# 24bit vs 16bit: How big is the difference?



## Brian loves music

Searched the forums and couldn't find a thread addressing this question. With my soon ariving emu 0404 usb - HD595 combo, I'm considering buying a few of my favorite albums (sigur ros's ( ) and radiohead's Kid A) off music giants in 24bit HD format. 

 Just wondering how big the difference is? Would it take better gear to notice it (say hd650s or something)? Has anyone compared the two? (not looking for technical specs like the increased range of 24 bit flacs but rather whether 24bit adds a considerable amount of enjoyment to everday listening)

 Anyone have experience switching over to 24 bit? How does this compare to the difference between say 320kbps (or less) and flac.

 Thank you!


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## krmathis

There are an audible difference between 24-bit/96kHz and 16-bit/44.1kHz to my ears. But it might not be audible for every person on all kind of gear.

 I suggest you give it a try yourself.
 You find some free 24-bit/96kHz samples at High Definition Tape Transfers.


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## mwofsi

Someone posted a comparison where the same track could be downloaded recorded in both 16/44 and 24/96 bit, (sorry I don't have the link).

 I can clearly hear the difference, finding the 24 far superior in detail and musicality, and that's with my Beyer231 which aren't particularly revealing.


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## HFat

... and that's a bogus test. You need to convert the sample yourself (and volume-match if necessary). Of course, it also needs to be a blind test.

 But yeah, do check the old threads.
 24-bit is mainly useful for recording... and possibly to playback extremely dynamic music real loud.


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## hciman77

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_... and that's a bogus test. You need to convert the sample yourself (and volume-match if necessary). Of course, it also needs to be a blind test.
_

 

X2

 The differences (very very slight) I noticed (in a blind test, I scored 13/15) between a 24/96 sample and a 16/44.1 sample downloaded from the same site vanished when I downsampled the 24/96 myself (score exactly 50%). I just did it again - same 50% score - this time through a better USB external soundcard outputting an optical digital to an external DAC.

 Ooops - my Entech DAC is only rated at 48K so I had set the sound card to 44.1 yesterday , I redid the test this time using the sound card at 24/96 but this time using the analog outs to my M^3 and Sennheiser HD580s - still the same result.

 Annoyingly I cant do a blind test of 24/96 vs 24/44 of the sound card itself as you have to physically set dip switches to change mode then re-initialise the card.

 Fwiw when I analysed the original 24/96 and 16/44.1 files in an audio package there were obvious (though admittedly small) visible differences in the waveforms these were far less apparent after downsampling.

 YMMV of course.


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## pompon

Try

Linn Records - specialists in Classical, Jazz and Celtic music

 You can try 24/44 and 88/192 on samples.


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## Tarkovsky

24 bit is very noticeable as it's a higher dynamic range. With a bit of volume you'd have to be deaf not to notice that. This means more space for bass beneath 40hz, which is difficult to achieve on CD because lower frequencies need larger amplitude for the same level of power (not to mention to achieve the same perceived level). The sampling is a bit different - I hear it, but I'm 19 and can hear beyond 20khz. Seeing as most music is recorded @ 192khz I think it's pretty sensible as you're then not loosing any detail and suffering any conversions.


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## hciman77

Quote:


  Originally Posted by *pompon* /img/forum/go_quote.gif 
_Try

Linn Records - specialists in Classical, Jazz and Celtic music

 You can try 24/44 and 88/192 on samples._

 

I downloaded a 24/88 and a 16/44.1 test sample from Linn of the same track - they were audibly indistinguishable to me in a blind test - but in my audio software the files were slightly different - the high res file was slightly shorter - might not make a difference. Once you adjust for this the waveforms were pretty damn identical - of course the high res recording has energy above 22K, not relevant for my 49 year old ears.

 Very nice recordings however.


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## Davesrose

Quote:


  Originally Posted by *Brian loves music* /img/forum/go_quote.gif 
_Searched the forums and couldn't find a thread addressing this question. With my soon ariving emu 0404 usb - HD595 combo, I'm considering buying a few of my favorite albums (sigur ros's ( ) and radiohead's Kid A) off music giants in 24bit HD format. 
_

 


 If you like modern rock then I'd say 24bit recordings aren't worth it. The few rock SACDs I've bought have not been that much different from CDs. Beck's Sea Change is nice on SACD, but all his other albums are CD (and through my DAC sound about as good).

 The areas that I find SACD sounds great at are DSD recorded material (modern classical) and older recordings where they didn't compress the bagebers out off the masters. So far, that tends to be classical and jazz as well.....haven't really found a rock SACD album that impresses me over CD. If a SACD is coming from a good master, I find it has a better soundstage and nicer detail over CD....not that CD is bad or anything. I have a lot of CDs that blow me away because of the performance and engineering that went into it.

 And my opinion about upsampling 16bit recordings to 24bit is that it doesn't inately do anything. The D/A might make things sound more pleasant or vibrant, and that's an effect of the processor itself: not with it being 16 or 24 bit.


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## facelvega

x2 on what the others have said so far. I'd worry more about SNR and dynamic range ratings than bits, and more about how one thing sounds against another than any of the measurements.


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## hciman77

Quote:


  Originally Posted by *Tarkovsky* /img/forum/go_quote.gif 
_24 bit is very noticeable as it's a higher dynamic range._

 

Except that I doubt that you will find a recording that has even a 93db dynamic range let alone the 120 - 140db that high res formats can have.

 You should try blind tests - FooBar 2000 lets you do these quite easily, you may be surprised at how hard it is to tell 24 bits from 16 bits.

  Quote:


 With a bit of volume you'd have to be deaf not to notice that. 
 

Meyer and Moran (2007) in a peer reviewed journal paper published by the AES concluded that nobody (in over 500 trials) could reliably discern a difference between high res and 16/44.1 - except that you could detect the higher noise floor when some music was played really loud. 


 See also : 

 Which Bandwidth is Necessary for Optimal Sound Transmission? 
 Plenge, G. H .; Jakubowski, H.; Schöne, P. 

 Sampling-Frequency Considerations in Digital Audio
 TERUO MURAOKA, YOSHlHlKO YAMADA, AND MASAMI YAMAZAKI
 (I have copy of this)


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## mwofsi

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_... and that's a bogus test. You need to convert the sample yourself (and volume-match if necessary)._

 

You have a point in that I wasn't switching between playback conversion modes, but these particular samples were indicated as being a rare instance of the possibility of direct comparison, in that it's uncommon for both versions to be carefully matched and recorded at the same time in both modes unless it had been done for comparative purposes as these had. There is an obvious difference, but perhaps I approached it incorrectly. 
	

	
	
		
		

		
			





  Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_ Of course, it also needs to be a blind test._

 

Only if I'm trying to sell you something. I just felt my experience was relevant to the op.


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## Brian loves music

are sacd-dts files, which i've been hearing about, in 24/196? how do you play them anyone know?


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## HFat

Quote:


  Originally Posted by *mwofsi* /img/forum/go_quote.gif 
_a rare instance of the possibility of direct comparison, in that it's uncommon for both versions to be carefully matched and recorded at the same time in both modes_

 

What does it matter that they've been recorded in both modes? All you need is a "high-res" sample: you can convert it.
 That would explain why the samples are different: they've been recorded with different gear (maybe even different mikes).

  Quote:


  Originally Posted by *mwofsi* /img/forum/go_quote.gif 
_Only if I'm trying to sell you something. I just felt my experience was relevant to the op. 
	

	
	
		
		

		
		
	


	


_

 

What does blind testing have to do with selling something?
 Blind tests aren't for convincing people (you could lie about the tests), they're for you: you know what your experience was like but you don't know what it is you're experienced until you've gone through the routine. The tests tell you very little, but the little they tell you is rather critical and quite useful in practice. You may also be able to apply what you've learned about your perception during the blind tests to sighted comparisons. Hell, I even apply it to other people's descriptions of what they hear.


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## Happy Camper

24 bit is beneficial to the recording engineer because it gives more dynamic headroom. Once mastered to 16 bit, increasing the bit length is more for DAC needs in processing information and reducing jitter. http://www.cambridgeaudio.com/assets...perwebedit.pdf

 As I understand it. Of course I have also mis-understood lots so I hope others would drop in for a more knowledgable answer.


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## lan

The difference is 8 
	

	
	
		
		

		
			





 You need to have good gear and ears to tell the difference. Nobody but yourself can tell you what you are experiencing.

 I personally like the extra nuances provided by the 24bit experience. Not everybody priorities such added detail though.


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## XaNE

Quote:


  Originally Posted by *krmathis* /img/forum/go_quote.gif 
_There are an audible difference between 24-bit/96kHz and 16-bit/44.1kHz to my ears. But it might not be audible for every person on all kind of gear.

 I suggest you give it a try yourself.
 You find some free 24-bit/96kHz samples at High Definition Tape Transfers._

 

dam thats really high quality 100mb for just a five minute song 
 sounds excellent 
	

	
	
		
		

		
		
	


	







 i can hear a really good diffrence the sound is soo sharp


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## mwofsi

Here is the source of the comparison tracks I listened to

Soundkeeper Recordings Format Comparison

 As pointed out as a possibility in the article, I have no idea as to what kind of resampling etc my system might be performing.

  Quote:


  Originally Posted by *Joeywhat* /img/forum/go_quote.gif 
_The problem with comparing the two is that one of them will have had been converted. Typically if you have a 24/96 recording, the 16/44 will have been converted from the higher rate. I do not know of anywhere that has true comparisons available. In order to do that one would have to simultaneously record into two separate sessions, one at 44.1 the other at 96...I do not know of any other way to do it.

 You can't just rerecord everything a second, separate, time since the live instruments may not be miked exactly the same, effects processors may have been changed, and many other factors._


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## powertoold

Wow, I tried to put the 24/96 flacs into my Rockbox Nano, and they work fine hehe


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## Brian loves music

where do you guys get your 24/96 versions of your songs? music giants is the only one that seems to have most popular indie rock...wish i had a invite to waffles/whatcd (though we probably shouldn't talk about those hehe).

 Is there a cheaper place than music giants? I'm not gonna go out and get a vinyl ripper...


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## Brian loves music

on second review, I'm not sure if music giants even offers 24/96 versions. 

 "The sound quality of this download is equal to a CD. It is a lossless digital file, unlike the MP3 which is compressed 7 times. The High Definition download is encoded in the Lossless WMA (Windows Media Audio) format. " 

 cd = 16bit if i'm not mistaken...guess i'll have to find another source...none of the other sites seems to have what i'm looking for (radiohead, animal collective, sigur ros, pink floyd, bjork, etc)

 any ideas? 

 thanks as usual guys


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## reprise

Just so everyone knows (as I'm sure most of you do,) Windows XP resamples everything to 48 KHz if you're not using kernel streaming or asio

 And Brian, you're looking for DVD-A discs


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## Brian loves music

Quote:


  Originally Posted by *reprise* /img/forum/go_quote.gif 
_Just so everyone knows (as I'm sure most of you do,) Windows XP resamples everything to 48 KHz if you're not using kernel streaming or asio

 And Brian, you're looking for DVD-A discs_

 

can i find a place that sell rips of these DVD-A's online?

 ya i'll be using winamp > asio > emu 0404 usb so i should be able to me use of 24 bit tracks.

 really wanna try them out, but can't find where to get them


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## hciman77

Quote:


  Originally Posted by *reprise* /img/forum/go_quote.gif 
_Just so everyone knows (as I'm sure most of you do,) Windows XP resamples everything to 48 KHz if you're not using kernel streaming or asio_

 

So it is likely that some of these _I hear a difference between 16/44.1 and 24/96 _are moot since you are by default getting 48K anyway, that is funny. My card does use ASIO.

 How do you get kernel streaming going in Winamp or FooBar and does it provide any advantages over ASIO ?

 What happens if you try and run them together ?


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## OverlordXenu

Quote:


  Originally Posted by *lan* /img/forum/go_quote.gif 
_The difference is 8 
	

	
	
		
		

		
		
	


	




 You need to have good gear and ears to tell the difference. Nobody but yourself can tell you what you are experiencing.

 I personally like the extra nuances provided by the 24bit experience. Not everybody priorities such added detail though._

 

So you need to be a 12 year old with thousands of dollars in equipment?

 Because, as we all know, our hearing degrades as we age...A lot.


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## HFat

Quote:


  Originally Posted by *reprise* /img/forum/go_quote.gif 
_Just so everyone knows (as I'm sure most of you do,) Windows XP resamples everything to 48 KHz if you're not using kernel streaming or asio_

 

Actually, it depends on your card/drivers.

 @OP: be wary of the advice of people who can't get the basic facts straight.


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## grawk

bit depth doesn't provide nuance. It provides dynamic range. 16bit vs 24 bit on most things (assuming the mastering engineer isn't aiming for FM) won't make any difference at all. 16bits provides 96db of signal to noise. Very few things in the signal chain can handle that, let alone more than that. 24 bits is handy when recording, because you can set your levels lower, and then set the loudest peak at 0db in post. 16 bits covers greater than the difference between normal conversation and the concord taking off.


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## HFat

Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_How do you get kernel streaming going in Winamp or FooBar and does it provide any advantages over ASIO ?_

 

Unsurprisingly, through plugins.

 Depending on your setup, it may provide some advantages (and disadvantages). For example, in my case, KS is buffered in a way that ASIO isn't (which has its pros and cons) and wastes more CPU cycles.


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## hciman77

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_Unsurprisingly, through plugins.

 Depending on your setup, it may provide some advantages (and disadvantages). For example, in my case, KS is buffered in a way that ASIO isn't (which has its pros and cons) and wastes more CPU cycles._

 

I just wasted an hour installing the Winamp ASIO plug-in - which on refelection might be redundant I think since my card as far as I can tell intercepts winamps output and does its own ASIO thing ?. My control panel shows the card using ASIO drivers, hmmm.

 Lets just say it caused a lot of problems with choppy audio and not just with winamp and I had to wipe all evidence of it, winamp, and re-install the card and which is now working fine again. If it aint broke dont fix it


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## HFat

I don't understand what you mean... ASIO is a software thing that sits between your apps and your soundcard.
 EDIT: you may be misreading the panel or it may be misreporting... or the drivers might indeed be ASIOing everything (why?)

 But yeah, depending on your card/drivers, DS is going to do a good job for the most part.


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## lan

Quote:


  Originally Posted by *OverlordXenu* /img/forum/go_quote.gif 
_So you need to be a 12 year old with thousands of dollars in equipment?

 Because, as we all know, our hearing degrades as we age...A lot._

 

You just need experience with your ears and various gear to formulate your own definition of good.

 $ says nothing about the quality of a component and certainly says nothing of system setup.

 Hearing (the tool) doesn't say anything about it's ability (the use of).

  Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_@OP: be wary of the advice of people who can't get the basic facts straight._

 

It doesn't matter what anybody else says. It's up to the original poster to determine things for themselves.


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## HFat

Quote:


  Originally Posted by *lan* /img/forum/go_quote.gif 
_It doesn't matter what anybody else says. It's up to the original poster to determine things for themselves._

 

So we should all shut up? Or copy&paste from Alice in wonderland?


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## hciman77

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_I don't understand what you mean... ASIO is a software thing that sits between your apps and your soundcard.
 EDIT: you may be misreading the panel or it may be misreporting... or the drivers might indeed be ASIOing everything (why?)

 But yeah, depending on your card/drivers, DS is going to do a good job for the most part._

 

Sorry, this is what I mean, the card uses ASIO by default if you install it with an advanced driver dipswitch set, but you can switch it out. My query was if it is using ASIO anyway is a plug-in required for the app ? - certainly when I installed the asio plug-in for winamp it really knackered the output, when I took it out it is back to working perfectly.


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## HFat

It isn't reporting anything then: this is just a setting.
 My understanding is that this controls whether the analog inputs are routed directly to the outputs and that this can be controlled in some software as well. There may even be a switch on your card... there's one on mine and I've always used that instead of the control panel so I'm not absolutely sure I got it right.

 I never bothered to use ASIO from winamp, only from foobar and it works well for me except for the odd buffering issue (see one of the few threads I started).

 EDIT: for clarity, it's not the card that uses ASIO... it (and its "advanced" driver) only support it. A plugin will be necessary if the application doesn't have ASIO support out of the box.


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## hciman77

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_It isn't reporting anything then: this is just a setting.
 My understanding is that this controls whether the analog inputs are routed directly to the outputs and that this can be controlled in some software as well. There may even be a switch on your card... there's one on mine and I've always used that instead of the control panel so I'm not absolutely sure I got it right.

 I never bothered to use ASIO from winamp, only from foobar and it works well for me except for the odd buffering issue (see one of the few threads I started).

 EDIT: for clarity, it's not the card that uses ASIO... it (and its "advanced" driver) only support it. A plugin will be necessary if the application doesn't have ASIO support out of the box._

 


 Thanks for the clarification. I installed the FooBar ASIO plug-in and it works fine - no drop-outs, the Winamp ASIO plug-in was seriously problematical with my set-up.

 Is yours the UA-1EX ?


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## Brian loves music

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_bit depth doesn't provide nuance. It provides dynamic range. 16bit vs 24 bit on most things (assuming the mastering engineer isn't aiming for FM) won't make any difference at all. 16bits provides 96db of signal to noise. Very few things in the signal chain can handle that, let alone more than that. 24 bits is handy when recording, because you can set your levels lower, and then set the loudest peak at 0db in post. 16 bits covers greater than the difference between normal conversation and the concord taking off._

 


 you've pretty much convinced me to drop this whole 24bit thing...you're argument makes sense because bit effects the length not detail of the source.

 It's still interesting that people write stuff like this:

 "Not at all. They simply need to turn it up! Seriously though, low-level listening will not allow you to easily hear the benefits of additional resolution of low level signals. The best way to perceive all of the benefits of a 24-bit recording is to listen as loud as possible without damaging your hearing. For concert recordings, this would be considered concert level (about 100-105dB maximum). When most people say it is hard to hear the difference between 16 and 24-bit recordings, they are referring to one of three things. One possibility is they either they don’t know what to listen for or how to listen for it. Another possibility is the fact that a high quality playback system is essential, and your average car stereo or boombox just won’t cut it. Finally, difficulty in perceiving the difference has also been said in reference to 24-bit recordings that are cut down to 16 bits for CD’s before listening, and this is only partially true, as performing processes at 24-bits before converting to 16 bits as a last step before listening does yield a more accurate recording.

 The differences to expect are greater realism and more accurate portrayal of the source event. Not so ironically, it may take several listenings, and even a variety of source material before the differences are recognized. We have had the enjoyable opportunity to sit with many listeners and observe their reactions. Sometimes the recording of a rock ‘n’ roll event, a well-defined bass line, a stunning cymbal shot followed by persistent chimes, or the feeling of being back in the venue that is the kicker that turns them on. For others, it is the dog barking, the door bell ringing or the car we have recorded that really moves them. Either way, with a few listenings, everyone soon hears the impact that 24-bit recording delivers over its 16-bit counterpart."


 ^from The 24-bit Field Recording FAQ - November 3, 2001

 what are people's thoughts?


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## grawk

24 bits makes a huge difference in recording. It lets you avoid the use of a limiter or compressor, and lets you get the maximum SNR possible. 

 Higher resolution (44/48/88/96/176/192) is a subtle and potentially significant improvement in playback. I have no comment as to whether or not you'll hear a benefit there, that's a LOT more dependent on playback signal chain and your own hearing. But given a competent engineer and a 24 bit master, a 16 bit file for playback (cd, etc) is so far beyond adequate as to be moot.


