you really arguing that 16/44.1 isn’t audibly transparent? I’d like to see listening tests that show that if you know of any.


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## IanB52 (Dec 3, 2022)

bigshot said:


> Most commercial music has a dynamic range of around 35dB. The most dynamic commercial recordings don’t exceed 55dB and those can be uncomfortable to listen to. The noise floor of 16/44.1 is overkill when it comes to bit depth. I doubt even picky audiophiles really need much more than 12 bits in normal listening.
> 
> An impulse shorter than a 20kHz wave would be a higher frequency than 20kHz wouldn’t it? 20,000 cycles per second is a VERY short sliver of time. None of the transients of musical instruments are anywhere near that. Square waves do, but that is an illegal signal in digital audio and nothing like that exists in the real world.
> 
> Are you really arguing that 16/44.1 isn’t audibly transparent? I’d like to see listening tests that show that if you know of any.


If there is any audio content at all part of the original recording that is below that 35 or 55db, then you obviously need greater dynamic range. I'm not necessarily arguing that 16 bit is inadequate in this regard, but rather than many people make inappropriate analogies.

I think we can agree though, that if you listen to mono recordings of synthesized sine waves with attack and release of slower than 1ms, and at a volume lower than the recording RMS volume,  then 16/44.1 is absolutely audibly transparent.

I'd rather not get deep into it at this time, but there is research showing that humans can hear between 10-20us in binaural audio signals. 20khz is 50us if you use an NOS DAC with full aliasing, and more like 300us with standard 44.1khz brickwall filter.

An addendum: My actual point is just to promote looking at these things from more than one dimension and not to make overly simplistic equivalencies. I feel I haven't been able to do that without inserting some heterodox claims and thoughts. You guys have your forum, and I'll just leave it there.


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## bigshot (Dec 4, 2022)

IanB52 said:


> If there is any audio content at all part of the original recording that is below that 35 or 55db, then you obviously need greater dynamic range.



Air conditioning hum from the venue? The recording generally has a much higher noise floor than the format, especially if elements are boosted in the mix. That raises the venue's noise floor. That's why they do fade ins at the beginning of tracks. If they didn't, you'd hear the sharp division between 16 bit silence and the room tone of the recording.



IanB52 said:


> I think we can agree though, that if you listen to mono recordings of synthesized sine waves with attack and release of slower than 1ms, and at a volume lower than the recording RMS volume,  then 16/44.1 is absolutely audibly transparent.



If you compare a 24/96 recording of music to a properly bounced down 16/44.1 file, it's transparent too. Have you ever found a study that showed people can hear the difference? All I've seen is that guy who looped fade outs and cranked the volume to unreasonable levels to hear the noise floor.

Here is some simple math... Normal very loud listening volume is well under 80dB. That leaves 10dB in 16/44.1 for transitory peaks. A boost of 10dB is TEN TIMES LOUDER than an already painful listening volume. If a human being is listening to music at that volume level, his ears aren't going to hear anything below -40dB, because our ears need time to adjust to volumes over 40dB apart. More dynamic range isn't necessarily better. If the dynamic range of a recording gets beyond 50dB, your ears need a few minutes to adjust from peak level down to the quietest sounds. That means that when you listen to Bruckner and the volume goes from forte to piano in the wink of an eye, if there is no adjustment time, you'll be reaching for the volume knob because the dynamic contrasts will sound too loud or too soft for your ears.

As I said, recording studios are quiet, but they still have a noise floor of around 25dB, concert halls have even higher noise floors. Your living room has a noise floor of 30-35dB at the absolute lowest. In order to raise the noise floor of 16/44.1 up to an audible level over the natural noise floor in your living room, you'd need to raise it to 120dB. Do you know how loud that is?

This isn't just science, it's horse sense. 16 bit has enough dynamic range that if you wanted to hear all of it at once, you would incur hearing damage.

It's all well and good to look at measurements, but to put it in a real world context, you need to know what those numbers mean in actual sound.

Friends don't let friends buy NOS DACs.


----------



## bigshot (Dec 4, 2022)

By the way, timing variations are less apparent as frequencies rise. Group delay is generally measured between 500Hz and 8kHz because above that it doesn't matter with human ears. Generally, the JDD (just detectable difference) threshold for group delay is from 1 (500Hz) to 3 (8kHz) ms, depending on the frequency. That binaural figure you quote is a very specialized thing. It can't be applied to all sound in general.

I'd recommend you read the article in my sig file titled "CD Sound Is All You Need". It has a lot of useful information and goes into great detail... far beyond my ability to explain as a layman.


----------



## castleofargh

bigshot said:


> A boost of 10dB is TEN TIMES LOUDER than an already painful listening volume.


This doesn't work. It's really about twice as loud and 10 times the power which I guess is what you somehow mixed up( "they struggle to put food on the family") . It also doesn't work because you started by giving an 80dB listening level as starting point when 120 is usually the given painful threshold. 




IanB52 said:


> I'd rather not get deep into it at this time, but there is research showing that humans can hear between 10-20us in binaural audio signals. 20khz is 50us if you use an NOS DAC with full aliasing, and more like 300us with standard 44.1khz brickwall filter.


I'm curious about this. Most people on this forum can't hear anything at 20kHz, long or short. I suspect that you're mixing things that can't be mixed because what's used to detect the lowest delays isn't simple tone and AFAIK, it's actually the silence between monstrous short sounds with energy at just about all frequencies before and after the cut. Something we don't have in music.  
The other paper I know of is for interaural delays and 16/44 has much higher temporal resolution between channels than the its low pass frequency.

About your argument for dynamic range, I don't think it matters much once we account for auditory masking, noise on the track, and standard practice of sticking the loudest signal near or at 0dB. Of course there is nothing wrong with just wanting more resolution, but it's some of the justifications for wanting it that often don't seem realistic to me. I'd sooner accept someone telling me "I want it because I want it".


----------



## bigshot

I thought about 3 dB was double. This says that 10 dB is ten times louder. https://www.healthyhearing.com/report/52514-What-is-a-decibel

This site lists 120 dB as the threshold of pain and ear injury.
https://www.cdc.gov/nceh/hearing_loss/what_noises_cause_hearing_loss.html

Regardless, 16 bit is overkill by more than one order of magnitude.


----------



## sander99

bigshot said:


> I thought about 3 dB was double.


3 dB is double the power but it doesn't sound twice as loud.
10 dB is 10 times the power but it doesn't sound 10 times as loud.
And this is where the logaritmic nature of it comes in: if you add 3 dB twice, you have multiplied the power by 4 but it doesn't sound like 2 multiplications of the level, it sounds more like 2 equal additions to the level (2 equal increments of 3 dB).

But regardless indeed, 16 bit is overkill.


----------



## IanB52

bigshot said:


> If you compare a 24/96 recording of music to a properly bounced down 16/44.1 file, it's transparent too. Have you ever found a study that showed people can hear the difference? All I've seen is that guy who looped fade outs and cranked the volume to unreasonable levels to hear the noise


I used to actually do this conversion professionally in mixing and mastering, many, many times. I don't think I ever came across a down-coversion with any dither shape or noise shaping that was fully transparent to the original 24 bit. In fact, different types of dither sound very slightly different, even at 70-80db listening volume. Same with 96khz SRC down to 44.1. And different SRC filters produce different effects. It always changes, and the real X factor is whether or not these changes are relevant to the style of music or recording quality. Sometimes it really doesn't matter, and in some cases losing extraneous information benefits the music.

Most people that do this work are not recording or publishing their blind test stats.

With bit depth again, I think it is relevant to think about what is necessary to reproduce the dynamic range of a musical performance, and nothing else, and what is required to produce a realistic recreation of the acoustic space and recording signal. I will argue you need a significantly higher margin to capture the full audio picture.

My suspicion however, is that masking is more of an issue here than absolute momentary dynamic range, why people (including myself) report hearing differences far above the noise floor. This does not appear to be adequately studied, but I would propose that a higher degree or resolution is required to reduce subtle masking to a more realistic level.

Take these ideas as you wish.

You may have seen this. People always complain that they consulted with Peter Craven but I've not seen anyone manage to challenge the data in this meta-study.

https://www.google.com/url?sa=t&source=web&rct=j&url=https://www.aes.org/e-lib/download.cfm?ID=18296&ved=2ahUKEwio89_TruH7AhUXJEQIHcr9Cb8QFnoECBYQAQ&usg=AOvVaw26c3O9EzZRyXL2jFIJqY0Y

I think that is about all I can contribute here.


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## 71 dB

bigshot said:


> Here is some simple math... Normal very loud listening volume is well under 80dB. That leaves 10dB in 16/44.1 for transitory peaks.


Average loudness level being 80 dB is indeed very loud. However, if the listening room background noisefloor level is at 30 dB, you can have theoretically about 90 dB over that before the "16 bits run out" so your peaks could be as loud as 120 dB! That is if your gear can produce such loud sounds, but nevertheless such peaks would be painfully loud and the average level at 110 dB would lead to hearing damages in minutes.



bigshot said:


> A boost of 10dB is TEN TIMES LOUDER than an already painful listening volume.


Ten times more power is needed (square root of 10 = 3.16 times larger amp voltage and current), but 10 dB louder sound feels about 2-3 times louder for the listener. There is no precise amount as it depends on the listener, the frequency content and the loudness level (40 dB to 50 dB is different from 80 dB to 90 dB)


----------



## 71 dB

bigshot said:


> I thought about 3 dB was double. This says that 10 dB is ten times louder. https://www.healthyhearing.com/report/52514-What-is-a-decibel
> 
> This site lists 120 dB as the threshold of pain and ear injury.
> https://www.cdc.gov/nceh/hearing_loss/what_noises_cause_hearing_loss.html


3 dB doubles the noise dose the ears suffer because the physical acoustic power of the noise doubles, but how much louder that sounds to the listener is a psychoacoustic phenomena. So, even if the noise dose increases 100 %, the perceived loudness may increase only maybe 30 %.

120 dB is a common threshold of pain, but it depends on the frequency (at low frequency it is closer to 130 dB) and for example older people with hearing loss/damage experience pain at much lower levels so that 90-100 dB may already be painful. Pain is one thing and it simply gives a threshold of instant hearing damage, but for example 85 dB already causes hearing damage if you are exposed to it all the time in your work (8 hours a day). Even much lower environmental noise levels can be harmful for the health causing anxiety, depression, increased blood pressure etc. *Enjoy the silence when you can people, because it is good for your body and mind!*


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## 71 dB (Dec 5, 2022)

IanB52 said:


> I used to actually do this conversion professionally in mixing and mastering, many, many times. I don't think I ever came across a down-coversion with any dither shape or noise shaping that was fully transparent to the original 24 bit.


How have you compared? Blindly? At what level? Test signals or music? It would be extremely surprising if someone could hear the difference of properly done conversion of this type.



IanB52 said:


> In fact, different types of dither sound very slightly different, even at 70-80db listening volume. Same with 96khz SRC down to 44.1. And different SRC filters produce different effects. It always changes, and the real X factor is whether or not these changes are relevant to the style of music or recording quality. Sometimes it really doesn't matter, and in some cases losing extraneous information benefits the music.


Dither is not supposed to be heard at all. In fact even the lack of dither (which causes distortion to the sound in truncation) shouldn't be heard at 16 bit, but dithering is the correct (distortion-free) way to truncate bit depth.



IanB52 said:


> Most people that do this work are not recording or publishing their blind test stats.


Most people that do this work know CD quality is transparent. Only those who don't understand digital audio or are in the business of selling hi-rez audio say otherwise.



IanB52 said:


> With bit depth again, I think it is relevant to think about what is necessary to reproduce the dynamic range of a musical performance, and nothing else, and what is required to produce a realistic recreation of the acoustic space and recording signal. I will argue you need a significantly higher margin to capture the full audio picture.


Well, I can argue elephants can fly...



IanB52 said:


> My suspicion however, is that masking is more of an issue here than absolute momentary dynamic range, why people (including myself) report hearing differences far above the noise floor. This does not appear to be adequately studied, but I would propose that a higher degree or resolution is required to reduce subtle masking to a more realistic level.


Are you saying 16 bit dither causes masking? Really? If not then what exactly are you talking about?


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## IanB52 (Dec 5, 2022)

71 dB said:


> How have you compared? Blindly? At what level? Test signals or music? It would be extremely surprising if someone could hear the difference of properly done conversion of this type.



Music. Lots and lots of blind tests. Same with my peers, and the effects in every down conversion are predictable.


71 dB said:


> Dither is not supposed to be heard at all. In fact even the lack of dither (which causes distortion to the sound in truncation) shouldn't be heard at 16 bit, but dithering is the correct (distortion-free) way to truncate bit depth.


Whatever people say, but each shape and scheme has a slightly different signature at normal listening volume. I prefer Mbit+ (or triangular if I must), though noise shaping is also interesting.

Also truncation "zipper" effects are notorious and easily heard in the absence of dither.



71 dB said:


> Most people that do this work know CD quality is transparent. Only those who don't understand digital audio or are in the business of selling hi-rez audio say otherwise.


This isn't true. I'd say there is a vocal minority with your perspective, but overall no such consensus exists in the audio field, same as here. Probably about 40% of recordings are done 24/88.2 or higher, so this is not a niche market.



71 dB said:


> Are you saying 16 bit dither causes masking? Really? If not then what exactly are you talking about?


I think dither may mask a little bit, as differences between triangular and lower level Mbit are apparent, but the ultimate problem, as I've said is not momentary macro dynamic range, but having sufficient bit depth to resolve subtle contrasts that would otherwise be homogenized. I could be wrong though, because I never convert without dither, so I can't say what is dither, and what is inherent to bit depth.

This may also bleed into having enough fixed points to perform a better decimation/reconstruction process, even where analog dynamic range is limited (24 bit vs 20 bit, for instance).


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## bigshot (Dec 6, 2022)

I’ve done direct A/B checks of the bounce down compared to the board on every mix I’ve supervised and neither I nor the engineers have ever heard a difference. I don’t listen at 80dB though. That wouldn’t be fun. If I did hear a difference, the red flag would go up and we’d track down the reason our final output isn’t matching the board. It really should.

There is a discussion of dither with samples in the audio myths seminar in my sig file. It’s very interesting.


----------



## 71 dB

IanB52 said:


> Music. Lots and lots of blind tests. Same with my peers, and the effects in every down conversion are predictable.


That is weird...



IanB52 said:


> Whatever people say, but each shape and scheme has a slightly different signature at normal listening volume.


Certainly not what I have experienced.



IanB52 said:


> I prefer Mbit+ (or triangular if I must), though noise shaping is also interesting.


TPDF dither is the default for a good reason, but there are time and place for other types of dither. For example rectangular dither retains digital silence (the least significant bit remains zero even after bit depth truncation).



IanB52 said:


> Also truncation "zipper" effects are notorious and easily heard in the absence of dither.


You probably mean by "zipper" effects granulation? These are not so easily heard without special cases (low signal level listened to unnaturally high level). Dither is used to remove granulation completely by killing the correlation between the signal and the quantization noise. Anyway, the use of dither is assumed when comparing hi-rez to a 44.1/16 conversion of it, so in that sense how easy the absense of dither is to hear isn't important in this context.



IanB52 said:


> This isn't true. I'd say there is a vocal minority with your perspective, but overall no such consensus exists in the audio field, same as here. Probably about 40% of recordings are done 24/88.2 or higher, so this is not a niche market.


Vocal? Yeah, some people just want sanity restored. High-resolution formats have some benefits in production, in studio, but consumers don't benefit from it at all. This thread is about consumer audio. You can record in studio using 88.2/24 or even higher, but once the track is mixed and mastered, consumers don't need more than 44.1/16. 



IanB52 said:


> I think dither may mask a little bit, as differences between triangular and lower level Mbit are apparent, but the ultimate problem, as I've said is not momentary macro dynamic range, but having sufficient bit depth to resolve subtle contrasts that would otherwise be homogenized. I could be wrong though, because I never convert without dither, so I can't say what is dither, and what is inherent to bit depth.


If dither can mask, then _everything_ masks. Your refrigerator in the kitchen, the blood flowing in your veins, the insects in your house etc. People don't seem to understand what natural listening level means, Yeah, I can hear dither clearly in the very quiet part when I increase level by 50 dB, but that is not natural listening level. That's "destroy everything and kill yourself" level. I'd say at 10 bit the music starts to almost completely mask the dither. 16 bit makes the dither 36 dB quieter than that. That's why what you write makes zero sense to me.



IanB52 said:


> This may also bleed into having enough fixed points to perform a better decimation/reconstruction process, even where analog dynamic range is limited (24 bit vs 20 bit, for instance).


Sampling frequency reduction is more demanding in this sense. Adding dither and truncating bit depth is trivial.


----------



## IanB52 (Dec 6, 2022)

71 dB said:


> That is weird...
> 
> 
> Certainly not what I have experienced.
> ...



Everyone I know who records acoustic instruments uses 24/96. 44.1 still happens largely because of sample libraries, and modern productions with tons of DSP plugins.

The primary reason 16/44.1 still exists is pretty similar to why hospitals still give you your MRI images on a CD-R. It is customary holdover from the 80s and 90s. Even if you don't think it matters sonically, it doesn't make sense to add an extra down-coversion step when everything already supports 24 bit.

The other reason is because some music is enhanced by 16 bit and/or dither because glosses over flaws, which is useful for hip hop, electronic, and pop. This is not unlike how 12 bit samplers are still popular in hip hop etc for their sound texture. Technology and music evolve together which is why we have fat 70s drums on tape, and crunchy hip hop beats on 12 or 16 bit samplers.



71 dB said:


> Sampling frequency reduction is more demanding in this sense. Adding dither and truncating bit depth is trivial.


Indeed, and in a delta sigma converter there is a significant reduction in sampling frequency.


----------



## bigshot

Artifacting caused by DSPs has nothing to do with the file format the final mix is bounced down to. I’m sure we all agree that signal processing is signal processing. Do we all agree that 16/44.1 is able to contain everything that human ears can hear, and if properly done, dithering is audibly transparent? Even without dithering 16 bit should be plenty for eliminating noise at comfortable listening levels in normal home use.


----------



## The Jester

bigshot said:


> Artifacting caused by DSPs has nothing to do with the file format the final mix is bounced down to. I’m sure we all agree that signal processing is signal processing. Do we all agree that 16/44.1 is able to contain everything that human ears can hear, and if properly done, dithering is audibly transparent? Even without dithering 16 bit should be plenty for eliminating noise at comfortable listening levels in normal home use.


I’d agree with that with the caveat “at comfortable listening levels in normal home use” …


----------



## bigshot

…but not at ear splitting levels that no one in their right mind would subject themselves to.


----------



## The Jester

bigshot said:


> …but not at ear splitting levels that no one in their right mind would subject themselves to.


Indeed,
With average speakers achieving around the common spec of 90db/@1m/@2.83v most people wouldn’t realise how loud an average level of 90db is without a sound level meter, Even with a quiet listening room 40db is really quiet, so the softest sounds can still be easily heard at 45-50db with an average level at 90db there’s still another 40db of headroom for short dynamic peaks with a “measly” 90db signal to noise ratio that could hit 130db, not that you’d want to be in the same room, so the S/N ratio with 16 bit is already more than enough,
Sadly with most recent popular music recordings you won’t see a dynamic range anywhere near 90db, more like 10-20db if your lucky, as compression is used (overused) to lift the lower level sounds up into the mix so they are more audible on phones and other relatively low power devices, or in car audio with higher ambient road, wind and mechanical noise levels,
Finding something where some care has been taken in the mastering to preserve the dynamics of the original sound can be a revelation, and in my opinion part of the reason for the whole “Vinyl revival” where the relatively limited dynamic range is more fully used with skilful mastering specifically for the Vinyl medium which is going to be listened to on more “traditional HiFi “ systems, and nothing to do with the physical media, logically if you were able to obtain a digital copy of a Vinyl master mixed and mastered from PCM or DSD, and using a similar quality DAC as used to drive the cutting lathe how can the Vinyl record be in any way “better” if both were compared side by side, it must be worse ?
Thinking of doing some extensive listening tests between DAC’s and maybe even Vinyl but first I want to listen to the new Bowie BluRay in Atmos ..
😃


----------



## 71 dB

IanB52 said:


> Everyone I know who records acoustic instruments uses 24/96. 44.1 still happens largely because of sample libraries, and modern productions with tons of DSP plugins.


Recording instruments in studio/concert halls is different from consumer audio formats. This thread is about what bitrates the consumer needs to consume music. Music production is different and using higher sample-rates and bit depth has its benefits. Especially 24 bit gives flexibility and safety margin in production, but higher samplerates than 44.1 kHz (music productions) or 48 kHz (video productions) are really needed in rare instances. The biggest reason for the use of high sample-rates is because some clients offering hi-rez formats demand it, but that is what this thread is about: Music consumers do not need hi-rez. CD quality is enough. Heck, >200 kbps lossy formats are often totally enough. So, there is no sound quality reason to offer hi-rez. So, clients should not have a reason to demand music to be produced in hi-rez. So, music producers should use 24 bit for "workflow" reasons and 44.1 kHz/48 kHz sample-rates depending on the production type and higher sample-rates only in special cases (such as using recorded sounds at lower speeds in the production as an effect). 



IanB52 said:


> The primary reason 16/44.1 still exists is pretty similar to why hospitals still give you your MRI images on a CD-R. It is customary holdover from the 80s and 90s. Even if you don't think it matters sonically, it doesn't make sense to add an extra down-coversion step when everything already supports 24 bit.


CD doesn't support 24 bit. It is 16 bit. If you release music on CD, you HAVE TO truncate it to 16 bit. That is not a problem, because consumers need about 13 bits for even the most demanding reasonable listening scenarios, so even 16 bit is overkill. Sure, a lot of things do support 24 bit (in practice hardly even 20 bit due to the noise floor of the analogue circuits), but consumers _benefit_ NOTHING from that support, but more storage space/money is wasted. That's the point of this thread. 



IanB52 said:


> The other reason is because some music is enhanced by 16 bit and/or dither because glosses over flaws, which is useful for hip hop, electronic, and pop. This is not unlike how 12 bit samplers are still popular in hip hop etc for their sound texture. Technology and music evolve together which is why we have fat 70s drums on tape, and crunchy hip hop beats on 12 or 16 bit samplers.


You seem to confuse lo-fi as an aesthetic statement from technical aspects of music formats...



IanB52 said:


> Indeed, and in a delta sigma converter there is a significant reduction in sampling frequency.


Huh?..


----------



## 71 dB

bigshot said:


> …but not at ear splitting levels that no one in their right mind would subject themselves to.


At ear splitting levels the  threshold of hearing is at least temporarily and probably permanently raised so much that nobody would hear even 8 bit dithering... ...well, perhaps that could be barely heard, but not much more than that...


----------



## bigshot

I don’t think anyone would get close to 90dB in a speaker setup. 80 is VERY loud… loud enough to be difficult to listen to for any length of time.

Whenever someone starts arguing that 16/44.1 isn’t enough, they generally have no idea what the numbers mean.


----------



## 71 dB

bigshot said:


> I don’t think anyone would get close to 90dB in a speaker setup. 80 is VERY loud… loud enough to be difficult to listen to for any length of time.


Yes, if you talk about average level, but peaks can go 10-20 dB higher than that depending on how dynamic music you listen to. Those peaks eat up dynamic range, so you don't have 90 dB of dynamic range "under" the average level in 16 bit unless the music is super-loud overcompressed pop. Luckily about 50 dB of dynamic range under the average level is enough.



bigshot said:


> Whenever someone starts arguing that 16/44.1 isn’t enough, they generally have no idea what the numbers mean.


People often talk with unwarranted confidence about things they don't know much about. I guess we all have started from the position "of course bigger numbers mean better sound" and with time moved to the more educated/informed position "CD quality is actually enough."


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## The Jester

Using identical masters most wouldn’t tell the difference between 16/44.1, 24/96 or DSD,
So if someone is going to “remaster” a popular recording from the past and want to do as good a job as possible to make it a worthwhile purchase why not release it for download at 24/96 and charge a premium for it, that’s what they’re in business for, there are a few re released remastered CD’s that sound better than the original on a good system, but most are overly compressed and boosted which can make them sound better on portables, car audio etc while being no better, and sometimes worse on a good, well setup home system.


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## bigshot

This is someone who claims to hear a difference in his own bounce downs consistently all the time. I think that something is configured wrong in his software somewhere.


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## castleofargh

Options:
1/ He was able to perceive stuff that most people cannot thanks to great hears and listening skills.
2/ The equipment wasn't able to remain transparent for reasons. 
3/ No matter how reliant on hearing someone in the industry is, he doesn't stop being human. If the experiences weren't well controlled, the conclusions might just be wrong. 

1/ is rare, 2/ did and does happen sometimes,  3/ is almost everywhere.


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## redrol

I gotta push back on very loud, etc.  I have a SPL meter, in A weighted mode and do hit 90-100DB plenty of times.  Is it loud, kinda.. is it concert loud? no.


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## IanB52 (Dec 7, 2022)

71 dB said:


> Recording instruments in studio/concert halls is different from consumer audio formats. This thread is about what bitrates the consumer needs to consume music. Music production is different and using higher sample-rates and bit depth has its benefits. Especially 24 bit gives flexibility and safety margin in production, but higher samplerates than 44.1 kHz (music productions) or 48 kHz (video productions) are really needed in rare instances. The biggest reason for the use of high sample-rates is because some clients offering hi-rez formats demand it, but that is what this thread is about: Music consumers do not need hi-rez. CD quality is enough. Heck, >200 kbps lossy formats are often totally enough. So, there is no sound quality reason to offer hi-rez. So, clients should not have a reason to demand music to be produced in hi-rez.