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## lan

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_So we should all shut up? Or copy&paste from Alice in wonderland?_

 

Everybody can talk about whatever they wish but at the end of the day, your own ears and experiences will dictate what direction you'll be going in. Unless you are just convinced by words.


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## hciman77

Quote:


  Originally Posted by *Brian loves music* /img/forum/go_quote.gif 
_"Not at all. They simply need to turn it up! Seriously though, low-level listening will not allow you to easily hear the benefits of additional resolution of low level signals. The best way to perceive all of the benefits of a 24-bit recording is to listen as loud as possible without damaging your hearing. For concert recordings, this would be considered concert level (about 100-105dB maximum). When most people say it is hard to hear the difference between 16 and 24-bit recordings, they are referring to one of three things. One possibility is they either they don’t know what to listen for or how to listen for it. Another possibility is the fact that a high quality playback system is essential, and your average car stereo or boombox just won’t cut it. Finally, difficulty in perceiving the difference has also been said in reference to 24-bit recordings that are cut down to 16 bits for CD’s before listening, and this is only partially true, as performing processes at 24-bits before converting to 16 bits as a last step before listening does yield a more accurate recording.

 The differences to expect are greater realism and more accurate portrayal of the source event. Not so ironically, it may take several listenings, and even a variety of source material before the differences are recognized. We have had the enjoyable opportunity to sit with many listeners and observe their reactions. Sometimes the recording of a rock ‘n’ roll event, a well-defined bass line, a stunning cymbal shot followed by persistent chimes, or the feeling of being back in the venue that is the kicker that turns them on. For others, it is the dog barking, the door bell ringing or the car we have recorded that really moves them. Either way, with a few listenings, everyone soon hears the impact that 24-bit recording delivers over its 16-bit counterpart."


 ^from The 24-bit Field Recording FAQ - November 3, 2001

 what are people's thoughts?_

 

Well, if you need to play stuff at 100db to notice the difference - count me out, even 100db is loud and here is a quote from the NIH Noise-Induced Hearing Loss

_Long or repeated exposure to sounds at or above 85 decibels can cause hearing loss. The louder the sound, the shorter the time period before NIHL can occur. Sounds of less than 75 decibels, even after long exposure, are unlikely to cause hearing loss. _

 Before you argue about one sound being better than another you can ask, is there actually a difference, so far the best experimental evidence 

_Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback E. Brad Meyer and David R. Moran, 2007_

 is that when you properly downsample High res audio to 16/44.1 that the difference is inaudible except at really high volume levels where the noise floor comes into play. 

 The downsampling step seems to be the key. I downsampled a few 24/96 files and 16/44.1 files and if you dont do it right you get differences that should not be there. For instance when I converted a 24 bit wav to 16/44.1 with dither it altered the high frequency response, it smoothed it out so it rolled off gently , taking dither off kept the same response which had a sharp drop-off on the original 16/44.1 wav.


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## sejarzo

Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_Well, if you need to play stuff at 100db to notice the difference - count me out, even 100db is loud and here is a quote from the NIH Noise-Induced Hearing Loss

Long or repeated exposure to sounds at or above 85 decibels can cause hearing loss. The louder the sound, the shorter the time period before NIHL can occur. Sounds of less than 75 decibels, even after long exposure, are unlikely to cause hearing loss. 
_

 

My SPL meter shows that well-recorded orchestral music played back with peaks in the 100 to 105 dB SPL range probably averages 75 to 80 dB SPL. Note that the post above referred to "max" levels, and I take that to mean peak SPL's.


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## bigshot

Quote:


  Originally Posted by *Brian loves music* /img/forum/go_quote.gif 
_"Not at all. They simply need to turn it up! Seriously though, low-level listening will not allow you to easily hear the benefits of additional resolution of low level signals. The best way to perceive all of the benefits of a 24-bit recording is to listen as loud as possible without damaging your hearing. For concert recordings, this would be considered concert level (about 100-105dB maximum)._

 

If you have to turn the volume up to the edge of hearing damage to hear a difference, perhaps the difference doesn't matter.

 See ya
 Steve


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## bigshot

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_24 bits makes a huge difference in recording. It lets you avoid the use of a limiter or compressor, and lets you get the maximum SNR possible._

 

Actually it doesn't eliminate the need for compression, it makes it easier to compress, because the stuff you're pulling up is cleaner. Compression isn't a bad thing necessarily... it's a tool that can be applied well, or abused. It all depends on how it's used.

 24 bit recording is great for mixing. It gives you a lot of flexibility. But at normal listening volumes, it makes no difference at all.

 See ya
 Steve


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## Brian loves music

hey, i posted this in the music section i didn't get any responses.

 where do you guys get your 24 bit music (if you get any at all). Does music giants provide 24 bit or just uncompressed 16 bit flacs? That's the only website i've found that has indie rock. ANyone know where i can get a 24bit download of sigur ros (I'll pay of course).

 thanks


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## cconnaker

The format by itself makes no difference at all. It is the whole recording and mastering process that really matters. Some 24/96 files sound no different than their 16/44 counterparts because they were not done very well to begin with. Some SACDs actually use a 16/48 master. Garbage in to a 24 bit / 192 recording won't sound good.

 As far as an ABX test with a 16 and a 24 song it is really tough because often it is music you are unfamiliar with. If you are able to use the music you know extremely well and listen to it over time, then a personal conclusion could be made. Everyone is of course different.


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## HFat

Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_Is yours the UA-1EX ?_

 

No, I've got a UA-25 (bling bling).


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## HFat

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_My SPL meter shows that well-recorded orchestral music played back with peaks in the 100 to 105 dB SPL range probably averages 75 to 80 dB SPL. Note that the post above referred to "max" levels, and I take that to mean peak SPL's._

 

That would indeed make sense... except that it wouldn't be loud enough to hear the 16/24 difference, isn't it? From that perspective 100-105 averages would make more sense (note the reference to "rock and roll events").


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## hciman77

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_That would indeed make sense... except that it wouldn't be loud enough to hear the 16/24 difference, isn't it? From that perspective 100-105 averages would make more sense (note the reference to "rock and roll events")._

 

As I see it, I could be wrong, to tell the difference between 24 and 16 bits you have to have material that cannot be captured by 16 bits, i.e it must have a dynamic range of > 93db (real world not theoretical). In a pretty quiet room you will have background noise of, lets be optimistic, 25db. So to even stretch CD you have to be able to play back sound peaking at 118db. 

 All this of course is moot unless you have recordings with 93db+ dynamic range in the first place. Has anyone found any recordings with a real 93db + dynamic range ?


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## HFat

Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_As I see it, I could be wrong, to tell the difference between 24 and 16 bits you have to have material that cannot be captured by 16 bits, i.e it must have a dynamic range of > 93db (real world not theoretical)._

 

So you're counting half of the last bit? Why?

  Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_In a pretty quiet room you will have background noise of, lets be optimistic, 25db. So to even stretch CD you have to be able to play back sound peaking at 118db._

 

On the contrary, this is quite pessimistic. Even if there are louder noises, you can hear the somewhat quieter ones.
 I'm sure you've been annoyed by noises which would measure at less than 20dB. Based on other people's measurements of computer parts, I can easily hear a 20dB noise despite the nearby road and the other noises you'd expect to be saddled with in a regular building.

 You posted yourself a link to an experiment (BAS Experiment Explanation page - Oct 2007) where people have been able to ABX an SACD with its 16bit equivalent with a peak level of only 111dB. The report is not really clear about that but it seems the background noise was about 19dB. They apparently didn't even try to go as low as possible either given the result (15/15).

  Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_All this of course is moot unless you have recordings with 93db+ dynamic range in the first place. Has anyone found any recordings with a real 93db + dynamic range ?_

 

Again, see the experiment you posted. Did you not read what you linked to?

 In any case, I believe you are overstating your case.
 I agree with the thrust of your argument but I think it would be more honest to acknowledge that it's actually possible to hear the lower noise floor, albeit in a somewhat contrived test. If you also grant the point that ABX is hard, I think it's not that far-fetched to conclude that it may actually be possible to hear that noise floor (without necessarily being able to identify it reliably) at loud yet arguably reasonable levels provided you listen to extremely dynamic music (so-called experimental music maybe?).


----------



## grawk

Yes, it's possible to produce material that needs more than 16 bits worth of dynamic range. Realistically, no one listens loud enough to justify bothering. The benefit to 24 bits remains on the recording end, and not the final product end of things.


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## HFat

I think bigshot made a good point about mixing though. It would also apply to sampling isn't it?

 EDIT: I mean, what if you hear some noise in a recording and you want to know what it is? You could play it real loud to hear it clearly. Wouldn't it be nice to have 24bit for that? Like, to "pause and zoom in" to hear the quality of the paper when the conductor is turning the page.


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## bigshot

I think I'd rather hear the music the conductor is making than his baton tapping the podium.

 Maybe if we downsample to 4 bit, we won't hear Glen Gould's grunts!

 See ya
 Steve


----------



## Brian loves music

looks people pretty much agree that 24 vs 16 bit doesn't make a significant difference in everyday headphone music enjoyment then...I'm curious to see if many people who don't see too much of a difference in 16/24, think there is a significant audible difference in flac vs lossy audio files


----------



## cconnaker

Quote:


  Originally Posted by *Brian loves music* /img/forum/go_quote.gif 
_I'm curious to see if many people who don't see too much of a difference in 16/24, think there is a significant audible difference in flac vs lossy audio files_

 

Good question BLM. I would be very interested in that as well. I have a feeling that harsh responses and egos may prevent people from admitting they can't tell the difference between flac and lossy files. I could be wrong though.


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## bigshot

When you say lossy, you need to define the amount of compression. I can easily hear a difference with 128 and below, but 256 and above are the same as lossless for my use.

 See ya
 Steve


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## Crowbar

No difference.
AES E-Library: Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback by Meyer, E. Brad; Moran, David R.
 Very extensive blind testing study with hundreds of trials including golden ears and many others showed that 24/96 and 24/196 is useless; the only reason many recordings in the hi-rez formats seem to sound better is because they were mastered better, since they're intended for the audiophile crowd.


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## eruditass

bit depth doesnt matter except with turning down volume because of the sampling theorem.

 sampling rate on the other hand does matter. yes, more than 44.1khz, due to timing, not loss of frequencies. someone posted a link before.


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## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_No difference.
AES E-Library: Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback by Meyer, E. Brad; Moran, David R.
 Very extensive blind testing study with hundreds of trials including golden ears and many others showed that 24/96 and 24/196 is useless; the only reason many recordings in the hi-rez formats seem to sound better is because they were mastered better, since they're intended for the audiophile crowd._

 

That paper has already been brought into this discussion previously.

  Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_Meyer and Moran (2007) in a peer reviewed journal paper published by the AES concluded that nobody (in over 500 trials) could reliably discern a difference between high res and 16/44.1 - except that you could detect the higher noise floor when some music was played really loud._

 

 Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback E. Brad Meyer and David R. Moran, 2007_

 

Strictly speaking the paper doesn't really apply here as it is testing 24/96 or 24/192 against 16/44.1 not 16/44.1 vs 24/44.1. So you have two variables, bit-depth and sampling rate that are changed. To test whether bit-depth alone makes a difference you have to test with the same sample rate. Now it may seem intuitive to say that if 24/96 is indistinguishable from 16/44.1 that bit-depth doesnt matter but that is actually poor science since there may be an interaction effect between bit-depth and sample-rate. The way to do this is to do a set of tests varying both bit-depth and sample-rate and run an statistical analyses to see then if there is an interaction effect.


----------



## nick_charles

Quote:


  Originally Posted by *Brian loves music* /img/forum/go_quote.gif 
_looks people pretty much agree that 24 vs 16 bit doesn't make a significant difference in everyday headphone music enjoyment then...I'm curious to see if many people who don't see too much of a difference in 16/24, think there is a significant audible difference in flac vs lossy audio files_

 

 Quote:


  Originally Posted by *cconnaker* /img/forum/go_quote.gif 
_Good question BLM. I would be very interested in that as well. I have a feeling that harsh responses and egos may prevent people from admitting they can't tell the difference between flac and lossy files. I could be wrong though._

 

I will volunteer to answer that one. I did try testing some uber-res uncompressed files against prole-res uncompressed files. In an audio program they did look different , at least when you zoomed in really deeply. My personal audio processor, (ears and wetware) didnt detect a difference. However when it comes to WAV against high quality MP3 files the picture is entirely the same. With an average bit-rate of 240kbs I cant tell a difference, maybe it is the legacy of too many loud rock concerts in the 1970s... Ho hum.


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## Crowbar

There was testing around the turn of the century with the Orhpeus headphones and people failed to tell 216 kb/s MP3 from CD with even the then-poor quality MP3 encoders. With modern encoders, 192 should be enough for anyone.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_There was testing around the turn of the century with the Orhpeus headphones and people failed to tell 216 kb/s MP3 from CD with even the then-poor quality MP3 encoders. With modern encoders, 192 should be enough for anyone._

 

I remember that test. They used 256 and 128 vs CD and there was a lot of variability in performance 

c't 6/2000 cross-examination test - Mp3 vs CD


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## Crowbar

There generally is variability. Don't forget that if you test enough people, chances are that some will get it right most or even all of the time by blind luck alone.


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## gregorio

Wow, a lot of conflicting information, most of which is inaccurate.

 24bit vs 16bit. We use 24bit during recording simply to provide additional headroom for unforseen transient peaks. When mastered down to 16bit the dynamic range is still 96db. 24bit provides 144dB of dynaic range but is pointless for playback. There is no way anyone could tell a difference between 24bit and 16bit unless you are a golden ears listening in an anaechoic chamber to 60 grand monitors and even then there would be no point! If you think you can hear a difference this is purely down to the way your DAC is handling the reconstruction.

 Now on to the higher sample frequencies. Let me make this clear, there is not and cannot be any audible difference between a sample rate of 48kFs/s, 96kFs/s or 192kFs/s. No human being can hear above 22kHz so a sampling freq of 48kFs/s is more than adequate. So why was 96kFs/s invented? To con more money out of people who are understandably ignorant of digital audio theory. There is a technical reason for 96kFs/s, it makes the anti-alias filter on an ADC easier and cheaper to impliment. If you think that 96kFs/s sounds better than 44.1kFs/s then what you have proved is that the music was originally recorded using cheap ADCs.

 While I'm at it, Up-sampling. Forget it, it's another way to con money out of you. There is no way an upsampled recording can sound better than the original unless the manufacturers have deliberately created a DAC with a rubbish reconstruction filter at 44.1kFs/s.

 Hope this helps,

 Gregorio


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## frankR

I can certainly hear the difference in high res audio. My ears have been put to the test—and passed. I was always able to hear the difference between 16/44 "redbook" and SACD, in a variety of recordings, rock and jazz, from old analog masters, and direct DSD recordings. SACD is a good format for this test because the redbook track can be resample with high quality from the DSD master making A-B comparison under similar conditions possible through the same hardware, from recording and playback equipment.

 My father on the other-hand struggled, to his dismay, to consistently pick-out the high resolution material. The consolation for those who have money invested in high res. audio equipment is, they can perceive the difference, but for psychological reasons, they are unable to pass an A-B test, having nothing to do with technical reasons with the signal or hearing acuity.

 I must stress, hearing the difference very much depends on the quality of the recording and playback equipment. For my test it was on exceptional speakers, bi-amped VMPS RM40 with a high quality Marantz SACD player and an acoustically tuned listening room.

 The differences are very subtle, and mostly heard in high frequency tones with very complex harmonic structure. For example, someone a few pages back posted links which featured 16/44 and 24/96 tracks, see link below. I can pick-out the difference right away, from those opening guitar chords. The high res. material is capable of revealing the richness of the dense harmonic structure in the guitar, meaning the high res. material sounds more true to life.

Soundkeeper Recordings Format Comparison

 I believe most people when they quote the theoretical SNR and Nyquist frequency of digital audio, fall into a trap of thinking of music as being composed of well behaved pure tones (sinusoids). While in theory that is true, in reality, the harmonic structure, meaning the collective group of pure tones which describes the ‘tiber’ of that particular guitar, contains information that extends far beyond the frequency of that fundamental tone, which is determined by many factors, the wood species of the guitar body, the strings, the recording environment acoustics, ect. The superposition of all of these factors makes for a very complex signal, which is fractal in nature; meaning there are layers of higher frequency tones on top of the fundamental tone, that’s what makes subtle differences between guitars audible to a musician or critical listener. Image an ensemble of complex tones from instruments and vocals combining for an exceptionally complex signal.

 One tone that high resolution excels over lower res. audio is sibilance, which is a very complex high frequency tone made from the ‘s’ sound. I would bet most people could be able to pick-out the high res. track of sibilance, by virtue of their everyday familiarity of that sound. Any form of compression can’t coupe with sibilance, i.e. MP3. The visual analog of sibilance could be ocean waves, which the typical heavily compressed HD broadcast lacks the bandwidth to resolve this scene, and can only resolve but a few aliased blocks, especially during panning.

 I got into a heated discussion with my Fourier optics professor about this topic. I argued that human hearing does not extend much beyond 20 khz, but that many musically instruments do, and the inability to resolve those frequencies in lower res. audio results in distortion of the audible frequencies the ear does hear.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_No human being can hear above 22kHz so a sampling freq of 48kFs/s is more than adequate._

 

While direct conscious experience of it is not there, ultrasound present in audio does have an effect which influences conscious perception, according to peer reviewed studies:

 T. Oohashi, E. Nishina, M. Honda, Y. Yonekura, Y. Fuwamoto, N. Kawai, T. Maekawa, S. Nakamura, H. Fukuyama, and H. Shibasaki. *Inaudible high-frequency sounds affect brain activity: Hypersonic effect.* Journal of Neurophysiology, 83(6):3548–3558, 2000.

 At the same time, it's established that the effect bypasses the auditory system (if you dig in deeper, some research suggests that the path is bone conduction to the brain which demodulates the ultrasound; this is why the effect is minimized with headphones and speakers are needed to recreate it).

 Regardless, the point is that there IS an effect, and a perfect reproduction must, by definition, capture and reproduce it. Of course, while a 96 kHz or higher sampling rates are available, no recording actually contains anything this high because those are filtered during capture.

 In regards to bit depth, the ear does 120 dB from threshold of hearing to threshold of pain. Add to this margins for dynamic range, avoiding clipping due to inter-sample peaks post-reconstruction exceeding 0 dB, etc., 16 bit audio is definitely insufficient.


----------



## gregorio

Crowbar - you are forgetting about the noise floor, the extra dynamic range provided by 24bit will just give you an extra 8bits of noise floor. The only point of 24bit is to provide additional headroom during recording. You are not going to hear more than 96dB above the noise floor, let alone 144dB. Not to mention that most equipment in a recording chain (I'm talking high end professional gear here) cannot take advantage of 96dB (above the noise floor). So 16bit audio is definitely sufficient, unless of course you like listening to uncorrelated noise under music which is so loud that your ear drums would rupture! You said it yourself, 120dB is the threashold of pain, so why do you need a range of 144dB?

 There is evidence that high frequency sounds beyond human hearing do have an effect on brain activity but you need to use a relatively large amount of energy and there is no evidence to suggest that these frequencies are perceived as sound. Certainly most recording or playback equipment could not handle the energy levels at these frequencies to have any effect and most equipment couldn't reproduce those frequencies at any energy level. So while this research is mildly interesting it has no baring on consumer audio or for that matter professional audio!

 FrankR - Those tones you are refering to are called harmonics and it is the balance of those harmonics in relation to the fundamental which gives a sound or instrument it's timbre. However, once those harmonics exceed 20kHz (or about 17kHz when you're my age) they cannot be heard. There is nothing in a 96kFs/s recording (which can be heard) that cannot be captured in a 44.1kFs/s recording. There are no exceptions to this rule, it doesn't matter what equipment you are using, your listening environment, nothing. If you are hearing a difference, what you are hearing is the implimentation of the various different reconstruction filters and components in your DAC or possibly weaknesses in the anti-alias filters in the original ADC, not a difference in the audio file sample frequency. BTW, sibilance is found mainly in the 2kHz to 4kHz range, not the high frequencies.

 Your analogy with HD video is not valid because neither HD nor SD video resolutions exceed the capability of the human eye, whereas even 16bit, 44.1kFs/s exceeds the capability of the human ear!

 Gregorio


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Not to mention that most equipment in a recording chain (I'm talking high end professional gear here) cannot take advantage of 96dB (above the noise floor)._

 

So? That's a fault of the recording process. The fact remains that 96 dB is a subset of the audible dynamic range. That the common equipment and listening conditions are often inadequate simply says that they need to be improved. Clearly we need significant improvements in the recording process. As for the listening conditions, I've seen quite a few extreme room treatments which, while failing far short of anechoic chambers, result in much improvement in the higher frequencies.

  Quote:


 so why do you need a range of 144dB? 
 

You don't; you need 120 dB.
 By the way, the ear can detect signals several dB below the noise floor if they are narrower in bandwidth than the noise (which is generally the case).

  Quote:


 there is no evidence to suggest that these frequencies are perceived as sound. 
 

That is also false.
 Lenhardt, M. (2003). "Ultrasonic hearing in humans: applications for tinnitus treatment". Int. Tinnitus J. 9 (2): 69-75.
 From the abstract:
 "Masking of tinnitus is possible using high audio frequencies and *low-frequency ultrasound*. The *mechanisms* involved in reception and perception of both audio frequencies and ultrasound *are identical*"
 Moreover,
 "*speech discrimination* in deaf subjects was possible using *modulated ultrasound*. More recently, the primary auditory cortex has been implicated, using magnoelectroencephalography, in *processing ultrasound tonotopically, consistent with very-high-frequency hearing*"
 It is an audible effect that is being discussed here. This goes beyond the initial citation, which establishes that there is some effect on perception--which is important even if it were not directly audible, because it contributes to the overall experience created by sound. Thus, even if it weren't for the actually audible effect discussed here, if it affects perception as the first study mentioned established, it *must* be included to have a complete reproduction.

  Quote:


 Certainly most recording or playback equipment could not handle the energy levels at these frequencies to have any effect and most equipment couldn't reproduce those frequencies at any energy level. So while this research is mildly interesting it has no baring on consumer audio or for that matter professional audio! 
 

To paraphrase your argument: most current equipment is not good enough to capture and reproduce this, so we should forget about it.
 What the hell? The whole point of this site is the advancement of audio realism. I don't even know what the hell you are doing here, where people are striving to go beyond the restrictions of mere "consumer audio".

 There's yet another thing that gregorio here is missing, and that is the fact that the ear does both frequency- and time-domain analysis, and research has been mentioned on the prosoundweb.com forums that the latter can perceive differences in timing of transients that exceeds the 20 kHz limit of their Fourier analogs.


----------



## bigshot

If you want harmonics to be accurately presented, it's a lot more important to make sure that the frequencies you *can* hear are balanced than it is to worry about the ones you *can't* hear. See the psychoacoustic principle known as the "masking effect".

 By the way gregorio, when you say "We use 24bit during recording simply to provide additional headroom for unforseen transient peaks." you might just confuse people. Peak level is peak level, regardless of the bit depth. The added dynamics in 24 bit sound extends below the range of redbook, not above it.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_The fact remains that 96 dB is a subset of the audible dynamic range._

 

85dB is the highest safe volume to listen to music. Anything above that can cause hearing damage, depending on the length of exposure.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_As for the listening conditions, I've seen quite a few extreme room treatments which, while failing far short of anechoic chambers, result in much improvement in the higher frequencies._

 

You're missing the point of room treatment. It isn't to extend the range of the frequency response into super and sub-audible frequencies, it's to balance the levels of the frequencies you *can* hear. That's a common misconception. As I said in the previous post, the problem is the relative balance of frequencies. If there is an imbalance in one frequency, it can cancel out the frequencies an octave above. This is called the "masking effect". Masking is a much more serious issue than inaudible frequencies.

 See ya
 Steve


----------



## frankR

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Your analogy with HD video is not valid because neither HD nor SD video resolutions exceed the capability of the human eye, whereas even 16bit, 44.1kFs/s exceeds the capability of the human ear!_

 

Let me just be clear, your theory is on solid ground. However, if your statement that 16/44 exceeds the capability of the human ear, then listeners would not perceive high resolution audio as sounding better. Your contention is that these differences are coming from arifacts in the DAC?

 Do you believe standard 16/44 CD audio perfectly records the acoustic waveform as measured by a microphone?


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_85dB is the highest safe volume to listen to music. Anything above that can cause hearing damage, depending on the length of exposure._

 

It's not continuous! it's about allowing full peaks for large transients. Classical orchestras can, and often do, exceed 110 dB.

 The point of room treatment is to reduce reflections, which ruin a cohesive spatial image. However, I'm talking about the total surface room treatment some people do, which includes significant attenuation of outside noise of the mid and high frequencies.


----------



## Crowbar

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Let me just be clear, your theory is on solid ground._

 

Are you on drugs? All his arguments are about practical issues. The theory is clear: the ear has a 120 dB dynamic range, and, ultrasound causes both audible and other mental influences. 16/44.1 cannot capture these even in a perfect practical implementation.

 BTW, good things coming up on the recording end of the chain: Georg Neumann GmbH - Products/Current Microphones//
 130 dB microphone with A/D in the capsule. 28 bit output, 96 kHz. Set the anti-alias filter with -3 dB at around 40 kHz and we have something resembling what should be used to create proper quality recordings.


----------



## frankR

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_There's yet another thing that gregorio here is missing, and that is the fact that the ear does both frequency- and time-domain analysis, and research has been mentioned on the prosoundweb.com forums that the latter can perceive differences in timing of transients that exceeds the 20 kHz limit of their Fourier analogs._

 

No, the human ear and brain is far more capable then gregorio is claiming.

 I don't think you can think of the human ear and brain as simply as an electrical engineer would look at a machine.

 How about human sonar?
YouTube - Amazing Blind boy see's using radar!


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_While direct conscious experience of it is not there, ultrasound present in audio does have an effect which influences conscious perception, according to peer reviewed studies:

 T. Oohashi, E. Nishina, M. Honda, Y. Yonekura, Y. Fuwamoto, N. Kawai, T. Maekawa, S. Nakamura, H. Fukuyama, and H. Shibasaki. *Inaudible high-frequency sounds affect brain activity: Hypersonic effect.* Journal of Neurophysiology, 83(6):3548–3558, 2000._

 

The Oohashi study is interesting but has never been replicated and has been challenged by a few researchers, Kaoru and Shogo (2001) "Detection threshold for tones above 22 kHz", tried to replicate Oohashi's findings and could not, but did find effects caused by the super high frequency sounds interacting with normal frequency sounds and causing IMD in the audible band. So the Oohashi effect may not be a function of the musical frequencies as such but simple distortion. Which of course goes away when you have just the super-high frequencies or the others...


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## Crowbar

That study is supported by references cited in the second document I quoted.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It's not continuous! it's about allowing full peaks for large transients. Classical orchestras can, and often do, exceed 110 dB._

 

Do you know what 110dB sounds like? You can't even shout over that and hear yourself. You sure don't want that in your living room, even for short bursts. (Well, maybe you do, but your neighbors sure don't!)

 It's important to understand what the numbers really mean. The 96dB range of redbook is more than anyone ever uses in a home situation. And 96dB is much more than you would ever hear in a classical concert hall unless you're sitting on the stage facing the bell of the horns.

 By the way, the only studies I'm aware of dealing with ultra high frequencies say that it is possible to perceive high frequency sound, but it makes absolutely no difference to music. In fact, I read one study where they filtered off everything above 10kHz and asked people to compare it to the full range track. People didn't feel that either of the tracks sounded better than the other. The most important frequencies in music are the ones smack dab in the middle. And getting those clean and balanced isn't as easy as some people think.

 See ya
 Steve


----------



## frankR

Some are misunderstanding the concept of dynamic range, which is measure of the theoretical signal-to-noise ratio (SNR).

 When someone quotes 145 dB as the dynamic range for 24-bit audio that has nothing to do with how loud it is. How loud it is is determined only by how powerful your amplifier is.

 For example, take the following 1-bit signal, composed of a trail of 01010101010101010101010101010… sampled at 1 khz. That describes a really nasty sounding 1 khz saw-tooth wave. I can play that signal at whatever volume your audio playback equipment can handle by turning up the knob.

 What dynamic range really describes is how finely divided the numbers are that encode that sound which is a continuous analog wave, like a sinewave. For a 2-bit signal, the wave form can be represented by whole numbers from 1 to 4, because 2 squared is equal to 4. For 16-bit it’s 2^16, or 65536. As the bit-depth increases from 1 to 16, the sharp saw-tooth wave begins to represent that continuous sinewave. To understand how dynamic range is measured you must understand what the signal-to-noise ratio is.

 As the name implies, signal-to-noise ratio is simply the signal divided by the noise. A common way engineers express SNR is on the decibel scale, in units abbreviated dB, which is a convenient scale for comparing a signal to a given reference, i.e. signal compared to noise.

 The way this signal-to-noise or dynamic range (DR) is calculated for digital audio is as follows:

 DR = 20*log10(A_signal/A_noise)

 A_signal is the largest quantization size. For 16-bit audio this is equal to 2 to the power of 16, or 65536.

 A_noise in this case is the smallest quantization size, or simply 1.

 So you end up with a simple formula like this:

 DR=20*log10(2^bd), where bd is the bit depth, i.e. 16-bit or 24-bit.

 Having more DR means dynamic instruments that produce high intensity sound like drums or cymbols should be able to reproduce those sharp pulses and quieter sounds like vocals, at the same time, assuming the DR isn’t intentionally compressed during the recording process, which is most likely the case.


----------



## gregorio

Ahh Bigshot, someone who know what he's talking about, thanks for helping out!

 Frankr - you seem to think that just because I know a bit about the theory that's all I know. I was trained as a classical musician and was a professional orchestral musician for quite a few years. I agree with your explanation of SNR and bit resolution but you haven't explained why you think 16bit 44.1kFs/s is not capable of of perfectly reproducing a waveform. I would like to hear your proof, which flies in the face of 25 years of proof and implementation of digital theory.

 "if your statement that 16/44 exceeds the capability of the human ear, then listeners would not perceive high resolution audio as sounding better". - You are entirely missing the point. You are hearing an improvement at 24/96, so you've come to the logical but incorrect conclusion that 24/96 is better. What you are hearing is a smoother, better implimented reconstruction filter in your DAC, meaning that the reconstruction filter in your DAC at 16/44.1 is so poor that you can hear a difference when using a different filter. I absolutely maintain that with current playback and recording technology 16/44.1 does infact exceed human hearing, believe it or not, this is why CD was specified at 16/44.1.

 Crowbar - If you want to re-invent all the equipment used in a recording studio, go ahead and knock yourself out. It doesn't matter where the extra 8bits are when recording in 24bit. The point is that no mic pre-amp on the planet can get anywhere near 144dB dynamic range, only the very best ones can get close to 96dB. If you wack the mic-pre up so you are close to 0dBFS then the LSBs are all just noise. You are not correct when talking about the design of recording studio control rooms. The idea is not to reduce the RT60, it is to make an RT60 of about 0.5secs. As Bigtop says the treatment is also to get an equal frequency response throughout the spectrum, not to enhance specific areas of the spectrum and of course to eliminate exterior noise.


----------



## nick_charles

As a practical matter I have downloaded loads of high resolution files from the web. So far not one of them has shown a DR in excess of 96db. Only 1 has been near 96db.


----------



## gregorio

Hi Nick,

 Most pop music has a tiny dynamic range because the audio has usually been passed through a compressor and limiter a number of times! It's not unusual these days to find commercial releases where the entire song has a dynamic range of only a few dB.

 The chances are that the high resolution files you've downloaded are not infact high resolution files, they are much more likely to be 16/44.1 files that have been upsampled. Upsampling cannot increase the dynamic range of a 16bit file unless it also employs some form of expansion, definitely not what I would want to be hearing!

 Gregorio


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Hi Nick,

 Most pop music has a tiny dynamic range because the audio has usually been passed through a compressor and limiter a number of times! It's not unusual these days to find commercial releases where the entire song has a dynamic range of only a few dB.

 The chances are that the high resolution files you've downloaded are not infact high resolution files, they are much more likely to be 16/44.1 files that have been upsampled. Upsampling cannot increase the dynamic range of a 16bit file unless it also employs some form of expansion, definitely not what I would want to be hearing!

 Gregorio_

 

What you say may well be true, but some of these recordings have come from Linn records and SoundKeeper recordings and I would hope that they would not be so misleading especially when they are specifically advertised as high res, but who knows. My listening is normally classical which does tend to have a higher DR but I have never found any CD recording that stretches the capability of 16 bits


----------



## gregorio

Hi Nick,

 I am not saying for definite that they are not hires files but as I mentioned before, there is no point to hires for the consumer. As part of the mixing process compression is added, during mastering more compression is added, during broadcast more compression is added. All this compression reduces the dynamic range dramatically and completely makes a mockery out of so called hires recordings. In my opinion hires audio is simply another marketing ploy to get people to spend more money. As you've realised, 16bit is rarely ever utilised, so what is the point of 24bit for the consumer?

 Gregorio


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_When someone quotes 145 dB as the dynamic range for 24-bit audio that has nothing to do with how loud it is. How loud it is is determined only by how powerful your amplifier is._

 

In order to hear all 96dB of dynamic range on a CD... specifically to be able to hear the quietest thing recorded on the track... you would have to turn the volume on your amp up over 100dB. (The little extra bit would be to boost the sound over your normal room tone.) This would be more than just uncomfortably loud. It would damage your hearing after a while.

  Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Having more DR means dynamic instruments that produce high intensity sound like drums or cymbols should be able to reproduce those sharp pulses and quieter sounds like vocals, at the same time_

 

Having more dynamic range than what? It's all relative. 96dB is plenty of headroom to be able to reproduce drums and cymbals along with vocals. Increasing that to 140dB isn't going to make it sound better, because the only things benefitting from added resolution are sounds so quiet, you can't hear them anyway.

 Scientific theory is great, but you have to bring the numbers and concepts into the real world to decide if they are important or not. More and bigger numbers do not necessarily translate into better sound. There's a point where you hit the limits of human hearing.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_What you say may well be true, but some of these recordings have come from Linn records and SoundKeeper recordings_

 

It would be interesting to get a pro grade audio editing program, downsample and properly dither those hi-res tracks down to redbook, balance the line levels and see if they can be told apart in an A/B DBT. I'd but a twenty spot on the answer to that being no.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_It would be interesting to get a pro grade audio editing program, downsample and properly dither those hi-res tracks down to redbook, balance the line levels and see if they can be told apart in an A/B DBT. I'd but a twenty spot on the answer to that being no.

 See ya
 Steve_

 

I don't suppose Cool Edit Pro is good enough, I have that.


----------



## gregorio

I've always recorded at 24bit (since about 1996) and then noise shape dither down to 16bit, in fact this is what pretty much every commercial recording studio on the planet does. Even in a highly specified studio environment you can't tell a difference.

 Nick, cool edit pro is pretty good for the money but is not used by professionals much, most professional studios use DigiDesign ProTools.

 Gregorio


----------



## Jo-Vo

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I've always recorded at 24bit (since about 1996) and then noise shape dither down to 16bit, in fact this is what pretty much every commercial recording studio on the planet does._

 

I agree, that's the way to go.

 Greets
 Jonas


----------



## frankR

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I would like to hear your proof, which flies in the face of 25 years of proof and implementation of digital theory._

 

I don't need proof; I can hear the difference with my own ears.

 You want proof, here is data.

 Let’s test my hypothesis that loss in the ultrasonic frequency spectrum causes distortion in the audible frequency spectrum.

 I will use the following test tracks in 16-bit / 44 khz and 24-bit / 96 khz quantization that msofsi posted back in late Dec-07.

Soundkeeper Recordings Format Comparison

 I will analyze the frequency spectrum of a 6 ms sample of the opening guitar chord between approximately 0.078 and 0.084 seconds. My ears tell me the guitar chords from the 24/96 sample sounds more lifelike. I believe the complex harmonics present in the guitar are better rendered from the higher resolution audio.

 Let’s take a quick look at the waveforms from both samples in figure 1.






 Figure 1: Upper waveform is from the 16-bit/44 khz sample, the lower wave form the 24-bit/96 khz sample.

 The difference of these two waveforms is striking. The higher resolution audio is able to resolve the high-order harmonics present in the guitar chord. One could postulate that those high frequency oscillations in the high resolution waveform are out of the audible frequency band, beyond 20 khz, and therefore irrelevant to the ear.

 However, my hypothesis is that losing those harmonics results in distortion of the audible frequency band. Remember, the guitar is actually making those high frequency zigzags in the 24/96 waveform, they are not an artifact from the recording process! Therefore, we will use the 24/96 waveform as a reference and make a comparison of their power frequency spectrums.





 Figure 2: Power frequency spectrum, 24/96 compared to 16/44 audio samples.

 In figure 2 we see the 24/96 frequency spectrum extends well beyond the audible frequency cutoff at ~20 khz. However, even in the audible band we see quite striking differences between the two frequency spectrums.

 Now I want to add a big caveat to these frequency spectrums because of the way they were computed. The software I used, called ‘Audacity’ lacks precise control over how the frequency spectrum is binned. The software has an approximate 5% error between the frequency bin centers. However, I believe the error is negligible enough to make a qualitative comparison. And we certainly notice errors in the 16/44 frequency spectrum just by looking at the raw waveform.





 Figure 3: Frequency spectrum error (difference)

 Figure 3 shows what the 16/44 track is missing. We can see from several sharp peaks above 4 khz that the 16/44 track is missing some key harmonics in the audible band present in the guitar chord. These missing harmonics are evidence of what my ear is telling me. You can’t just throw away ultrasonic harmonics and expect the lower frequency information to render properly. In other words, the ear can’t hear the ultrasonic harmonics, but it can hear the distortion of lower frequency harmonics caused by their absence.

 I don't understand the logic of artificially throwing away information because some short-sidded theory about how the ear-brain works says it's irrelavent. Record and reproduce every bit of information possible and let your ears determine what is important!


----------



## frankR

I thought I might describe what it is I'm hearing in the guitar chords so you can go back and try to pick out the differences yourself.

 In the 24/96 track I can hear some slight beating in the guitar chords, like a slight mistune. It gives the chords a kind of flair that makes the guitar sound real.

 In the 16/44 track the beating is not there. In contrast the chords have a kind of metallic, digital sound, making the guitar sound flat and artificial. This caused by the smoothing of the waveform.

 There is a lot more that I hear each time I listen, but it's difficult to describe what I hear. The sound stage is also quite different, the high res material has a lot more depth.


----------



## frankR

Perhaps you've tried and are unable to hear the difference.

 I know you can see the difference:





 24-bit / 96 khz, 500 microsecond sample




 16-bit / 44 khz, 500 microsecond sample

 These are graphic reprentations of the frequency spectrum from the same samples I presented in the ealier analysis generated in AVS Audio Editor. The horizontal axis is time and the verticle axis is frequency. The color table represents relative power frequency spectrum intensity. I vertically compressed the 24/96 graph in the 0-24 khz range to match the scale of 16/44 audio frequency range in Photoshop. Notice how small the time domain sample is, less then a half a milisecond, the pixels in the 16/44 graph represent samples.