Just because someone thinks the lowest common denominator you can get away with should be forced on everyone, doesn't mean some won't like it.



71 dB said:


> CD doesn't support 24 bit.



CD is obsolete and practically nobody uses them. There is no reason to go through an extra, lossy step.


71 dB said:


> You seem to confuse lo-fi as an aesthetic statement from technical aspects of music formats...


Any format produces an aesthetic if you know how to push it in a certain direction, because none are truly transparent. Thus, the format tends to promote certain aesthetics.



71 dB said:


> Huh?..


You seem to be the kind of person who knows how sigma delta conversion and oversampling work. You have something like a 5 bit modulator, sample at 11.2mhz, then decimate and apply a 1000db/oct linear phase brickwall filter to get the output format like 24/ 44.1khz. Plenty of both sample rate conversion and filter DSP going on even before the digital signal leaves the ADC, and the reverse in a DAC.


----------



## IanB52

The Jester said:


> Using identical masters most wouldn’t tell the difference between 16/44.1, 24/96 or DSD,
> So if someone is going to “remaster” a popular recording from the past and want to do as good a job as possible to make it a worthwhile purchase why not release it for download at 24/96 and charge a premium for it, that’s what they’re in business for, there are a few re released remastered CD’s that sound better than the original on a good system, but most are overly compressed and boosted which can make them sound better on portables, car audio etc while being no better, and sometimes worse on a good, well setup home system.


I suggest asking the people who produce the masters.

Or, if so inclined you can obtain a high resolution file and convert it yourself and compare each at their native rate.


----------



## redrol

I've done the high-res to 44.1/16 and yeah, you cant tell a damn thing.  No chance.  Zero.  Nor does high res sound better.


----------



## IanB52

redrol said:


> I've done the high-res to 44.1/16 and yeah, you cant tell a damn thing.  No chance.  Zero.  Nor does high res sound better.


I'm honestly envious. Sounds like an easier audio life.


----------



## bigshot

Are you
Measuring peak or average?


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## Davesrose (Dec 8, 2022)

bigshot said:


> I don’t think anyone would get close to 90dB in a speaker setup. 80 is VERY loud… loud enough to be difficult to listen to for any length of time.
> 
> Whenever someone starts arguing that 16/44.1 isn’t enough, they generally have no idea what the numbers mean.



90dB peak isn't hard to reach.  I haven't gone overboard with having powerful monoblocks for each of my speakers...but my "140 watt" per channel setup and I'm no where near going to the max of my volume (when I've accidentally set volume to near max, there hasn't been distortion while still being very uncomfortable).  I'd say for many home cinema setups with speakers and room size, they can easily reach the comfortable range (of having most content being below 90dB and just have very short swings hitting 90dB or even more).  You can even go louder, but when it comes to dynamic range it's relative.  When you consider time is also a factor, 16 bit is enough for a source medium.  Seems most of the argument about 24bit or 32bit is with authoring audio.  Where the intent is different in which you might need to adjust levels that would bring out noise with 16 bit (and where the recording was adjusted for having best range in that space: where from what I've read is less than ideal scenarios that are different than being in a recording studio).


----------



## bigshot (Dec 8, 2022)

90dB for a fraction of an instant is obviously different than an averaged 90dB, and the perceived volume of a peak level of 90dB could sound incredibly loud or quite soft, depending on how compressed the music is. The best way to talk about the loudness of sound produced by a stereo system is averaged.

With a speaker system, you've got a room tone of at least 30dB. So if your music was mixed to the most dynamic level commercial music is commonly mixed... 50dB, 80dB would still be the volume level that would reproduce everything you need to hear. And of course, 50dB fits on a CD with tons of room to spare.

16/44.1 is overkill when it comes to dynamic range.


----------



## Davesrose

bigshot said:


> 90dB for a fraction of an instant is obviously different than an averaged 90dB, and the perceived volume of a peak level of 90dB could sound incredibly loud or quite soft, depending on how compressed the music is. The best way to talk about the loudness of sound produced by a stereo system is averaged.
> 
> With a speaker system, you've got a room tone of at least 30dB. So if your music was mixed to the most dynamic level commercial music is commonly mixed... 50dB, 80dB would still be the volume level that would reproduce everything you need to hear. And of course, 50dB fits on a CD with tons of room to spare.
> 
> 16/44.1 is overkill when it comes to dynamic range.


So I've even got a SPL meter with my Apple Watch.  The average SPL for my receiver's volume is 65-70dB.  Certainly watching a blockbuster movie, there's more swings in having peak loudness beyond 90dB (and as I've pointed out, this is in my level that's comfortable listening: my system can go much higher in volume).  We've had some discourse about Atmos and music.  Actually, Tár is a great movie to watch in Atmos with classical music.  The movie itself is a bit too long, but I was really impressed with the sound mix of the scenes with the symphony.  There was quite a realism with dynamics and percussion that I haven't heard with typical albums.  I suspect that the total swings is greater than 30dB, but again when we look at time, 16 bit can still be adequate for the delivery medium.  Again, when it comes to the debates of 24bit or 32bit, it seems a lot is scenarios with audio engineering.


----------



## bigshot

16 bit is more than enough.


----------



## bigshot

16 bit is more than enough.


----------



## IanB52 (Dec 8, 2022)

_Deleted my original post because was mostly me thinking out loud and had a couple mistakes._


----------



## Ryokan

I had to admit I can't hear a difference between 1GB+ hi-res albums and 16/44.1, and I wanted to hear a difference. Pleased as it's saved me money (and space). 
Same if I double up an album by mistake where one track is flac and the next mp3, too close to call after A-B'ing a few times. I have an album that actually sounds better in mp3 format than flac, which I bought to replace it, realised it's because they are different mastering's.


----------



## bigshot

There is no audible improvement in transients with sampling rates over 44.1. Transients in music are several orders of magnitude larger than anything that would be affected by sampling rates. Square waves aren’t anything that exist in the real world.


----------



## 71 dB (Dec 8, 2022)

IanB52 said:


> Just because someone thinks the lowest common denominator you can get away with should be forced on everyone, doesn't mean some won't like it.


What is forced? As I said, use as high resolution you want when you record, but consumers don't benefit anything.



IanB52 said:


> CD is obsolete and practically nobody uses them. There is no reason to go through an extra, lossy step.


No it is not. For me it is the main source of listening to music and I am certainly not the only one*. Don't you FORCE your hi-rez files on me. I want physical media. I have bought music on files and I don't like it much. Not for me. I was born in 1971. There is nothing lossy about CDs for me. I don't need information my ears can't hear. I need the information my ears can hear. CDs can do that easily.

* This isn't even about what is the most popular way to consume music or even CDs. This is about how much sample-rate and bit depth consumers need. CD is allready enough in that sense.



IanB52 said:


> Any format produces an aesthetic if you know how to push it in a certain direction, because none are truly transparent. Thus, the format tends to promote certain aesthetics.


Doesn't this mean they are transparent without pushing them to certain direction?



IanB52 said:


> You seem to be the kind of person who knows how sigma delta conversion and oversampling work. You have something like a 5 bit modulator, sample at 11.2mhz, then decimate and apply a 1000db/oct linear phase brickwall filter to get the output format like 24/ 44.1khz. Plenty of both sample rate conversion and filter DSP going on even before the digital signal leaves the ADC, and the reverse in a DAC.


How is 11.2 MHz less than 44.1 kHz? I don't claim to be an expert on sigma delta modulators. Oversampling makes it possible to use relaxed reconstruction filters, so no need to use 1000 dB/octave brickwall filters if one doesn't want to.

People like you make me feel useless in this world. I got myself an education that gives me the understanding that hi-rez formats are useless for consumers, but what can I do with that understanding? Absolutely nothing. Some people continue buying hi-rez music and placebo makes them think it sounds better. Meanwhile I'm spinning my old (and new) CDs and all I hear is the quality of the music production, mixing and mastering, because to my ears the music format itself is transparent. Writing online gives the feeling and illusion of having some sort of part in this World, but the sad fact is I am a lonely loser whose life has been a massive failure. 2022 has been horrible. First a couple of years of covid-19  and the war in Ukraine (as a Finn this war means massive loss in feel of security and it doesn't help Hungary and Turkey takes time to ratify our NATO membership). Now I have health problems, perhaps because the 2020's has been so bad. Life sucks so much. People like you are least of my problems. I could ignore you totally unlike other problems.


----------



## 71 dB

IanB52 said:


> I'm honestly envious. Sounds like an easier audio life.


You can make audio life as difficult to yourself as you want, but audio life can be easy.


----------



## The Jester

IanB52 said:


> I suggest asking the people who produce the masters.
> 
> Or, if so inclined you can obtain a high resolution file and convert it yourself and compare each at their native rate.


Hardly practical,
Luckily I still buy CD’s now and then and checkout carefully anything that states on the cover “remastered” or 24 bit remastered” and nothing else,
But if I want say an early David Bowie album I’m missing (mainly cos back in the day there was more good music around than I could afford) and something comes up “Remastered by Tony Visconti” who produced a lot of his albums at the time, same with Beatles remasters and some others where it’s fairly obvious that sound quality was an important factor in releasing various remastered albums I’m more confidant …
I’ll continue to buy CD’s, and the odd all analogue produced Vinyl LP”s as needed, and that’s the important point for me at least, “buy”, not rent via a streaming service or lease a downloaded album.


----------



## 71 dB

The Jester said:


> Hardly practical,
> Luckily I still buy CD’s now and then and checkout carefully anything that states on the cover “remastered” or 24 bit remastered” and nothing else,
> But if I want say an early David Bowie album I’m missing (mainly cos back in the day there was more good music around than I could afford) and something comes up “Remastered by Tony Visconti” who produced a lot of his albums at the time, same with Beatles remasters and some others where it’s fairly obvious that sound quality was an important factor in releasing various remastered albums I’m more confidant …
> I’ll continue to buy CD’s, and the odd all analogue produced Vinyl LP”s as needed, and that’s the important point for me at least, “buy”, not rent via a streaming service or lease a downloaded album.


When it comes to popular music of the 80's, it is often wise to get the "original" CD release (used) rather than a new remastered version. Why? Because the new remasters try to make the music sound like music of today sounds, but it doesn't make sense because it is not music of today, but music of the 80's. It is totally different aesthetics. The correct way to remaster an 80's release is to make it sound like brilliantly mixed and mastered releases did back in the day. The whole music production process was based on THAT aesthetics. If you remaster such music to have similar sound signature as music today, it can totally destroy the music, its soul, its groove, its everything. Good remasters exist of course, but I'm afraid most of them are not. At worst a new remaster can sound as perverse as an old technicolor movie would look if its colors were processed digitally into modern "orange and teal" aesthetics.


----------



## The Jester

71 dB said:


> When it comes to popular music of the 80's, it is often wise to get the "original" CD release (used) rather than a new remastered version. Why? Because the new remasters try to make the music sound like music of today sounds, but it doesn't make sense because it is not music of today, but music of the 80's. It is totally different aesthetics. The correct way to remaster an 80's release is to make it sound like brilliantly mixed and mastered releases did back in the day. The whole music production process was based on THAT aesthetics. If you remaster such music to have similar sound signature as music today, it can totally destroy the music, its soul, its groove, its everything. Good remasters exist of course, but I'm afraid most of them are not. At worst a new remaster can sound as perverse as an old technicolor movie would look if its colors were processed digitally into modern "orange and teal" aesthetics.


Indeed,
That’s why I’m careful with buying remasters vs re releases, those early CD’s sound so good on modern digital equipment,
If it’s you kind of music, one I can recommend is Jewel’s “pieces of you” 25 anniversary 2CD release, all the originals sounding as good or better than ever with the second disc the real gem, couple of live tracks as well as a couple of direct studio takes before any processing plus various outtakes.


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## bigshot (Dec 8, 2022)

I find that recent remasters, especially those done by Sony are fantastic. There were whole record lines back in the days of LPs that sucked. The initial CDs released didn’t sound a lot better. But new remastering breathes new life into the recordings. For examples of this look at David Bowie’s early albums on RCA or Columbia classical records like Bernstein in NY.

I’ve also found that CD releases in the last five years don’t have the problems with hot mastering that they used to. A lot of old, crappy hot mastered CDs are still being sold, but it’s being fixed as they go back and remaster again.


----------



## Ryokan

bigshot said:


> I find that recent remasters, especially those done by Sony are fantastic. There were whole record lines back in the days of LPs that sucked. The initial CDs released didn’t sound a lot better. But new remastering breathes new life into the recordings. For examples of this look at David Bowie’s early albums on RCA or Columbia classical records like Bernstein in NY.
> 
> I’ve also found that CD releases in the last five years don’t have the problems with hot mastering that they used to. A lot of old, crappy hot mastered CDs are still being sold, but it’s being fixed as they go back and remaster again.



What were the engineers thinking? some were almost unlistenable. I bought a re-master that was done probably ten years ago and there were bits of the songs actually chopped off, and when you forwarded to the next track it started halfway the song. Another had double endings if you turned up the volume, or fade outs abruptly ended, totally ruined the album and I went back to the original cd which sounded very good as long as you turned it up.


----------



## The Jester

Ryokan said:


> What were the engineers thinking? some were almost unlistenable. I bought a re-master that was done probably ten years ago and there were bits of the songs actually chopped off, and when you forwarded to the next track it started halfway the song. Another had double endings if you turned up the volume, or fade outs abruptly ended, totally ruined the album and I went back to the original cd which sounded very good as long as you turned it up.


Being able to turn it up was the issue,
When portable digital players came along, as well as car CD players the whole “loudness war” started, and as often said in comparative testing level matching is important, the loudest one will usually sound better, with Digital 0db is the absolute limit, so the dynamic range was compressed to enable the average level to be increased, as well as sometimes lifting the lower level sounds to make them more audible, and with some music it sounds fine like that, even with most music it can sound ok as far as noise, distortion etc, that is until you find possibly an earlier version that hasn’t been heavily compressed and the difference is obvious.


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## bigshot

People are buying digital downloads and not ripping CDs, so CDs have started to go back to the way they were.


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## 71 dB

Yesterday I got *B12*'s _Electro-Soma I + II_ (WARPCD9R) releases in 2017. I also have the original _Electro-Soma_ (WARPCD9) from1993, but it seems to be one of those infamous bronzing CDs by PDO. That's why I wanted a new version of it, because the old one might become unplayable. The new version is remastered and sounds different. It has crazy amount of bass (which kind of works with this kind of techno music, but there was nothing wrong with the original sound so the remastering in my opinion has been totally unnecessary and even harmful). I'm thinking of ripping the old CD (if it can still be ripped), but I hate music as digital files. So non-tangible...


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## 71 dB (Dec 8, 2022)

The Jester said:


> If it’s you kind of music, one I can recommend is Jewel’s “pieces of you” 25 anniversary 2CD release,


I wouldn't know. I don't know 80's popular music that well. I didn't care* much about music until about 1987-8 and I have explored 70's and 80's music later in life. Just recently I explored the albums of Roxy Music (on Spotify) and even bought the album _Avalon_ on CD (original!) since I liked that the most.

* Because most of the stuff I heard on radio/elsewhere felt crap so I though it all must suck, but later in my life I realized my taste in music varies a lot from the masses and so the stuff I like is often more obscure and played much less on radio. Once I started to explore the World of music in order to find my own favorites I have discovered amazing things.


----------



## bigshot

IanB52 said:


> I used to actually do this conversion professionally in mixing and mastering, many, many times.


hmmm...


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## 71 dB

bigshot said:


> People are buying digital downloads and not ripping CDs, so CDs have started to go back to the way they were.


What does this mean? What way CDs have been because people ripped them?


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## bigshot

Because they were ripping them to MP3s and playing them in shuffle mode... That encouraged producers to hot master so their song didn't sound quiet next to other random songs from other albums. People are going back to listening to CDs as albums again, which means that the levels of songs can vary because the mastering transitions from loud to quiet, not just an abrupt switch in a  random shuffle. People are playing albums all the way through from beginning to end.


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## The Jester

That and the “must have” car accessory the “10 stack player”, load 10 CD’s and press play on the head unit and they’d play sequentially, but there was the “shuffle” mode that played a track at a time randomly across all 10 discs, and with a mix of older and newer releases there could be 10db or more difference in average loudness …. 🗯️📢


----------



## redrol

bigshot said:


> Are you
> Measuring peak or average?


Hard to say, meter has slow and fast mode.  Clearly peaks get averaged out with slow mode.  I'm using slow mode:


----------



## sander99

bigshot said:


> hmmm...


Hmmm indeed. Since you often don't remember usernames let me remind you that this is the guy who claimed to be a professional recording engineer, claimed that pcm is not audibly transparent regardless of sampling frequency and bit depth, that is why he recorded in dsd, that it is better to apply effects and perform mixing analog to avoid conversion to pcm at any cost, etc. etc. Luckely we had @gregorio there to tear all that nonsense to shreds.
(in this thread https://www.head-fi.org/threads/how-do-you-master-a-dsd-recording.963405/)


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## bigshot (Dec 8, 2022)

OK. I remember him now. Why do these jokers keep coming back? Do they enjoy being humiliated? We need a troll scorecard with stats on each one so I don’t forget and start out polite every time.

He asked to be banned. I hope the admins obliged him and didn’t wimp out. We don’t need him back here.


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## IanB52

71 dB said:


> How is 11.2 MHz less than 44.1 kHz? I don't claim to be an expert on sigma delta modulators. Oversampling makes it possible to use relaxed reconstruction filters, so no need to use 1000 dB/octave brickwall filters if one doesn't want to.



This is false. Oversampling allows you to avoid analog brickwall anti alias filters, which incur phase shift. When downsampling happens you still must use a 1000db/octave digital filter, down at least 100db between 20 and 22khz. The only difference is that in digital you can use linear phase filters and it is cheaper to build to build such an extreme filter.


71 dB said:


> People like you make me feel useless in this world. I got myself an education that gives me the understanding that hi-rez formats are useless for consumers, but what can I do with that understanding? Absolutely nothing.


I understand the feeling of futility trying to communicate with other people. In particular, you are never going to convince me that high rez is useless because it is a daily part of my life and I can't unhear what I hear, and I'm not willing to pretend I don't hear something I do in order to conform .

Look, I struggle coming up with technical reasons why I would be right. But I am, and whether you like it or not a large number of engineers and consumers are never going to come around to your point of view because there are differences and there is a large consensus that they exist and people hear them.

On the flip side, you might empathize with people who feel like their knowledge is dismissed or gaslighted, and who are passionately in favor of better audio quality and feel frustration that there are those invested in not only stopping progress, but even lower the bar for audio quality further.


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## IanB52 (Dec 8, 2022)

sander99 said:


> Hmmm indeed. Since you often don't remember usernames let me remind you that this is the guy who claimed to be a professional recording engineer, claimed that pcm is not audibly transparent regardless of sampling frequency and bit depth, that is why he recorded in dsd, that it is better to apply effects and perform mixing analog to avoid conversion to pcm at any cost, etc. etc. Luckely we had @gregorio there to tear all that nonsense to shreds.
> (in this thread https://www.head-fi.org/threads/how-do-you-master-a-dsd-recording.963405/)


It sounds an awful lot like you are claiming that I am lying about my professional history...

Also, if tearing me to shreds amounts to simply repeating false assertions again and again, while falsely pretending expertise, yes gregorio did that. I went as far as posting my education and work history, he did none of that, yet you imply I am the fraud and accept everything he says simply because he tells you what you want to hear and for no other reason. You are like one of these people who think Robert Malone is the world expert on mRNA, listening to a fringe person because they reinforce what you already comitted to believe.

Anyway, I ventured into this thread really trying not to be confrontational and hashing out some ideas and different perspectives. I think I played nice. But as usual, you guys never change and keep resorting to toxic behavior and character attacks when someone gives you information that you don't like. That is not the way someone who actually uses their intellect operates. If you can't engage in ideas and don't want any insight into the topic as it plays out in the real world, just baselessly attack the person and turn off your brain.

But I guess that is "Sound Science" is about, avoiding learning at all costs.


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## bigshot (Dec 8, 2022)

I think you're a duffer who thinks he is professional. The fact that you didn't recognize Gregorio's experience is proof of your duffer status. You probably record the local church choir for free coffee and doughnuts and call that a career.


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## sander99

gregorio said:


> IanB52 said:
> 
> 
> > You have a polyannish view of everything PCM and make inappropriate equivalencies between analog and digital artifacts …
> ...


@IanB52: Based on the above, there is no way I can take you serious.


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## IanB52 (Dec 8, 2022)

sander99 said:


> @IanB52: Based on the above, there is no way I can take you serious.


For what it's worth, there is no way I can take you seriously either if you can't tell the difference between a string of willful misunderstandings, and baseless ad homonyms, and a coherent counter argument.

Most of what I said went totally over his head, and then he thinks his own lack of comprehension is some kind of gotcha, inventing contradictions that dont exist, that he can use to baselessly accuse me of being a fraud involved in marketing. What kind of person goes around calling people a liar before even doing the diligence of understanding what is being said?

Case in point, I claimed that because of my experience I understand in broad terms views and practices within the industry.  I also have my own views and niche, which is small, and I acknowledged that. Gregorio right there thinks that is a contradiction! And beyond that, that this imaginary contradiction is both proof that his _own_ opinion is universal, and that I am a fraud! Use you brain just a little bit. jfc I feel like you would if you weren't more emotionally invested in personal takedowns than genuine inquiry.

He also dismisses a prestigous and thorough academic paper I posted claiming that the data was cherry picked, when it literally contained the entire field of published studies on the subject. It doesn't seem like he bothered to even read it and just threw a generic attack before even verifying if it was true or not. If garbage like that is convincing to you, well it's not my business.

Anyway, please to dont quote reply anything back at me. I am not interested in remaining in this discussion. Let it revert back to the circle jerk it has been 197 pages.


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## bigshot (Dec 9, 2022)

We've all patiently explained all of this to you. You don't listen. You aren't discussing things with us. You are posting nonsense soliloquies. We can reply, but you'll just ignore what we say and repeat the same soliloquy again. You've discouraged us from interacting with you... you've discouraged me from even reading your TL/DR nonsense. There's no reason to read it. It would be like hearing a joke that you already know the punch line to.

By the way, you are the one he accused of cherry picking. He said that the paper you cited was spurious. You chose that one spurious paper over a mountain of evidence to the contrary.


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## 71 dB (Dec 9, 2022)

> This is false. Oversampling allows you to avoid analog brickwall anti alias filters, which incur phase shift. When downsampling happens you still must use a 1000db/octave digital filter, down at least 100db between 20 and 22khz. The only difference is that in digital you can use linear phase filters and it is cheaper to build to build such an extreme filter.



How steep filter you need to use in downsampling depends on how much content you have above Nyquist. How often do you have audio content over 22.05 kHz at 0 dBFS level? Also, 22.05 kHz to 24.1 kHz band gets folded into 20 kHz to 22.05 kHz band, which is inaudible for pretty much everyone except for a few freaks of nature. I don't need hi-rez audio just because someone's dad happened to be a dolphin. In this sense, only the aliasing of audio content over 24 kHz is relevant to young ears and maybe over 27 kHz or so for older ears. Considering how fast ultrasonic sound attenuates in the air, how strong content in this audio band is natural to begin with? -30 dBFS? -50 dBFS? How much do you have in your productions?



> I understand the feeling of futility trying to communicate with other people. In particular, you are never going to convince me that high rez is useless because it is a daily part of my life and I can't unhear what I hear, and I'm not willing to pretend I don't hear something I do in order to conform .



Okay, but some other people reading this thread might be convinced. I can't unhear the weird stuff I have heard in life either, but I have learned to accept it was all placebo, in my brain.



> Look, I struggle coming up with technical reasons why I would be right. But I am, and whether you like it or not a large number of engineers and consumers are never going to come around to your point of view because there are differences and there is a large consensus that they exist and people hear them.



A lot of sound engineers are on "my" side. Consumers on the other hand are for the most part quite ignorant about these things, so their opinions tell nothing about the truth.



> On the flip side, you might empathize with people who feel like their knowledge is dismissed or gaslighted, and who are passionately in favor of better audio quality and feel frustration that there are those invested in not only stopping progress, but even lower the bar for audio quality further.



Better sound quality comes from producing, mixing and mastering better. When those things are done well, "CD quality" is all consumers need. A song doesn't sound bad because there is a brickwall filter at 20 kHz. It sounds bad because it is overcompressed, the balances of tracks are off, the effects are bad and so on... ...you should know this if you work in the profession.


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## bigshot (Dec 9, 2022)

Why are you quoting him with a tag, 71 dB? He asked to not be tagged. He doesn't listen. Your arguments are falling on deaf ears. I don't think anyone wants him to reply. Let it die. You can quote him without tagging him in the post.... like this:



> Anyway, please to dont quote reply anything back at me. I am not interested in remaining in this discussion.


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## 71 dB

bigshot said:


> Because they were ripping them to MP3s and playing them in shuffle mode... That encouraged producers to hot master so their song didn't sound quiet next to other random songs from other albums. People are going back to listening to CDs as albums again, which means that the levels of songs can vary because the mastering transitions from loud to quiet, not just an abrupt switch in a  random shuffle. People are playing albums all the way through from beginning to end.