 If the 16/44 audio was not missing anything in the audible frequency range then these two graphs should look identical. Obvious they do not.

 The conclusion here is the same as before. The audible frequencies in the 16/44 audio is distorted all the way down to DC frequencies.


----------



## Crowbar

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_I don't need proof; I can hear the difference with my own ears._

 

I think you need to eliminate the possibility this is a filtering issue before you draw any conclusions.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_The point is that no mic pre-amp on the planet can get anywhere near 144dB dynamic range, only the very best ones can get close to 96dB._

 

There's no preamp involved in the example I gave! The ADC is right in the capsule, with just an anti-alias filter before it. I'm quite sure a pro-audio company wouldn't claim 130 dB if they can't meet their own specs!
 If I can build a DAC I/V with noise and THD below 120 dB (as I have), I'm quite sure that professional engineers can build an analog front end to an ADC with at least as good performance.

 You're incredibly short sighted by talking about limitations of typical recording and mastering processes, and missing the goal: the goal is to capture sound and then reproduce it to a point where it is not physiologically possible for a participant to distinguish it from the original performance, under any possible listening condition, and including even subconscious effects such as from the Hypersonic paper, or other effects of sound quality which are sometimes noticeable only after very prolonged experience to a system (because they are just at the conscious threshold).


----------



## frankR

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I think you need to eliminate the possibility this is a filtering issue before you draw any conclusions._

 

Any comments on the frequency spectrum analysis?

 I've shown cutting off ultrasonic frequencies results in distortion of audible frequencies.


----------



## Crowbar

How were they cut off though?
 And the Fourier transform is done with some windowing to create those spectral plots; it's not mathematically ideal.


----------



## frankR

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_How were they cut off though?
 And the Fourier transform is done with some windowing to create those spectral plots; it's not mathematically ideal._

 

They're not cut-off, the're just not resolved. The 16/44 digitization is not recording the actual signal. It's smoothing it out, that is distortion! 

 I can't take accurate data of a signal that has much of its information above 20 khz with an instrument only capable of making a measurement below 20 khz.

 The optic analogy is trying to resolve an object which has spatial frequencies above the resolving capability of the optics. What you get is a blurred image from the point spread function. Blurring high frequencies objects lowers the contrast of adjacent objects it can resolve. That's what happening to the audio, it's being blurred.

 Have you listen to the audio samples? I'm shocked you can't hear the difference, it's becoming night and day for my ears the more I listen to them.

 If you watch the graphical frequency spectrum in AVS Audio Editor in realtime you can see there is a lot of frequencies that the guitar makes that extend to 50 khz, and probably beyond. You think 20 khz is adaquate?

 If I could find an app that can output the raw wave form I can import it into IDL and do some better numerical analysis.

 What do you mean by Fourier windowing? Are you refering to the frequency bin? The bin sizes are very close, like I said there is some error attributed to this binning discrepency, but the second post that showed the graphical representation really gives you a sense of the difference.


----------



## Febs

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_If you watch the graphical frequency spectrum in AVS Audio Editor in realtime you can see there is a lot of frequencies that the guitar makes that extend to 50 khz, and probably beyond. You think 20 khz is adaquate?_

 

A 24/96 recording cannot have frequencies that extend to 50 kHz.

 Moreover, the fundamental frequency of the E on the 24th fret of the "E" string on a guitar--which is about the highest note that some guitars can play and higher than many guitars are even capable of playing--is just 1.4 kHz. The guitars on the recording you linked to are playing at least an octave lower than that at their highest. What sort of musical content do you imagine exists at 50kHz, more than five octaves above the highest fundamental frequency the guitar is capable of producing?


----------



## frankR

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_A 24/96 recording cannot have frequencies that extend to 50 kHz._

 

The Nyquist frequency is 48 khz, or nominally 50 khz.

  Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Moreover, the fundamental frequency of the E on the 24th fret of the "E" string on a guitar--which is about the highest note that some guitars can play and higher than many guitars are even capable of playing--is just 1.4 kHz. The guitars on the recording you linked to are playing at least an octave lower than that at their highest. What sort of musical content do you imagine exists at 50kHz, more than five octaves above the highest fundamental frequency the guitar is capable of producing?_

 

Something, according to the data.

 The modes of a guitar string(s) and accoustic body are very complex.

 The following article shows how many octaves with signifigant energy an open A bass guitar open string detuned to a 50 hz fundamental has. There is signifigant energy out to 14 harmonics, or 4 octives.

http://arxiv.org/ftp/physics/papers/0605/0605154.pdf

 Imagine multiple strings resonating together along with the guitar body how complex the frequency spectrum is. Harmonic of harmonic of harmonics. I wasn't implying that frequencies as high as 50 khz are critical for reproducing a guitar.

 22.4 khz is only 4 octives from 1.4 khz. 44.8 khz is only 1 octive from that.


----------



## nick_charles

I did a blind test on the 24/96 and 16/44.1 files and scored 13/15, but the difference was minute, real straining at gnats stuff. However when you downsample the 24/96 to 16/44.1 and compare the 24/96 to 24/96 downsampled they are not audibly different. I did two lots of unsighted tests to make sure. Thus it may be that the original 24/96 and 16/44.1 files are not fundamentally the same recording, i.e they differ in more than sample-rate and bit-depth.


----------



## Febs

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_22.4 khz is only 4 octives from 1.4 khz. 44.8 khz is only 1 octive from that._

 

And as I indicated, the guitar on the recording you linked to is not playing anywhere close to 1.4 kHz. It's playing more than an octave below that, so 22.4 kHz is 5 octaves or more from the fundamental. 

 Regardless, look at Figure 3 of the article you linked me to. How much energy is shown in the fourth octave above the fundamental (880 Hz)?


----------



## frankR

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_And as I indicated, the guitar on the recording you linked to is not playing anywhere close to 1.4 kHz. It's playing more than an octave below that, so 22.4 kHz is 5 octaves or more from the fundamental. 

 Regardless, look at Figure 3 of the article you linked me to. How much energy is shown in the fourth octave above the fundamental (880 Hz)?_

 

Nevertheless, the data indicates there is frequency content there.


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_I will use the following test tracks in 16-bit / 44 khz and 24-bit / 96 khz quantization that msofsi posted back in late Dec-07.

Soundkeeper Recordings Format Comparison_

 

Don't use their 16/44.1 track. Resample the 24 bit down using the proper dither.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_http://arxiv.org/ftp/physics/papers/0605/0605154.pdf

 Imagine multiple strings resonating together along with the guitar body how complex the frequency spectrum is. Harmonic of harmonic of harmonics. I wasn't implying that frequencies as high as 50 khz are critical for reproducing a guitar.

 22.4 khz is only 4 octives from 1.4 khz. 44.8 khz is only 1 octive from that._

 

The charts in that paper are pretty useless, they have no scale on the Y axis, there is no way of knowing how much energy there is on those harmonics...


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_The following article shows how many octaves with signifigant energy an open A bass guitar open string detuned to a 50 hz fundamental has. There is signifigant energy out to 14 harmonics, or 4 octives._

 

What you aren't taking into account is that with each octave the harmonic is significantly quieter than the octave before it. By the time you get up to 4 octaves of harmonics, you aren't going to even be able to hear it. Add to that the fact that most musical instruments (with the exception of gongs and cymbals) produce fundamental tones that are at least four octaves within the upper limit of human hearing (20kHz). That means that for a harmonic to exceed the capabilities of redbook, it not only will be too quiet to hear, it's at a frequency you can't even hear. Totally unimportant.

 See ya
 Steve


----------



## frankR

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Don't use their 16/44.1 track. Resample the 24 bit down using the proper dither.

 See ya
 Steve_

 

I'll try that.


----------



## bigshot

Make sure the line levels are matched when you compare the two too.

 See ya
 Steve


----------



## frankR

I don't need to prove that the guitar is making high frequency sounds, it's in the frequency spectrum I've already shown.

 Do you even both to look at the data, or are you going to go on a tanget?






 Down sampled to 16/44




 Native 24/96 audio

 Again, same conclusion. The down sampling distorted the audible spectrum and the details I was hearing are now gone. My ears hear it, the data shows it...


----------



## frankR

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_The charts in that paper are pretty useless, they have no scale on the Y axis, there is no way of knowing how much energy there is on those harmonics..._

 

The purpose of the article was to show guitar strings create higher frequency sound then the fundamental from high order harmonics. A relative amplitude y-axis is sufficient. It even labels it as a log scale. What’s your point?

 I know they have much less energy then the fundamental. So where is the disagreement?

 And you can't gloss those harmonics over as irrelevant.

 What makes a C on a guitar sound different then on a piano, on a bassoon, on a clarinet, out of Celine Dion's vocal chords? It's not the fundamental, because they have the same frequency, more or less. It's those weaker harmonics!


----------



## eruditass

no, it's not the weaker harmonics, it's the stronger overtones. An overtone may or may not be an integer multiple of the fundamental, so it includes harmonics. however weaker harmonics/overtones obviously don't have as much amplitude as stronger harmonics/overtones, which play more of a role of defining an instrument. after a certain point, there is a lot less energy in the upper overtones depending on the instrument and the fundamental.

 i don't see how 24bit/16bit relates to this, unless we're talking about sample rates, which are completely different. regardless, our ears really can't hear much above 20Khz, and that is a fact. most cant even hear around 20 Khz. the point of higher sample rates is to get timing down better.


----------



## frankR

Quote:


  Originally Posted by *DoomzDayz* /img/forum/go_quote.gif 
_i don't see how 24bit/16bit relates to this, _

 

It doesn't, it's a tanget.

 My argument is that high resolution audio increases fidelity of the audible frequency spectrum.

 Claiming that a higher cut-off frequency is not necessary because we can't hear that high is the same as saying we don't need see the texture of her skin because we can already see the outline of her face. Higher resolution reveals finder details in the stuff you already can see, or hear.


----------



## indikator

how 24 bit output in foobar?

 will it somehow interpolate the 16 bit mp3 to become better?


----------



## eruditass

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_It doesn't, it's a tanget.

 My argument is that high resolution audio increases fidelity of the audible frequency spectrum.

 Claiming that a higher cut-off frequency is not necessary because we can't hear that high is the same as saying we don't need see the texture of her skin because we can already see the outline of her face. Higher resolution reveals finder details in the stuff you already can see, or hear._

 

your analogy isn't quite accurate, you need to extend it to the limits of vision as it relates to the limits of hearing. a more accurate analogy would be you don't need put the atoms of her skin in an image because we can't see at that level. or only a certain resolution is necessary for a screen at a certain distance because the viewer only has 20/20 or worse vision. there is absolutely nothing in our inner ears that responds to anything above 20Khz. If you're counting other senses, I suppose it could be different. I can certainly accept that argument for sub bass and feeling it.


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_My argument is that high resolution audio increases fidelity of the audible frequency spectrum._

 

I don't think anyone is arguing with you about that. We're just pointing out that the added resolution is so far down the dynamic range, you aren't going to be hearing it when you sit down in front of your stereo to listen to music.

  Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Claiming that a higher cut-off frequency is not necessary because we can't hear that high is the same as saying we don't need see the texture of her skin because we can already see the outline of her face. Higher resolution reveals finer details in the stuff you already can see, or hear._

 

Yes, but you have to *blow the image up* to see the pores in the skin on the girl standing forty feet away. In an audio mix, you might need to blow up details like that, so added resolution is useful. When you are listening to a stereo, you aren't going to be able to pull up those tiny details without making the loud portions unbearably loud.

 It's all relative. You need to understand the perspective.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_The purpose of the article was to show guitar strings create higher frequency sound then the fundamental from high order harmonics. A relative amplitude y-axis is sufficient. It even labels it as a log scale. What’s your point?_

 

If the 19th and 20th harmonics are at - 120db from the fundamental they are utterly irrelevant.

  Quote:


 I know they have much less energy then the fundamental. So where is the disagreement? 
 

It is your insistence that capturing these distant low level harmonics is important.

  Quote:


 And you can't gloss those harmonics over as irrelevant.
 What makes a C on a guitar sound different then on a piano, on a bassoon, on a clarinet, out of Celine Dion's vocal chords? It's not the fundamental, because they have the same frequency, more or less. It's those weaker harmonics! 
 

But you don't need to extend the harmonics up to 15 or 20 to tell the difference. Back in 1978 Muraoka, Yamada and Yamazaki (JAES 26) ran some experiments. They took speakers capable of rendering 40Khz, music with high frequency components of well over 30K (including moog synthesizer tweedling and cymbals crashing) plus trained audio professionals. They ran music without filtering and with filters at 14, 16 , 18 and 20K. All the filters would remove plenty of the higher harmonics known to be present in the music. 

 Not one person could tell the difference when the filter was at 20K to a standard 0.05 significance level, none could manage it at 18K or 16K either only at 14K did the filter make the unfiltered and filtered music differentiatiable at the 0.05 level .........for 9/32 audio pros. They concluded that a 20K filter was perfectly safe but that pragmatically 15K would probably be fine too.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Not one person could tell the difference when the filter was at 20K to a standard 0.05 significance level_

 

Any meaning derived from this obviously depends on ability and training of the subjects. Moreover, the much more recent studies I mentioned contradict your implied claim that >20 kHz has no perceptually measurable effect.


----------



## Crowbar

Quote:


  Originally Posted by *DoomzDayz* /img/forum/go_quote.gif 
_there is absolutely nothing in our inner ears that responds to anything above 20Khz._

 

The studies cited above prove there is something, whether in the ears or elsewhere in the head, that DOES respond to >20 kHz. It is flippant and insulting of you to pretend things posted in this thread were not posted 
	

	
	
		
		

		
			




 Next time bother to read a thread completely before you decide to add your two cents--they might have been discredited before you even posted them, as is in fact the case here.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Any meaning derived from this obviously depends on ability and training of the subjects. Moreover, the much more recent studies I mentioned contradict your implied claim that >20 kHz has no perceptually measurable effect._

 

Recency really is not an issue, inaudible in 1978 is inaudible in 2008, we have not suddenly evolved better hearing in the last 30 years. Sure some listeners *are* better than others, but we are talking about trained audio engineers here, not Joe and Mary from the Corner shop.

 The Oohashi paper has already been rendered very dubious by the discovery of the IMD effect of the two frequency ranges being played back together and te fact that nobody has ever been able to replicate it and there have been attempts tpo do this.

 The Lenhardt paper *relies on bone conduction *- they are talking about *making the brain resonate *- this is a parlour trick and so tangential to the argument as to be irrelevant. For human listening without cochlear implants or direct contact with the source these frequencies are irrelevant.


----------



## grawk

Trained audio engineers are more likely to have hearing damage than the average person, I'd be willing to bet...


----------



## Febs

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_I don't need to prove that the guitar is making high frequency sounds, it's in the frequency spectrum I've already shown._

 

Can you explain to me how it is that your frequency spectrum graph is showing data from the 16/44.1 sample that is *above *the Nyquist frequency, 22.05 kHz? Perhaps I'm misinterpreting the graphs.


----------



## frankR

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Can you explain to me how it is that your frequency spectrum graph is showing data from the 16/44.1 sample that is *above *the Nyquist frequency, 22.05 kHz? Perhaps I'm misinterpreting the graphs._

 

Anything above 22 khz is obiously cut-off in the 16/44 audio. However, the frequency spectrum of the 24/96 (red-line) audio shows signifigant spectral power just past where the blue line ends. This is from an approxminate 10 ms sample of the guitar chord. There are others with musical training in this thread, maybe they can describe the chord being played. My guess from doodling on guitar is a C-major chord? Is that standard tunning?






 I want to remind everyone, my hypothesis was not that the high resolution audio is better because of extend frequency response above ~20 khz, it's because the audible band is rendered with much increased resolution. I've posted several graphs now that successfully demonstrate this.

 The point is I DO hear a difference in the high resolution audio.

 The question is why? My efforts yesterday may explain why.

 Those who believe there is no difference, you must think that all those audiophiles that spend money on high res. digital equipment like SACD are wasting their money; because it's impossible for them to expirence increased fidelity.


----------



## Febs

Sorry, I wasn't clear. I was referring to this graph:


----------



## frankR

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Sorry, I wasn't clear. I was referring to this graph:




_

 

You're wondering why the scale shows information that extends beyond 22 khz.

 The 24/96 graphs also goes beyond 48 khz. The scale is incorrect?

 The freeware software I used to generate my data aren't exactly scientific grade. If I can find a way to extract the audio waveform I could do much more careful analysis in a proper scientific platform.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Any meaning derived from this obviously depends on ability and training of the subjects._

 

Is that anything like training yourself to fly by jumping off taller and taller ladders?

 There's a point where our hearing just can't hear it, no matter how much we struggle and strain in a vain attempt to educate our ears.

 This all comes down to whether you are building a system for bats to listen to or one that does the job for you. As I said before, the balance of the frequencies between 200Hz and 8kHz is infinitely more important to sound quality than the frequencies that lie outside the range of human hearing.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_The studies cited above prove there is something, whether in the ears or elsewhere in the head, that DOES respond to >20 kHz_

 

But it has nothing to do with listening to music.

 When I was a kid, I hated to go to Sears, because they had banks and banks of florescent lights that would put out a high pitched squeal that would give me a headache. I couldn't really hear the squeal, but I could feel the pain it caused.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_Trained audio engineers are more likely to have hearing damage than the average person, I'd be willing to bet..._

 

I've found that a lot of them aren't very good at judging mixes either. I remember one mix I was supervising where the director heard a bump where the mixer had messed up with one of his pots. He asked the engineer to fix it, and he ran over and over the section leaning over the board peering at his VU meters. "Nope. No bump there." The director growled at me, and I gently suggested that the engineer stop looking at the meters and just sit back and listen. "Oh! I hear it now!"

 If it wasn't on his meter, he couldn't hear it.

 See ya
 Steve


----------



## nick_charles

I have played around with the high-res and 16/44.1 files a lot, more than I can really afford to. From what I see the only real difference and the only thing that made the two files audibly different was the opening 1/10th of a second of the opening guitar chord, for the remainder of the tracks the waveforms are identical unless you zoom to the 1/10,000 of a second scale at seconds 10ths of seconds 100ths of seconmds and 1000ths of seconds these are identical. My feeling is that the difference and it really is tiny is an artifact rather than an indication of better resolution. 

 Also the zooming is a bit of a red-herring, when you zoom to the maximum both 24/96 and 16/44.1 waveforms become completely flat lines and there is no difference between them at all, well give or take about 0.001db 
	

	
	
		
		

		
			





 When however you use Audacity to look at the first 0.15 of a second and view it as waveform DB the opening chord is drastically different on the two samples for the first 1/10th of a second, from that point of view these recordings simply do not look the same, for the first 1/10 of a second the 24/96 sample shows a lot of squaring off of the bottom of the wave like it is truncated while the 16/44.1 the wave descends normally (?). 






 This is interesting. After that point though they are identical.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_But it has nothing to do with listening to music.

 When I was a kid, I hated to go to Sears, because they had banks and banks of florescent lights that would put out a high pitched squeal that would give me a headache. I couldn't really hear the squeal, but I could feel the pain it caused.

 See ya
 Steve_

 

It contributes to the experience that the sound creates. Whether it is pleasant or not is a purely subjective evaluation and so has no relevance to the discussion.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It contributes to the experience that the sound creates. Whether it is pleasant or not is a purely subjective evaluation and so has no relevance to the discussion._

 

Listening to music is a subjective experience. Figuring out how to make your stereo sound good is an objective one. You shouldn't confuse those two. Super high frequency sound in recorded music is almost always noise.

 See ya
 Steve


----------



## CyberTheo

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Listening to music is a subjective experience. Figuring out how to make your stereo sound good is an objective one. You shouldn't confuse those two. Super high frequency sound in recorded music is almost always noise.

 See ya
 Steve_

 

May I suggest that figuring out how to make your stereo sound good is also an subjective one.


----------



## bigshot

Putting together a stereo isn't creative. It's about achieving optimal fidelity. You do that by understanding how sound works and how your equipment works and applying what you know to make the system perform at its best. That isn't a subjective process.

 If you try to put together a stereo system randomly based purely on subjective reactions, you'll be at the mercy of how you feel at any particular time and what you ate for dinner. You'll also be prime prey for commissioned salespeople. (They don't want you to think. They just want you to put your feet up and take their word for it.)

 See ya
 Steve


----------



## frankR

It's a shame this disscussion is distracted by wether sound beyond 20 khz is important. I happen to believe that it is not.

 I found a way to extract the raw waveform last night and have some nice data comparing the power frequency spectra (PFS) of the 16/44 and 24/96 audio.

 What I found is similar to what I showed in the first post, only this time with greater detail. Of the first few second of the audio track most of the PFS energy is in the 64 to 1024 hz range. The magnitude and distribution of the 16/44 accoustic energy at several key harmonics is distorted relative to the 24/96 audio.

 Also what was surprising is there is several harmonics with signifigant energy that are artifiicial. These effects are probably what is refered to as quantization noise.

 Bottom line is, there is signifigant evidence which suggest that the enhanced resultion in the audible band found in the 24/96 audio is indeed critical to fidelity. My ears are not misleading.


----------



## scompton

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Putting together a stereo isn't creative. It's about achieving optimal fidelity. You do that by understanding how sound works and how your equipment works and applying what you know to make the system perform at its best. That isn't a subjective process.

 If you try to put together a stereo system randomly based purely on subjective reactions, you'll be at the mercy of how you feel at any particular time and what you ate for dinner. You'll also be prime prey for commissioned salespeople. (They don't want you to think. They just want you to put your feet up and take their word for it.)

 See ya
 Steve_

 

But that's not the audiophile way as witnessed in the cables forum
	

	
	
		
		

		
		
	


	




 Of course, I've never considered myself an audiophile and these forums have reinforced that.

 I must say that this is an incredibly interesting thread, as well as intimidating. I'm planning to start digitizing some of my LPs in the not too distant future. While I knew it would be some work, this thread is making me realize that it's a lot more work than I knew. And maybe cost more money than I thought. It's not going to stop me from doing it, just raising the intimidation factor a bit.


----------



## gregorio

Interesting arguments. Interesting because some of them are based on a perception of fact rather than the reality. For example, when looking at a waveform on a computer screen, what are you actually looking at? You're looking at a graphical representation of the digital data stored in the audio file. What you are not looking at is a representation of what the analogue waveform will look like once it's come out of a DAC. This is obvious if you think about it, how could a piece of software emulate the effects of all the different processes that take place in every DAC on the market? In other words, of course a 44.1kFs/s file is going to look less detailed than a 96kFs/s file, the question is, is it any less accurate once it's converted back to an analogue waveform? The answer is no! The answer has to be no, otherwise the whole theory of digital audio is wrong and digital audio doesn't exist!! You need to have two sampling points per waveform in order to perfectly recreate that waveform. Having more than two points is not going to make the recreated waveform more perfect. Hence why the Nyquist theorem states that you need to have twice the sampling frequency to encode a given audio frequency.

 In response to freqs >22kHz having an affect on the freqs in the hearing range. Possibly but what difference does it make? If anything is affected in the hearing range, those effects would be encoded at 44.1kFs/s as well as they would at 96 or 192.

 I routinely use a system that has 48bit resolution, that's 288dB dynamic range. So it must sound wicked compared to 24bit when listening to completed mixes, err no. It makes no difference whatsoever, nor does comparing my 48bit system with 16bit. It's not unusual to find pop songs with a dynamic range of less than 10dB, for classical it's usually less than 50dB. 96dB (16bit) is more than enough, why do you need 144dB (24bit)? Maybe you want to hear the tuba player's nose hairs vibrate, just before he plays a note and vapourizes your eardrums!

 Why do some of you refuse to believe that intrinsically 24/96 as a consumer format is no better than 16/44.1 and that any perceived difference is an effect of a DAC? 

 My guess is it's because it's difficult to get past the logical (but incorrect) assuption that more data must mean more detail and therefore better quality.

 Crowbar - The idea of recording is not to make the recording sound identical to the live performance. Most pop music cannot be performed acoustically. Even with classical music this statement is incorrect. You go listen to a french horn, tuba or even flute, up close. It sounds nothing like it does in a big concert hall from an audience point of view, but we can't put the mic too far away or we'll get SNR problems and no clarity. So we have to fake it, we make value judgements about the perception of our target demographic and then we work to thier expectation. We also have to fake it because recording equipment is far from perfect (sometimes deliberately so) and we have to make compensations.


----------



## CyberTheo

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Putting together a stereo isn't creative. It's about achieving optimal fidelity. You do that by understanding how sound works and how your equipment works and applying what you know to make the system perform at its best. That isn't a subjective process.

 If you try to put together a stereo system randomly based purely on subjective reactions, you'll be at the mercy of how you feel at any particular time and what you ate for dinner. You'll also be prime prey for commissioned salespeople. (They don't want you to think. They just want you to put your feet up and take their word for it.)

 See ya
 Steve_

 

Not exactly randomly though. How a system sounds is largely subjective. Everyone's ears are different, and each has ones own perception of "optimal fidelity". Optimal to one does not mean optimal to another. But then lets not confuse a "clinical setup" to one that suits each person's "subjective" taste. Both types of setups are equally difficult to achieve, one by using test equipments and one by using ears. As well, lets remind ourselves of experience that a showroom system sounds very different in our homes. That's why the hi-fi industry has continued to survive over the generations and continued to pump out "better" products.


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Interesting arguments. Interesting because some of them are based on a perception of fact rather than the reality. For example, when looking at a waveform on a computer screen, what are you actually looking at? You're looking at a graphical representation of the digital data stored in the audio file. What you are not looking at is a representation of what the analogue waveform will look like once it's come out of a DAC. This is obvious if you think about it, how could a piece of software emulate the effects of all the different processes that take place in every DAC on the market? In other words, of course a 44.1kFs/s file is going to look less detailed than a 96kFs/s file, the question is, is it any less accurate once it's converted back to an analogue waveform? _

 

So basically the zooming in on the waveforms tells you precisely nothing since the software does not show the effects of the reconstruction filter which makes the output whole again. 

 Just one question then, when I zoomed into the 24/96 I saw what looks suspiciously like truncation, a flattening of the energy (see my post a page or so back) on the 24/96 while the 16/44.1 descends gracefully, what does this mean ?, is this an artifact, this seems to be the only thing different on the two samples, could it be that the 24/96 ADC isnt responding correctly on the first guitar crash ?


----------



## gregorio

Nick - "So basically the zooming in on the waveforms tells you precisely nothing since the software does not show the effects of the reconstruction filter which makes the output whole again."

 Correct. The graphical display is a real approximation. Zoom all the way in so you can see the individual samples. The very fact that you can see individual samples is telling you that what you are looking at digital data, not the smooth analogue waveform that is going to come out of your DAC. Every DAC has different filters, different processes and different analogue circuitry. There is no way for your software to know what is going to happen to the digital datastream once it has passed out of RAM and is routed to a DAC.

 A squared off wave usually indicates that clipping has occurred somewhere in the recording or mixing process. Clipping is when the amplitude (gain) of the waveform exceeds 0dBFS. Normally there is an obvious click or digital distortion when the signal has been clipped.


----------



## bigshot

Quote:


  Originally Posted by *CyberTheo* /img/forum/go_quote.gif 
_Not exactly randomly though. How a system sounds is largely subjective. Everyone's ears are different, and each has ones own perception of "optimal fidelity"._

 

Everyone's ears are different on different days and under different situations. That's why using them as your only guide will lead you back and forth randomly.

 Optimal fidelity has nothing to do with how we perceive sound. It is the ability to reproduce recorded sound as close to the way it was intended as possible. You do that by trying to get low signal to noise and distortion, accurate dynamics and balanced frequency response.

 Everything is tempered by practicality. For example, it might be possible to reproduce beyond the range of human hearing, but why waste effort on things you can't hear?

 If you want me to just say, "Whatever floats your boat is OK for yourself" then consider it said. But if you want to offer advice to other people about how they might achieve optimal sound with their particular ears, you're going to be a lot more useful to them if you stick to objective things than if you dwell on subjective ones that probably don't apply to them.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Super high frequency sound in recorded music is almost always noise._

 

Doesn't matter. If it was there in the original sound being recorded, it should be there in the reproduction as long as there is potentially any effect detectable by human physiology.


----------



## bigshot

My stereo produces the cleanest sounding noise on the block!

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_You go listen to a french horn, tuba or even flute, up close. It sounds nothing like it does in a big concert hall from an audience point_

 

I know that. I've sat at the piano performing in a hall just as I've sat in the audience. But the point is that though the recording can sound different depending on location, it's not the same type of difference as the distortions added by the recording-reproduction process.

  Quote:


 We also have to fake it because recording equipment is far from perfect (sometimes deliberately so) and we have to make compensations. 
 

That just shows that the recording equipment needs improving.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_That just shows that the recording equipment needs improving._

 

You really don't understand what's involved in recording and mixing music. You're talking to several people here who have had professional experience doing just that and you are arguing a fundamentally wrong premise.

 I corresponded with a well known audio engineer on the internet once, and he told an interesting story. He said that he was at a carnival being held in his town, and he was standing at the top of the "chute", with games and barkers extending down to a roundabout (aka merry go round) at the end. All around him was the sound of people chatting, music playing, barkers calling out, machinery clanking and the calliope of the carousel in the distance. As the breeze wafted, the sounds mixed and moved around along with it.

 He closed his eyes, and as he stood there soaking it all in, he came to the realization that even if he had the best audio equipment in the world and all the money, expertise and time he needed at his disposal, he still couldn't reproduce the sounds he was hearing at that particular instant.

 Anyone who has worked in recording knows that you can't reproduce reality. There are too many complex distance, timing and directional cues involved. It's impossible today and it will be impossible for the foreseeable future. However, you CAN create a mix that is balanced and aesthetically pleasing. It can indicate the reality of the sound the same way a photograph can indicate the beauty of light. Creative selections determine how the music is blended, and focus the attention of the listener on the composition of the sound field.

 Reproducing sound is a mechanical process which involves issues of fidelity, but the presentation of music is a purely creative one. Compromises and creative decisions are how the art is created. They aren't a limitation to be eliminated by better technology.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_There are too many complex distance, timing and directional cues involved. It's impossible today and it will be impossible for the foreseeable future._

 

That's ludicrous. You only have two years, so all the spatial imaging is ultimately reduced to a direction-sensitive frequency response and timing filter that is the auditory mechanism. You need only three conditions to capture and reproduce reality so that it's indistinguishable--perfect to some level of precision.
 a) very low distortion microphones and recording equipment
 b) very low distortion canal-phones and playback equipment
 c) proper recording and playback geometry: this basically means a binaural setup with dummy head either sufficiently close to that of the intended listener, or, since that's not practical in most cases, use of DSP and knowledge of listener's HRTF to compensate for difference between him and the generic dummy head used for recording (HRTF can be determined by a laser scan of the head with an couple thousand dollar scanner, or you can go to the computer vision or graphics lab at your local university, and subsequent finite boundary method simulation, which runs overnight on a modern computer).

 c) is obviously a difficult part, but even without personalized HRTF the results of a binaural setup can be excellent. a) and b) are what's under discussion here.
 Your argument: (not c) ) implies no ability to approximate reality, therefore a) and b) irrelevant
 Refutation: not c) does not have to be the case. Just because you've got a mediocre recording setup doesn't mean one can't, and shouldn't, strive for doing it the right way.

  Quote:


 Compromises and creative decisions are how the art is created. They aren't a limitation to be eliminated by better technology. 
 

Then why bother with ultra-high resolution displays, autostereoscopic displays, and wide gamut and high dynamic range displays? A combination of these technologies (each of which are already starting to mature and I have experienced them at SIGGRAPH a few years ago) will create a window into a captured and reproduced reality that is near-indistinguishable from the original. Your argument is analogous to arguing for living with the compromises of PAL/NTSC. No one would be silly enough to call that art being made, yet you're saying the same thing for audio! *Art is not the function of the recording and playback process and technologies; it's only rightful place is held in the original real world presentation by the artists. A recording engineer and an audio product designer should not play an artist because it's not their job, despite what their ego would prefer*.


----------



## bigshot

OK. Let me know next time you get into the studio for a mix. I'd like to watch.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Doesn't matter. If it was there in the original sound being recorded, it should be there in the reproduction as long as there is potentially any effect detectable by human physiology._

 

That is a big if, as I tried to point out the evidence for high frequency sound being detectable by humans seems to rely on a mix of IMD artifacts and/or direct coupling to the sound source.

 This is great for Beethoven who can feel the Piano vibrate and for others with serious hearing loss who can have it compensated for by bone conduction and I applaud it, but of little practical import to normal hearing listeners. 

 There is an interesting question as to why this IMD should produce such physiological effects, but it isnt anything to do with accurate sound reproduction.


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## bigshot

Beethoven listened to the piano by biting the sounding board. Perhaps some audiophile company could come up with speakers that have some sort of football teeth guard connected to them!

 HEAR BEETHOVEN THE WAY BEETHOVEN HEARD IT WITH THE EROICA MANDIBULAR AUDIOLIZER! (Inquire about the new limited edition "woodie" mandibular audiolizer by email!)

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_OK. Let me know next time you get into the studio for a mix. I'd like to watch._

 

If my doctor does a poor job in a surgical procedure and I criticize it, it is not acceptable for him to tell me to do it myself.
 Yet this is a perfect analogy for your argument. It shows great intellectual laziness on your part.


----------



## gregorio

Crowbar - Bigshot is correct, you are arguing from an incorrect standpoint. There has never been and probably never will be any piece of studio equipment which is perfectly transparent. Baring in mind we have mics, mic-pre amps, mixing consoles, EQ, compression, reverb and a whole raft of other tools commonly used on every production. None of them are perfect, they all cause some sort of artefacts and they all colour the sound in some way. Sorry if you don't like this information but these are the facts, it's always been this way and it probably always will be.

 Binaural mics definitely have their place in the recording arsenal but they are just a single tool, good at some things not so good at others. Just sticking up a binaural mic, a spaced pair, an XY pair or an M&S pair is how orchestras used to be recorded 40 years ago, things have moved on a fair bit since then.

 During a recording session at the Hit Factory in London many years ago I switched out a balanced pair of Neuman M50's (mics) worth about $200,000 for 4 Shure SM57s costing about $120 each. Why? Because the SM57's gave the frequency response and sound I was looking for, whereas the M50s, though fantastic mics, were picking up too much high freq and not representing the instrument as I wanted. I tried this same trick in another studio and it sounded terrible. Moral of the story is that recording, production and mastering are all art, it's not a science or a technical proceedure.

 Bottom line is that you can't just stick a mic up where a listener would sit and get an usable recording. Mics don't respond to sound the way our ears do. Mics have polar patterns, freq response graphs, off axis responses and precedence effect, just to name some of their features, all of which are different to the way your brain works. We're dealing with capacitance differentials, analogue electrical signals, digital data and the ballistics of magnetic coil movement, your ear deals with sound pressure levels and electrical impluses, almost totally different functions and mechanics.


----------



## scompton

An analogy to what you're saying about recording, is photography, something I know something about, unlike recording
	

	
	
		
		

		
			





 Current cameras and any camera in the foreseeable future, cannot take a picture that looks like what your eye sees. 

 Unless you use "normal" lens (approx. 50mm equivalent), the angle of view will most likely be different. 

 Foreground to background will be compressed with a telephoto lens or elongated with a wide angle lens. 

 Depth of field can be different. 

 White balance probably will be different. Your brain automatically corrects white balance in most light conditions. This can be approximated in digital with white balance settings or in film with different films and filters, but it's not perfect.


----------



## gregorio

Scompton - I don't know much about photography but it sounds like a good analogy. I should imagine that upgrading to a GigaPixel camera is not going to solve the problems you mentioned anymore than increasing CD music to 24bit/192k.


----------



## Crowbar

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Current cameras and any camera in the foreseeable future, cannot take a picture that looks like what your eye sees._

 

Patently false. There are already HDR cameras (at least in machine vision product lines), and stereoscopic cameras. But even with a regular camera you can take HDR photos by doing multiple exposures, and stereo ones by taking two parallaxed photos. Presenting these on an HDR, autostereoscopic, high resolution display, as I mentioned in a previous post, will be nearly indistinguishable from the original. The various technologies already exist; it's merely a problem of combining them.

 Even varying depth of field of the eye (in physiology known as accommodation) can be captured. It's been known for years in the field of computer graphics that an array of camera can capture data allowing reconstruction of ANY depth of field; you can search SIGGRAPH conferences for papers dealing with this. And VR helmet can easily then adjust its display lens to match varying eye accommodation based on pupil tracking--again, all of these technologies already exist and is a matter of combining them (it would be possible to do this with a screen that is a dual of the camera array I just mentioned, a compound display also discussed in computer graphics literature.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Scompton - I don't know much about photography but it sounds like a good analogy. I should imagine that upgrading to a GigaPixel camera is not going to solve the problems you mentioned anymore than increasing CD music to 24bit/192k._

 

You seem not to read what has been posted. The analogy is obviously that increasing sampling rate is just like increasing resolution of a display. But this is hardly the only thing one does with a display, as I pointed out in just a few post above, discussing stereoscopy and raising the color gamut and dynamic range! The hell? Do you people really lack reading comprehension, or have a memory that does not last more than two posts in the past?

 In any case, obviously his analogy actually helps my argument, since the same argument I'm using to show that in fact visual reality can be captured, however difficult for the moment, applies to aural one as well. It is a practical, not a theoretical problem.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_and probably never will be any piece of studio equipment which is perfectly transparent._

 

It only needs to be transparent to the resolving power of extensive blind testing under optimal conditions. Now, that's difficult and a lot of work to do, but we can infer conservative bounds on it based on human physiology, and design equipment and recording/playback setups to those instead.

  Quote:


 None of them are perfect, they all cause some sort of artefacts 
 

And each can be minimized below any possible perceptible threshold. It's a matter of engineering.

  Quote:


 and it probably always will be. 
 

You cannot possibly back this up with a logical argument, thus I reject it outright. This is the same old **** that we've been hearing for thousands of years that things don't change and things can't be done, no matter how many times they're proven wrong.
 Thomas J. Watson, in 1958 while IBM chairman: "There is a world market for about five computers."
 Or how about Charles Duell in 1899 while Patent Office commissioner: "Everything that can be invented has been invented."
 and my favorite, Lord Kelvin's 1895 "Heavier-than-air flying machines are impossible."
 But here we're not even talking about future technologies; we're talking about refining current ones.

 By the way, your anecdotes are pointless. Anecdotes are not valid arguments in any sense. All they show is that you are so afraid of change that you subconsciously are holding on to the past and I can tell you really hope there is no change and that we don't reach the transparency goal. And I mean, that would surely be a big blow to your ego as a recording technician who doesn't know his place and is instead an artist wanna-be.