Okay, thanks! I am bad at knowing what other people do, because I just do what I want to do. Most of the time I listen to CDs as albums and I have never been a fan of shuffle modes. To me shuffle mode is for people who own just 5 albums and have to listen to them in random orders to stop getting bored. I have about 2000 CDs (not a _massive_ collection, but a decent one anyway) and I listen to various genres of music. I may "shuffle" between music genres: From Ke$ha to Herbie Hancock to J. S. Bach for example. That's a healthy (and fun) musical diet!

It is great if albums are mastered more artistically (again) thanks to this behavioural shift.


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## 71 dB (Dec 9, 2022)

bigshot said:


> Why are you quoting him with a tag, 71 dB? He asked to not be tagged. He doesn't listen. Your arguments are falling on deaf ears. I don't think anyone wants him to reply. Let it die. You can quote him without tagging him in the post.... like this:


He asked to not be tagged? Huh? Whatever. I removed the damn tags........


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## bigshot

If you continue to engage him, YOU will be the problem. This forum could be so much more than stupid arguments about fundamentals everyone should already know and trivia time sharing superfluous footnotes that don’t change the overall point being made. But the discourse won’t be raised if you dive down to the levels of the stupids. We all need to respect the forum we’ve been given and not crap all over it.


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## Ryokan

A few years ago I had an email conversation with an artist who also produces music, I asked if he could put up flac files (I dismissed lossy at that time as inferior) as he only offered his first album in mp3. He replied 'Why do you want flac?'  It was then that the 'penny began to drop'.


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## 71 dB (Dec 9, 2022)

Is there a list of people I can engage with without being a problem? I thought I was replaying to his posts properly, but instead I am accused of "tagging" him, being a problem and diving  down to the levels of the stupids. What the hell?


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## bigshot

Don’t feed trolls.


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## 71 dB

bigshot said:


> Don’t feed trolls.


My idea of who is a troll may differ from your opinions, but if it is a "fact" that he is a troll then so be it. I can stop "feeding him". It saves my energy and time.


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## bigshot (Dec 9, 2022)

Here are some clues to look for in troll posts...

This one tells you that nothing you say will make any difference because he isn't listening:


> you are never going to convince me that high rez is useless because it is a daily part of my life and I can't unhear what I hear, and I'm not willing to pretend I don't hear something I do in order to conform .



This one tells us that he blames all of his problems on us as a collective evil group:


> Anyway, I ventured into this thread really trying not to be confrontational and hashing out some ideas and different perspectives. I think I played nice. But as usual, you guys never change and keep resorting to toxic behavior and character attacks when someone gives you information that you don't like. That is not the way someone who actually uses their intellect operates. If you can't engage in ideas and don't want any insight into the topic as it plays out in the real world, just baselessly attack the person and turn off your brain.



Here he reveals that he doesn't know the most fundamental things about audio formats:


> CD is obsolete and practically nobody uses them. There is no reason to go through an extra, lossy step.



And here he takes your comment, juggles the words around to make it say something that shows he completely rejected everything you said without providing any proof:



> Any format produces an aesthetic if you know how to push it in a certain direction, because none are truly transparent. Thus, the format tends to promote certain aesthetics.



He throws in just enough cut and paste technical stuff from hoodoo audiophool sites to keep you replying with technical refutations, but he ignores every bit of your evidence and keeps repeating the same incorrect belief (as opposed to understanding) over and over. It really isn't hard to tell if someone is trying to play you. The fact that he has come back here multiple times with the same schtick makes it clear. He isn't here to contribute. He's here to poke the sound science lion in the cage with a stick.

People online play games. You have to read between the lines.


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## 71 dB

Yeah, but I was almost done with him when that example got my attention... ...he started with less troll-like posts and I took the bate.


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## bigshot (Dec 9, 2022)

That's what they do... when they sense they've pushed a little too far, they pull back and try to draw you in again. Sander99 posted a link to a previous thread where he blatantly embarrassed himself and revealed his motives. So he reeled back in and tried to sound more rational in the next post because he realized he was losing us. A person reveals himself by the totality of all his posts, not just one single post where he is behaving himself to get the reaction he wants from you.

It's entirely possible that he is trolling without realizing it. Some people's minds are wired in sociopathic ways. This sort of manipulation and self absorption comes naturally to them. They don't need to plan it. The internet attracts an awful lot of unbalanced people.


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## 71 dB

I'm too lazy to keep track of everyone's post history, but actually I did have this voice in my head telling me this person means trouble.


----------



## castleofargh

Not sure that declaring anybody with a different point of view who doesn't give up in 3 posts, a troll, is what a science section needs. 
I don't know him outside this forum, but I've seen him at least once respond to my overly aggressive comment by not getting mad at me(when I would have deserved it) and even acknowledging that he had perhaps gone too far with his post. I won't say that this completely defines his personality or motivation on the forum because I don't have the psychology level of a fashion magazine, but most of the people I see resorting to trolling seem quite incapable of considering that they might sometimes be wrong about anything. Only time they do, it's sarcastically because, you know, what's more hilarious and preposterous than them saying they made a mistake...

Ian has not been like that IMO, but I admit that I have given up on reading a bunch of overly long back and forth with gregorio so maybe I missed hot stuff? I'll probably disagree with him on so many threads that it's maddening, but the vibe I got so far is that of someone I can and should respect. Out of everything, what causes me to doubt my view the most are @sander99's posts because he's like a technical care bear on HeadFi and we basically never see him being confrontational or saying bad stuff to someone. That does make me question my own judgement a little.

 Anyway, if not being able to agree on something was a deal breaker, who would be safe in here?


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## bigshot (Dec 9, 2022)

Oh come on. He did a lot more than just not agree with an opinion. He denied facts. If people can just not agree by denying facts, then this isn't the sound science forum. We have always been able to demand that opinions be supported.

The quality of the conversations in this forum have fallen in the past six months to a year. The forum is overrun by invaders with opinions that deny facts. They specifically target the people who they think are the "bosses" of the forum. That is exactly what drove away Steve Eddy, Pinnahertz and now I assume Gregorio. The frustration of having to deal with the same dumb cable, HD audio and "everything sounds different" arguments makes us get short with each other and we fight among ourselves over things that don't matter.

The sound science regulars who have been here for years aren't the problem. We *are* sound science. The duffers and dolts who come in here pretending to be things they aren't and claiming things that aren't true are the problems. "Benefit of the doubt" has been stretched to ludicrous extremes.

We all know certain things for sure. We've seen the peer reviewed tests, we understand the concepts behind why they're true, we've even done informal controlled tests ourself to verify them... Why do we have to argue about those things all the time? Isn't there anything better to discuss? We're so tolerant and broadminded of "opinions to the contrary" which are nothing more than bias and subjectivity, we've allowed the whole focus of the forum to slide backwards.


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## 71 dB

What should we be discussing about exactly? Echo chambers can be problematic too:

*Me:* 2 + 2 = 4.
*You: *Yes! That's a provable FACT!
*Me:* Exactly and also 3 + 5 = 8.
*You:* Correct! Also a provable FACT!
:

Some things are controversial and art is also subjective. Opinions require justification depending on what they are about. If I say I like strawberry ice-cream, I don't need to give any support for the claim, but if I say _everybody_ likes strawberry ice-cream, then I must obviously be able to support the claim!

I'm sorry if I have driven someone away from this board and made the quality of the conversations lower.


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## Ryokan

71 dB said:


> Once I started to explore the World of music in order to find my own favorites I have discovered amazing things.




Would it be de-railing the thread by much to ask what amazing music you've discovered?


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## bigshot

We shouldn’t have 300 page arguments about whether wires have a sound.

When the discussion is 99.9% arguing about dumb stuff that is basically self evident, and the whole forum’s attention is focused on one troll who runs us all in circles, this isn’t properly described as Sound Science. A better title would be Stupid crap.


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## bigshot

Ryokan said:


> Would it be de-railing the thread by much to ask what amazing music you've discovered?


In this case, off topic is welcome! I haven’t gotten it yet, but I ordered a Japanese SACD with a quad surround mix of Santana’s Caravanserai. I’m very excited to get to hear it. I’m watching the shipping notices from halfway around the world.


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## The Jester

Maybe it’s more we are at cross purposes ?
CD audio properly done, 16/44.1 with dithering to reduce quantisation noise below audibility is fine for distribution,
Early CD’s that were analogue mixed and mastered before digitising for CD sounded ok too, it was the first generation all digital that had issues until 24 bit mixing and mastering was released in the late 90’s, (Protools etc) only then could multiple DSP plug ins be used with no effect on the end sound quality, record, mix, master at 24/48 or 96 and then down sample to 16/44.1 for CD release, 
Just about any home theatre amp in the last 10years, possibly more, will have 32bit DSP processors for similar reasons, complex multi processing without affecting audio quality.
I’m not immune to the possibility that with the right equipment I may be able to hear some subtle difference between 16 and 24 bit, even if I do where do I buy a hard copy of a 24/96 Album ?
“Digital Download” ?
No thanks, I may be old fashioned, or a collector,  but I prefer a hard copy with sleeve notes and artwork, lyrics etc, and song and album charts where someone had to buy a copy to be counted.


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## The Jester

71 dB said:


> What should we be discussing about exactly? Echo chambers can be problematic too:
> 
> *Me:* 2 + 2 = 4.
> *You: *Yes! That's a provable FACT!
> ...


Most of what I’ve read of yours can only improve things .. 👍


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## The Jester

bigshot said:


> In this case, off topic is welcome! I haven’t gotten it yet, but I ordered a Japanese SACD with a quad surround mix of Santana’s Caravanserai. I’m very excited to get to hear it. I’m watching the shipping notices from halfway around the world.


Might as well join in,
Listened to Moonage Daydream earlier today, an interesting look at David Bowie’s long career,
And while not something I’d use as a demo for the capabilities of Atmos, the live concert clips in particular gave a sense of “being there” as part of the crowd instead of just watching a concert video, that alone made the Atmos soundtrack worthwhile ….


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## 71 dB

Ryokan said:


> Would it be de-railing the thread by much to ask what amazing music you've discovered?


I have no problem listing what I have found, but what is amazing to me might be crap to you and vice versa. Everyone should find their own favourites. That's why I don't recommend (anymore) music to other people. Anyway, the history of my music journey has been something like this:

1987: I started listening to radio and noticed some pop music is actually to my liking.
1988: I discover British Acid House and become a fan of electronic dance music, especially the music of *S-Express*.
1991: Hardcore Rave is the name of the game. I hear "Charly" by *The Prodigy*.
1992: The magic year of Breakbeat: So much awesome stuff such as "Hurt You So" by *Jonny L*.
1993: I discover* Autechre* when I buy "Incunabula" because it has so cool cover art.

At this point the best most creative and intereting years of electronic dance music are over and the genre starts to fracture into more and more sub-genres with less creativity and more emphasis on establishing the genre style which restricts creativity. Good stuff still emerges, but my interest starts to drop overall. I start to look elsewhere for interesting music. My friend tells me how cool the melodies of classical music can be and get a little bit curious, but in my ignorance I am sceptic if hundreds of year of music can be compatible to my modern ears that have heard electronic music. In 1996 I start listening to a classical music radio station as background music while studying (I was in university at this time)

1996: December: I hear *Elgar*'s "Enigma Variations" on radio and I am totally blown away! Still my favorite composer!
1997: I explore classical music like there was no tomorrow. *J. S. Bach *and many many others.
2001: I start watching MTV and discover some nice Nordic pop/soft rock music.
2002: I discover New Age music.

At this point I have discovered so much stuff I need to take a break from intense exploring.

2008: BANG!! I discover *Tangerine Dream* and *King Crimson*! Two mega-discoveries (very unknown groups in Finland).
2011: I discover *Carly Simon* and I am amazed by how much I can enjoy this kind of music.
2012: I am amazed by how good American pop music has gotten, for example *Katy Perry*. Also *Herbie Hancock*.
2013: *Ke$ha. *I think she is mega-talented, a genius even.

Unfortunally pop music started to decline in 2013, but the stuff made in 2011 and 2012 is AMAZING in my opinion!

2015: I discover contemporal classical music (not atonal cacophony as I had assumed).
2016: Re-discovery of *Jean-Michel Jarre*. An idiotic compilation CD had killed my interest in early 90's.
2021: To my shock I realised I really like the hyper-syrupy yacht rock of *Air Supply*!

This is kind of the highlights only. So much various stuff in between. Especially the list of favorite classical music composers is extensive. What I have learned is to not be prejudiced and to trust my own instincts. Music that is considered complete junk by many (e.g. Ke$ha) can be amazing music to me while many highly regarded artists may mean nothing to me. I like both mainstream and underground music. I find favorites in many music genres, but that doesn't mean I enjoy those genres overall. To me some artists just "operate within a music genre" in wonderful ways.  Anyway, my listing may feel pretty anti-climactic to you. Sorry.


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## bigshot (Dec 10, 2022)

I don’t have Atmos, but in 5.1, the beginning animated scenes of the film had sound objects in the center of the room and crossing from one corner to another.

I made a playlist of Brian Eno ambient stuff on Amazon and I play it through my Alexa at low volume when I sleep. Music to be unconscious to!


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## 71 dB

The Jester said:


> Most of what I’ve read of yours can only improve things .. 👍


Thank you! 👍


----------



## castleofargh

bigshot said:


> Oh come on. He did a lot more than just not agree with an opinion. He denied facts. If people can just not agree by denying facts, then this isn't the sound science forum. We have always been able to demand that opinions be supported.
> 
> The quality of the conversations in this forum have fallen in the past six months to a year. The forum is overrun by invaders with opinions that deny facts. They specifically target the people who they think are the "bosses" of the forum. That is exactly what drove away Steve Eddy, Pinnahertz and now I assume Gregorio. The frustration of having to deal with the same dumb cable, HD audio and "everything sounds different" arguments makes us get short with each other and we fight among ourselves over things that don't matter.
> 
> ...


Then how come you spend so much time replying and talking about people you don’t want to see posting? That I will never understand. If we have a true desire for fact based conversations and we don’t wish to ”feed the trolls”, how about we act like it?
Who cares if a guy makes a few empty claims, they’re empty claims, they don’t weight anything. Let that stuff fly away with the wind.
I said the same in the off topic thread, it takes 2 to tango. Nobody so far has posted for pages without one of ”us” being his very motivated dance partner. What I see most of the time looks like this:
- "you hang up"
-"no U"
-"oh stop it, I don't want to talk to you anymore"
-"then hang up!"
-"no you hang up first!"
etc.  


The actions don't match the will.



Now that I've shamelessly participated in the off topic, I will naturally tell you all to go discuss music discoveries in another thread(in the pub or the one about Sound science idea of good music that @Steve999 made a while back that made me feel like an idiot. Or make a new one. We have like 10000 billion threads(perceived number) on burn in and cables, we can afford 2 or 3 on music).
 Anyway, do what I say, not what I do!


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## redrol

Bigshot is an old grumpy fart that likes the Beatles.  Enough said.


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## The Jester

redrol said:


> Bigshot is an old grumpy fart that likes the Beatles.  Enough said.


No, no .. I won’t have that !
He likes David Bowie too … 😬


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## 71 dB

It is important to make the distinction between audio formats in studio and consumer audio formats. Those things have quite different requirements. I'd use this analogue:

*You can deliver a DAC to a customer inside a small box, but you need production facility to make them.*


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## bigshot (Dec 10, 2022)

redrol said:


> Bigshot is an old grumpy fart that likes the Beatles.  Enough said.


I like Snoop Dogg too!


The Jester said:


> He likes David Bowie too … 😬


Actually, I only like certain Bowie albums, not others. And I don't care for him as a person. I like Frank Zappa better!

I reserve the right to call a spade a spade.


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## Ryokan

71 dB said:


> I have no problem listing what I have found, but what is amazing to me might be crap to you and vice versa. Everyone should find their own favourites. That's why I don't recommend (anymore) music to other people. Anyway, the history of my music journey has been something like this:
> 
> 1987: I started listening to radio and noticed some pop music is actually to my liking.
> 1988: I discover British Acid House and become a fan of electronic dance music, especially the music of *S-Express*.
> ...



Thank you for the detailed reply. I don't often listen to pop but like A-Ha from Norway. I was into classical heavily for about a year but then discovered extreme bands (some from Scandinavia) around 2003 and hardly play much classical now, sometimes Holst's The Planet Suite, Scheherazade by Rimsky Korsakov, Borodin: Polovtsian Dances, the modern composer: Zbigniew Preisner some Brahms: symphony no.4 Tchaikovsky: 1812 Overture, Stravinsky: Firebird suite, Shotaskovich: The gadfly, and Rachmaninov. I didn't listen to the Prodigy until recently as when younger Keith Flint looked too scary, turns out he was a really cool guy.
I bought In The Court Of The Crimson King when I was about 14 because I liked the cover art, didn't realise it was a classic. Also try Camel you might like them.

Any way just a quick reply as castleofargh rightfully pointed out this isn't a music thread.


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## bigshot

I prefer the period when King Crimson sloughed off the art rock pretensions and with Adrian Below became more like a Talking Heads sort of band. And I think that the three Robert Fripp solo albums and his work with Darrell Hall and Peter Gabriel are better than anything he ever did with King Crimson. The new Fripp box set reminded me of how good that period of his career was and gave me newfound respect for his Frippertronics, which is infinitely more musical than Eno's experiments along those lines. In general, I prefer cogent, concise musical statements. Long, meandering noodling doesn't hold my interest, especially if it pretends to be more than it actually is. The best thing that the new wave movement did was to put punch and directness back in a sub-genre of rock that had become slack and lazy. In fact, new wave did for art rock what punk did for heavy metal.



I recently watched this documentary along with several others on YouTube and it put into perspective what prog rock was... it was basically a middle class music that combined the the lower classes' rock n' roll and blues with high brow classical music- essentially taking hot and cold and making lukewarm. Frank Zappa was correct in saying "Art rock is neither." Prog rock was neither rock n' roll, nor high brow classical.  It eschewed the best aspects of both: the directness and honesty of rock n' roll and the structural architecture of classical. New wave took modern art as the model instead of classical music and did a much better job of becoming "art rock" and punk did a much better job of recapturing the directness and honesty of rock n' roll and translating it to a more modern, urban context. Both were like a breath of fresh air after dinosaur hard rock and album side long ELP classical noodling that had slowly ground the late 70s down to a halt. It's funny to see what happened to prog bands like Yes and Genesis when they finally saw the handwriting on the wall and realized that new wave and punk had completely replaced them. They were gobsmacked and had no idea what had happened to them.


----------



## megabigeye

Coughcoughsoundsciencemusicthreadcoughcough

Oh, excuse me. I had a tickle in my throat.


----------



## The Jester

Point taken,
Posted there for 71db on music I mentioned earlier …


----------



## T 1000

You have repeatedly stated the fact that you at Sound Science are not interested in what anyone hears if it cannot be measured.
OK, let's apply that rule of yours here; 99% of people don't listen, but what about the facts.
As a completely unprofessional person, I will ask you a question, and I am interested in your answer (everyone is welcome).
--Why are files in HI-RES much "heavier" than 16bit/44.1kHz
--Can all DACs process all DSD samples (need DAC for such processing)
Personally, I couldn't tell the difference between 44,1-48-96..., but what I hear or not is not a scientific fact.
So what are the scientific facts about HI-RES


----------



## bigshot

See my sig file link, CD sound Is All You Need. Thanks for playing! Have a nice day! We have some lovely parting gifts for you!


----------



## The Jester

Probably need someone experienced in the hardware application of sampling theory for a definite answer,
Maybe an analogy would be to go back in the history of analogue recording via tape, early examples were 2 and maybe 4 channel, so with those limitations editing or “mixing” either involved modifying the original audio and putting it back onto tape again with physical losses, or physically cutting and splicing the tape itself, move on to more modern 24 track recorders and various channels could be mixed in at will without altering the individual channels, if you look up the sample numbers between 16/44.1 and 24/96 there’s a huge increase with resolution way above the normal audio band, so plugging in various high rate digital filters and effects in the mixing environment has no audible effect, 
So 24/96 is virtually immune to correct application of various filters in the recording chain, but attempting to mix a 16/44.1 digital recording has severe limitations,
So 24/96 for recording/mixing/mastering,
16/44.1 “CD quality” for consumer,
Don’t ask me any further questions as I’m like you, not a professional just a curious end user.


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## T 1000

Anyone else have an explanation?
I'm really interested in the scientific explanation
Personally (as I already said) I don't hear any differences, but no, I can claim that someone with better hearing and better equipment than me can't also not hear.
So scientifically, not subjectively


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## T 1000

The Jester said:


> Probably need someone experienced in the hardware application of sampling theory for a definite answer,
> Maybe an analogy would be to go back in the history of analogue recording via tape, early examples were 2 and maybe 4 channel, so with those limitations editing or “mixing” either involved modifying the original audio and putting it back onto tape again with physical losses, or physically cutting and splicing the tape itself, move on to more modern 24 track recorders and various channels could be mixed in at will without altering the individual channels, if you look up the sample numbers between 16/44.1 and 24/96 there’s a huge increase with resolution way above the normal audio band, so plugging in various high rate digital filters and effects in the mixing environment has no audible effect,
> So 24/96 is virtually immune to correct application of various filters in the recording chain, but attempting to mix a 16/44.1 digital recording has severe limitations,
> So 24/96 for recording/mixing/mastering,
> ...


@ The Jester- I didn't see your post when I posted mine.
so I wasn't thinking of you


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## T 1000

@ castleofargh - what is your position on these issues?


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## T 1000 (Dec 12, 2022)

If I don't hear, no one can?
I came to my hometown on vacation.
I have a Marantz HD-DAC 1i in my apartment, the same headphones I have where I work, but with a more advanced system than the HD-DAC
When I listen to music with a more advanced system, every time I put headphones on I am fascinated by the sound, always, but here with the Marantz the stage presentation is so poor, the sound is so thin, there is no bass, simply sad.
My point is, can many who have far better devices than me (from my main  system) and more refined hearing be able to hear the higher resolution contributions?


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## bigshot (Dec 12, 2022)

Mixing 16/44.1 doesn’t have severe limitations at all. Some great sounding albums were mixed 16/44.1. It’s better sound than analog tape on every metric, and therefore easier to mix with than 24 track tape. The only reason mixers use 24/96 is to allow more room for processing. It’s always preferable to have extra room. Even if you don’t end up using it, it’s nice to know it’s there. But that doesn’t mean mixing without it is a severe limitation.


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## T 1000 (Dec 12, 2022)

bigshot said:


> 16/44.1. It’s better sound than analog


I remember the first comparisons that were imposed on me and my generation, which sounds better, vinyl or CD
The conclusion was that the CD has passionless (cold) sound


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## The Jester

bigshot said:


> Mixing 16/44.1 doesn’t have severe limitations at all. Some great sounding albums were mixed 16/44.1. It’s better sound than analog tape on every metric, and therefore easier to mix with than 24 track tape. The only reason mixers use 24/96 is to allow more room for processing. It’s always preferable to have extra room. Even if you don’t end up using it, it’s nice to know it’s there. But that doesn’t mean mixing without it is a severe limitation.


Depends on the effort used I guess,
Just going back to buying the early CD’s when there was need for the SPARS code, the majority of CD’s that sounded harsh and bright were from the late 80’s to the mid-late 90’s, some of those “AAD” recordings were only recently done better with a good remaster.


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## bigshot (Dec 12, 2022)

Vinyl is inferior to CD sound in the areas of frequency response, signal to noise, wow and flutter and distortion. It can sound good if done right, but 16/44.1 is audibly transparent from the get go. CD sound is perfect for human ears.

Jester, listen to Donald Fagin’s The Nightfly or Dire Straits (I think it was Making Movies). Those were recorded and mixed 16/44.1 and they are among the best sounding albums ever made. That whole “digital glare” thing was BS. People’s systems just weren’t accustomed to handling upper frequencies properly because they weren’t present on LPs. LPs had high end roll off to reduce distortion due to record wear.

The CDs from the 80s that people claimed had “digital glare” are the ones that people now say sound better because they haven’t been remastered. CDs have always been perfect sound and they still are. We’ve just got better DACs now than in the days of NOS DACs.


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## The Jester

Not arguing that CD sound is more than enough,
All the early CD’s sounded fine generally, there are some pretty ordinary exceptions in all formats, but there were 3 changes in the sound of CD’s,
“AAD”, “DDD” and then the SPARS code disappeared in the late 90’s when computer processing caught up with the requirements of digital audio manipulation with 24 bit mastering software and pure digital plug ins.


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## bigshot

Those are differences in how the albums were mastered. It didn’t reflect the sound quality of digital as a medium. Aside from oversampling DACs, the red book format is basically the same as it always has been. CDs haven’t gotten better. That whole digital glare thing in the beginning was a crock.


----------



## The Jester

Indeed, decimation was always the easy part,
Reconstruction was the more difficult part, but the ability to manipulate mix and master in the digital format followed the advances in computer technology,
Did you have a PC when CD was first released ?
My first was a 286 with 4K of ram and a 20MB Hdd in the late 80’s


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## bigshot

The only thing missing in the early days of digital was editing and processing. My first job in the film business was working with a sound mixer. We did the first TV show recorded digitally using a Sony beta portapack. I believe that was 16/44.1. We transferred it to magstripe because that was the only way to edit and mix film. All that changed within a year or two.


----------



## old tech

T 1000 said:


> I remember the first comparisons that were imposed on me and my generation, which sounds better, vinyl or CD
> The conclusion was that the CD has passionless (cold) sound


That's interesting anecdote as it is the complete opposite for my generation. I was in my early 20s when CD players were first released and I remember the hifi shows where they showcased the new technology. Nearly everyone and at every show were blown away on how much better CD sounded compared to the high end record players and cassette decks. Most of my friends were into hifi but I was one of the first to purchase a CD player (1984) and they would come around and be amazed by the clarity and dynamics compared to LP records.