  Quote:


 Mics don't respond to sound the way our ears do. Mics have polar patterns, freq response graphs, off axis responses and precedence effect, just to name some of their features, all of which are different to the way your brain works. 
 

Brain doesn't enter the recording/playback subchain, so that's irrelevant.
 Listener at live event: sound sources->ears->brain
 Listener listening to binaural recording: sound sources->microphones in dummy head canals->storage->headphones in listener ear canals->brain

 The listener provides the brain in either case; you don't need to worry about it with your microphones. Precedence. Moreover, polar response etc. is irrelevant in this geometric setup for a very simple reason: once in the ear canal, sound has NO directionality, because for all practical purposes the canal is now a one dimensional channel. The 3D sound field in a live listening is filtered by the head/ear and the positional information is encoded in the frequency response of the sound; in essence, the HRTF is a dimensionality-reducing operator: the input has directional information but the output doesn't.
 In the case of recorded/playback listening, the dimensionality reduction is provided by the dummy head. The microphones in the dummy head ear canals only get sound coming into the canal; there's no directional information. So the polar response is irrelevant, as long as the microphones are oriented to point towards the outward openings of the canal.

 As you can see, I've gone an point by point addressed what you posted. Better than bigshot's flippant remarks, which simply tell me that he had to resort to cheap derision after he ran out of logical counterarguments.


----------



## gregorio

I take your point Crowbar and look forward to hearing all about your new recording theory that does away with transducers. I can't wait to play with all the new recording equipment you are going to invent that's going to be so much better than the million dollar environments we currently creat music in. At last, we're going to have equipment which doesn't owe anything to Thomas Edison. May I suggest a name for your new technology ... How about ethereal osmosis?


----------



## Crowbar

This has been disappointing in the sense that I have not gotten any proper rational arguments in response. All I'm seeing is hand-waiving explanations such as "well look at this equipment and this one here I used and they have such and such problem, so it cannot be done, or why don't you show us how it's done"--gregorio's post just above this one a perfect example. Are you really dumb enough not to understand that whether I can construct something better or not does not in any way invalidate my arguments? That's like a physicist in the early 40s saying that atomic bombs can be made and will work, and then your analogue arguing that as an engineer he doesn't see how to make it so it must be impossible.
 I couldn't even count the logical fallacies that have been committed here just in the last two pages alone. It's sad that a course in critical thinking is not mandated in many countries' educational systems.
 By the way, I never said anything about doing away with transducers; don't put words in my mouth or I will request moderation in this thread.


----------



## gregorio

crowbar - honestly, I've never seen anyone as determined to maintain and broadcast their ignorance. Your arguments are based on what you believe should be possible but they have no basis in fact and are completely divorced from the reality of creating a recording. To then go on about other peoples's "logical fallacies" is completely nonsensical. Are you on some kind of medication?


----------



## Crowbar

It's not on what I "believe should be possible". It's based on what is possible given current technology, or at most with engineering refinements of current technology.
 All your arguments ultimately hinge on practical difficulties and "well I haven't done it this way so it can't be done".

*I have already described at a high level a system that can present sound transparently to human physiology in the three-requirements post earlier, and refuted your objection to microphone issues a couple of posts back.*

 You have not addressed a single of my points adequately; the thread is a record of this, and I'm still waiting. And questioning my qualifications is not a valid argument. Logic 101!

 The one that is determined here is you--determined to preserve the ego-stroking status quo that the recording technician should leave his signature on the sound instead of realizing his position as lowlier than that of an artist, and thus blocks in his mind any possibility that transparency can be arbitrarily increased by incrementally improved engineering, for that would take out the creativity of his job.

 As an aside, I'm involved in two projects on the playback side that cover correspondingly the electronics and the transducer elements and meet criterium b) as defined previously. If b) can be done a) can as well since the concepts are duals of each other. c) has already been done--there are papers dealing with the HRTF-from-simulation, etc.


----------



## colonelkernel8

This situation is clearly one of two conflicting perspectives. Neither is more valid than the other. Crowbar's position takes more of a theoretical stance on the limits of recording and suggests that the recording industry (represented here by gregorio) is ignorant because he doesn't see that today's technology has far surpassed the traditional methods of recording and mixing. Gregorio defends his position with an equally important fact, that Crowbar doesn't have recording industry experience and doesn't know first hand the recording industry environment. Both sides need to relax a bit on the insults, read the arguments, and respond cordially instead of making attacks on each other's intelligence.


----------



## gregorio

There you go again. You're quoting your belief of what current technology is capable of, not what technolgy is actually capable of. If you don't have the signature of the recording engineer and producer you do not have a recording, it's that simple. Only in science fiction and apparently in your mind does transparent recording equipment exist. It's got absolutely nothing to do with a recording engineer's ego.

 A good analogy would be like blaming the airline company for it taking 7 hours to fly from London to New York instead of 10 minutes. You can't blame the airline, they are stuck with the aircraft manufacturers and the fact that the technology doesn't exist to do the journey in 10mins. You can rant at the airline and call it ego-stroking but that doesn't detract from the fact that what you want is not possible. Do you understand, it is NOT POSSIBLE!! One day it may be possible but at the moment what you want doesn't exist in reality, only in your head.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It shows great intellectual laziness on your part._

 

It isn't laziness. I'm happy to share my experiences with anyone who makes an effort to understand what I am saying. They don't have to agree... just try to understand. I just didn't see any point in continuing after two or three examples of blind absurdity in a row.

 Records don't get produced by theory alone. Mixing is an art form of selected emphasis, just like photography and painting and drawing and every other artform. You can paint a picture mechanically perfect, but it will suck anyway because it takes brains behind the technology to know how to apply it creatively.

 In any case, I'd be very interested in hearing your miking theories, gregorio. I'd done a lot of work with recording vocals and I've always preferred the Neumann U87, but I've found the performance of the mike pres and gates makes a huge difference in the quality of the recording. Vocals are a lot more dynamic than anything else it seems.

 See ya
 Steve


----------



## gregorio

colonelkernal8 - It's even worse than that. I've been recording hi-res digital since 1992, I should imagine well before crowbar or anyone else here had even heard of hi-res digital. I still work today at the cutting edge of technology. The system I currently use is capable of 48bit 192kFs/s. It is my job to know exactly how the latest professional digital audio technology should be used as I'm a certified expert and instructor for ProTools, the hardware/software that is most commonly used worldwide for the recording and production of both music and sound for film/TV.

 I'm not trying to blow my own trumpet here, just show how ridiculous the argument is.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I've been ....... music and sound for film/TV._

 

argumentum ad verecundiam
 It's a logical fallacy. Again, basic stuff.
 Your qualifications have no bearing upon the truth-value of my arguments.


----------



## grawk

Ok, in fairy land, things can be better than in worlds with physics.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_It isn't laziness. I'm happy to share my experiences with anyone who makes an effort to understand what I am saying._

 

Your experience has no bearing upon the theoretical limits of the technology.
 The practical limits are are asymptotic to the theoretical ones--it just takes more careful engineering.

 And gregorio, saying "what you want is not possible" needs to be backed up. Why is it not possible? You tried arguing that already with microphone deficiencies but I countered it. Then?


 I'm still not sure what parts of my argument exactly is being disagreed with, and how you people support your disagreement, with anything besides anecdotal experiences which cannot be generalized to what is possible.

 I'm going to try a constructive approach.

 If an air pressure waveform can be created at the ear canals that is sufficiently close to what the person would have experienced had he been in a given location at the performance, then would you agree that that is transparent?? (let's say that sufficiently close is that the difference between waveforms would be < 10 ppm, or whatever number you feel is below any possible physiological ability to resolve)


----------



## grawk

Why not move this discussion of what is possible in fairlyland out of the discussion of hte differences between 16 and 24 bit?


----------



## gregorio

Hi bigshot,

 Ah yes, the good ol' U87. A great mic but a little weird in that it sounds great on an analogue mix but when using it with high quality digital system it's surprisingly noisy. Significantly higher noise floor in my experience than say a Beyer MC740. The U87 used well can give a lovely rich sound, whereas the Beyer is more clinical. The Beyer is quite a rarely used mic though for some reason. The other big player in the large diameter condenser market is the AKG C414 which is a good work horse but I can't think of many occasions where I would prefer it to a U87. If it's the richness of the U87 you like, have you tried the U47?

 When you start using quality gear like any of the mics above then the effect of the mic-pre becomes more pronounced. I always liked the Avalons but they were far from transparent. I really liked the DigiDesign mic-pre, it was based on the Grace Designs pre, which was one of the most transparent pres I've used. You've still got to go a long way before you can beat the quality of the Neve pres, the Focusrite Reds were great too.

 Vox are always awkward in a mix, there is usually no option but to use a fair bit of compression to get it to cut through, especially if there are some distorted guitars in the mix. I personally don't use a gate when recording vox. I find the noise floor of the Beyer MC740 doesn't warrant the artefacts of using a gate. On the very rare occasion when there is a bit too much noise, I would rather process it out in the software.


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_Why not move this discussion of what is possible in fairlyland out of the discussion of hte differences between 16 and 24 bit?_

 

I have no interest on what's possible in fairyland. Your mention in a previous post of physics shows how ignorant you are, since you cannot make one argument from physics against what I've said--all of which is well within the realm of physics. As a scientist, I resent that you would imply I've said anything that does not fit within a scientific worldview.

 I intend to describe in more detail what I see as a manifestation of a transparent system.


----------



## grawk

This isn't for your manifestation of a theoretical system, tho, it's for the differences between 16 and 24 bit audio. Using real world technology. If you go to a more theoretical discussion topic, you'll get people who are interested in theory, rather than practice.


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_This isn't for your manifestation of a theoretical system, tho, it's for the differences between 16 and 24 bit audio. Using real world technology. If you go to a more theoretical discussion topic, you'll get people who are interested in theory, rather than practice._

 

The theory sets the limits. Then practice needs to strive to maximize what can be done within them. That is behind everything I've said. I'm myself working on two of the things I've identified as necessary for transparency as I defined it previously.


----------



## grawk

do they involve 16 or 24 bit currently existing recording equipment?


----------



## Crowbar

No, but neither do anyone else's recent posts in this thread. Everything has already been said in the 16 v. 24 bit issue.


----------



## grawk

ok, I'll just remove a discussion I'm interested in (current real world technology) from my list so you can talk about fairyland in a thread about the real world.


----------



## Crowbar

How am I talking about "fairyland"? I can describe in technical detail a transparent implementation--using parts one can buy and make now.


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## grawk

Until it goes from "I can describe" to "Here is the product for recording, here is the product for playback, and here's how it's better", it's in fairyland, but this is my last visit to this thread.


----------



## gregorio

Crowbar - "If an air pressure waveform can be created at the ear canals that is sufficiently close to what the person would have experienced had he been in a given location at the performance, then would you agree that that is transparent??"

 This part of your statement is where your problem lies crowbar, with the If an air pressure wave can be created... It 's the word "if" that's creating all the problems because you are talking about a hypothetical not about what technology is actually capable of. An air pressure wave cannot be created which is transparent. It is simply not possible to capture all the reflections present in an acoustic environment, even it it were, all microphones and all the other pieces of equipment in the recording chain have slightly different responses to frequencies and therefore colour or change the sound. The same is true of your DAC, amp, speakers and listening environment.






 This is a typical frequency response chart of a mic. The graph would have to be completely flat though all frequencies for it to capture transparently and even then the graph changes and worsens drammatically when you go off-axis, so all the reflections in a concert venue captured by the mic are going to have an incorrect amplitude and frequency relative to what your ears would perceive during the performance. Reflections hitting the mic outside of it's polar pattern will not be captured at all. In this scenario it's not uncommon to have to generate reflections electronically, the unit that does this is called a reverb unit.

 Crowbar - This is the proof that you asked for. You could encode this mic's output at 1000bit and ten million samples per second, it's not going to change the fact that no piece of gear in the recording chain acts transparently, ever!


----------



## Crowbar

Yes, it is a big if. I wanted to establish the rest first, then explain how that if can be achieved. I'm started going step by step because then the areas of agreement and those of disagreement become clear, so we don't have a misunderstanding.
 Do microphones not work on inverse principles as transducers? That is, various driver technologies have duals as various microphone technologies (dynamic, electret, etc.)? If so, can I assume that a driver technology that would produce much flatter response than others can be used to build a flatter microphone?

 In the case of a no, is there anything fundamentally wrong with equalization (both magnitude and phase) using high precision in the digital domain?

 (I'll ignore the fact here that inverse distortion can be used to deal with nonlinear distortions as well and has been done in RF applications for some time, by assuming harmonic distortion and other nonlinearities in an otherwise excellent microphone driver would be very very low)

 The reason for my question is that a technology from the 80s actually allows two orders of magnitude less distortion driver than current dynamic drivers for reasonable power consumption, with the ability to almost directly decrease distortion further by lowering efficiency. I'm referring to glow discharge drivers based on Hill's patent, and including recent developments which make them more practical (and also being able to make them into headphone drivers which allows covering the full frequency band without need for dynamic woofers). Then my IF becomes simply that a dual version of this can be created as a microphone technology.

 Anyway, I'm off to bed for the moment. I'm sorry if I used strong words before but it's easy to get frustrated on head-fi


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Your experience has no bearing upon the theoretical limits of the technology._

 

I'm sorry. I'm just not interested in discussing theoretical sound quality. I'm only interested in the practical application of technology. You can theorize about dividing the fractions further and further in hopes of making a reproduction more theoretically "real". But to me, that's like jumping off higher and higher ladders in hopes of being able to learn to fly. I'm only interested in how people can make their home stereos sound better.

 In the home, the difference between redbook and higher bitrate sound is inaudible and totally unimportant.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I intend to describe in more detail what I see as a manifestation of a transparent system._

 

Knock yourself out, champ. I think I'll go get a sandwich.


 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I personally don't use a gate when recording vox. I find the noise floor of the Beyer MC740 doesn't warrant the artefacts of using a gate._

 

It isn't for the noise in the system, it's for the tiny inhales, lipsmacks and teeth clicks. We always use a gate for VO and looping for TV. It's a LOT easier than editing all that stuff out manually. Otherwise it sounds like the announcer is making love to your ear. Super creepy!

 See ya
 Steve


----------



## gregorio

The mic graph I gave earlier is for a condenser (capacitor) mic. Condenser mics have the most elegant ballistics and therefore are generally considered to be the most transparent but as you can see from the graph above, even a condenser mic is far from transparent. We commonly use mics of all different types of design; dynamic, ribbon and condenser being the most common. Mics are just tranducers, picking up the SPLs and converting them to electrical waveforms.

 Remember though the mic is just one piece of equipment in the chain. Mics output a tiny signal that requires massive amplification before the signal is usable, a 50db boost is not uncommon. So then the frequency response and colouration of the mic-pre becomes an issue. The output of the mic pre then goes to a mixing desk or ADC, these both colour the sound, so on almost ad infinitum.

 The is nothing fundamentally wrong with using EQ to equalise out the mic's imperfections, in fact this is exactly why EQ was invented in the '40s. Afterall, that is why it's called Equalisation! However, EQ like all other studio processes have artefacts.

 The problem is how can you measure what the ear is hearing? You need to capture the sound so you can analyse it. How are you going to capture that sound without using a microphone which is going to add some colouration. You can't find out exactly what is going into the ear, so you've got nothing to compare your mic response to. It's a vicious circle.


----------



## gregorio

Bigshot - For VOs and ADR I often use the strip silence feature in ProTools, which I find usually less obtrusive than a gate. Careful playing around with the axis of the mic can sometimes reduce inhales, lip smacks, etc., to a point that is acceptable. I can edit in ProTools pretty quickly and although still slower than chucking on a gate, I much prefer the results. I've just so rarely managed to find a gate setting that doesn't compromise audio quality.

 The other option that has worked well for me is a very quiet and very dead VO booth and placing something like the Beyer MC740 further away from the "talent's" mouth. This eliminates most of the intimate mouth sounds and a good compressor can easily bring back some of the presence if required. Providing the booth is really dead you don't suffer from the problems of picking up the acoustic, so there's no noticable problems with mixing in some room tone with the ADR or indeed if the dubbing (re-recording) engineer wants to apply echo/reverb.


----------



## Manny Calavera

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Wow, a lot of conflicting information, most of which is inaccurate.

 24bit vs 16bit. We use 24bit during recording simply to provide additional headroom for unforseen transient peaks. When mastered down to 16bit the dynamic range is still 96db. 24bit provides 144dB of dynaic range but is pointless for playback. There is no way anyone could tell a difference between 24bit and 16bit unless you are a golden ears listening in an anaechoic chamber to 60 grand monitors and even then there would be no point! If you think you can hear a difference this is purely down to the way your DAC is handling the reconstruction.

 Now on to the higher sample frequencies. Let me make this clear, there is not and cannot be any audible difference between a sample rate of 48kFs/s, 96kFs/s or 192kFs/s. No human being can hear above 22kHz so a sampling freq of 48kFs/s is more than adequate. So why was 96kFs/s invented? To con more money out of people who are understandably ignorant of digital audio theory. There is a technical reason for 96kFs/s, it makes the anti-alias filter on an ADC easier and cheaper to impliment. If you think that 96kFs/s sounds better than 44.1kFs/s then what you have proved is that the music was originally recorded using cheap ADCs.

 While I'm at it, Up-sampling. Forget it, it's another way to con money out of you. There is no way an upsampled recording can sound better than the original unless the manufacturers have deliberately created a DAC with a rubbish reconstruction filter at 44.1kFs/s.

 Hope this helps,

 Gregorio_

 



 One of the more well respected MOT's on here told me almost the exact same thing very recently.Also,I have a close friend who's an electrical engineer,who spent 17+ years working on DAC's and DSP's for TI,and he
 told me nearly the same thing.Lots of snake oil,and BS in the HiFi world.


----------



## bigshot

gregorio- I'm afraid my talent just doesn't stand still and the spit shield would be dripping by the end of the session!It was quite a trick. The dial on Ren & Stimpy would go from a whisper to the most extreme yell at the drop of a hat. The engineer was trained by Sam Horta (of Star Trek fame)- a real pro. When you work with a really good engineer, it's quiet behind the board and everything just works. I seem to remember hearing him mentioning that the mike pre, compressor and gate cost more than a very nice house. I think we tried a Beyer once, but the voice actors kept blowing it out. (We recorded a lot of screaming.)

 See ya
 Steve


----------



## gregorio

Bigshot - absolutely. I've worked with a few top engineers and I've always been impressed at their professionalism, speed and skill. When you move to the high end of pro-audio gear it gets really expensive really quickly. Most of the mic-pres I mentioned are going to cost close to 5 figures for just 4 channels. I could easily imagine spending close to $30,000 for a great mic-pre, compressor and gate.


----------



## Crowbar

Quote:


  Originally Posted by *Manny Calavera* /img/forum/go_quote.gif 
_One of the more well respected MOT's on here told me almost the exact same thing very recently.Also,I have a close friend who's an electrical engineer,who spent 17+ years working on DAC's and DSP's for TI,and he
 told me nearly the same thing.Lots of snake oil,and BS in the HiFi world._

 

You talked to engineers--they're missing half of the equation. You need to talk to physiologists and otoneurologists as well.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_as you can see from the graph above, even a condenser mic is far from transparent_

 

Moreover, I assume that response is not identical when direction for the source to the microphone changes. That would make equalizing impossible. However, in the binaural case all sound is channeled through the ear canal, so only a very narrow range of directions matters.
 Equalizing artifacts are a function of how good your filter is. In the case of digital filtering, while current DSP chips are not quite there, a computer can do in real time very high quality filtering in double precision floating point, such that any artifacts would be far below the audible range.
 Any filter that can match a target response (in this case inverse of the microphone one) followed by phase correction, such that any artifacts are below -120 dB can be regarded as doing its job transparently.

  Quote:


 Remember though the mic is just one piece of equipment in the chain. Mics output a tiny signal that requires massive amplification before the signal is usable, a 50db boost is not uncommon. So then the frequency response and colouration of the mic-pre becomes an issue. The output of the mic pre then goes to a mixing desk or ADC, these both colour the sound, so on almost ad infinitum. 
 

Yes, but distortion is not magic and can be engineered to whatever level necessary. I have built a DAC prototype with THD of -120 dB over most of its range (-115 dB at 20 kHz full-scale; with lower signals it's less actually, though eventually noise floor limits that). If I can do that, someone can build a comparably good ADC (and probably has already). Now, obviously THD is only part of the story. Distortions like PIM are likely not captured in that measurement, various harmonics are important to different extents perceptually, and there's stuff like thermal memory distortion. But when one is aware of the various mechanisms, they can be dealt with. Right now I'm trying to figure out how to minimize the thermal distortion in my analog design as I suspect is very important issue. But ultimately, again, there's no magic here--it's physics and any distortion can be measured and then decreased by reengineering the equipment. I'm definitely NOT of the school that a piece of reproduction or playback gear should add anything to the sound, even if some people might perceive certain colorations as euphonic.

  Quote:


 The problem is how can you measure what the ear is hearing? You need to capture the sound so you can analyse it. How are you going to capture that sound without using a microphone which is going to add some colouration. You can't find out exactly what is going into the ear, so you've got nothing to compare your mic response to. It's a vicious circle. 
 

On the side of electronics, of course, those parts of the chain can be optimized easily since John Curl and Bob Cordell have described various ways to measure distortion approaching -140 dB.
 With the microphones and speakers/headhones, it's another matter though.
 Still, don't forget that microphone was measured, the one you posted 
	

	
	
		
		

		
			




 One can always do iterative refinement. The process is basically identical to iteratively solving a linear system represented as a matrix.
 You can create an extremely accurate sound source of a given frequency. Frequency generators exist that can create very pure sine waves. Any physicist is able to take a source and measure to extreme precision the total acoustic power that is put out by it (not by measuring with microphones, so no chicken-and-egg problem here). Then this is a reference source. Repeat for a bunch of frequencies. But this is overkill; I would bet real money that iterative refinement is sufficient.

 Here's the response of corona discharge headphones: http://membres.lycos.fr/plasmapropul...y_response.jpg
 Now if you look at speakers, the glow discharge speaker is about an order of magnitude flatter than good corona discharge ones. So one would expect the same improvement for headphones, and I bet more can be squeezed out if you design the enclosure with good sims in Mathcad. After all Hill didn't have such tools back in the 80s and still got amazing result with the speaker.


----------



## gregorio

Crowbar - It's not about slightly improving specs, to make your concept work you would have to redesign the whole of recording theory. Let's say you create an EQ to do your filtering and your EQ has a noise floor of -120bB. In a modern mix you could have 100 audio paths, in film sometimes over a 1,000 audio paths. If you put an EQ on each one of these paths, you are summing together the noise that each EQ is creating. If you were to sum together say 100 of them, your noise floor would have rizen to the point where you now have 8 bit resolution audio! Then you've got to add all the other processors used in a digital mix, plus the noise floor of the mics, amps, cables, connectors and of course the noise floor of the venue. All told, that's a lot of noise, that's why pretty much any type of music recording you listen to has a dynamic range of less than 50dB (above the noise floor), some of it only has a range of 10 or so dB. Even the most dynamic of symphonies will rarely exceed 50dB and if we made the dynamic range bigger it would put the quietest sections of music well below the noise floor of all but the very finest audiophile systems. So as you can see, even with a dramatic improvement in both studio equipment and consumer equipment we are still no where near the 96dB range of 16bit, let alone the 144dB range of 24bit.

 There were quite a few misconceptions in your last message, for example: "while current DSP chips are not quite there, a computer can do in real time very high quality filtering in double precision floating point". I was using DSP chips which did high quality real-time 48bit precision filtering a decade ago! 

 "Yes, but distortion is not magic and can be engineered to whatever level necessary." No it can't, there are always going to be errors but more importantly, there is always going to be a noise floor, unless you can work out a way to record in a vacuum.

 "However, in the binaural case all sound is channeled through the ear canal, so only a very narrow range of directions matters." Yes but the polar patterns of mics are very different to pick up pattern of the ear. The ear is suprisingly good at hearing sounds coming from different directions in a horizontal plane.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Let's say you create an EQ to do your filtering and your EQ has a noise floor of -120bB. In a modern mix you could have 100 audio paths, in film sometimes over a 1,000 audio paths._

 

OK, but I'm only interested in acoustic music--classical and some jazz. But even not limiting to that,

  Quote:


 If you put an EQ on each one of these paths, you are summing together the noise that each EQ is creating. If you were to sum together say 100 of them, your noise floor would have rizen to the point where you now have 8 bit resolution audio! 
 

Say you sum them in the digital domain, so there's no noise from the mixer itself. 100-fold the noise is a 40 dB increase. However, you forget that the signal also adds up. If you have two high level signals from two sources at the same time, obviously you can't just add them without attenuating them as it will clip--but when you attenuate it, that also attenuates the noise down. It's easy to see that the attenuation needed is to some extent proportional to the number of sources you're summing (though each one would vary based on its level). That mostly counters the noise addition, so the actual noise increase will be a fraction of that 40 dB.

  Quote:


 Then you've got to add all the other processors used in a digital mix, plus the noise floor of the mics, amps, cables, connectors and of course the noise floor of the venue. 
 

I already showed an example of a microphone with the ADC in the capsule itself, so the rest of the path to storage would be digital only, thus no new source of noise exists.
 And I already mentioned that in a classical concert, the peaks can do 110 dB from the quietest tones--which are themselves a bit louder than the noise floor, so you have more like 115 dB range in a good concert hall for some music.

  Quote:


 if we made the dynamic range bigger it would put the quietest sections of music well below the noise floor of all but the very finest audiophile systems. 
 

First you argue that playback systems are not needed with high dynamic range because recordings don't have it, and now you argue that recording systems with high dynamic range aren't needed because playback ones don't have high dynamic range... it's circular.
 The more one side is improved, the more there is reason to improve the other.

  Quote:


 There were quite a few misconceptions in your last message, for example: "while current DSP chips are not quite there, a computer can do in real time very high quality filtering in double precision floating point". I was using DSP chips which did high quality real-time 48bit precision filtering a decade ago! 
 

How many taps in the filter? I wouldn't consider FIR filtering with less than 2000 taps per band in the equalizer--that adds up to a few thousand taps. And you definitely cannot do that with even the most recent SHARC DSP--unless you use a bunch of them.

  Quote:


 No it can't, there are always going to be errors but more importantly, 
 

The ear is a machine and subject to the laws of physics. The fundamental limitations of the ear can be matched by an artificial creation because they're both subject to the same laws of physics.

  Quote:


 there is always going to be a noise floor, unless you can work out a way to record in a vacuum 
 

No disagreement here. The ear has a noise floor too--and below that, air itself has thermal noise. The equipment only needs to be as good as the ear, and no better. I keep saying 120 dB because that guarantees you can't hear it in any circumstances (unless you turn the volume up to the point where 0 dB peaks will cause hearing damage).

  Quote:


 Yes but the polar patterns of mics are very different to pick up pattern of the ear. The ear is suprisingly good at hearing sounds coming from different directions in a horizontal plane. 
 

You don't seem to understand binaural.
 That's why in binaural setups the microphones are in a dummy head's ear canals. The 3D geometry of the ear is a system that takes in 3D sound and outputs 3D sound filtered with a frequency response depending on each input source's direction, sending sound to the inner ear (or microphone, in the case of a binaural setup) that is in one direction only. See sketch below--all the 3D processing is done by the ear and encoded in 1D sound.


----------



## gregorio

Crowbar - Binaural mics approximate what the ear will pick up. The are a useful tool but not a perfect tool.

 Processing power, I was refering to an EQ unit's filter not a fir filter. If you want to build a system which costs hundreds of thousands just to provide the processing power for a whole bunch of fir filters to counter the frequency response of mics, go right ahead, no one will buy it though.

 Even you are now changing your tune, first you started on about 24bit audio, now you are talking about 120dB capable system, well that is a 20bit system. 120dB range (above the noise floor) is way too high. Even a world class professional studio is not capable of representing that range. Even if you could represent that range, who would want to? To hear the quietest sounds you would have to put the level so high that the loudest sounds would put you in hospital! No one in their right mind is going to do that, which means that everyone will have to turn down their amps and not hear the quiet sections. I can see an advantage for audiofiles of increasing the current recording range of symphony orchestras from the 40-50dB range to say 70dB but that will mean the majority of listeners will be out of range with their systems. Even if we were to make the leap to 70dB we are still well within the range of 16bit audio. From an audience perspective, during a live performance, they are rarely if ever going to hear peak levels from a symphony orchestra of more than about 90dB (including noise floor). Your would only experience 115dB if you were a viola player sitting directly in front of the trombone section blasting away at FFF.

 Not many professionals would choose a mic with a built in ADC. You are tied to an ADC which is unlikely to be of the quality of a standalone studio ADC, same with the mic pre-amp. I should imagine the benefits of this mic to be more apparent in the live sound market rather than the studio market. Beyer quote a range of 115dB for their digital mic, I very much doubt if this figure is attainable in practical use. Many mics are theoretically capable of this dynamic range but in practice about a 70dB range above the noise floor is all they can manage. You have to learn with audio equipment that quoted specifications cannot usually be obtained in a practical recording situation.

 Enough of this now crowbar, it's getting boring. I've given you perfectly good explanations of a number of issues but you seem unable or unwilling to comprehend them. You still have no idea about the practicalities of producing music for a mass market. You think you can revolutionise the recording industry and the consumer market, be my guest, go knock yourself out.


----------



## bigshot

Equalization isn't applied mechanically in a mix. It's used creatively to create contrast, separation and balance between various instrumental voices. You don't mix to flat, you play back flat. I'd explain why, but I suspect no one is interested except those who have worked in the business and already know the answer.

 See ya
 Steve


----------



## scompton

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_From an audience perspective, during a live performance, they are rarely if ever going to hear peak levels from a symphony orchestra of more than about 90dB (including noise floor). Your would only experience 115dB if you were a viola player sitting directly in front of the trombone section blasting away at FFF._

 

Last night, I saw the NSO play Mahler's 2nd. The crescendo at end was just short of pain level where I was sitting in row DD. The orchestra was so loud you could barely hear the pipe organ. My guess is that is was probably over 115dB. I don't know how the musicians can stand it. I imagine the sound was over the pain threshold on stage, at least for me.


----------



## bigshot

Would you listen to it at home that loud?

 See ya
 Steve


----------



## scompton

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Would you listen to it at home that loud?

 See ya
 Steve_

 

Not a chance. I wouldn't have been happy if it was that loud for more than the minute or so at the very end. I've pretty much stopped going to rock concerts because of how loud they are. I recently bought the Etymotic ear plugs to wear at loud concerts. 

 Because my wife is hard of hearing and doesn't like noise, I tend to listen exclusively to headphones. I measure how loud I'm listening periodically and it's usually between 65-70dB. I've never measured a loud peak. Last night's concert makes me think I should. I probably will this week end. I'm pretty sure that the crescendo on the CD I have of Mahler's 2nd is not as loud as the live concert.


----------



## bigshot

I had some digital Karajan records that were so broad in dynamics it made them a pain to listen to. I kept jumping up to adjust the volume. 50dB is really as much dynamics as you really need for home listening. Most recordings are much less and they sound better for it. Compression isn't a bad thing, over-compression is.

 See ya
 Steve


----------



## scompton

I remember hearing stories back in the 80s about turntables having problems tracking the Telarc digital LP of the 1812 overture. Supposedly, the needle would jump the track when the cannons went off. Who knows if it was true, but it did make a good story.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Equalization isn't applied mechanically in a mix. It's used creatively to create contrast, separation and balance between various instrumental voices._

 

Again, only some music types depend on mixing. I'm more concerned with recording acoustical music, and I want to be fooled into thinking I'm listening to it live.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Binaural mics approximate what the ear will pick up._

 

And the approximation can be improved by engineering until a difference is unresolvable by the ear/brain.
 Microphones can be improved. ADCs can be improved. Ballistic gel can be used for the head. etc. There's active research in all these areas.

  Quote:


 Processing power, I was refering to an EQ unit's filter not a fir filter. 
 

Er, you can build an equalizer out of FIR filters.

  Quote:


 If you want to build a system which costs hundreds of thousands just to provide the processing power for a whole bunch of fir filters to counter the frequency response of mics, go right ahead, no one will buy it though. 
 

Why would I? A modern computer will do just fine. Which is what I said already!

  Quote:


 Even you are now changing your tune, first you started on about 24bit audio, now you are talking about 120dB capable system, well that is a 20bit system. 
 

I was supporting 24 bit over 16 bit. Actually, if you didn't have to use linear PCM and could use say a log-encoded PCM, then 16 bits would be plenty (this is because the size of a difference between two levels detectable by the ear varies depending on the absolute loudness). But from available standards, 24 bit is the one to choose (HDCD is not true 20 bit, of course, so I'm ignoring it).

  Quote:


 Your would only experience 115dB if you were a viola player sitting directly in front of the trombone section blasting away at FFF. 
 

You're talking about sustained tones; I'm talking about peaks of occasional transients.

  Quote:


 You still have no idea about the practicalities of producing music for a mass market. You think you can revolutionise the recording industry and the consumer market, be my guest, go knock yourself out. 
 

I'm not interested about the practicalities of mid-fi productions! Considering the ever-falling quality of modern music, I'd rather it not even be recorded.
 What I am interested in is in pushing the very top end and being able to have transparent archival recordings of great performances.

 People already record live events themselves. The continued spreading of personal technology allows people to capture more and more of their experiences, visually and audibly. Some enthusiasts actually do care about quality and will strive for transparency. You're so entrenched in your old ways of thinking that you're failing to anticipate the very near future. But that's your mind protecting you from the fact that in a dozen years you'll be obsolete.


----------



## mikeg88

Quote:


  Originally Posted by *Davesrose* /img/forum/go_quote.gif 
_If you like modern rock then I'd say 24bit recordings aren't worth it. The few rock SACDs I've bought have not been that much different from CDs. Beck's Sea Change is nice on SACD, but all his other albums are CD (and through my DAC sound about as good).

 The areas that I find SACD sounds great at are DSD recorded material (modern classical) and older recordings where they didn't compress the bagebers out off the masters. So far, that tends to be classical and jazz as well.....haven't really found a rock SACD album that impresses me over CD. If a SACD is coming from a good master, I find it has a better soundstage and nicer detail over CD....not that CD is bad or anything. I have a lot of CDs that blow me away because of the performance and engineering that went into it.

 And my opinion about upsampling 16bit recordings to 24bit is that it doesn't inately do anything. The D/A might make things sound more pleasant or vibrant, and that's an effect of the processor itself: not with it being 16 or 24 bit._

 

whoa where do you actually physically shop for SACDs?
 do they sell them in stores?


----------



## TheMarchingMule

Quote:


  Originally Posted by *mikeg88* /img/forum/go_quote.gif 
_whoa where do you actually physically shop for SACDs?
 do they sell them in stores?_

 

Yup, they're here and there...hard to come across though in brick-and-mortar stores, of course.


----------



## bigshot

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_I remember hearing stories back in the 80s about turntables having problems tracking the Telarc digital LP of the 1812 overture. Supposedly, the needle would jump the track when the cannons went off. Who knows if it was true, but it did make a good story.
	

	
	
		
		

		
			



_

 

It was actually the Carmen suite. I had that record and there was a bass drum wallop that distorted no matter how you tried to track it. It was cut so far out of spec it was pitiful.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Again, only some music types depend on mixing._

 

Yeah, right. Some music records itself untouched by human hands.

 See ya
 Steve


----------



## gregorio

Crowbar - "Again, only some music types depend on mixing." You are talking about revolutionising recording technology but you don't even have the level of knowledge of a school child. If you don't mix music you have two choices, output it using ethereal osmosis or record the two mic input straight to CD. The first of course doesn't exist and the other would put recording methodology back 50 years.

 It's really unbelievable. "You're so entrenched in your old ways of thinking". That's right, I'm entrenched in cutting edge technology and methods of production and your progressive idea is to take recording methodolgy back to the 1950's. If having a good understanding of how the most cutting edge recording studios work is an old way of thinking, I'll take that over complete and utter ignorance any day!

 Your problem is that you are trying to consider advanced recording technology and methodology when you haven't got even the faintest idea of the basics.

 Others on here have said that 50dB is perfectly sufficient yet you crowbar want to increase this a few thousand times. Who on earth is going to buy a recording which could not be played back even in a world class recording studio. If you could design somewhere to play it back then whoever listened to it would not just slightly damage their hearing but would utterly destroy it forever. Your true 24bit master would put most people in hospital and kill a percentage. So you carry right on developing a system which no one in their right minds would listen to and which is far too dangerous to be allowed in the marketplace.

 Just to make it clear from what Daverose wrote, upsampling 16bit to 24bit is not going to make a blind bit of difference. Can anybody explain to me the difference between a 16bit audio file and a 24bit audio file with the 8 least significant bits set to zero? Another question, hands up anyone who thinks that 24bit gives any more audio resolution than 16bit or 8bit.

 Scompton - I've worked with the NSO and know Mahler's 2nd pretty well. These big post-romantic symphonies provide some of the biggest challenges to recording. If you think 115dB was too loud, multiply that by a factor of 50 or so and you're approaching the level that 24bit digital (144dB) is capable of encoding. See how ridiculous the argument about putting out recordings in 24bit really is.


----------



## Crowbar

50 dB? LOL! That's a range covered by 9 bits. By your logic, it's not about 16 vs 24 bit; we should be going back to 8 bit audio really!

 You seem to be confusing the envelope of the music, which indeed may only vary 50 dB or less in many cases. I'm talking about the range of individual components, which is much greater. The ear masks low order harmonics of a given component, but has little masking of other components of the signal even if they're quieter--and even when not perceived as independent components, they can be perceived as a different coloration of the sound.
 Your comments about consumer listening setup limitations are misleading because you are not including the fact that signals can be heard below the noise floor, because noise is spread throughout a wide spectrum, whereas individual signal components are narrowband and it's been proven the ear can hear them several dB below the noise floor. This has already been pointed out.

 By the way, if a binaural recording is to preserve proper sound, it needs to do exactly that--go straight from microphone to CD. At most it will be attenuated and dithered down to make the best use of 16 bits, if that's the output format. If you mix it with anything else, then you immediately ruin the soundstage--and a binaural recording listened through headphones is the ONLY way you can capture and reproduce the 3D sound environment to a high degree.


----------



## P_A_W

Hi Guys,

 Really interesting thread... My ears aren't even good enough to require 16/44 for music. 
	

	
	
		
		

		
		
	


	




 I guess the sensitivity of our ears falls precipitously at the edge of the audible range (20Hz-20kHz). While no one needs the dynamic range of 24bits for a middle c, what about sounds at the lower end of the spectrum where we have very low intrinsic sensitivity? I guess I am thinking explosions in movies played in home theater more than music. 

 --PAW


----------



## Crowbar

People like gregorio/Bigshot are what's holding up back progress in audio. The crappy recordings they make means when you make some improvement in your playback system, you get to hear more how crappy their recording is, and think why bother. gregorio/Bigshot and their kin are the fountain of mediocrity.