If you look at hifi publications back then, nearly all of them were extolling the virtues of CD and how it was a significant jump in high fidelity compared to the analogue media of the day. The only exceptions were a few cranks like Fremer. It wasn't until the very late 90s/early 2000s that we started to hear claims such as yours and not coincidentally it aligned with the loudness wars which were giving CDs a bad rap in the pop/rock genre.

If you consider that the greatest demand on fidelity is in the classical music genre (particularly full orchestral music), listeners overwhelmingly took to CDs and never looked back, even during the loudness wars era as this genre of music tends to be mastered well. Indeed, it was enthusiasts of this genre that led the development of digital audio as they were never satisfied with the limitations of analogue tape and records.


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## bigshot (Dec 12, 2022)

I was around back then too and was similarly impressed. I wasn’t keen on replacing my whole LP collection though, so I grumbled and griped for a year or so before giving in.

I do remember reviews in the audio magazines claiming that LP sound had something CD didn’t however. At the forefront of that were Lincoln Mayorga and Doug Sax who were heavily invested into producing direct to disk LPs at the time. That’s where I first read the hogwash about stair steps and digital glare. The got pummeled in the letters to the editor the next month, and there was an article about how nyquist actually worked. They continued to argue the point for a couple of months, then I guess Doug Sax got more familiar with digital pro audio, and they quietly issued an apology and retraction. Sheffield Lab started backing up their D2D sessions with digital, but they kept flogging the obsolete tech for years after conceding.

The bologna they started got picked up and repeated by audiophools, and stair step myths keep getting mentioned long after it was proven incorrect.

My theory is that the current variety of pseudoscience is based on the way analog used to be (ie: generation loss, veils on sound, added detail and resolution, and wider soundstage) when the causes for those things in the analog era no longer exist. It’s become like old wife’s tales.

Classical music has led the charge towards high fidelity for over a century.


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## Ryokan

I wasn't too bothered giving up vinyl and going over to cd's as my well played albums had lots of distracting surface noise.


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## redrol

Vinyl sucks.  I have no idea why anyone cares about that potato tech.  I didn't mind tapes but CDs are where its at.  Cold?  My a55.  That is a meaningless statement.


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## bigshot

There's an ocean of music that never made it to CDs. If you have an interest in music from the past, obsolete formats are the best place to find it.

I saved up in high school and bought a Thorens turntable. Almost all my LPs (I still have them all) are in good shape.


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## redrol

All the old vinyl I care about from 60-70s on has been ripped to PCM wave.  I have tons and tons of old Jamacian reggae albums ripped off of vinyl.  I still wouldn't use vinyl unless I somehow suddenly got thousands of old albums from that era.   I get your point, but im only 47.  I collected tapes as a young kid but those are dead now, so CD it is.


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## bigshot

That's probably a great reggae collection! That is exactly the sort of stuff where CDs only got the "greatest hits" and not the bulk of the album releases.


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## T 1000 (Dec 13, 2022)

Good times...
But to return to the present.
44.1 vs HI-RES
The music in HI-RES are more voluminous in bytes, how is this voluminousness reflected in the audio?
Does HI-RES have its theoretical advantage?


----------



## bigshot

In frequencies beyond the range of human hearing and noise floors so low you would have to turn the volume up to ear splitting levels to hear.

See the link in my sig file “cd sound is all you need”. Then you won’t have to keep asking the same question over and over.


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## T 1000

I just came across this explanation about the size of the file


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## T 1000 (Dec 13, 2022)

?...
What bothers me is that, despite the fact that HI-RES has no effect on audibility, there are many reports of differences, and not small ones compared to 44.1, from people who have no interest in HI-RES production.
Ignore my comments
I don't want to open pandora's box
I'm sorry


----------



## bigshot

See the link in my sig file.


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## 71 dB

redrol said:


> Vinyl sucks.  I have no idea why anyone cares about that potato tech.  I didn't mind tapes but CDs are where its at.  Cold?  My a55.  That is a meaningless statement.


Vinyl has its own issues and problems, but I wouldn't say it sucks. It is not _my_ chosen format, but I don't mind other people using it. The distortions of vinyl make the format less transparent, but that can be beneficial as it warms up the sound and masks harshness caused by bad production, mixing and mastering. So, vinyl protects against "cold harsh sound." whereas CD protects against surface noise and other things that are problematic for vinyl. CDs of course can sound as warm as you want. All you need to do is to put warm sound on them and they will transparently play back just that. For some reason that's not done very often despite of consumers loving warm sound. Especially in the genre of metal music people constantly complain how cold CD sounds compared to vinyl. Why not emulate vinyl distortions on the CD master to make it sound closer to vinyl? The "magic" of vinyl is that it introduces non-linear distortion in mid/side form. This can be emulated in music production by processing the sound in M/S mode and adding different kind of non-linear distortions to the Mid and Side channels. The sound gets warmed up, the spatiality gets smeared more relaxed/diffuse and when applied to the downmixed track the different tracks get glued together increasing "cohesion" of the music, but I believe the benefits depend on the genre of the music and how the music has been produced, mixed and mastered in the first place. So, I wouldn't add vinyl distortion emulation on carefully executed classical music recordings for example.


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## bigshot (Dec 13, 2022)

Artifacts don’t improve sound. Lack of artifacts do. All that stuff about analog warmth and euphonic crackle is just excuses. People don’t really like Lps for their weaknesses. LPs sound good depending on how good the pressing is… free of excessive surface noise and distortion, well balanced response, etc. No different than any other format. There’s lots of great sounding LP records of all types, including classical. As a format, it’s capable of high fidelity sound. It just isn’t perfect and convenient like CDs.


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## 71 dB

bigshot said:


> Artifacts don’t improve sound.


The question is what are artefacts and what aren't? When a mixer adds a tube amp emulating saturation plug-in on a track, is he/she adding artefacts or making the mix sounding better (in his/her own opinion anyway)? In my opinions the latter.


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## bigshot (Dec 13, 2022)

Artifacts are deviations from fidelity. Tube amps are liked by people for their artifacts. People like LPs that are as free of artifacting as possible.

Records don’t all sound warm, and the high end roll off is above 10kHz, which is high enough to not really matter much. In fact if anything, LPs are lacking sub bass compared to CDs, not high end.

Sound quality isn’t judged by the edges of the spectrum. It’s judged by the most common frequencies. LPs do a great job of that if care is taken in the pressings.

I wouldn’t go back to LPs because they weren’t convenient. But they were capable of sounding good.


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## VNandor

T 1000 said:


> 44.1 vs HI-RES
> The music in HI-RES are more voluminous in bytes, how is this voluminousness reflected in the audio?
> Does HI-RES have its theoretical advantage?


The reason hi-res takes up more space is because it contains more bits that have to be stored. A 1 hour long song isn't six times better than a 10 minute song just because it takes up six times more space.

You could take a 44.1kHz/16bit wave file and convert it to 192kHz/24bit. Do you think that would somehow add more information to your file? It would encode the same information but less efficiently (as in using up more bits you have to store). If you converted a wave file to a FLAC file, it would take up less space, yet it would represent the same information as the wave file.

Hi-res does have their theoretical advantages. A higher sample rate lets the digital signal represent a higher bandwidth analog signal, and a higher bit depth lets it to do so with less quantization error. They also have some practical advantages but they aren't the ones I mentioned because in practice there is nothing to gain with recreating a frequency greater than 20kHz, or having a better signal to noise ratio than 96dB when listening to music.


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## T 1000 (Dec 13, 2022)

The basics are clear to me, but I am having trouble with errors that are reported from the theoretical point of view and the listening test.
I don't want to discuss certain disagreements between my experience and the opinions of the Sound Science members, but we agree on this issue for now. I want to give myself a perspective on the possibilities of HI-RES with adequate equipment, and whether it makes sense, or is it just theoretical and there are no auditory benefits.
In my searches, I came across comments like this, and there are many more that point to the usefulness of HI-RES

"Don’t worry, we’re not simply dismissing the faithful old CD. CD quality is still great. The problem for us is that while some material sounds incredible, quite a lot of it doesn’t! Much has been written about recent problems with the mastering and recording of music with reductions in the amount of dynamic range in some albums making them sound terrible on a good hifi system. We feel that our ATF upsampling and adjustable filters can alleviate some of these issues but obviously good quality recordings to start with will always come out on top.

Therefore beyond the measurements and arguments over human hearing, we feel that high res music has a role to play. As well as the technical benefits of 24 bit mastering which you can choose to believe is useful or not, they are generally recorded and mastered with far greater care and attention than many mainstream releases. For example, The 2nd Law by Muse. This is available on CD and as a 24/96kHz high res version and it is fair to say that the difference between the two versions is not small. This is in no small part because the high res version has completely different mastering and vastly greater dynamic range that gives the album a scale and depth that simply isn’t present on the CD mix recorded with radio broadcast and MP3 conversion in mind. Is this cheating? It may well be but we know which version we would rather listen to!"


I am not willing to actively participate in clarifying the problem without having my own built-in attitude, that's why I feel absolutely inferior in this topic


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## T 1000 (Dec 13, 2022)

For now, this question has only an optional meaning for me.
My practical interest is related to the removal  noise and various other disturbances like RF, that mask the original audio information


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## T 1000 (Dec 13, 2022)

.


----------



## VNandor

Oh man, NOT THE STAIRSTEPS AGAIN, I'm going to lose it! The picture is completely wrong, digital signals aren't continous, they are made up by momentary samples. Playing connect to the dot with them like the 2nd and 3rd picture do it is completely nonsense. The second and third pictures are meaningless made up squiggles not "MP3 16bit/44.1kHz" nor "Hi Res Audio 24bit/192kHz". Think of the digital signals as a lollipop graph, because they aren't defined between the sample points, something like this:



The DAC uses these points along some additional information to recreate the original analog waveform. The recreated waveform will not have any of the stairsteps your picture shows. Here is a much better visual representation of how the reconstructed analog signal is going to look like after converted from digital to analog.





There are benefits to using Hi-res audio, but they aren't present when listening to music. Hi-res music often sounds better than their 44kHz 16bit counterparts but even your quote points out it's because Hi-res is produced with greater care and not due to the different file format. Noone wants to buy a worse produced version of the same music so of course people are going to buy the hi-res version. But that has very little to do with the "science" of hi-res.


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## T 1000 (Dec 13, 2022)

Yes, OK, we have come to a positive development in my education, and that is that what I thought I knew about this, I was mistaken.
now I feel really useless.
Please ignore me

BTW-- thanks for pointing


----------



## T 1000

In any case
I could educate myself about these issues on the Internet, but interactive is more interesting, and it led to positive knowledge for me.
Once again, thank you


----------



## sander99

T 1000 said:


> This is in no small part because the high res version has *completely different mastering* and vastly greater dynamic range that gives the album a scale and depth that simply isn’t present on the CD mix


(This is all explained in “cd sound is all you need” in bigshot's sig, that he pointed at several times...)
If you convert this different master to cd format, it will sound exacly the same with the same great dynamic range scale and depth (under normal listening conditions, so for example not listening at crazy high volume to extremely soft parts of the recording). The official cd release sounds less dynamic only because another master was used.


----------



## T 1000

sander99 said:


> (This is all explained in “cd sound is all you need” in bigshot's sig, that he pointed at several times...)
> If you convert this different master to cd format, it will sound exacly the same with the same great dynamic range scale and depth (under normal listening conditions, so for example not listening at crazy high volume to extremely soft parts of the recording). The official cd release sounds less dynamic only because another master was used.


YES, it is clearer now, new information has emerged that has helped me understand the basics and what makes a digital record more effective.
For now, the new perspective is enough for me


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## Ghoostknight (Dec 13, 2022)

i hear a improvement with high res over cd quality (specially 16bit vs 24bit) but i think also higher samplerates have some benefit
1. it moves the filter of the dac far behind audible
2. (maybe) it makes it easier for the dac to reconstruct the wavewave if there are more than 2 samplerates per full frequency cycle in the case of 20-24khz

and why not just go with the best quality available today? it makes it 
1. more futureproof if things are "happening/changing" in the future 
2. storagesize isnt really a issue today, maybe with your phone somewhat still but with a phone dac + mediocre headphones you can also just go with mp3 
3. you can always downgrade music but not upgrade so specially masters should ALL be done in 24bit imo


----------



## T 1000

Specifically, I have Master quality on Tidal, which I pay more for, but I don't feel the benefits with my system. I was wondering if my system was problem...
Of course, size is not a problem


----------



## Ghoostknight

T 1000 said:


> Specifically, I have Master quality on Tidal, which I pay more for, but I don't feel the benefits with my system. I was wondering if my system was problem...
> Of course, size is not a problem


dont worry to downgrade if you dont hear a improvement -at all- but i also have to say the changes are very subtil with cd quality vs high res but there is a little difference with revealing systems, so best to start from the source and go with the best which is high res 

(tho i tried Tidal some time ago where it was possible to stream in flac instead of mqa and Qobuz sounded better in comparision (dont ask me why but there are also few other people that report the same), this made probably even more of a change compared to cd quality vs high res)

i also dont hear the difference with my DT880 which are not bad but i hear this little difference with my room corrected studio monitors


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## bigshot (Dec 13, 2022)

Ghoostknight said:


> i hear a improvement with high res over cd quality (specially 16bit vs 24bit) but i think also higher samplerates have some benefit
> 1. it moves the filter of the dac far behind audible
> 2. (maybe) it makes it easier for the dac to reconstruct the wavewave if there are more than 2 samplerates per full frequency cycle in the case of 20-24khz


1.) If you are using an oversampling DAC (which you almost certainly are) the filter is already moved way beyond where it needs to be to be transparent.

2.) According to the Nyquist theory, two points are all that are needed to *perfectly* recreate the waveform. More points don't increase the resolution. The range between 20Hz and 20kHz is exactly the same in 16/44.1 as it is 24/96.

All of the improvements above 16/44.1 are beyond the range of human hearing. In a blind, level matched, direct A/B switched listening test of the same audio in 16/44.1 and 24/96, you would not be able to hear a difference. The difference you think you hear is likely due to differences in mastering. A song on Tidal won't necessarily sound the same as the same song on CD or on another streaming service. But that is due to mastering, not the file format. High data rate lossy, lossless and HD audio all can be audibly transparent. Done properly, they should all sound the same.

The article in my sig titled "CD Sound Is All You Need" clearly explains how it all works in as much detail as you would want to know.


----------



## Ghoostknight

bigshot said:


> 1.) If you are using an oversampling DAC (which you almost certainly are) the filter is already moved way beyond where it needs to be to be transparent.
> 
> 2.) According to the Nyquist theory, two points are all that are needed to *perfectly* recreate the waveform. More points don't increase the resolution. The range between 20Hz and 20kHz is exactly the same in 16/44.1 as it is 24/96.





bigshot said:


> The article in my sig titled "CD Sound Is All You Need" clearly explains how it all works in as much detail as you would want to know.


i know the theory tho it still makes a difference and depending on the dac the filter already starts at the audible range

and im pretty sure it was because of tidal, not the master, i tried with many albums and it kinda doesnt make sense to offer different masters for different streaming services


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## T 1000 (Dec 13, 2022)

Ghoostknight said:


> dont worry to downgrade if you dont hear a improvement -at all- but i also have to say the changes are very subtil with cd quality vs high res but there is a little difference with revealing systems, so best to start from the source and go with the best which is high res
> 
> (tho i tried Tidal some time ago where it was possible to stream in flac instead of mqa and Qobuz sounded better in comparision (dont ask me why but there are also few other people that report the same), this made probably even more of a change compared to cd quality vs high res)
> 
> i also dont hear the difference with my DT880 which are not bad but i hear this little difference with my room corrected studio monitors


There are differences between stream services. I also heard that Qobuz is better than Tidal. I hear songs better on Tidal than on Spotify Premium.
The best quality I get with Tidal is when I play it through Audirvana Studio
And as far as financial savings are concerned, in the quality differences on Tidal, I'm guided by logic, it's a few euros more, I don't care


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## bigshot (Dec 13, 2022)

Ghoostknight said:


> i know the theory tho it still makes a difference and depending on the dac the filter already starts at the audible range
> 
> and im pretty sure it was because of tidal, not the master, i tried with many albums and it kinda doesnt make sense to offer different masters for different streaming services


Nope.  Oversampling DACs don’t have audible filtering by definition.

Streaming services have requirements for the mastering of the files that are delivered to them. Apple has a whole list of technical requirements for acceptance into their music store.

The only way to know if the difference in sound is due to the file format or the mastering is to take a master file and bounce it down to compressed and then compare using a blind, level matched, direct A/B switched listening test. I’ve done this. You haven’t.

Sorry to let you down like this. You can believe whatever you want, but you’re wrong on these points.

For more detailed info, see the link in my sig "CD Sound Is All You Need".


----------



## Ghoostknight

bigshot said:


> Nope. Oversampling DACs don’t have audible filtering by definition.


oh i didnt think about that, i get it, well it was just a guess to be honest, tho it doesnt make the differences i hear dissappear (i hear more differences in 16 vs 24 bit instead of samplerates tho)



bigshot said:


> Streaming services have requirements for the mastering of the files that are delivered to them. Apple has a whole list of technical requirements for acceptance into their music store.
> 
> The only way to know if the difference in sound is due to the file format or the mastering is to take a master file and bounce it down to compressed and then compare using a blind, level matched, direct A/B switched listening test. I’ve done this. You haven’t.


i think you you are talking mostly about volume requirements, i level matched as best i could, tho the strange thing was that tidal sounded like more treble and less bass (if i remember right) compared to Qobuz, Qobuz sounded the same compared to CD, maybe this changed by now, i did the comparision in 2017/2018 i think and just stuck with Qobuz since then,  i kinda wonder if qobuz has any of those upload requirements

beside that some time after my comparision tidal introduced/enforced MQA and i dont wanna support it, its properitary bs that is inferior to plain old flac/wav in many ways with the only benefit of being a little smaller imo



bigshot said:


> You can believe whatever you want


will do, i kinda trust my ears (most of the times), specially if i can hear the same things over and over again on different days etc
tho i dont wanna go into a objective vs subjective fight here, take my findings with a grain of salt i guess


----------



## 71 dB

Ghoostknight said:


> i hear a improvement with high res over cd quality (specially 16bit vs 24bit) but i think also higher samplerates have some benefit
> 1. it moves the filter of the dac far behind audible
> 2. (maybe) it makes it easier for the dac to reconstruct the wavewave if there are more than 2 samplerates per full frequency cycle in the case of 20-24khz
> 
> ...


Hi Ghoostnight! You are pretty new here I see and probably a young dude am I wrong? This post of yours indicate that you do not have very deep knowledge and understanding of digital audio. We all start from zero and this place is a chance to learn from those who know better. It is up to you if you use that opportunity.

I am not saying 44.1 kHz/16 bit can do _anything_, because it of course can't, but considering what the technical requirements for audibly transparent audio are for human hearing, that format fulfil those requirements and then some. In fact, the bit depth could be dropped to just 13 bits, and it would still be enough for consumers in any practical listening scenario.

People hear differences between CD quality and hi-rez quality because:

- The masters aren't same.
- Placebo effect can easily fool us.
- Level differences

Differences tend to disappear when proper double blind tests are conducted.


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## GoldenOne

bigshot said:


> Nope. Oversampling DACs don’t have audible filtering by definition.


The audibility of reconstruction filters is very much up for debate and there is a surprising lack of conclusive study into the area.
Though also worth noting that DAC reconstruction filters vary greatly and oversampling alone doesn't mean a filter is going to be similar. One could oversample with a zero order hold approach to get a result that behaves the same as NOS for example.

Also many DACs reconstruction filters do not adhere to nyquist, and for the ones that do there is the argument of audibility of lost information due to early rolloff. 
Some of them have treble rolloff that would absolutely be audible.


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## bigshot (Dec 13, 2022)

There are filters that create audible differences. I have no idea why anyone would want to use one of those. I don't know why anyone would want a reconstruction filter that doesn't follow nyquist or a NOS DAC either. You have to go out of your way to deliberately mess up the sound to create a situation with an audible difference. All of these are pathological exceptions. They don't represent how a typical DAC or player performs.

Properly designed and manufactured DACs and players that are configured to perform to spec are audibly transparent. It isn't hard to find one of those. I have a $40 Walmart DVD player that fits that bill. Apple makes one for $8. Just about any DAC or player you can find at Amazon is the same. You have to spend a lot of money to get compromised performance!

In Sound Science it's common to spend more time discussing exceptions to the rule that are rare and specialized than to talk about the sorts of things consumers will actually run into. This misplaced focus gives newbies a distorted view of what the world of home audio is actually all about. I'd bet a quarter that bringing that stuff up just confused T1000 and ghoostknight.


----------



## GoldenOne

bigshot said:


> I don't know why anyone would want a reconstruction filter that doesn't follow nyquist


Because to fully adhere to nyquist would require infinite computing power. Something we cannot achieve.

We can get closer and closer by throwing more compute power at the problem like products such as the chord DACs or software like HQPlayer and PGGB do. But we can never achieve truly perfect reconstruction




bigshot said:


> Properly designed and manufactured DACs and players that are configured to perform to spec are audibly transparent. It isn't hard to find one of those. I have a $40 Walmart DVD player that fits that bill. Apple makes one for $8. Just about any DAC or player you can find at Amazon is the same


Personally I don't find the audibly transparent part to be the case.

But in terms of being 'to spec' on the topic of filters. Most dacs reconstruction filters don't even attenuate to 16 effective bits by the nyquist frequency let alone do so with a particularly steep rolloff.

Also the apple dongle's reconstruction filter isn't even linear phase


bigshot said:


> In Sound Science it's common to spend more time discussing exceptions to the rule that are rare and specialized than to talk about the sorts of things consumers will actually run into. This misplaced focus gives newbies a distorted view of what the world of home audio is actually all about. I'd bet a quarter that bringing that stuff up just confused T1000 and ghoostknight.


Whilst it's certainly not a good idea to confuse or mislead people about the benefits of spending huge amounts of money on stuff that doesn't make a difference, the same applies to making absolute statements about things NOT mattering or making a difference when either there isn't sufficient evidence to make that claim, or in some cases even evidence to the contrary.


On the topic of this very thread for example:
https://www.aes.org/e-lib/browse.cfm?elib=18296

"400 participants in more than 12,500 trials. Results showed a small but statistically significant ability of test subjects to discriminate high resolution content, and this effect increased dramatically when test subjects received extensive training"


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## bigshot (Dec 13, 2022)

What commonly available DACs or players have you found to be clearly not audibly transparent in a controlled listening test using music? Or measurements of the line output to show they aren't clean and flat to human ears? I haven't found any except for the pathological exceptions.

It seems to me that if a DAC or player can't produce 20 to 20 flat with no distortion, it isn't much of a DAC or player. It would be good to know which ones those are so they can be avoided.

As for super audible frequencies. I don't doubt that they can be perceived. That has been proven using brain waves. But perception isn't the same as hearing. They add nothing to the quality of sound when listening to home audio equipment in a living room. And it's certainly possible to hear noise floors if you loop fadeouts and gain ride. But no one listens to music that way. For the purposes of listening to music in the home, HD audio is pretty much useless unless the mastering has been done with more care.


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## GoldenOne (Dec 13, 2022)

bigshot said:


> What commonly available DACs or amps have you found to be clearly not audibly transparent in a controlled listening test using music


I've found various dacs to be discernably different. Actually a friend and I made a device to perform unassisted controlled blind tests which has been really useful when testing both DACs and amps.




bigshot said:


> Or measurements of the line output to show they aren't clean and flat to human ears?


This is the important part.

To be clear, I'm not a believer in magic. If two things sound different, then they will measure different and there will be something to explain the audible difference.

I wouldn't have purchased an APx555 if I didn't believe measurements were important.

The key thing is that the measured differences might not be intuitive or obvious. I don't believe that basic measurements like SINAD @ 1Khz alone can actually describe/characterise the sound of a device or guarantee transparency nor do I feel this makes objective sense.

I've not found any DACs with a variation in frequency response beyond early treble rolloff due to the choice of reconstruction filter, but FR was never the issue. There are soooo many factors beyond this that can have an audible impact

Jitter
Phase linearity
THD level
THD structure/composition
IMD
DIM
Noise Modulation
Reconstruction filter
Crosstalk
THD vs frequency
And many more

If two things 'measure the same' they will sound the same. But you'd be surprised how incredibly rare it is to find things that could really be described as 'measuring the same'.



bigshot said:


> It seems to me that if a DAC or player can't produce 20 to 20 flat with no distortion


Producing it flat is easy.
With no distortion is impossible


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## bigshot (Dec 13, 2022)

You're new, so I need to state what I've made clear in the past. I'm not talking about sound that is measurable, but not audible to human ears. And I'm not talking about sound that can only be heard in lab situations with test tones. I'm always speaking from the position of a consumer of home audio, not a sound engineer and not a scientist, and definitely not an anal retentive audiophile... because I'm always looking to improve the sound quality of my own system for the purpose of home listening.

So here's my question... In normal home use without employing pathological settings, listening to commercially recorded music, which DACs or players are not audibly transparent? Just to make it simple, which ones are the most obviously colored?


----------



## GoldenOne

bigshot said:


> I'm not talking about sound that is measurable, but not audible to human ears


Which is exactly the issue.

"Audibly transparent" is not something we can currently conclusively say for any product.

There are a lot of factors or combinations of factors where we simply do not have a concrete understanding of what the thresholds of audibility and/or audible effects are.

Even if we take something prominent, like jitter.
This gets talked about a lot and there are whole market segments of products dedicated to reducing it.
And yet we don't actually have a conclusive answer for what the thresholds of audibility are. Let alone for the different structures/types of jitter or the ways in which the effects may vary depending on the dac topology itself.

The main study (linked here) on jitter audibility was done back in 1998. But was only done with a small sample size of 9 subjects and iirc no info on the subjects themselves in terms of background, listening ability or experience etc.
Additionally even if the study was more thorough and conclusive then, the answer might need updating or be influenced by other aspects like the DS modulator. Improved modulators since 1998 (theyve come a long way) may allow for lower thresholds of audibility for jitter due to reduced quantization noise.

But the TLDR is we can't say with certainty a device is 'audibly transparent' when we don't know the actual thresholds of audibility for so many things.




bigshot said:


> DACs or players are not audibly transparent


As said above. I do not think audibly transparent is something we can currently determine. 

But another answer would be that I rarely find dacs that sound the same. They're often similar, but so rarely identical


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## bigshot (Dec 13, 2022)

The more I try to define my question for you, the more you dodge it.

There is such a thing as "it doesn't matter". Whenever someone has told me that they can hear differences it goes from "it's night and day, even my wife can hear it!" to "we can't know everything, so we can't know anything" as I try to get a straight answer out of them about exactly what they are hearing and under what circumstances.

All I want to know is the make and model of a DAC or player that will sound different in normal use. I can rack it up next to my Oppo or my iPod Classic, level match and see if it sounds different with a simple blind test playing some of my best sounding CDs. Which make and model should I do this with? Make it easy on me. Pick one that sounds quite different, just not the pathological examples we've already discussed.


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## GoldenOne

bigshot said:


> All I want to know is the make and model of a DAC or player that will sound different in normal use


As said above. I find that dacs rarely sound the same/usually sound different.

So I guess you could say just about any dac I've tried


----------



## The Jester

GoldenOne said:


> As said above. I find that dacs rarely sound the same/usually sound different.
> 
> So I guess you could say just about any dac I've tried


For someone who’s compared more than a few DAC’s, do you hear a noticeable difference between CD audio and say 24/96,
“noticeable difference” and “improvement” being hard to quantify, but if you do, does the “difference” increase, decrease or remain fairly constant as you go up the “technology/quality/price” ladder ?


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## bigshot (Dec 14, 2022)

GoldenOne said:


> So I guess you could say just about any dac I've tried


Even from copy to copy of the same make and model?

Which DACs or players have you found that sound the same?

Are they all different by the exact same amount, or are some more different than others?

Are there any videos on your YouTube channel of you conducting a listening test like this (level matched, blind, a/b switched, music)?


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## 71 dB (Dec 14, 2022)

GoldenOne said:


> The audibility of reconstruction filters is very much up for debate and there is a surprising lack of conclusive study into the area.


If the audibility of reconstruction filters was a clear thing, there wouldn't be a debate and if it really mattered to the consumers, one would think there would be more studies conducted on it.

The problem I have with people saying there is a "debate" like this is it can make music consumers doubt things they shouldn't be doubting. Consumers should be enjoying music. That's it. It is that simple. If a consumer starts needlessly worry about if his/her DAC's reconstruction filter sounds audibly bad, placebo effect can easily make it sound bad doing harm to the "enjoying music" aspect.

Also, audibility of reconstruction filters is different from audio formats, especially when we compare 16 bit to 24 bit.



GoldenOne said:


> Though also worth noting that DAC reconstruction filters vary greatly and oversampling alone doesn't mean a filter is going to be similar. One could oversample with a zero order hold approach to get a result that behaves the same as NOS for example.


This is more of a DAC _implementation_ discussion. ZOH is typically used when oversampling film or video frames to make it less computationally taxing. Oversampling audio isn't computationally that hard anymore and windowed sinc interpolation is used.



GoldenOne said:


> Also many DACs reconstruction filters do not adhere to nyquist, and for the ones that do there is the argument of audibility of lost information due to early rolloff.


Audibility is one thing. Does it matter is another. Does for example 3 dB rolloff at 20 kHz matter from the point of music enjoyment for someone older who can't hear anything above 17 kHz?



GoldenOne said:


> Some of them have treble rolloff that would absolutely be audible.


Yeah, completely ruins music for me! I need to throw my 2000 CDs away and buy everything again at 192 kHz files (I wonder how much of my music is available even at 96 kHz...) In audio the things that really matter are room acoustics, speakers, speaker/listener placement and headphones on the consumer end and production, mixing and mastering on the other end. Amp power and some impedances have also some role, but beyond those, things just don't matter almost at all.


----------



## Ghoostknight

71 dB said:


> If the audibility of reconstruction filters was a clear thing, there wouldn't be a debate and if it really mattered to the consumers, one would think there would be more studied conducted on it.


i think most of this stuff "on debate" is that its kinda a chicken/egg problem in regards of studys and probably the wrong people get tested

i can switch between reconstruction filters on my Aune X8 and there is a small difference, very very small actually but its there, tho i think it has more todo with phase/impulseresponse than frequencys 



71 dB said:


> Audibility is one thing. Does it matter is another. Does for example 3 dB rolloff at 20 kHz matter from the point of music enjoyment for someone older who can't hear anything above 17 kHz?


probably not but you are not instantly "def" on higher frequencys either, its more of a roll off, a roll off in the dac just makes it worse and is bad practice if we speak objectively



71 dB said:


> The problem I have with people saying there is a "debate" like this is it can make music consumers doubt things they shouldn't be doubting. Consumers should be enjoying music. That's it. It is that simple. If a consumer starts needlessly worry about if his/her DAC's reconstruction filter sounds audibly bad, placebo effect can easily make it sound bad doing harm to the "enjoying music" aspect.


we can also keep denying everything and there will be no advancements anymore

be happy with what you have, then it also doesnt kill your enjoyment, it kinda sounds depressing to have this kind of mindset

--

i think a good analogy is that fps advancements over 24fps shouldnt matter since for the brain its fluid already, but many report they even see a difference between 120hz and 200<hz, why is that? low fps "stress" the brain more since it has todo more work to reconstruct a fluid video from low fps (atleast thats my thinking), i think its the same for audio (tho in other points than bitrate/samplerate), how is it that many report no audible fatique with high end systems anymore?

i think its pretty much the same with the usual stuff from objectivists " its not audible " but many report differences, strange coincidence imho

one example i have done myself is checking which lowest volume difference you can detect, (audiocheck.net) there is a common word around 1db (because studys....) but after some practice i was able to detect 0,5db (9/10 result if i remember right) and im even able to hear the difference in a 0,1/0,2db EQ change, its not like i hear those differences directly, but the tonality overall changes

also im able to detect absolute phase while people keep claiming it doesnt make a difference

imo many things matter where no research has been done (or invalid research/studys) about but for many people it still matters, yes there is true snakeoil out there but also things that is snake oil under objectivists but it still makes a difference, you have to hear difference to believe them

and please dont start with placebo... yes the mind can fool you
i actually think a blind test can fool you just as much with the right/wrong mindset
beside that i often detect changes way more obvious if i listened to the same system a few weeks and then change something, its way more obvious than fast switching back and forth


----------



## Elegiac

I hear a difference between DAC's, and especially DAP's. 
How can two devices made up of totally different parts, and different tunings/implementations not sound different?


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## bigshot (Dec 14, 2022)

There are filters that change the sound. Don’t use those.

71 dB is right. Tests aren’t being done on different sounding DACs for the same reason it isn’t done on other audiophile hoodoo… it’s already established.


----------



## bigshot

Elegiac, have you compared DACs with level matched, direct A/B switched, blind listening tests?

Most electronics are made from off the shelf parts made in bulk overseas. You’ll find the same parts being used in different units.


----------



## bigshot

Saying a blind test can fool you as much as bias is like saying car insurance can make you get in car accidents. When you start arguing points like that, you’re just revealing how strong your own validation bias is.


----------



## Ghoostknight

bigshot said:


> Saying a blind test can fool you as much as bias is like saying car insurance can make you get in car accidents. When you start arguing points like that, you’re just revealing how strong your own validation bias is.


if we go down the whole mindset and bias route it probably can!


----------



## Elegiac

bigshot said:


> Elegiac, have you compared DACs with level matched, direct A/B switched, blind listening tests?
> 
> Most electronics are made from off the shelf parts made in bulk overseas. You’ll find the same parts being used in different units.


And then there's source impedance of the device ect ect... perhaps, probably, with every extraneous variable removed it'll all sound basically the same, as it should do, but all these small differences make different flavours of sound. And I like different flavours. I wouldn't want to dismay myself with level matching and such, hahaha. 

Two different DAC's using two different op amps, each with a sound different to the other, that the maker intended, _should_ sound different. The overall sound signatures of my various bits of equipment all sound quite different, all of them using different DAC's and op amps. Actually, it is quite difficult for me to believe that volume matching would make, say, a difference in soundstage width disappear. I have two DAP's with different DAC's but the same op amp, and I can tell which is which through the same headphone. I put that down to implementation.


----------



## GoldenOne

The Jester said:


> For someone who’s compared more than a few DAC’s, do you hear a noticeable difference between CD audio and say 24/96,


Yes, though it's dependent on the recording.


The Jester said:


> “noticeable difference” and “improvement” being hard to quantify, but if you do, does the “difference” increase, decrease or remain fairly constant as you go up the “technology/quality/price” ladder ?


Not necessarily.
The best sounding stuff I've come across has tended to be quite expensive, but there's PLENTY of expensive products that sound no better than far cheaper ones to me.
IMO if you want the best, it will probably cost a fair bit, but price is FAR from any guarantee of quality.

Plus there's been so many truly excellent products coming out recently that are very affordable.
Singxer SA-1 as a headphone amp for example objectively and subjectively puts many more expensive amps to shame IMO.


bigshot said:


> Even from copy to copy of the same make and model?


No I've not heard two units of the same make/model sound different. I suppose maybe they could if there was something that might cause massive unit variance (Poor tolerance in an R2R ladder for example), but that would be easily measurable too.


bigshot said:


> Which DACs or players have you found that sound the same?


Typically stuff that follows very similar designs. There's a lot of more affordable DACs that are using the same ESS chip, output stage opamps/design, clocks and meanwell PSUs. Different manufacturers, different products, but so close in design and for me not possible to audibly discern between.

As you start going to the products that are using more varied designs even if using the same DAC, say a Gustard X26 Pro vs a Topping D90SE, or at the higher end typically, fully proprietary designs that both perform well but operate fundamentally differently like a Chord DAC vs a dCS DAC, then differences are more pronounced.


bigshot said:


> Are they all different by the exact same amount, or are some more different than others?


Some more different than others.


bigshot said:


> Are there any videos on your YouTube channel of you conducting a listening test like this (level matched, blind, a/b switched, music)?


No, this is something I've debated and am still unsure as to whether it's a good idea to do.
For some context, a while ago Amir at ASR challenged me to do a blind test between two devices generally considered 'audibly transparent' and offered $1000 to charity if I could pass.
I accepted, and accepted various controls and test limitations, as I thought it would be an interesting thing to do and for a good cause, offered to let him send someone in person to observe the testing etc, but eventually the paramaters kept getting changed until he didn't want me listening to the two devices at all but instead a recording of each device...quite different.
That thread got quite hostile and it's best to leave it at that, but afterwards, a friend and I set about making a device to allow for unassisted blind ABX testing.

This is the current iteration of it:





Explanation of how it works and avoids tells here: https://streamable.com/tuuyzg

But this has been incredibly useful for me personally when testing stuff.
I've found that in a level matched blind test I cannot for example discern between a Gustard X26 Pro and Gustard X18, but can discern between an X26 Pro and a Chord DAVE, or a Chord DAVE and a dCS Bartok, or dCS Bartok and Holo May etc.
Part of me wants to post some videos of blind tests, but the reason I've not done so is simply because it will do nothing but cause arguments.

Anyone who believes that there are audible differences in DACs will at best simply say 'well duh'. And at worst will use my video as ammunition during online arguments and probably rope me into fights I do not want to be part of. Constructive debate is great, and the conversation we're having here is good. But unfortunately the majority of 'discussions' in this topic quickly devolve to bickering.

And anyone who is adamant that there are no differences will not be convinced anyway, and will conclude that I've cheated, did the test wrong, or am in some way misleading things. It won't convince them no matter how conclusive it is, and so nothing of use will have been achieved.

I think a better way will be to organise for some blind testing with a group overseen by a number of people, and this is something I am currently working to do. That way there shouldn't be any issue about the limited number of test subjects, and there will be some very reputable people who can verify that the test was done fairly to dispel any accusations of cheating.



71 dB said:


> If the audibility of reconstruction filters was a clear thing, there wouldn't be a debate and if it really mattered to the consumers, one would think there would be more studies conducted on it.


Yes, one would think there would be more studies conducted into it, but unfortunately there aren't and so at the moment it is indeed up for debate.
Though as previously mentioned, there is evidence that higher sample rate content and/or higher performance reconstruction filters are indeed audible.

https://www.aes.org/e-lib/browse.cfm?elib=18296


71 dB said:


> Consumers should be enjoying music. That's it. It is that simple. If a consumer starts needlessly worry about if his/her DAC's reconstruction filter sounds audibly bad, placebo effect can easily make it sound bad doing harm to the "enjoying music" aspect.


I agree that enjoyment of music is the most important thing, and tbh this is why there likely isn't much study into this and various other areas. There simply isn't the need or financial incentive for anyone to do it when differences if proven to exist are quite subtle and 99.99% of consumers couldn't care less about it. 
Doesn't mean that we should have 0 discussion about it or just go with the assumption that they make no difference when the evidence doesn't support that.


71 dB said:


> Also, audibility of reconstruction filters is different from audio formats, especially when we compare 16 bit to 24 bit.


Sort of. Audibility of reconstruction filters isn't really linked to bit depth, but is linked with native sample rate.


71 dB said:


> This is more of a DAC _implementation_ discussion. ZOH is typically used when oversampling film or video frames to make it less computationally taxing. Oversampling audio isn't computationally that hard anymore and windowed sinc interpolation is used.


EXACTLY
The point is that the implementation discussion applies to basically all DACs.
There is no possibility to create a perfect reconstruction filter and therefore the oversampling implementation and its merits can be discussed for any product, because there will inherently be limitations or tradeoffs.


71 dB said:


> Audibility is one thing. Does it matter is another. Does for example 3 dB rolloff at 20 kHz matter from the point of music enjoyment for someone older who can't hear anything above 17 kHz?


Whether something is audible enough to 'matter' is entirely up to the listener.
Most consumers don't care about audio quality much at all, some are obsessive about it. But so long as an audible difference is present it's valid to discuss it no?

Additionally, when discussing filters and treble rolloff, it's not so much the frequency domain rolloff itself (or aliasing influenced by it), but how this lack of high frequency information or unwanted aliased content impacts the timing of transients.

There's some quite interesting research shows that whilst the human frequency domain accuracy is indeed for most people up to about 20khz (though some can hear a bit above that, particularly when much younger), the temporal resolution is much higher. We might not be able to hear sounds above 20khz, but we can discern time domain differences that would require frequency domain components above 20khz in order to accurately reproduce.

https://www.tnt-audio.com/casse/temporal_resolution.pdf
Because of the above, as well as the fact that other research has shown that people are able to discern between 44.1khz and 96khz content, the current evidence suggests that yes, high sample rate content and/or higher performance reconstruction filters do indeed have an audible effect. Whether that 'matters' is upto the listener. 



71 dB said:


> In audio the things that really matter are room acoustics, speakers, speaker/listener placement and headphones on the consumer end and production, mixing and mastering on the other end.


I'm not at all disagreeing that these things are more important, they certainly are.
But again, dismissing something as pointless or harmful to even discuss simply because it's low on the ladder of how much things matter seems silly to me.
Why shouldn't we be able to discuss anything if it can make a demonstrable improvement?


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## Ryokan

GoldenOne said:


> I think a better way will be to organise for some blind testing with a group overseen by a number of people, and this is something I am currently working to do. That way there shouldn't be any issue about the limited number of test subjects, and there will be some very reputable people who can verify that the test was done fairly to dispel any accusations of cheating.



Maybe ask someone not much into music to be a tester?


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## danadam (Dec 14, 2022)

GoldenOne said:


> Doesn't mean that we should have 0 discussion about it


Sure, we can have plenty of discussion about how insignificant it is  



GoldenOne said:


> We might not be able to hear sounds above 20khz, but we can discern time domain differences that would require frequency domain components above 20khz in order to accurately reproduce.
> 
> https://www.tnt-audio.com/casse/temporal_resolution.pdf
> Because of the above,


Sigh... Kunchur again. From the end of this post:


> The 0.7 dB threshold is assumed without checking. Kunchur says in the FAQ that he can't redo everything and that evaluating again the Just Noticeable Difference would take about two years.
> 
> However, since this point annoyed me, I ran an ABX test between two 7 kHz sines. I chose the level difference that Kunchur got for the 5.6 µs lowpass experiment : 0.25 dB.
> 
> ...


And if this is the type of papers they used in this meta-analysis, then, well, no comment


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## castleofargh

The biggest problem with the meta-analysis is how small the impact found is. The entire conclusion could change just by replacing a paper or 2 in the selection. And that's troubling because not everybody was happy with the selection process of those papers.


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## bigshot (Dec 14, 2022)

GoldenOne said:


> Whether something is audible enough to 'matter' is entirely up to the listener.


That is true if you are a theoretical scientist. They don't have any responsibility to an end user. They are just recording facts. However if you are offering advice to other people as a reviewer of consumer products, your responsibility is different. If you are saying that a difference exists, but you're completely unable to quantify that difference... to the point of just saying "everything sounds different and none of it sounds more different than any other", you are making a statement that is completely useless to a person who is listening to a review of a piece of equipment they are considering the purchase of. They aren't interested in theoretical measured differences. They want to know which one sounds *better* with the highest *perceivable* fidelity. If everything sounds different to the same degree, then you aren't recommending anything. You're just vomiting out contextless facts.

This is my complaint about audiophiles. They point at numbers on a page without the slightest clue what those numbers mean in actual sound we can hear. And then they assume that because the number shows a difference, there must be an audible difference. You admit that you don't know the JDD thresholds, and I believe you. I don't think you've ever researched human hearing as carefully as you research measurements. I can read between the lines and know that you've never taken a good sounding CD and racked it up through two DACs, level matched and blind switched between them, regardless of your claim to the contrary. I'm not saying you're lying. You may have done this with test tones. You may be basing your belief that they sound different on measurements, and you may be absolutely correct that there is a difference. I don't deny that. But the only way to determine whether that difference *MATTERS* is to do a real-world comparison that can *QUANTIFY* the degree of difference. Is it a huge difference that would affect your enjoyment of Beethoven while you sit on the couch and listen? Or is it a minute difference that you would have to create non-real-world situations and strain very hard to hear? A consumer reading a review wants to know what that extra $500 is buying him. Does it matter? Not answering that question and not quantifying the differences is ducking the most important job as a reviewer.

Audiophile equipment salesmen want consumers to believe that any difference is an important difference. They argue that super audible frequencies and noise floors far below the noise floor in your living room are life and death matters. They argue that jitter is an important consideration when every single example of measured jitter in every single digital consumer audio product is more than an order of magnitude below the threshold of audibility. No one even knows what jitter sounds like because it doesn't exist in audible amounts! The reason there isn't a lot of research into many of the bugaboos that audiophiles fret about is because sound engineers know they don't matter. They have bigger fish to fry than to chase down fractions of a fraction of a fraction of a percent.

Audiophilia is a progressive disease that attacks common sense and logic. It sets a line for "good enough", then immediately tries to argue that good enough isn't good enough. It doubles down with minute differences in unlikely situations over and over again. It searches for solutions to problems that don't exist. It gives weight to distortions where there's no evidence they can even be heard. It judges the sound quality of equipment by numbers on a page, not the sound reaching your ears. I happen to agree with the crackpots who invade this forum and say "trust your ears". Our ears are a vital part of the chain and their limitations are a bottleneck for all the sound being produced upstream. You need to consider that to make any kind of practical decisions about sound quality. Thresholds matter. If you throw your hands in the air and say "thresholds haven't been established, so we had better consider everything we can measure as being important", then you're abrogating the value of controlled listening tests. Establish your own thresholds then and do it with real-world controlled listening tests. What we can and can't hear matters... a LOT.

All of this is simply solved by just doing a blind, level matched, direct A/B switched listening test USING REAL WORLD CONTEXT. People in the audio industry are generally OK with most of that sentence, but the part in all caps escapes them. They focus on exceptions that don't exist in the real world and numbers that represent sound without any clue about how that relates to sound our ears can hear. The sound our ear hears matters. The sound that is produced using our own music in our own living room matters. Every listening test doesn't need to be to the rigid standards of peer reviewed journals. Controlled listening tests are a valuable tool for casual hifi nuts too. There are two kinds of sound science- theoretical ivory tower sound science that exists for the purposes of theoretical talk and technical understanding, and real-world sound science that helps consumers optimize their stereo systems to achieve better sound in the music they listen to for enjoyment in their home. A scientist might be interested in the former, but someone in an audio forum is 100% going to be interested in the latter. That's what people look for in a review of consumer audio equipment.


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## Ryokan

When I was getting into high end audio (for me) a salesman set up a demonstration and after a few minutes I turned to him and said 'sounds good doesn't it?', to my mild annoyance he gave a gesture like a shrug. I realised then I was listening to expensive gear and I was on my own, he sold 'stuff' to people which they thought was good and that was what mattered. At least he didn't try to steer me towards anything and let me know it was my choice.


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## old tech (Dec 14, 2022)

GoldenOne said:


> On the topic of this very thread for example:
> https://www.aes.org/e-lib/browse.cfm?elib=18296
> 
> "400 participants in more than 12,500 trials. Results showed a small but statistically significant ability of test subjects to discriminate high resolution content, and this effect increased dramatically when test subjects received extensive training"


Geez, people still quoting this meta-analysis when it has been thoroughly discredited? It was Stuart's attempt to create an environment for his MQA marketing by selectively choosing the studies in the meta, some of them with very poor controls while omitting the ones that stood the test of time (such as below). Even after all that cherry picking, the best the paper comes up with is a minuscule difference of listener's ability to distinguish between high res and MP3.The paper never got past peer review and hence there is a reason why the AES does not charge access to read it. Have a read of a properly controlled study such as the one below.

http://drewdaniels.com/audible.pdf

And for your claim that you can hear a difference between 16bits and 24, have a listen of this video which compares 8bits and 16 and then reflect whether it is all in your head or indeed you are listening to different masters.


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## GoldenOne (Dec 14, 2022)

old tech said:


> And for your claim that you can hear a difference between 16bits and 24


I never said this.
My opinion in relation to audibility with hi-res content is to do with the sample rate not bit depth. I don't think 24 bits vs 16 bits makes an audible difference.


old tech said:


> Geez, people still quoting this meta-analysis when it has been thoroughly discredited? It was Stuart's attempt to create an environment for his MQA marketing


What does this have to do with Bob Stuart? Genuine question, I'm no fan of Bob Stuart or MQA as may be obvious from some of my previous videos


But it's not clear to me how he relates to this?


old tech said:


> Have a read of a properly controlled study such as the one below.
> 
> http://drewdaniels.com/audible.pdf


This is a good paper but does not actually mention what the specifics of the 'High Res' content being played was.
SACD is usually DSD64 is it not? It'd be more relevant to the discussion if higher sample rate PCM content was used. (Though additionally when doing so, it's important to detail the recordings themselves, there are a lot of recordings available in >44.1khz that were actually recorded/produced at 44.1khz)

An easier way would be to simply take two 44.1khz files, then apply a nyquist reconstruction filter using a tool like HQPlayer. Once with a very high performance filter, and once with a more basic filter akin to what many DACs use.
Then use a tool like Foobar ABX to conduct an audibility test between them. If they are audibly discernable, this shows that reconstruction filters and high sample rate content can indeed make an audible difference.

Here's one I did just now, will do another later with some more runs and have a go at getting a lower P-Value, but I think P=0.019 is sufficient for now and as a first attempt:





(Text copy attached to post for easier access to checksums)

File A is upsampled to 705.6khz using the 2 million tap 'Sinc-L' filter, and a 15th order noise shaper
File B is upsampled to 705.6khz using a more typical linear phase sinc filter, and standard TDPF dither

Both files have -3.3dB attenuation to prevent intersample clipping.

*Source files (verifiable via checksums above) available for inspection here*: https://mega.nz/file/e7pEhRyJ#QH576dKnZeTsUAGh7jbWv7E252tZmiH4hj3op3C1R1s

*My Setup used for the test:*
Holo Audio May (Connected directly to PC, being fed 705.6khz PCM in both instances from Foobar via ASIO)
Zähl HM1
Hifiman Susvara

Here are the filter responses of 44.1khz white noise fed through with the same configs as files A/B above

*File/Filter A:*





*File/Filter B:*



I chose this filter as it's fairly similar to what you see in many consumer or professional DACs regarded as 'audibly transparent' such as an SMSL DO100 for example.





(Note this graph adjusts offsets Y axis so that the audible band whitenoise spectrum is at 0dB, the two graphs above do not, so make sure to take that into account when looking at a particular level of attenuation)


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## GoldenOne

If anyone would like to try the above themselves, this is the plugin I use: https://www.foobar2000.org/components/view/foo_abx


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## bigshot

Both of those white noise graphs look like they are audibly transparent. What am I supposed to be seeing?


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## GoldenOne (Dec 14, 2022)

bigshot said:


> Both of those white noise graphs look like they are audibly transparent. What am I supposed to be seeing?


See the part above where a blind test was passed with a statistically significant result, evidencing that they are in fact not audibly transparent


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## bigshot (Dec 14, 2022)

That shows 20-20 basically flat. That is transparent. Most people can't hear above 15kHz, much less up above 20kHz. Everyone tested could hear the difference? Listening to white noise? That makes no sense whatsoever. Something's wrong with that listening test there, or there's something wrong with that measurement.

Can you tell the difference between white noise with those two filters" How about with music? If there's a lot of super audible content in the white noise sample, I can see your ears ringing or getting a headache, but super audible content at high volumes wouldn't be a good thing. It certainly wouldn't sound better, it would sound worse.

I think you (or whoever you got this from) are interpreting these measurements incorrectly. 20-20 clean and flat is audibly transparent by definition. (Not counting bizarre things like bone conduction or ultra high frequencies at high volumes creating nausea.)


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## GoldenOne

bigshot said:


> That shows 20-20 basically flat. That is transparent. Most people can't hear above 15kHz, much less up above 20kHz. Everyone tested could hear the difference? Listening to white noise? That makes no sense whatsoever. Something wrong with that interpretation there.


Please read the post thoroughly.

Source files are also provided


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## Ghoostknight

bigshot said:


> Both of those white noise graphs look like they are audibly transparent. What am I supposed to be seeing?


well the blind test kinda shows that its in fact not "audibly transparent", just the frequency response is (till 20khz)

tho i think @GoldenOne  your efforts are kinda pointless for argueing with people that didnt heared the differences themself since you cant convince them anyway
but its great to see people take the abx challenge, even if i think abx is somewhat flawed too but i guess its the best approach if you wanna go down the scientific road to show your results


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## bigshot (Dec 14, 2022)

Most SACDs are 24/96 because you can't edit or process DSD natively. It's very difficult to find a native DSD SACD. I did an A/B blind test with one I found and failed dismally to tell any difference between the SACD and CD layer. That was using an SACD deck and a CD player of different brands too.

Save me a little reading please... in the test you are citing, was a statistical majority of people able to discern the difference between those two filters? This is something the average person can discern?

Here is Ken Rockwell's measurement of Apple's $8 dongle. This is the sort of cutoff I am used to seeing.


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## GoldenOne (Dec 14, 2022)

bigshot said:


> Save me a little reading please


In order to discuss the topic it really would be ideal if you read the post.

Hard to have a discussion when one side is unwilling to even look at points or evidence put forward.

Music was used, played through the same chain at the same level. The only difference being the reconstruction filter applied. One version being a normal linear phase sinc filter very close to what is found in various DACs, the other being a very high performance 2 million tap linear phase sinc filter.
Both are flat to 20khz

In a blind ABX test the two were audibly discernable with a P-value of 0.019 in that instance. Therefore showing that reconstruction filters or high sample rate content are audibly discernable using regular music and standard sinc filters found in most DACs cannot be conclusively considered 'audibly transparent'


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## bigshot

What kind of music? Did they cite the recording used?


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## GoldenOne

bigshot said:


> What kind of music? Did they cite the recording used?


PLEASE read the post


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## bigshot (Dec 14, 2022)

You'll have to forgive me, I'm not an engineer nor a scientist, and I know nothing about statistics. A p value of .019 means that the test subjects got it right in 90% of the trials... is that correct? How many test subjects and how many trials were in this test? Once I figure out the statistics, I'll puzzle out the testing method. I appreciate your help with sorting it out. Scientists are not good communicators. I require translation.

I only see a download, I don't see any description of the music used. I also see multiple studies being mentioned without clearly identifying which is which. Am I looking at two separate things here? I would like to be stepped through it so I can wrap my head around it in an orderly way.


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## bigshot (Dec 14, 2022)

Let me see if I can organize this myself...

If someone is willing to help me understand, my first questions are to define the basic outlines of the test and tofind out the odds of being able to hear the difference between the two samples.

If I understand correctly, the test subjects were given the same bit of music using two different filters, which both measure 20-20 flat. The only differences between the response of the filters were frequencies above 20kHz, and the volume level of those super audible frequencies were at natural levels for recorded music.

Is this correct?

How many people were tested, and how many tests did they conduct with each person?
What percentage of all the tests conducted were correct in identifying which sample was which?

If I get the answer to those two questions, I'll be able to move on to puzzling out the next bit.


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## GoldenOne (Dec 14, 2022)

bigshot said:


> A p value of .019 means that the test subjects got it right in 90% of the trials... is that correct


A P Value is a number that represents the likelihood that the result was obtained by chance.

So in this instance 0.019 means that the likelihood this result is the result of a genuine difference and not chance/luck is 98.1%

I'll be doing further testing with many more runs to obtain a lower P-Value over the next few days. This was just a quick test to contribute to the discussion at hand with a result that is fairly noteworthy/conclusive.


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## bigshot

You don't need to go out of your way to do that. I'd just like to get help understanding the study you're pointing to.


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## GoldenOne

bigshot said:


> You don't need to go out of your way to do that. I'd just like to get help understanding the study you're pointing to.


This was my own testing done today in response to the previous discussion.

The music used is linked in the post


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## bigshot

Ah OK. It was hard to separate the study and your personal test because they were being discussed simultaneously. Your download link is to the music they used in the study, right? You were taking their test yourself if I understand correctly. If so, then my understanding of the parameters of the test is correct.

I have the files, and I'll listen to them when I get to that. If you could please answer my questions on the statistics that the study found first, that would be helpful. Thanks.


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## GoldenOne

bigshot said:


> Ah OK. It was hard to separate the study and your personal test because they were being discussed simultaneously. Your download link is to the music they used in the study, right? You were taking their test yourself if I understand correctly. If so, then my understanding of the parameters of the test is correct.
> 
> I have the files, and I'll listen to them when I get to that. If you could please answer my questions on the statistics that the study found first, that would be helpful. Thanks.


No the study linked previously was different.

That one I thought was interesting but also lacked quite a bit of info about the music used and didn't even specify the actual music format being used so didn't really provide much in the way of a conclusive answer.

I set up my own test, using a song I've been listening to recently since I'm familiar with it. I took the 44.1khz version and created two 705.6khz versions. One with a basic sinc filter extremely similar to what many dacs might use. And another that was a 2 million tap high performance filter.

If an audible difference was shown, then that would demonstrate that reconstruction filters and hires content do indeed make audible differences

I have provided the original files and the ABX log contains checksums of the files used so that anyone can verify the results are legitimate


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## bigshot

Ah, I'm interested in the published study. I tried to read it and it was a bit too specialized for me. So we don't know what music was used... OK. That is fine. Could you help me understand it?

What were they comparing? 16/44.1 to 24/96? Or the different filters like your samples?
If so, how many test subjects, how many trials and how many correct answers about which was which?

Thanks


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## GoldenOne

bigshot said:


> Ah, I'm interested in the published study. I tried to read it and it was a bit too specialized for me. So we don't know what music was used... OK. That is fine. Could you help me understand it?
> 
> What were they comparing? 16/44.1 to 24/96? Or the different filters like your samples?
> If so, how many test subjects, how many trials and how many correct answers about which was which?
> ...


We don't know. They don't specify what the comparison content sample rate was. Hence my issue with it


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## bigshot

OK. I thought it was being put forward as proof that super audible frequencies were audible. Thanks for clarifying it for me.


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## 71 dB

GoldenOne said:


> I never said this.
> My opinion in relation to audibility with hi-res content is to do with the sample rate not bit depth. I don't think 24 bits vs 16 bits makes an audible difference.


Not long ago I suggested that 58.8 kHz / 12 bit with heavily shaped dither would have the same bitrate as 44.1 kHz / 16 bit, enough dynamic range for consumers and much easier anti-alias/reconstruction filter requirements.


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## 71 dB

bigshot said:


> That shows 20-20 basically flat.


Actually not. The attenuation at 20 kHz varies 0-12 dB. However, the noise spectrums are "flat" up to 18 kHz, which should be good enough for most ears.


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## Ghoostknight (Dec 15, 2022)

71 dB said:


> Actually not. The attenuation at 20 kHz varies 0-12 dB. However, the noise spectrums are "flat" up to 18 kHz, which should be good enough for most ears.


One thing i didnt quite understand was that applying a lowpass filter which starts for example at 18-21khz (i hear flat to 15-16khz) quite changes the "sensation" of music listening (frequencys above 15khz seem to add mostly "air/sparkling" to the music), did someone had a similar expierence/can test it/or has a explanation?

a -3db point at 20 to 22khz is quite obvious in my expierence, not sure if i could tell in abx/different dacs but its kinda obvious if you just add a lowpass to the chain (camilladsp in my case)

my chain is quite simple -> Rpi4 -> Usb -> Aune X8 XVIII -> XLR -> Presonus Eris E8


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## danadam

GoldenOne said:


> Here's one I did just now, will do another later with some more runs and have a go at getting a lower P-Value, but I think P=0.019 is sufficient for now and as a first attempt:


In order to get such result, is it required for the content in the original file to go right up to the 22'050 Hz or can you do that also when it has some headroom there?


Zoom at >20'000:

Spectrogram:


----------



## 71 dB

Ghoostknight said:


> One thing i didnt quite understand was that applying a lowpass filter which starts for example at 18-21khz (i hear flat to 15-16khz) quite changes the "sensation" of music listening (frequencys above 15khz seem to add mostly "air/sparkling" to the music), did someone had a similar expierence/can test it/or has a explanation?
> 
> a -3db point at 20 to 22khz is quite obvious in my expierence, not sure if i could tell in abx/different dacs but its kinda obvious if you just add a lowpass to the chain (camilladsp in my case)
> 
> my chain is quite simple -> Rpi4 -> Usb -> Aune X8 XVIII -> XLR -> Presonus Eris E8


What kind of lowpass filter at 18-21kHz? One thing to keep in mind is that digital filters having cut off frequency very near Nyquist can have a funny response depending on how they have been designed.


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## VNandor

castleofargh said:


> The biggest problem with the meta-analysis is how small the impact found is. The entire conclusion could change just by replacing a paper or 2 in the selection. And that's troubling because not everybody was happy with the selection process of those papers.


I mean, the papers' title is "A Meta-Analysis of High Resolution Audio Perceptual Evaluation" and then for some reason papers from 1980 and 81 got included in the analysis. That's before even the standard digital audio format was a thing. The infamous Meyer and Moran study is also there. It certainly makes me go  but since this is from AES, it must have been peer reviewed by people who are much smarter than a bunch of head-fi posters so there's that.


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## Ghoostknight

71 dB said:


> What kind of lowpass filter at 18-21kHz? One thing to keep in mind is that digital filters having cut off frequency very near Nyquist can have a funny response depending on how they have been designed.


12dB/oct, thats the only thing i see on the camilladsp page, i also tried lowshelf filters and the effect was similar (but not as pronounced)


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## bigshot (Dec 15, 2022)

Generally, at 18kHz and above, the level in commercial music is so low, it would probably be masked anyway. I suppose with a synth you could create something with audible 19kHz, but even if you were 9 years old and you could hear it, it wouldn’t sound like much.

I would call both of those graphs transparent for 95% of the public. Audiophiles worry about all the wrong frequencies.


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## 71 dB

44.1/16 is transparent in the sense that the rest of the audio chain is much less transparent so that 44.1/16 can't bottleneck transparency. Even if some individuals with superhuman hearing can tell 44.1/16 apart from hi-rez, it is meaningless, because hi-rez doesn't quarantee transpancy either. Transducers are far from transparent and without transducers there is no sound. When 44.1/16 is compared to hi-rez in listening tests, the rest of the audio chain remains the same, but it is not transparent! Even if 44.1/16 wasn't 100 % transparent (to golden ears), it makes the audio chain just a little bit less transparent and without revolutionary advance in transducers that is meaningless. That's why we can say 44.1/16 is transparent no matter what people say.


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## bigshot

Frequencies at the top end of the spectrum and beyond may be perceived in some sense of being felt somehow, but they are not heard, and they add nothing to the perceived sound quality of music. You could do a gradual roll off starting at 10kHz and although you would likely be able to barely hear a difference in a direct comparison, it wouldn’t sound any worse. It would just sound different.

10-20 kHz is one octave, and 18-20 is just one whole note on the musical scale. It doesn’t add up to a hill of beans. Stuff like that just doesn’t matter.


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## 71 dB

Ghoostknight said:


> 12dB/oct, thats the only thing i see on the camilladsp page, i also tried lowshelf filters and the effect was similar (but not as pronounced)


12dB/oct butterworth? That's a very gentle filter in this context and will affect frequencies far below Nyquist. For example if you have one at 20 kHz (-3 dB), the lower frequencies are affected like this (rounded to 0.1 dB accuracy):

19 kHz........ -2.6 dB
18 kHz........ -2.2 dB
17 kHz........ -1.8 dB
16 kHz........ -1.5 dB
15 kHz........ -1.2 dB
14 kHz........ -0.9 dB
13 kHz........ -0.7 dB
12 kHz........ -0.5 dB
11 kHz........ -0.4 dB
10 kHz........ -0.3 dB

So, even below 15 kHz such filter makes a clear difference.


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## Ghoostknight

71 dB said:


> 12dB/oct butterworth?


hmm im not sure, but you can set the Q which is probably the thing you mean, right? i think i was testing with 0,7071Q which ends up at around -0,08db at around 16khz (20khz, 0,7071Q, 12db/oct) (atleast this is what the camilladsp graph shows and i also tried 22khz because of the "roll in", but this test is also a few weeks back)

hmm i guess it could be possible that i was hearing either this or the groupdelay i guess, tho its still somewhat unclear what i was hearing in the end but i was not ending up using the filter, it was just a "silly" test


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## GoldenOne

danadam said:


> In order to get such result, is it required for the content in the original file to go right up to the 22'050 Hz or can you do that also when it has some headroom there?
> 
> Zoom at >20'000:
> 
> Spectrogram:


Can also do it when the file does not go up that high.

I cannot hear above 20Khz when testing that standalone, so at least in my case the most likely answer as to why reconstruction filters have an audible difference even if one cannot hear above 20khz is to do with the fact that higher sample rate native content OR higher performance reconstruction of 44.1khz content would allow for more accurate reconstruction of transient timing.

Time domain and frequency domain are both affected.


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## 71 dB

Ghoostknight said:


> hmm im not sure, but you can set the Q which is probably the thing you mean, right? i think i was testing with 0,7071Q which ends up at around -0,08db at around 16khz (20khz, 0,7071Q, 12db/oct) (atleast this is what the camilladsp graph shows and i also tried 22khz because of the "roll in", but this test is also a few weeks back)
> 
> hmm i guess it could be possible that i was hearing either this or the groupdelay i guess, tho its still somewhat unclear what i was hearing in the end but i was not ending up using the filter, it was just a "silly" test


Well, Q = 1/√2 ≈ 0.7071 means Butterworth filter, but -0.08 dB at 16 kHz is confusing and suggests a 9th order (-54 dB/octave) Butterworth filter.


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## Ghoostknight (Dec 15, 2022)

71 dB said:


> Well, Q = 1/√2 ≈ 0.7071 means Butterworth filter, but -0.08 dB at 16 kHz is confusing and suggests a 9th order (-54 dB/octave) Butterworth filter.


16012hz is actually -0.046db
18043hz is -0.266db
20000hz is -3db

im not sure what it is but camilladsp page says second order lowpass
if i use a Q of 0.4 i end up getting around the db`s you suggested before here:


71 dB said:


> 19 kHz........ -2.6 dB
> 18 kHz........ -2.2 dB
> 17 kHz........ -1.8 dB
> 16 kHz........ -1.5 dB
> ...



Edit: but camilladsp also shows in the graph with a Q of 0.0701 that at 16012hz the phase is shifted by -26.812, so possibly i was just hearing the phase shift? im not saying im a bat and hear the frequency loss at 16khz and up (and my hearing just goes to around 15-16khz) but it was audible like there was a loss "in something"


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## 71 dB

Ghoostknight said:


> 16012hz is actually -0.046db
> 18043hz is -0.266db
> 20000hz is -3db
> 
> ...


I know nothing about camilladsp, so you need to tell more if you want me to comment further. Nothing about this tastes like a 2nd order filter. Your data suggests a filter of about 10th order or -60dB/octave.


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## Ghoostknight

71 dB said:


> I know nothing about camilladsp, so you need to tell more if you want me to comment further. Nothing about this tastes like a 2nd order filter. Your data suggests a filter of about 10th order or -60dB/octave.


hmm thats the only thing the CamillaDSP github says:

Highpass & Lowpass
Second order high/lowpass filters (12dB/oct)
Defined by cutoff frequency freq and Q-value q.
HighpassFO & LowpassFO
First order high/lowpass filters (6dB/oct)
Defined by cutoff frequency freq.


i also found this which may be related:

To build more complex filters, use the type "BiquadCombo". This automatically adds several Biquads to build other filter types. The available types are:

ButterworthHighpass & ButterworthLowpass
Defined by frequency, freq and filter order.
LinkwitzRileyHighpass & LinkwitzRileyLowpass
Defined by frequency, freq and filter order.
Note, the order must be even

so it seems the "default" lowpass isnt a butterworth lowpass


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## 71 dB

Ghoostknight said:


> hmm thats the only thing the CamillaDSP github says:
> 
> Highpass & Lowpass
> Second order high/lowpass filters (12dB/oct)
> ...


Q-value dictates what type the (12dB/oct) filters are. First order (6dB/oct) filters are too simple to have a Q value or "type" so they just are what they are (all the types at once). The Biquad-based  filters are just higher order filters made cascading many biquad blocks. Maybe you have been using 10th order ButterworthLowpass? That would more of less fit the data you gave at 16, 18 and 20 kHz.


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## Ghoostknight

71 dB said:


> Maybe you have been using 10th order ButterworthLowpass? That would more of less fit the data you gave at 16, 18 and 20 kHz.


i was using the normal default biquad "Lowpass" with Q0.7071
im not sure if camilladsp is using a wrong filter but i would imagine that someone would have reported it by now if its indeed the wrong type

here is a pic (ignore that the filter is called subsonic )


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## Sinister Whisperz (Dec 15, 2022)

Glad I stumbled over this. Now I understand why my 24 bit 192kHz Pink Floyd collection is indistinguishable from several 16/44.1 files I have of the same albums. All this time I thought my equipment/ears sucked.  Is there a way to convert the 24 bit stuff down to 19/44.1 or do I need to replace them?  The 24 bit take up a ton of space on my devices.


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## bigshot

You can use an app like Audacity to bounce down your large files, or you can convert them to high data rate lossy which is also audibly transparent and even smaller in file size. I use AAC 256 VBR for everything.


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## Sinister Whisperz

bigshot said:


> You can use an app like Audacity to bounce down your large files, or you can convert them to high data rate lossy which is also audibly transparent and even smaller in file size. I use AAC 256 VBR for everything.


Awesome, Thank you.


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## bigshot

No problem. Happy to help.


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## old tech

GoldenOne said:


> I never said this.
> My opinion in relation to audibility with hi-res content is to do with the sample rate not bit depth. I don't think 24 bits vs 16 bits makes an audible difference.
> 
> What does this have to do with Bob Stuart? Genuine question, I'm no fan of Bob Stuart or MQA as may be obvious from some of my previous videos
> ...


I recall Bob Stuart (or Meridian) had something to do with commissioning the meta analysis.  It's been a while but I think there are a few references in one of the threads on this forum.

Sort of what you alluded to, the main criticism of the Meyer and Moran test was that there were no controls on the source material, i.e. were they actually hi res or as we mainly find, just upsampled recordings marketed as hi res. Mark Waldrop in particular argues that unless the production chain from recording through to final master is at least 24/96 then it is not really hi res and that rules out all analogue recordings.

Now Mark's business is the production of hi res recordings so presumably he has a vested interest in hi res. He has always maintained that there is a clear audible diffference to a proper hi res recording to 16/44 or analogue. So he himself conducted controlled tests between his own hi res recordings and the same bounced down to 16/44 and opened the test to his readers.  The results confirm all the other results over the past 30 odd years and he now agrees there are no audible differences for most listeners and only under non-natural listening conditions.

https://www.realhd-audio.com/?p=6993

https://www.realhd-audio.com/?p=7037


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## bigshot

See Mark Waldron’s link in my signature file. He has changed his tune.


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## old tech

bigshot said:


> See Mark Waldron’s link in my signature file. He has changed his tune.


That's what I said... I admire the guy because his business model is based on the sound quality difference between hi res and 16/44 and analogue recordings and against his business interests he now says there is no real difference between CD quality and Hi Res, instead he now focuses on his end to end production processes that result in superior sound.

The other elephant in the room that seems to be ignored is why after 30 odd years are we still debating this? Even if there was some difference that could be audible in certain circumstances it must be so immaterial that the debate rages on. Surely our energies should be focused on other area that do make a material difference. Personally I'd prefer to hear more discussion on various nuances in the recording/mixing/mastering techniques and processes that actually do make a difference and have the potential to drive sound quality higher.


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## Ghoostknight (Dec 16, 2022)

old tech said:


> The other elephant in the room that seems to be ignored is why after 30 odd years are we still debating this? Even if there was some difference that could be audible in certain circumstances it must be so immaterial that the debate rages on. Surely our energies should be focused on other area that do make a material difference.


because why even bother to NOT use high res? the only downside is storage which in most cases isnt really a problem anymore

even if we cant detect changes in (most) systems, high res is just "better" than CD quality, yes you can argue that cd quality is enough for the human hearing but why not go beyond since its theoretical better? like i said... storage is the only minus about it... either way it cant be worse than cd quality so why even keep on doing this discussion?

if you wanna make sure to get the best possible audioquality starting from your playback chain -> go with high res
if storage is a limiting factor for you -> go with cd quality


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## bigshot

Why not double down and make files even bigger than HD? Because as soon as a format is established, they create an even bigger one so you buy Dark Side of the Moon for the umpteenth time.

That’s assuming you have a phone with massive internal storage, and you don’t want to do any kind of Bluetooth wireless connection.


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## 71 dB (Dec 16, 2022)

Ghoostknight said:


> i was using the normal default biquad "Lowpass" with Q0.7071
> im not sure if camilladsp is using a wrong filter but i would imagine that someone would have reported it by now if its indeed the wrong type
> 
> here is a pic (ignore that the filter is called subsonic )


Interesting that I don't see filter order selection anywhere. The phase response is about -90 degrees at 20 kHz which indicates 2nd order filter.


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## 71 dB (Dec 16, 2022)

Ghoostknight said:


> because why even bother to NOT use high res? the only downside is storage which in most cases isnt really a problem anymore


Aren't hi-rez files more expensive than lower rez files? Isn't that the whole point of hi-rez business?



Ghoostknight said:


> Even if we cant detect changes in (most) systems, high res is just "better" than CD quality


Technically it is far superior to CD quality, practically less superior (because taking fully advantage of for example 24 bit is quite impossible) and for human ears it is practically just as transparent as CD quality. For bats and dogs with hearing that goes an octave or two higher than our hearing hi-rez offer clear benefits.

If there was a 32 kHz / 10 bit format out there most people wouldn't hear anything wrong with it! Audiophiles would and could pretty easily tell it apart from CD quality of course. That's why it makes sense to have 44.1/16 instead of 32/10, because making only people with not so great listening skills listening to DR 6 pop with beats audio headphones on their smartphones in a noisy bus happy just isn't enough. That's why having 44.1/16 is important, but that's where the benefits on consumer audio ends (actually this happens at about 13 bit) and having bigger numbers stops making sense.



Ghoostknight said:


> Yes you can argue that cd quality is enough for the human hearing but why not go beyond since its theoretical better? like i said... storage is the only minus about it... either way it cant be worse than cd quality so why even keep on doing this discussion?


Why not be contempt with something that is good enough and spent the energy/time/money on something that isn't? It is unnecessary to make people with lesser understanding and knowledge into this subject to doubt things. Hi-rez has its place in production/studios. They aren't unnecessary, but consumers don't need them.

Hi-rez can be worse than CD quality, if for example it contains strong ultrasonic noise (undetected in production because nobody can hear it!) can cause trouble with some analog gear not engineered to handle strong frequencies that high. CD quality offers protection against such "ultrasonic garbage" thanks to everything above 20-22 kHz being filtered out.



Ghoostknight said:


> if you wanna make sure to get the best possible audioquality starting from your playback chain -> go with high res
> if storage is a limiting factor for you -> go with cd quality


And if you are old-fashioned like me go with CD.


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## Ryokan

Walkman playing times for one model using SE, most other players will be a lot less:

MP3 (128 kbps) 30 hours 
AAC (256 kbps) 27 hours 
FLAC (96 kHz/24 bit) 26 hours 
FLAC (192 kHz/24 bit) 23 hours 
DSD (2.8224 MHz/1 bit) 19 hours 

Why use a large format which shortens battery time on a dap which can't even be distinguished from a high mp3 bit rate? I mostly use flac which is more than adequate and as I have a few players so can rotate when the battery gets low.


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## 71 dB

Ryokan said:


> Walkman playing times for one model using SE, most other players will be a lot less:
> 
> MP3 (128 kbps) 30 hours
> AAC (256 kbps) 27 hours
> ...


LPCM 44.1/16 files such as wav probably consume batteries the least.


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## Ghoostknight

bigshot said:


> That’s assuming you have a phone with massive internal storage, and you don’t want to do any kind of Bluetooth wireless connection.


i probably wouldnt bother with phones, its either 1. bluetooth or 2. a crappy dac, maybe if you use DLNA to stream content from your phone it matters



71 dB said:


> Interesting that I don't see filter order selection anywhere. The phase response is about -90 degrees at 20 kHz which indicates 2nd order filter.


there is a "lowpassFO" option which is first order or those BiquadCombo filters which i havnt tried yet
but i took a look at the "ButterworthLowpass" BiquadCombo filter (where you can set order instead of Q) with second order set and it seems identical to the "lowpass Q0.7071" i was using, i guess Q is the only mattering factor then?
i think if you wanna go with a 4th order filter you have to just stack two 2th order filters



71 dB said:


> Aren't hi-rez files more expensive than lower rez files? Isn't that the whole point of hi-rez business?


hmm true but most the time they are not that more expensive i think or you are using the wrong site to buy them



71 dB said:


> And if you are old-fashioned like me go with CD.


i have to say i kinda like the CD-culture, specially for the tradeability/second hand market, i wish there would be a "modern" equivalent, i even thought about the idea of getting a cd drive for my RPI4 
(i kinda wonder right now if you can resell digital files, tho i didnt saw a market for that or it would be easly flooded with pirated stuff if it would exist)

but i think digital files are superior 
1. you get a original copy (in most cases) 
2. i actually compared ripped CD files with "original" digital files and there was something strange going on with most ripped CDs idk why ( the difference was even more apparant then high res vs cd quality digital files imo ) im not sure how a cd player sounds/compares since i didnt own one (yea the cd rips were pirated, but just for test purposes ^^)

But overall, of course, dont feel pressured from me saying high res matters, i mainly go for it to just have piece of mind and not bother with asking the question myself tbh 
i think there is a (very small) difference but it probably is in the 1-5% range, mp3 vs cd quality is far more audible


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## bigshot (Dec 16, 2022)

My phone measures as good as high end audio players.

MP3 LAME 320 and AAC 256 are indistinguishable from lossless and higher data rates.


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## Ghoostknight

bigshot said:


> My phone measures as good as high end audio players.


maybe from a frequency response pov but it doesnt sound the same



bigshot said:


> MP3 LAME 320 and AAC 256 are indistinguishable from lossless and higher data rates.


for me mp3 just doesnt sound as crisp, more unnatural highs, sloppier bass, voices are more sibilant are probably some of the things i hear in mp3 compared to cd quality/high res
tho i just hear those differences on a good setup (room corrected studio monitors), i dont hear them with my DT880 either


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## bigshot (Dec 17, 2022)

Ghoostknight said:


> maybe from a frequency response pov but it doesnt sound the same


My Mac equipment sounds the same as my Oppo HA-1 and BDP-103D. That's as high end as I get.

You wouldn't be able to discern a difference between lossless and AAC 320 VBR for sure, and I would like to find someone who can discern 256 VBR in a controlled test. I have yet to find someone who can do that. (The only one who could later admitted to gain riding and looping fadeouts to generate ABX logs to convince others of the golden quality of his ears.) Now I ask that the test files be created and the test be administered by a neutral third party.

You are probably thinking of lower data rate Fraunhofer MP3. Codecs have come a long way since then.


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## Ryokan

71 dB said:


> LPCM 44.1/16 files such as wav probably consume batteries the least.



wav use less battery even though they're often twice+ the size of standard flac? That surprises me. I used wav a few times but they took up too much space, couldn't tell them apart from flac and can't embed album art to them.


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## Ghoostknight

Ryokan said:


> wav use less battery even though they're often twice+ the size of standard flac? That surprises me. I used wav a few times but they took up too much space, couldn't tell them apart from flac and can't embed album art to them.


flac needs to be decompressed, just like mp3, but flac is LOSSLESS compression, it needs cpu power to be decompressed
wav on the other hand is completely uncompressed so it uses more space and doesnt need decompression/encoding


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## Ryokan

Ghoostknight said:


> flac needs to be decompressed, just like mp3, but flac is LOSSLESS compression, it needs cpu power to be decompressed
> wav on the other hand is completely uncompressed so it uses more space and doesnt need decompression/encoding



 Thank you for the explanation.


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## 71 dB

Ryokan said:


> wav use less battery even though they're often twice+ the size of standard flac? That surprises me. I used wav a few times but they took up too much space, couldn't tell them apart from flac and can't embed album art to them.


As Ghoostknight nicely explained, it is not so much about how much data you have, but how much work you need to do with it.

Portable music is a compromise between sound quality, battery life and memory usage. All three can't be "great" at the same time.


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## Ryokan

Yes @71 dB portable music is often a compromise though is fine for me, devices have come a long way since my first dap with 256mb memory and I used to buy albums in mp3 192kbps format thinking they were cd quality, and which I've since replaced.


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## 71 dB

Ryokan said:


> Yes @71 dB portable music is often a compromise though is fine for me, devices have come a long way since my first dap with 256mb memory and I used to buy albums in mp3 192kbps format thinking they were cd quality, and which I've since replaced.


192 kbps mp3 can be (perceptually) close to CD quality, especially in more challenging listening environments typical to portable music. Above 200 kbps bitrates lossy formats start to be hard to tell apart from original non-compressed versions and even if you can hear differences when comparing, you are not comparing anything when listening to music on the go! Differences don't automatically ruin music enjoyment unless you convince yourself it does and somehow our "greedy" mind tells us constantly that only the best/perfect can be good enough. It doesn't have to be that way. I'm old enough to remember the days of Sony Walkman C-Cassette players and crappy portable headphones with no bass below 100 Hz. I listened to hissy, compressed and distorted sound with wow and flutter recorded from analog FM radio and I ENJOYED* it! Compared to that kind of sound quality, even 128 kbps mp3 is extremely high quality and 192 kbps mp3 is significantly better. Audio is about the rule of diminishing returns. It is okay and even recommended to make some effort for increased sound quality, but at some point the efforts get you too little and it is time to be contempt with what you have.

* Perhaps nowadays I would find that kind of sound comically low in quality, so there is that.


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## Ryokan

71 dB said:


> 192 kbps mp3 can be (perceptually) close to CD quality, especially in more challenging listening environments typical to portable music. Above 200 kbps bitrates lossy formats start to be hard to tell apart from original non-compressed versions and even if you can hear differences when comparing, you are not comparing anything when listening to music on the go! Differences don't automatically ruin music enjoyment unless you convince yourself it does and somehow our "greedy" mind tells us constantly that only the best/perfect can be good enough. It doesn't have to be that way. I'm old enough to remember the days of Sony Walkman C-Cassette players and crappy portable headphones with no bass below 100 Hz. I listened to hissy, compressed and distorted sound with wow and flutter recorded from analog FM radio and I ENJOYED* it! Compared to that kind of sound quality, even 128 kbps mp3 is extremely high quality and 192 kbps mp3 is significantly better. Audio is about the rule of diminishing returns. It is okay and even recommended to make some effort for increased sound quality, but at some point the efforts get you too little and it is time to be contempt with what you have.
> 
> * Perhaps nowadays I would find that kind of sound comically low in quality, so there is that.




I rarely take my portable device further than the garden. 256kbps is the minimum I'd listen to. A few years ago it was flac which is nice knowing your 'covered' but isn't necessary.


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## bigshot

The difference in file size between 192 and 256 is very small compared to the difference between lossy and lossless.


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## Ryokan

And yet you use 256kbps (vbr) to be on the safe side. I'd hazard a guess 192 is the region where you'd start to notice an occasional downgrade?


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## bigshot (Dec 18, 2022)

I did a lot of testing with all kinds of recordings and found a “killer track” that was harder to compress without artifacting. At 192 the artifacting was barely audible. At 256, it was completely gone. I added VBR because that just makes sense. I probably would have done fine at 192 VBR, but the difference in file size was negligible in the grand scheme of things.


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## Ghoostknight

bigshot said:


> I did a lot of testing with all kinds of recordings and found a “killer track” that was harder to compress without artifacting. At 192 the artifacting was barely audible. At 256, it was completely gone. I added VBR because that just makes sense. I probably would have done fine at 192 VBR, but the difference in file size was negligible in the grand scheme of things.


im quite sensitive to sibilance and found its one of the easiest things to spot in flac vs mp3, the sibilance itself just sounds more "artifical/worse" with compression


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## bigshot (Dec 18, 2022)

It may be that your encoder is boosting the level a hair into clipping. I found that was the cause of my harshness. Compression is more a matter of artifacting. A good experiment is to take a track and encode it in even data rates from 64 to 320. The lower data rates will show you where the artifacting occurs and you can trace it as it gets less and less until it goes away.

As I said before though, the codec matters. An MP3 is not necessarily the same as another MP3. The more recent codecs, like MP3 LAME and AAC perform transparently at higher data rates, while Fraunhofer MP3 just barely scratches the line of transparency when it's topped out.

If you are concerned, the best one to use is AAC 320 VBR. AAC is the best codec I've found. It's transparent at 256, and if you select 320 VBR, it can actually go above 320 if necessary. (Which isn't likely.)


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## gregorio (Dec 21, 2022)

Looks like a lot has been added while I’ve been gone. I’ll respond to some of it but with paraphrasing rather quotes, to hopefully avoid re-awakening those who are/can be trolls:

Ian: _I have a degree … I can’t explain technical things._ - These two statements are incompatible because to get a degree you have to write numerous essays which specifically require you to “explain technical things”. And of course if you can’t or if you write audiophile myths/marketing explanations rather than the correct technical explanations, then you do not pass. The rest of it I dealt with in another thread.

T1000: _If you can’t hear the difference, what’s the point of HiRes (eg. 24/96), what are the technical advantages?_ - Good question. Contrary to some of the responses, 24bit has nothing to do with and no advantages when mixing or mastering. Mixing and mastering are not and have never been done at 24bit. Typically it’s done these days at 64bit (float) and prior to that 32bit (float) or 54bit (fixed). The first commercially available digital mixer (1987) used 28bit (fixed). The advantage of 24bit is purely for headroom when recording. 20dB (for example) of headroom when recording in 16bit can impinge on the 16bit noise floor and make it audible during mixing when the headroom is removed and compression is applied. It also can be advantageous (for the same reason) when transferring a mix for further processing, EG. Transferring a final mix to the mastering studio. In the case of higher sample rates, 96k was/is very useful in the film industry, when designing sound effects. It’s quite common to have SFX elements which are pitch-shifted down by 2 octaves (or even more) the roughly 20kHz cut-off with the 44.1k sample rate becomes 5kHz, which is problematic. With a 96k sample rate, a 2 octave pitch down puts the cut off point at 12kHz, which isn’t a problem. Additionally, certain effects used in music production benefit from the higher sample rate (modelled compressors, soft synths and others) and within a window of a few years, around and just after the millennium, it was possible to hear the difference between 96k and 44.1k recordings (when these effects were employed). Then increased processing power enabled the plugin programmers to upsample internally and there was no longer any audible difference. So there are/were some technical benefits but not for reproduction, only for the recording/production side.

GoldenOne: _We can’t achieve Shannon/Nyquist perfectly. Filters are audible, jitter is audible, there are audible differences between DACs._ - Correct, we can’t achieve Shannon/Nyquist perfectly, we can only achieve it down to about -160dB. However, audibility isn’t an applicable question because the voltage that represents is way below the noise floor of any amp or speaker, far too small to have an effect on a driver and it can’t be audible if it can’t be reproduced to start with. Audibility of filters used to be an easy subject to deal with, a standard linear phase filter with a transition band of about 2k starting around 19-20kHz was inaudible. Then a few years ago chip manufacturers started putting in the facility to switch to different filters, some of which did have audible effects, for example a roll-off starting at 10kHz or no longer being linear phase. So now the answer is a rather unsatisfactory: “Filters should be inaudible but sometimes they’re not, it depends”. IMO, if they are audible then they’re defective! Jitter - Far more clear cut. Quite a few studies, starting AFAIK in 1974 (BBC). It became quite a topic in pro-audio circles in the 1990’s due to a marketing campaign by Apogee for a master clock and then an infamous public spat between some of the top music engineers and Apogee’s marketing manager. Quite a few studies and industry articles around the late ‘90’s put the issue to bed for good. One of them was the Gannon paper you mentioned, only 9 subjects it’s true but those 9 were selected as the most sensitive to jitter artefacts after two rounds of training from a larger cohort. The absolute threshold using a test signal was about 3ns, using music it was just under 30ns, about 200 times higher than the average jitter produced by standard/cheap CD/DVD players and digital TVs from around the mid 1990’s. Audible differences between DACs - Bit of a similar story to filters. They should all (even cheap ones) have reduced artefacts to below audible levels by around the turn of the millennium, therefore all sound effectively perfect and the same. However, that doesn’t help the audiophile manufacturers who want to charge a great deal more than the cheap ones. So either they increased the specifications and falsely implied/stated they had audibly better fidelity or a few actually used retro-designs (sticking tubes in them, R2R or NOS for example) with audibly poorer fidelity but marketed them as being somehow better. Poorer fidelity, higher price, sweet! So generally there’s no audible difference between DACs but very occasionally there is, it depends. Fortunately, this only affects the audiophile world. In pro-audio we just use the good old audibly perfect (linear phase) filters and ADCs/DACs without audible artefacts.

There were various other comments about mastering. As a general rule, the recording sample rate is mastered and we get a virtual/session master, say 48/64, 96/64 or 192/64 from which the distribution masters are recorded/bounced, say 192/24 and 44/16. Although we can apply more processing (EG. More compression) to a bounce/version. The use of 192 or higher for recording is rare because of the space and processing requirements. Of course you can get around this issue by compromising, using fewer channels and then marketing to audiophiles that this minimal 1940’s approach is actually better, sweet!

G


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## GoldenOne

gregorio said:


> GoldenOne: _We can’t achieve Shannon/Nyquist perfectly. Filters are audible, jitter is audible, there are audible differences between DACs._ - Correct, we can’t achieve Shannon/Nyquist perfectly, we can only achieve it down to about -160dB. However, audibility isn’t an applicable question because the voltage that represents is way below the noise floor of any amp or speaker, far too small to have an effect on a driver and it can’t be audible if it can’t be reproduced to start with. Audibility of filters used to be an easy subject to deal with, a standard linear phase filter with a transition band of about 2k starting around 19-20kHz was inaudible. Then a few years ago chip manufacturers started putting in the facility to switch to different filters, some of which did have audible effects, for example a roll-off starting at 10kHz or no longer being linear phase. So now the answer is a rather unsatisfactory: “Filters should be inaudible but sometimes they’re not, it depends”. IMO, if they are audible then they’re defective!


I'd ask that you read my post which included testing on audibility.
I've got some more stuff in the works and am going to be publishing further testing results, as I have now managed to achieve P-Values of <0.0001 between two filters, one of which is a 'correct' filter in that it begins rolling off at 20khz and fully attenuates by nyquist, and the other which is a 1 million tap filter with near instantaneous attenuation. (Both of which are linear phase I should mention, and additionally, I am unable to hear >20khz)
Both of which are 'correct' and what many would assume to be 'audibly transparent', and yet are audibly discernible in blind ABX tests to a beyond statistically significant degree on identical hardware.
I'll have a proper writeup coming on the topic soon which should hopefully serve as a good base for some interesting discussion and further exploration of the topic.


In regards to DAC audibility difference, the above in and of itself would demonstrate that DACs do sound different, as not all use the same filters and therefore if the filters themselves are audibly different the DACs will be audibly different.
But I'll be doing further controlled ABX testing once the first post is out.
I should mention that ABX testing between DACs is something I actually do fairly regularly when reviewing stuff. I've just not posted results because they can't be verified remotely anyway and as I've described elsewhere would simply lead to arguments and likely wouldn't actually be too helpful, previous attempts to set up a monitored/observed test have devolved into less than productive arguments and so I'd like to complete and publish the above testing on filters given as that CAN be verified remotely. And then will have the physical DAC ABX testing as a followup.

The device itself I'm quite happy with: https://streamable.com/tuuyzg
(Had a couple revisions since then but the video shows the way in which it works)





As to jitter audibility, I don't have a particular view on what I think the audibility threshold might be. My issue was simply that people are quick to call many studies concrete and conclusive, when actually I think it would be fair to assume that for various reasons that might not be the case.

- Many of them provide little information on participants
- Many of them use small numbers of participants
- Advances in things such as DS modulators themselves could potentially have an effect on the audibility of jitter. DACs have moved on quite a bit since the main study in 1998
- Other issues are often present such as not even properly describing what the content being tested was in the instance of one study someone linked earlier

I'm not saying "I think the studies are wrong". Just simply "I think more study needs to be done before we can actually be certain"
I do have other specific concerns about the previous studies, but I'll discuss those in more detail once I've done some thorough testing and see if results vary at all from current literature.


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## bigshot (Dec 23, 2022)

Would it be possible for you to prove a difference for yourself between two basic DACs or CD players? If you do it all the time, that would be easy to set up, and you could document your comparison as you do it. It would be good to prove that a difference exists in practice and try to quantify it to put it in context before trying to figure out theories about why there is a difference. You could do the test you say hasn't been conducted properly and prove that before moving on to refining the test to test for specific reasons.

If you jump straight to testing filters you create, then the question would immediately be "is the filter you created really typical" and you would be going back to this anyway. Just showing a picture of the electronic device you made doesn't prove that a consumer CD player sounds different. It only means that your device sounds different.

It's good to track stuff down methodically and not jump too far ahead of yourself.


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## GoldenOne

bigshot said:


> Would it be possible for you to prove a difference for yourself between two basic DACs or CD players? That would be easy to set up, and you could document your comparison as you do it. It would be good to prove that a difference exists in practice before trying to figure out theories about why there is a difference. You could do the tests you say haven't been conducted properly and prove that before moving on to refining the test to test for specific things.


Yes, as described previously the issue is that whilst I might be able to do this test and get a very conclusive result, without it being able to be verified, anyone who feels strongly that DACs make no difference will simply assume something was done wrong or cheated and nothing of much value will have been achieved.

I'll be publishing this anyway, but want to get the reconstruction filter stuff out first for a couple reasons

1) It kills two birds with one stone, demonstrating that reconstruction filters even when 'correct' are audibly discernible, and therefore by extension DACs which don't use the same filters can be audibly discerned.

2) It's verifiable remotely, testing is done digitally with the same hardware, with original files provided, and checksums for both the files themselves so anyone can verify the files I provide for anyone to inspect were the ones used, and also a hash that verifies the results of the ABX itself including the files, output device, and score. Therefore there can't really be any 'well you probably didn't volume match' etc or other doubts about the test setup (which is fine, test methodology SHOULD be scrutinized). The results will be concrete.


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## bigshot (Dec 23, 2022)

If you don't think you can prove that two CD players sound different, what makes you think that you can prove that a filter you create yourself isn't coloring the sound? It really isn't a blind test if one person is designing the equipment being tested, conducting the test and reviewing the results, is it? You admit that you have a vested interest in proving that your theory that filters all sound different is true, don't you? I think by avoiding the first step, you're just creating more problems for every step after that. You can go ahead and do this if it interests you and you learn something from it, but it doesn't seem at all like a test that would convince someone else of your theory. Peer review is even worse than Internet forums when it comes to people questioning details about how the test was conducted. If that is your aim, I would think you would want to not avoid crossing t's and dotting I's


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## gregorio (Dec 23, 2022)

GoldenOne said:


> Both of which are 'correct' and what many would assume to be 'audibly transparent', and yet are audibly discernible in blind ABX tests to a beyond statistically significant degree on identical hardware.


If accurate, that would be an anomaly. I’ve done significant testing on filters over the course of several decades, quite a few of my colleagues have, plus they’ve been tested by the industry for even longer. Obviously some filters are audible but the typically linear phase 2k transition band ones (starting around 20kHz) have been demonstrated to be inaudible for many years. This doesn’t mean it’s impossible, just very unlikely and one anomalous result isn’t going to change that without rigorous testing (and of course repeatability). I would first be looking for some methodology fault in your testing or some fault/anomaly in the filters you tested.


GoldenOne said:


> In regards to DAC audibility difference, the above in and of itself would demonstrate that DACs do sound different


It would, assuming it is accurate and repeatable. However, I did not claim all DACs sound the same, some pathological ones don’t.


GoldenOne said:


> As to jitter audibility, I don't have a particular view on what I think the audibility threshold might be. My issue was simply that people are quick to call many studies concrete and conclusive


First published study was in 1974, I don’t think that nearly 50 years is quick. Of all the studies and again, all my and my peers’ and industry testing, I’ve never seen any reliable evidence that jitter artefacts are even anywhere near audibility. Again, I would need some exceedingly robust evidence to overturn everything up to this point.


GoldenOne said:


> DACs have moved on quite a bit since the main study in 1998


Exactly! Would you expect a modern DAC to have significantly poorer performance than say a cheap CD drive from 1995?

G


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## GoldenOne

bigshot said:


> If you don't think you can prove that two CD players sound different


As I said, I DO feel I can prove this and have achieved more than conclusive P-Values in blind testing


bigshot said:


> what makes you think that you can prove that a filter you create yourself isn't coloring the sound


Because I'm not creating the filter. I'm using readily available tools that anyone is free to inspect for themselves if they feel there may be some other issue at hand.



bigshot said:


> but it doesn't seem at all like a test that would convince someone else of your theory


Which is exactly why I'm starting with methodology that can be completely picked apart and verified by anyone who reads it. Rather than a physical test that leaves a lot up to trust.
A more conclusive result with no issues is good is it not? This isn't about being right/wrong. This is about doing things properly and not making assumptions.


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## Ghoostknight (Dec 23, 2022)

one funny thing i noticed with blind testing lately was this:
for fun i wanted to repeat the absolute phase blind test of audiocheck.net to see how good i can get it, i got the first 9 out of 9 right, and it went like this after that: wrong, right, right, wrong, wrong, right, wrong, right, wrong and downhill from here, i reloaded the page to start freshly and all this switching back and forth made my ability to discern absolute phase like "vanish" i got even more false then right the second time

if i switch the absolute phase in camillaDSP for normal playback i think it has a very discernable difference, treble sounds more natural/realistic and you can actually hear/feel the direction in bass notes

this is for me one other point to the thing i mentioned earlier.... blind tests can actualy fool you with the fast switching back and forth and i dont believe the first 9/9 i got right was luck (yea there is a chance but this doesnt explain why i hear such differences if i switch absolute phase in normal playback (with "test-tracks" tho, it depends on the song if it makes a big difference))


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## bigshot (Dec 23, 2022)

All I'm recommending is a logical progression to your testing to isolate the parameter you're actually testing.

My experience is the same as Gregorio's. In order for two DACs to sound different, there needs to be something "pathological" as he puts it. Specifically NOS DACs, filters that are designed to color the sound, defects in design or manufacture, etc.

I find that most people who report differences in sound between DACs are referring to DAC/amps, not specifically the DAC chip. And the difference is generally caused by impedance issues between the DAC/amp and the particular headphones they are using, not the DAC chip itself. There are some IEMs that just don't work properly with most of the headphone amps on the market. You have to buy their brand's specific amp to get the proper impedance characteristics to achieve transparent sound.


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## GoldenOne

gregorio said:


> Exactly, would you expect a modern DAC to have significantly poorer performance than say a cheap CD drive from 1995?


I would not, but it would not be fair to assume that the audible effects of jitter are not masked by anything at all.
If we see improvements in various areas, it's worthwhile to check whether this may have been a bottleneck in previous testing would it not?




gregorio said:


> I would first be looking for some methodology fault in your testing or some fault/anomaly in the filters you tested.


And I would welcome any feedback on test methodology.
As said before, I want to do this properly, and I want what I put out to be as concrete as possible. If yourself or anyone has genuine feedback on the test methodology I would welcome it and will iterate on it to address concerns.

But posts in the realm of "Well I don't think it matters" or "No one else has shown a difference so why are you trying" are not constructive at all.


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## bigshot

Ghoostknight said:


> i dont believe the first 9/9 i got right was luck (yea there is a chance but this doesnt explain why i hear such differences if i switch absolute phase in normal playback (with "test-tracks" tho, it depends on the song if it makes a big difference))



If the first test hadn't validated your existing belief would you have felt the same? Probably not. This is a great example of validation bias. The chance of getting 9 out of 9 on your first test by random chance is the same as getting it on your tenth. The fact that it was your first test is only relevant to your personal bias.

If you think that you have a better chance of detecting it on the first try, do the test every day one time for the next 30 days and see if you get 9 out of 9 every time.


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## bigshot (Dec 23, 2022)

GoldenOne said:


> And I would welcome any feedback on test methodology.



I would recommend a logical progression to isolate the particular theory you are testing.

Prove a difference exists in practice using two different consumer CD players and comparing their line out, level matched, direct A/B switched, blind.

Then once you're proved a difference, test to see if the same electronics produces a difference. Test two copies of the same make and model to see if the difference is random due to manufacturing tolerances or if it is dependent on the design of the device.

Once you have proved that the two copies of the same make and model sound the same, you have eliminated manufacturing error and you can start focusing on the design. The next step would be to prove that the filter you designed to be typical sounds the same as the CD player you just proved was free of manufacturing error.

If these two are the same, then the filter you created is actually typical. You can compare that to your enhanced filter design and see if you can discern a difference.

Then you would need to determine with measurements and listening tests whether it is an improvement. Assuming listeners can hear a difference between your enhanced filter and the typical filter, do they prefer one over the other?

Then you have proven your point.

The problem is that you argue that every DAC and player sounds different, and then you point at the typical filter that they all employ as being inferior to your enhanced filter. How is your version of a typical filter in any way typical if every DAC or player sounds different? How can you claim a filter is the cause of a difference in sound when all players have the same kind of filter yet they all sound different?

I don't think rigorous testing procedure is the problem. The problem is the logical chain of proofs that lead up to what you are trying to prove.


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## GoldenOne (Dec 23, 2022)

In regards to methodology, this is the setup:

- A piece of music (in 44.1khz sample rate) is upsampled using Signalyst HQPlayer to 705.6khz (16x). Both filters will be linear phase sinc filters, but with differing numbers of coefficients.
I'll provide the filter responses themselves to show that they both permit <20khz content without attenuation, and fully attenuate by Nyquist, but you can also download HQPlayer Pro to upsample any file you like upto 60 sec long if you'd like to inspect/test anything further yourself if there are any doubts.

- Hardware being used will be:
DAC: Holo Audio May
Amplifier: Zaehl HM1
Headphones: Hifiman Susvara

- Both files will be played to the DAC bitperfect via ASIO using Foobar. The ABX plugin (https://www.foobar2000.org/components/view/foo_abx) will facilitate the test over 30 runs. (Allowing for a maximum possible P-Value of 0.0000000009)

- Results from the ABX tests will be provided, accompanied by:
1) A checksum that verifies the score, output device, and files, and can be verified here: https://www.foobar2000.org/abx/signaturecheck
2) The original 44.1khz file
3) Both upsampled files (verifiable via above checksum)

_____________________________


The above test will be repeated multiple times, to test the following:

1) 'Standard' Reconstruction + TDPF Dither VS 'High Performance' Reconstruction (1 Million Taps) + A high performance 15th order noise shaper
This should in theory be the easiest to pass if a difference is shown to be audible, as it affords every advantage to the listener in regards to comparison of 'industry standard' vs 'extra mile' signal processing that DACs might utilise.

2) 'Standard' Reconstruction VS 'High Performance' Reconstruction (1 Million Taps), but both using the same dithering method.
This is the same as test 1, however is testing ONLY the filter itself and does not include standard TDPF dither vs advanced 15th order noise shaping as a factor.

3) 'High Performance' Reconstruction (1 million tap) VS 'Stupid' Reconstruction (Maximum possible tap count possible with the number of samples in the track)
This is to test potential limits of audibility in the event that tests 1 and 2 show statistically significant results. This will test whether a 'high performance' reconstruction with 1 million taps is audibly different to a filter that is as mathematically perfect as we are able to do with the number of samples in the track. Both filters will have the same cutoff frequency.


If anyone has any genuine feedback on this methodology or feels there is an issue. Please let me know.


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## bigshot

What kind of test subjects do you plan to use and who will be administering and recording the test?


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## GoldenOne (Dec 23, 2022)

bigshot said:


> What kind of test subjects do you plan to use and who will be administering and recording the test?


Myself at first, some other audiophile friends when we meet up.
But as I've said, the whole point of doing it this way is that it is entirely transparent and verifiable remotely. Not just by one person, but by ANYONE who reads the post and wishes to do so.

The scores, device used, and files used are cryptographically signed and verifiable.
The files used will be available for inspection.

It's worth pointing out though that when proving something CAN be audible, you don't actually need a large sample size.
When you're demonstrating something is extremely unlikely to be possible, you need a large sample size so you can say "Look, we got 1000 people and there was no one who could do this to a statistically significant degree. Therefore it's extremely unlikely anyone can"

When you're demonstrating that something IS possible, you just need to show that someone can do it to a statistically significant degree.
The P-Value is a statistic that denotes the probability a result could have been obtained by chance. So if you get a P-Value of 0.00001, that means there is a 99.999% certainty that the result is due to a genuine ability to hear a difference. And only a 0.001% chance it was luck.


Whether filters are something that most people can hear is not a question I'm looking to answer. I'm looking to answer whether they can be audible even if that's just by one person.


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## bigshot

If you want other people to accept your findings, get other people to administer the test and other people to be the test subjects. Let the chips fall where they may. Don't consciously or unconsciously skew the results. That's my advice.

An informal test with friends is fine to give you the info you need, but that isn't going to be enough to convince anyone else.


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## GoldenOne

bigshot said:


> If you want other people to accept your findings, get other people to administer the test and other people to be the test subjects. Let the chips fall where they may. Don't consciously or unconsciously skew the results. That's my advice.
> 
> An informal test with friends is fine to give you the info you need, but that isn't going to be enough to convince anyone else.


You can't consciously or unconsciously skew the results if the blind test is properly controlled.

If you have genuine criticism of the methodology please let me know. But there isn't any reason why I wouldn't be able to participate in the test when controls are appropriate. As I believe they are.


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## bigshot

You aren’t really interested in doing a controlled test at all. Your purpose is to prove your gizmo audibly improves sound no matter what.

Okie doke. I understand now.


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## GoldenOne

bigshot said:


> You aren’t really interested in doing a controlled test at all.


Yes I am. Hence why I've described the methodology and structured it to be transparent and verifiable.
And why I've asked for any constructive criticism on it, because I want it to be thorough.

You having a preference for the test to be done a certain way is not a valid reason to change the test and you've not provided a reason as to why the methodology I laid out is problematic. 




bigshot said:


> Your purpose is to prove your gizmo audibly improves sound no matter what.


I've no idea what 'gizmo' you're referring to. I do not make or sell products and have no financial interest in any company with anything to do with this topic


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## bigshot (Dec 23, 2022)

OK. Whatever. You asked for suggestions. I gave you several and you ignored every one without comment.  It sure doesn’t look like you’re intending to do a focused, fair test.


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## Ghoostknight

bigshot said:


> If the first test hadn't validated your existing belief would you have felt the same? Probably not. This is a great example of validation bias. The chance of getting 9 out of 9 on your first test by random chance is the same as getting it on your tenth. The fact that it was your first test is only relevant to your personal bias.
> 
> If you think that you have a better chance of detecting it on the first try, do the test every day one time for the next 30 days and see if you get 9 out of 9 every time.


while you are right that the chance is the same wether its the first or tenth test i just found it "funny" that it was like it was
well you can see it your way, i see it my way, that fast switching back and forth messed things up (i already got a high score in this test like 1 year ago btw)

the "song"  "Circio`s Rapture - Polarity Test" (and apparently a quite new one "Polarity Check") is one of the good examples i have in my list and makes it kinda easy to hear the difference if someone is interested if absolute phase makes a difference 

for me its out of question if absolute phase makes a difference, tho many people say it doesnt matter, that was also one of the first thing which made me really careful with objectivists or general opinions in general regarding audio, one other thing was "motherboard dac is fine" which just baffeled me after trying "good" external dacs
i cant imagine how many people go by these "standards" and havent tried it by themself and probably will not 

maybe i will repeat this like you said, probably not every day, but to see if i come to the same conclusion that ABX tests maybe just dont work for me that good, and probably for others as well
but i think the provided sample of audiocheck.net is a kinda bad one anyway


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## bigshot

Do you always get it right on the first try?


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## Ghoostknight

bigshot said:


> Do you always get it right on the first try?


i haven`t done many blind tests tbh it was a one time thing so far in this constellation, atleast where i noticed/aknowledged it since it was kinda obvious this time
but i done this test already sucessfully, so, atleast for me, its enough proof that i -really- hear a difference, so the last test i did is more than strange imo

so for me its this conclusion : you cant rely fully on subjective tests, you cant fully rely on objective (ABX) tests, both as a one-off test, so you have to go with a "higher sample-rate" either way


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## Steve999

Ghoostknight said:


> i haven`t done many blind tests tbh it was a one time thing so far in this constellation, atleast where i noticed/aknowledged it since it was kinda obvious this time
> but i done this test already sucessfully, so, atleast for me, its enough proof that i -really- hear a difference, so the last test i did is more than strange imo
> 
> so for me its this conclusion : you cant rely fully on subjective tests, you cant fully rely on objective (ABX) tests, both as a one-off test,



Okay, I am bad at this, but here we go.

True, you can't fully rely on a subjective test or an ABX test as a one-off test. I don't think anyone here would would say you could. For more reasons than I could count.



Ghoostknight said:


> so you have to go with a "higher sample-rate" either way



Exactly, or even generally, what do you mean? Do you mean that because you can't fully rely on a subjective test or an ABX test as a one-off test, that you must use a higher sample rate for playback? How do you get to that conclusion? How does one follow from the other?

Or do you mean you would have to conduct more tests, so you have a larger sample size from which to draw conclusions? If so, that appears correct, as far as it goes, it's a start.


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## Steve999

GoldenOne said:


> Yes I am. Hence why I've described the methodology and structured it to be transparent and verifiable.
> And why I've asked for any constructive criticism on it, because I want it to be thorough.
> 
> You having a preference for the test to be done a certain way is not a valid reason to change the test and you've not provided a reason as to why the methodology I laid out is problematic.
> ...



Well, again, I am not good at this stuff, I hate arguing on the net.

However, if I were trying to prove something generally in a sound and credible way about human perception (even if I were just looking for one rare bird with exceptional ability) I do not think I would make myself the main subject of my own ABX tests for the audibility of a filter that I advocated in advance would sound different from another filter, against general conventional wisdom. Surely if you can hear it you are not the only one who could ever hear it? So why include yourself as a subject of the testing at all? It truly just introduces the thorny subject of bias. Problems with bias are rampant among professional scientists who conduct real-world ABX tests. So I do not buy that you are somehow immune to this extremely real problem. 

Can you imagine a study along these lines earning respect among one's peers as reliable proof? Your objectivity about your own testing of yourself to prove or disprove your own beliefs is going to be unconvincing. 

If you are doing this to educate yourself as to just noticeable differences for yourself and that sort of thing then that is a constructive exercise, it will give you a more realistic lay of the land I think. And if you can hear a difference, and you educate yourself to know that you have proven this to yourself well, then you really have something to build on in terms of putting together a credible set of data with other folks as test subjects from which you might draw conclusions. As far as your test of yourself being reliable evidence worth outside consideration and inspection to prove something in particular to the outside world, not so much, IMHO. Once you are sure you are hearing it for yourself, go to someone in the outside world as a test subject. But until you do that, it's really just you learning about your own perceptions and it's not proving anything to anyone else in a meaningful way, IMHO. If other folks are skeptical, that would be the intelligent position for them to take, IMHO.

Does that make sense to you?


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## bigshot

Ghoostknight said:


> i haven`t done many blind tests tbh it was a one time thing


Well, it's a start. But logic and science isn't something you try once and congratulate yourself for a job well done. It's a process of breaking down problems and understanding them so you can solve them. Random subjectivity yields random results. And randomness leads to chaos. Science and logic is a process that points the way to accomplishments.

If you want to actually know how things work so you can make a plan for improving the sound of your system, you need to make the effort to think logically about the problem and do research. If you just wish it to be the way you want it to be, and you just ignore anything that goes against your preconceived notions, you're going to be sorely disappointed. That is true of most things in life. But a lot of people lead miserable lives because they don't realize that.

Wishing accomplishes nothing without well thought out action. Of course it doesn't matter if you don't really care, but if that's the case there are a lot better things to do than to talk about stuff you don't care about on the internet.


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## bigshot (Dec 24, 2022)

Steve999 said:


> Can you imagine a study along these lines earning respect among one's peers as reliable proof?



I think you inadvertently touched on something there that is pertinent to what this forum is experiencing right now in the word "respect". There is a profound amount of disrespect flying around. And some people seem to be craving it without the first idea of how to get it.

A lot of people seem to want to be authorities because it will get them the respect of the community. But those aren't things you get just by posting in an Internet forum or making YouTube videos. Respect is something that is gifted to you by others. You can't demand or even expect it. You just receive it if you've earned it, and you're not the one to judge whether you deserve it or not- other people do.

The best reason to post or create videos is in the spirit of generosity- sharing information that you've picked up through experience and research. Even better is to share resources and tools to help other people generate their own experience and research. If someone shares info with you, that is a plus, because you can add that to your own tool kit and have more to share with others. The value of your information isn't judged by whether it makes you look right. It's in the truth of it.

Some people flat out refuse any input because they accepting help from others as weakening their "authority". Some have the goal of- truth be damned- making themselves right at all costs. Other people think that by tearing down someone else who has been given a certain amount of respect , they will gain that respect themselves. Those people are clueless and probably will never achieve the respect that they crave.

Why you share is more important than what you share.


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## The Jester

Logical progression,
Once you eliminate the probable, possible, improbable and unlikely any issue is in what you have left.


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## bigshot

The Jester said:


> Logical progression,
> Once you eliminate the probable, possible, improbable and unlikely any issue is in what you have left.



What you have left is the impossible, don't you?


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## The Jester

bigshot said:


> What you have left is the impossible, don't you?


Theoretically.


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## castleofargh

My bias is that I don't want to trust sighted impressions but I want to trust controlled tests. I say it's a bias because while it seems like a rational approach, it is obviously possible for sighted impressions to come up with the right conclusion while it's possible for controlled tests to be flawed and give false conclusions. Even more so when we're talking lonesome amateur ABX results.
 And yet, without any other data, I'll always lean toward the guys who put in some work toward trying to remove/control variables. I do believe that such people are, on average, wrong less often than the lazy guys. When someone tells me he passed an ABX for something I can't pass, it's strange but I don't doubt it. Same thing coming from a sighted impression and I think "cool story bro". 
I'm biased like that, I want more abx and I want to trust them.


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## bigshot

Bias isn’t the only consideration. How well you’ve isolated the variable you’re testing for is just as important.


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## 71 dB

I have CDs with poor sound quality, CDs with stunning sound quality and everything in between. If CD quality didn't allow awesome sound quality then obviously those ones with stunning sound quality wouldn't exist. Obviously the difference in sound quality comes from other aspects than from the CD consumer format. 

I don't even need 100 % _objective_ _transparency_. Such a thing doesn't even exist outside live music! I need _subjective transparency_ which means the sound is "good" enough to "fool" me its transparent and only when (carefully) compared to even more transparent sound is revealed to be less than 100 % transparent.

I'd say the purpose of this thread is to convince people that 44.1/16 digital format is:

(1) Perhaps not 100 % transparent _objectively*_, but nevertheless insanely close to that.
(2) Easily 100 % transparent _subjectively,_ which is what should matter to people.

Also, the transducers alone make sound reproduction less than 100 % transparent _objectively_, so demanding 100 % _objective_ transparency from consumer audio formats isn't rational to begin with when the _objective_ transparency is limited anyway by some other parts of the audio chain.

We like the though that things can be better. It is a fascinating thought that we can "improve" things: Purchase favorite albums as hi-rez files to replace CDs and business-oriented people know how to take advantage of such feelings. 

* Individuals with excellent/super-human hearing may hear the effects of brick-wall anti-alias and reconstruction filters on certain signal types.


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## bigshot

Transparency does exist. If the sound is audibly the same, it’s transparent. Transparency is a word with both a technical meaning and a connotative meaning. When we talk about audible transparency, we are talking about the denotation, not the connotation.


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## gregorio (Dec 24, 2022)

GoldenOne said:


> I would not, but it would not be fair to assume that the audible effects of jitter are not masked by anything at all.


First of all, what audible effects of jitter? You haven’t determined there are any and Secondly, audio masking only happens in the ear and requires a similarly or higher level signal to mask it, both of which would be apparent in various measurements or more obviously in a null test.


GoldenOne said:


> If we see improvements in various areas, it's worthwhile to check whether this may have been a bottleneck in previous testing would it not?


Assuming there are any audible bottlenecks.


GoldenOne said:


> But posts in the realm of "Well I don't think it matters" or "No one else has shown a difference so why are you trying" are not constructive at all.


If any improvement is below the level of audibility then it doesn’t matter as far as listening is concerned. It will only matter to those who place importance on better inaudible specifications. I didn’t say and don’t believe that you shouldn’t try just because no one else has shown a difference. If you did really demonstrate a difference, I for one would find that interesting, although I wouldn’t know if that result had any practical implications until it was investigated formally and reproduced by others.


GoldenOne said:


> In regards to methodology, this is the setup:


I have a few issues/potential issues of varying concern.

Why are you converting to 705kHz? With such a relatively high data rate, there’s the potential concern of output/transfer errors. EG. Are you hearing differences between filters or errors due to an abnormal digital audio transfer rate, from your computer or DAC? This isn’t a big concern because even if there are errors they should be random and not bias the result but it is potentially a point of failure (of methodology).

The Holo May is also a slight concern. The R2R topology is one of those esoteric/pathological design topologies which can indeed be audible. It’s only a slight concern because the Holo May appears to perform virtually as well as standard topology DAC but again, why not eliminate this potential “point of failure”/criticism if you can?

A more serious point of concern, please state the test SPL reference. Obviously, one could ABX a small, quiet section (say the fade out), whack up the amplification by XdB and then pretty much any artefact at any level can be made audible.

In your second section, I think you should not combine tests, say a filter + dither test. Test for them individually, the idea being to try and make sure that the only single variable changing is the one you’re testing. Having 2 simultaneous variables introduces more potential points of failure and if you manage to falsify the null hypothesis, you don’t know why, which of the variables caused it.

Again, the SPL reference please. TDPF dither is trivially easy to hear (and distinguish) at even just a fairly modestly elevated level.

Also, very high and “stupid” number of taps, are you absolutely certain you’re actually testing the difference between filters and not the difference between artefacts caused by the software/computer while trying to calculate them?

G


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## Eagle_Driver

I can see how 24-bit/192 kHz digital audio can sound worse than even 8-bit/32 kHz audio, much less 16-bit/44.1 kHz:

If a DAC is capable of 24-bit/192 kHz processing, then the analog output stages must be capable of rejecting this IM distortion caused by extraneous ultrasonic frequencies. Unfortunately, most analog outputs in consumer devices are mediocre to poor at this job. This makes a proper 24/192 device (that is, one that sounds even meaningfully better than a device that's restricted to 16/48) prohibitively expensive to manufacture - so much that only the extremely affluent can afford it.

That said, 24- or 32-bit recording does have its advantages in production and mastering as dithering the final output down to 20- or even 16-bit will not affect the sound all that much. But recording at too high of a sampling rate will be pointless, as many of the artifacts caused by ultrasonic frequencies will carry over into the downconverted-to-44.1 final output file. That makes it much harder on analog output stages to handle. Consumer, and even some audiophile/high-end, analog outputs just aren't good enough.


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## castleofargh

bigshot said:


> Bias isn’t the only consideration. How well you’ve isolated the variable you’re testing for is just as important.


Yes, that's why I say it's my cognitive bias to favor controlled test even before I know how well it was setup. A better man would only look at each test independently without a priori.


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## bigshot

Yeah. Sometimes it’s best to break a question down into several tests leading to the hypothesis, rather than one with a bunch of variables all at play at once.


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## Ryokan

I bought a remastered cd from Amazon and got a free mp3 download which I listened to while waiting for the cd to arrive and after initially liking it switched to the original mp3 file (256kbps) which sounded better! Hoped the flac equivalent sounded better as this is a 25th anniversary remaster and it does thankfully. So this isn't that flac sounds better than mp3 but Amazon's software is obviously worse than mine - Jetaudio. Anyone buying just the mp3 file won't be getting a good quality rip.


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## gregorio

Ryokan said:


> So this isn't that flac sounds better than mp3 but Amazon's software is obviously worse than mine - Jetaudio. Anyone buying just the mp3 file won't be getting a good quality rip.


It all depends when the mp3 was created and by whom. It’s often quite difficult to verify the provenance of an MP3, they can sometimes be a transcode from a lower bit rate or created quite a while ago when encoders were of poorer quality. 

G


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## Ryokan (Sunday at 8:01 PM)

gregorio said:


> It all depends when the mp3 was created and by whom. It’s often quite difficult to verify the provenance of an MP3, they can sometimes be a transcode from a lower bit rate or created quite a while ago when encoders were of poorer quality.
> 
> G



I thought buying an mp3 download from Amazon would be the best quality. In future I'll try and buy the cd and rip it myself, they're often only slightly more expensive and sometimes cost the same as other download sites which is crazy.


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## gregorio

Ryokan said:


> I thought buying an mp3 download from Amazon would be the best quality.


Maybe it was when it was created (if it’s an old mp3). Maybe it was batch created by an intern using inappropriate settings or transcoded from another mp3 or other format. There’s no way to know for sure, sometimes even an mp3 provided by the record label is dodgy. 


Ryokan said:


> In future I'll try and buy the cd and rip it myself, they're often only slightly more expensive and sometimes cost the same as other download sites which is crazy.


That’s really the only way to be sure but even then, you need to make sure the CD is authentic and not a dodgy knock off. 

G


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## Ryokan

gregorio said:


> Maybe it was when it was created (if it’s an old mp3). Maybe it was batch created by an intern using inappropriate settings or transcoded from another mp3 or other format. There’s no way to know for sure, sometimes even an mp3 provided by the record label is dodgy.


In this instance it's an anniversary remaster 2021 I expected it to sound better than the original.


gregorio said:


> That’s really the only way to be sure but even then, you need to make sure the CD is authentic and not a dodgy knock off.
> 
> G



I mostly only buy cd's from reputable companies or the artists themselves, sometimes from second hand shops where the price is low enough not to worry.


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## bigshot (Monday at 2:01 AM)

Ryokan said:


> I thought buying an mp3 download from Amazon would be the best quality.


A lot of music has been mastered and remastered multiple times. Who knows which mastering they used when Amazon generated the MP3? They might use the same MP3s for all the releases of that album. Or the digital download might be mastered differently than the CD. Amazon uses MP3 256 VBR, which is a transparent format. So if it doesn't sound the way you want it to, it's probably due to mastering.


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## Ghoostknight (Monday at 2:07 AM)

bigshot said:


> A lot of music has been mastered and remastered multiple times. Who knows which mastering they used when Amazon generated the MP3? They might use the same MP3s for all the releases of that album. Or the digital download might be mastered differently than the CD. Amazon uses MP3 256 VBR, which is a transparent format. So if it doesn't sound the way you want it to, it's probably due to mastering.


well i would hope that if amazon offers mp3 that go with the CD that its actually the same mastering as the CD (preferably derived from the CD directly or from the original master of the CD) if not than this is nonsense by amazon

tho, isnt this easly checkable by audacity?


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## bigshot

I don't know if you can tell for sure with any download service if the lossy file is made right off the CD. If the files are supplied by the record label, they might have a specific kind of mastering for digital downloads that is different than for SACD/blu-ray or CD or radio play.


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## Ghoostknight (Monday at 2:25 AM)

what kind of different mastering are we talking about?

i just know that for example for streaming service the loudness level gets adjusted but everything else stays the same, atleast to my knowledge CD is always the "original" mastering and most of the time there is different LP Mastering, but not much beside that (well maybe a radio mastering too?)

its not like that for each album 10 different mastering exist, well volume adjusted ones for the different streaming services but not like "completely different" masterings like LP vs CD (and im excluding "remastered" here since its always very obvious to tell from the album title)

atleast my assumption would be that download portals also offer the normal "original" CD Mastering

Edit: but maybe amazon gets their mp3 from amazon music, which may be indeed a different mastering, atleast volume-wise


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## bigshot

It's my understanding that there are multiple masterings for different venues and purposes.


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## Ryokan (Monday at 5:38 AM)

If you heard the mp3 version you may well be put off buying the vinyl or cd, had to turn the volume up a few steps which isn't an issue but it sounds muddy compared to the first release. Not many copies will be produced, nothing like main stream music so can't imagine why there would be multiple remasters, though I've just read that the independent record label was bought by Sony music.
Somethings gone amiss when the original sounds clearer when comparing both Vbr 256 files. When buying music on Bandcamp different formats are offered and they've always sounded the same (flac and mp3 anyway). Haven't noticed this before after buying many cds from Amazon hopefully it's a one off.

Just checked on Discogs and there are multiple listings for the new release from different labels. Also the original which was released over 20 years ago wasn't clear sounding (which some say adds to its charm and actually dislike the remaster) so maybe Amazon's mp3 file is the original with the new album art? though it comes with new bonus material so that doesn't fit.


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## bigshot

Try comparing the new material and see if that is poorer sounding too.


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## Ryokan

bigshot said:


> Try comparing the new material and see if that is poorer sounding too.



Both cds (album + live recordings) sound great as you'd expect with a re-master, just the mp3 offering of the same release sounds compressed? muddy by comparison.


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## bigshot

So if it’s the case with the new material too, they aren’t using an older copy of the album itself.


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## gregorio (Tuesday at 9:43 AM)

Ghoostknight said:


> what kind of different mastering are we talking about?


Good question and unfortunately there’s not a simple answer to this. There can be many masters, although this is relatively rare. Certainly in the past, there was often a “radio edit” and occasionally a different radio edit for certain different radio stations. The vinyl was often mastered differently to the digital version but not uncommonly they’re the same master. Sometimes separate masters are made for film and TV use. There can also be different lossy masters, say one for YouTube and one for Apple Music, with the YouTube version having significantly more compression due to higher loudness level specs. Even the CD version maybe mastered differently to the HD version, for example the label/distributor might want an audible difference between them to justify the different price points.

So, there could be just one master version or there could be a dozen, although in most cases it would be closer to one than a dozen.

G

Edit: Not uncommonly, the mastering engineer doesn’t provide a lossy version of the master, in which case some intern at the record label might or the distributor might.


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