----------



## gregorio

Crowbar - I'd rather be a fountain of mediocrity than an ignorant moron. The real shame is that your ignorance is so deep and so complete that you don't even realize how ignorant you are. You then argue from your standpoint of complete ignorance with people who aren't ignorant. It's so ridiculous that it's funny. Have you ever heard any of mine or bigshot's recordings? Then how do you know they are crappy, just more ignorance. It's like tryng to explain to a 4 year old that even though metal is heavier than air, a plane can still fly. At least a four year old is capable of learning and can at some point get past their ignorance!

 "50 dB? LOL! That's a range covered by 9 bits." Exactly, my god, we're runnning into the possibility that you might just be starting to understand the principles. Why do you think I started off by saying that even 16bit is far in excess of what even the most hardened (and rich) audiophile would find useful. There are probably very few if any recordings (analogue or digital) on the market which exceed 50dB dynamic range. At this point in time, with current playback technology, extending this range will make the music unplayable on the average home system.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_People like gregorio/Bigshot are what's holding up back progress in audio. The crappy recordings they make means when you make some improvement in your playback system, you get to hear more how crappy their recording is, and think why bother. gregorio/Bigshot and their kin are the fountain of mediocrity._

 

Seriously, where are your excellent recordings ? - I will nip out and buy a few. Do you have any published works on this topic that we can analyze ?

 For an adult you have some real self control issues, that is of course assuming you are an adult, something which we have little evidence of so far. That Bighot and Gregorio mock you gently is no excuse for such childish conduct.

 EDIT: I posted the above before I saw Gregorio's _ignorant moron _comment.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Seriously, where are your excellent recordings ?_

 

Seriously, whether I record or not has no bearing on my argument. It's basic logic.
 If my car runs poorly and I complain, do you ask me to build a better car? Of course not; yet your comment above is equivalent to that.

  Quote:


 mock you gently 
 

One cannot mock gently; mocking implies contempt, and so this is an oxymoron.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_ignorant moron_

 

Ran out of arguments, so has no choice but to resort to name calling. 
	

	
	
		
		

		
		
	


	




  Quote:


 Have you ever heard any of mine or bigshot's recordings? 
 

I should hope I never do.


----------



## Leporello

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Who knows if it was true, but it did make a good story.
	

	
	
		
		

		
		
	


	


_

 


 It is true. I have personally heard that happen in a record store.


 Regards,

 L.


----------



## grawk

For playback, bit depth is pure marketting. If you were to make use of even the dynamic range of cd, your ears would bleed. Therefore, any improvements to be found in playback won't involve more bits available for playback. 

 Binaural recordings will remain a niche, because they only sound good if you're listening with headphones. Most people don't listen that way, and therefore, most recordings won't be made that way. 

 You complain that you haven't heard a believable accoustic recording. I'd suggest that you aren't trying very hard. I've heard recordings made with even modest equipment, played back on good equipment that to my ears are indistinguishable from the original performance. So if you're going to continue down the path of insulting everyone who's ever made a commercial recording as being "part of the problem", you're going to have to do something to demonstrate that your fairyland solutions are better.


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_this is my last visit to this thread_

 

Either you're a liar, or you have Alzheimer's.


----------



## grawk

Or I changed my mind. It's comments like that that point out just how out of touch with how adults interact you really are.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Seriously, whether I record or not has no bearing on my argument. It's basic logic.
 If my car runs poorly and I complain, do you ask me to build a better car? Of course not; yet your comment above is equivalent to that.


 One cannot mock gently; mocking implies contempt, and so this is an oxymoron._

 

The best way to illustrate that your view of the world has some inherent advantages is to show them, in this context the best way you can illustrate the benefits of your world view is to do better than those you mock. Your comment is not like saying my mechanic is crap, to which the answer is find another mechanic, your comment is like saying ALL mechanics are crap.

 MOCK

 It also means ridicule, to laugh at or make fun of. Also gently mock has definitely entered common usage. Parody also _makes fun of_ but this does not mean contempt is necessary, look at Mel Brooks' parodies, many stem from an underlying affaction for the genre he mocks.

 From your failure to answer my second question can I assume that your theory of the world is still embedded in your head and not published, even as an IEEE opinion piece ?


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_People like gregorio/Bigshot are what's holding up back progress in audio. The crappy recordings they make means when you make some improvement in your playback system, you get to hear more how crappy their recording is, and think why bother. gregorio/Bigshot and their kin are the fountain of mediocrity._

 

For someone so quick to point out alleged logical fallacies in others' posts, you certainly are quick to engage in _ad hominem_ attack. 

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Ran out of arguments, so has no choice but to resort to name calling. 
	

	
	
		
		

		
			



_

 

What a hypocrite you are.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Either you're a liar, or you have Alzheimer's._


----------



## Crowbar

nick_charles:
 But I'm not saying all, just a large portion. That is certainly not an unreasonable position. Just like most music that comes out is crap as well--I think many people will agree with that sentiment.
 I am already working on improving some aspect in audio, but for the moment it happens to be on the playback side, and this makes financial sense since the consumer market is large. However, I do suffer from the rarity of good recordings and I might consider the challenge as a further project.

 As for publishing: what theory? I made some points and backed them up. There's no theory here; it's common sense and you can't publish something trivial.

 grawk:
 That most people use speakers is irrelevant. Since when do we set the highest goals by the lowest common denominator?! (not to mention the irony of posting this on a headphone-centric site)


----------



## Crowbar

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_For someone so quick to point out alleged logical fallacies in others' posts, you certainly are quick to engage in ad hominem attack._

 

Ah, but I only invoked it in response to the other side using it; it was anything but quick. No point in a logical reply when there isn't one from the other side. The important thing is not to be the first one to quit a rational discourse.


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Quick? I only invoked it in response to the other side using it. No point in a logical reply when there isn't one from the other side. The important thing is not to be the first one to quit a rational discourse._

 

I wasn't aware that there was a "but mommy, he did it first" exception to the _ad hominem_ fallacy.


----------



## Crowbar

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_I wasn't aware that there was a "but mommy, he did it first" exception to the ad hominem fallacy._

 

It ceases to be a fallacy when the discussion is no longer one based on rational arguments. So in fact I did not use an ad hominem fallacy. For something to be a fallacy it must be by definition a part of a rational argument.


----------



## gregorio

Crowbar - "If my car runs poorly and I complain, do you ask me to build a better car? Of course not".

 In your case that's exactly what we are saying. If someone like you comes along demanding that their ford station wagon accelerates to 1,000mph, what is the mechanic supposed to do? The mechanic is going to realise that you're completely ignorant of how cars work. Blaming the mechanic for not making your Ford go at 1,000mph proves that you are an idiot and that even wanting your Ford to be capable of 1,000mph proves that you're mentally unbalanced! Either way they are going to laugh at you and tell you to go away and built the car yourself.

 Thanks for bringing up this analogy crowbar, hopefully you will understand it well enough to stop making such an idiot of yourself!


----------



## Crowbar

You're abusing the analogy, since what I'm talking about is doable by incremental improvements in engineering, whereas that is not the case with making a ford go 1600 km/h.


----------



## Febs

Your analogy wasn't a good one to begin with. Your claims in this thread are roughly equivalent to driver who says, "my car's performance is bad. I can build a better car." 

 "Well, what experience do you have designing cars."

 "My experience designing cars is irrelevant. But every other car in the world is inferior to the hypothetical car that I could design, and the fact that you all drive inferior cars makes you incapable of intelligently discussing cars at all."

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Thanks for bringing up this analogy crowbar, hopefully you will understand it well enough to stop making such an idiot of yourself!_

 

gregorio, I appreciate the experience and perspective you bring to this thread and I've enjoyed reading your posts. I would encourage you to stay focused on the merits of the arguments you are making and to avoid this type of attack.


----------



## gregorio

Crowbar - Building a car that can go at 1,000mph is just as do-able with incremental improvements in engineering as what you're suggesting with audio, only you are too ignarant to realise it! What's more, unlike your car analogy, you are wanting to take the audio technology to an extreme which would seriously injure or kill your audience. The only difference with the car analogy is that 1,000mph would not in theory seriously injure or kill your passengers, although it probably would in practice!


----------



## bigshot

Quote:


  Originally Posted by *P_A_W* /img/forum/go_quote.gif 
_While no one needs the dynamic range of 24bits for a middle c, what about sounds at the lower end of the spectrum where we have very low intrinsic sensitivity? I guess I am thinking explosions in movies played in home theater more than music._

 

This is a common misconception about dynamics. Peak level is peak level no matter what the bitrate. A 16 bit boom is the same as a 24 bit boom. The difference in dynamics extends downward, not upward. 24 bit is likely to have better resolution down in the range of the quietest whisper. But in a normal listening situation, you'd have to turn the volume up so high to hear the difference, you'd blow out your speakers when normal volume stuff comes along.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_People like gregorio/Bigshot are what's holding up back progress in audio._

 

I'm going to give you a little lesson that will probably be ignored, because you seem more interested in manufacturing reality than observing the real reality. But I got this lesson pointed out to me long in the past and I learned from it. Perhaps you can too.

 Ignorance is cool. It's just not knowing. No one can know everything. You obviously don't know a lot about music production. I have absolutely no doubt that you've never set foot in a recording studio and you have no idea how the records you listen to are produced. That's fine. It's not your job to know all that.

 But when someone comes along whose job it is and he takes the time to share his experience with you, you should listen and process what he has to say. Even if you disagree with his approach, you should be respectful because he is a guy who is actually doing it, while you are just typing about it on the internet.

 As I said before, ignorance is OK. Mark Twain wrote, "Everybody is ignorant- just on different subjects." The line that ignorance crosses where it becomes not OK is when it becomes willful ignorance. That's the refusal to know. There's a word for that- stupidity. Don't be stupid, kid.

 Disrespectful people don't end up learning very much, except that no one seems to want to spend much time with them.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Seriously, whether I record or not has no bearing on my argument._

 

The fact that I've never flown an airplane has no bearing on me telling a jet pilot that he is doing everything wrong.

 By the way, it's not inconceivable that you might have heard something that I recorded. But I'm quite confident that I've never heard anything you've recorded; and if you keep on like this, I probably never will.

 See ya
 Steve


----------



## gregorio

Quote:


  Originally Posted by *P_A_W* /img/forum/go_quote.gif 
_While no one needs the dynamic range of 24bits for a middle c, what about sounds at the lower end of the spectrum where we have very low intrinsic sensitivity? I guess I am thinking explosions in movies played in home theater more than music. 
 --PAW_

 

According to digital theory, 24bit provides absolutely no more accuracy, or resolution than any other bit depth. What 24bit gives you is a lower noise floor, with results as described by Bigshot.

 The audio you hear in cinemas is Dolby Digital, the bit depth specification for Dolby Digital is 16bit. There is no LFE on the market and certainly no consumer sub woofer that gets anywhere near the dynamic range of 16bit.


----------



## shinew

Just want to say that I've enjoyed reading this thread(minus the last 2-3 pages) & I've learned a lot!

 So just want to be sure that I now have corrected my previous misconceptions of 16bit vs 24 bit:
 1 - 24 bit will only increase the dynamic range at the very very low end(soft), not the loudest. So the loudest part of a passage(without causing permanent hearing damage) in a 16bit audio will sound _exactly_ the same as in the 24bit.
 2 - in the "intermediate dynamic", let's say 70-80db, a violin, cello, piano, horn, or any other instrument will not sound any "fuller", or produce more definition, resolution or details in sound in a 24bit audio than a 16 bit one. As opposed the "higher bit rate" in photographic term usually means it provides more shades/details in any given dynamic range(although most beneficial at the extremes).

 am I correct?


----------



## gregorio

Shinew - Absolutely correct. There is much confusion due to the term resolution. We often talk about bit resolution but it leads to misunderstanding because higher resolution in visual images makes them "look" better, the same of digital audio is not true. The lower bit rate just defines a lower noise floor.

 It's the dithering quantizer in the digital conversion process which creates a perfect waveform at any bit resolution. Here's an explanation of how a dithering quantizer works if you're interested:

 When using a limited number of values to represent a continuosly varying waveform there are always going to be measurement errors which represent values bewteen the quantisation steps. These errors are called quantisation errors. The whole point of a dithering quantizer is that the quantisation errors present at any bit depth are converted into noise. The result is a perfectly linear system with noise. Adding another bit (MSB) of data doubles the quantisation points and results in half the number of quantisation errors, and therefore halves the noise. So each additional bit used to encode the audio results in the noise floor being reduced by about 6dB. So in 16bit audio the noise floor of the system is about 96dB below peak level, while in a 24bit system it is at 144dB. Of course in reality no piece of equipment can match the theoretical noise floor of a 24bit system.


----------



## sschmeichel

I feel embarrassed to be posting here... since i know nothing about audio. But since there seems to be people willing to "teach" here, I would like to ask questions to what I have read so far...

 So do you guys mean only at really low volumes... does 24bit "could" make a difference, and if I could hear it, everything else that was recorded louder would make my ears bleed?

 Would i get less "background hissing"... (have to REALLY crank the volume to hear it... if i played back at 24-bit? 

 (i am sorry... i have no clue what floor noise means...)

 Why do i hear differences when i switch from computer --> usb 16/44.1khz --> dac to computer --> external sound card --> optical 24/96khz --> dac?

 I know i have added an extra component... i just want to know more what's causing the difference...

 I apologize if these are really noob questions... but i know if there are nice people who will answer my questions, i will be less of one tomorrow... thank you


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_24bit provides absolutely no more accuracy, or resolution than any other bit depth._

 

And this 8-bit image provides no more accuracy than any other bit depth.


----------



## shinew

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Shinew - Absolutely correct. There is much confusion due to the term resolution. We often talk about bit resolution but it leads to misunderstanding because higher resolution in visual images makes them "look" better, the same of digital audio is not true. The lower bit rate just defines a lower noise floor.

 It's the dithering quantizer in the digital conversion process which creates a perfect waveform at any bit resolution. Here's an explanation of how a dithering quantizer works if you're interested:

 When using a limited number of values to represent a continuosly varying waveform there are always going to be measurement errors which represent values bewteen the quantisation steps. These errors are called quantisation errors. The whole point of a dithering quantizer is that the quantisation errors present at any bit depth are converted into noise. The result is a perfectly linear system with noise. Adding another bit (MSB) of data doubles the quantisation points and results in half the number of quantisation errors, and therefore halves the noise. So each additional bit used to encode the audio results in the noise floor being reduced by about 6dB. So in 16bit audio the noise floor of the system is about 96dB below peak level, while in a 24bit system it is at 144dB. Of course in reality no piece of equipment can match the theoretical noise floor of a 24bit system._

 

Thanks gregorio! I had no idea that's how the noises were introduced! 
	

	
	
		
		

		
		
	


	




 so in a way the 24bit vs 16bit can be compared to a person's field of view, as long as the scenary is broader than a person's FOV, what's outside of the person's FOV makes no difference to what he sees.

 I have another question which is probably more relevant from a consumer point of view. You mentioned few times that the only difference(or improvement if there is any) someone hears from an upsmpling 24bit DAC vs a 16bit one is because the 24bit DAC probably has a better filter at 24bit. You also said you've heard an insanely expensive 16bit DAC with a superb filter which outperforms most of the upsampling 24bit DAC, apparently it is also a very difficult thing to implement a decent 16bit filter. 
 How do they compare in a much more consumer friendly price range($200-$500, $600-$1200)? is it better to get a DAC with 16bit playback or an upsampling 24bit one?
 I know this can cause a lot of controversy and it is very much depending on the hardware implementation. Nevertheless I would still like to know your opinion on this.


----------



## b0dhi

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_According to digital theory, 24bit provides absolutely no more accuracy, or resolution than any other bit depth. _

 

I'm amazed that anyone is still taking you seriously at this point.

 Also, the analogy of the bit depth of audio being like the field of view of vision is _not_ correct. The FOV (spatial domain) analogue of vision is the _time_ domain of audio, _not_ bit depth. Having a wider field of view would be analogous to having a longer piece of audio. Bit depth in audio and bit depth in colour are basically the same thing, except colour is (usually) represented by 3 values for red, green and blue, whereas audio has the one amplitude measurement.


----------



## grawk

Upsampling dacs change the sample rate, not the bit depth.


----------



## gregorio

Shinew - Not quite, upsampling the bit depth from 16bit to 24bit will make no difference whatsoever. There is no difference between a 16bit file and a 24bit file with the last 8 LSBs set to zero. Upsampling the sample frequency from 44.1 to 96k may have a positive or negative perceivable effect depending on the filters in your DAC. I don't really know enough about the digitisation of visual images to comment much on specific analogies with digital audio, except to mention my own observation that increasing the bit depth in images does make them look better.

 Crowbar - Read what I have written before stating the obvious. "We often talk about bit resolution but it leads to misunderstanding because higher resolution in visual images makes them "look" better, the same of digital audio is not true."

 b0hdi - Just because you are unable or unwilling to understand basic digital audio theory, doesn't mean that other people reading this thead feel the same way.

 sschmeichel - Essentially your understanding is correct. In theory, you would have to crank up your volume level on your amp higher to hear the noise floor on a 24bit recording than on a 16bit one. I say in theory because in practice the noise floor is low enough even in a 16bit system that other factors are likely to have more effect than the theoretical noise floor of 16bit. Other factors would include noise floor of the recording venue and the cumulative effects of the noise introduced by all the equipment in the recording and playback chain. Given exactly the same situation and equipment the 24bit system will have less noise but of course the situation and equipment is never the same.


----------



## bigshot

Quote:


  Originally Posted by *sschmeichel* /img/forum/go_quote.gif 
_Why do i hear differences when i switch from computer --> usb 16/44.1khz --> dac to computer --> external sound card --> optical 24/96khz --> dac?_

 

Assuming that you are using the same audio file, DAC and computer in both chains, the most obvious suspect would be the external sound card

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_"We often talk about bit resolution but it leads to misunderstanding because higher resolution in visual images makes them "look" better, the same of digital audio is not true."_

 

Higher numbers don't mean a better image in pictures either. Assume you have a photo which is 600 dpi. Display that on a 72 dpi monitor at the same size on the screen as an image that is 72 dpi. They are going to look exactly the same, because you can't view the added resolution without blowing the image up to see.

 That's exactly the same as added dynamic range. In order to hear the improvement you have to increase the volume to incredible levels. That isn't a very practical way to listen to music.

 See ya
 Steve


----------



## gregorio

Bigshot - I'll take your word for it, I don't know enough about how images are processed. From a layman's point of view, if I have a digital image which is encoded using a certain number of bits and then blow it up it will look less pixellated than the same image encoded using fewer bits. So at least in theory the added resolution exists, even if you can't see it. Is that correct? If so, this is not the case with digital audio. The waveform will be perfectly linear at any bit depth. In other words, in digital audio the resolution will be identical (perfect) regardless of the bit depth, provided the amplitude of the waveform exceeds the noise floor.


----------



## shinew

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_ Quote:


 We often talk about bit resolution but it leads to misunderstanding because higher resolution in visual images makes them "look" better, the same of digital audio is not true. 
 

Higher numbers don't mean a better image in pictures either. Assume you have a photo which is 600 dpi. Display that on a 72 dpi monitor at the same size on the screen as an image that is 72 dpi. They are going to look exactly the same, because you can't view the added resolution without blowing the image up to see._

 

Assuming we're still talking about bit depth. It's true that higher bit in image does not always mean higher _perceivable_ image quality, but foundamentally it has nothing to do with dpi or ppi(doesn't matter if you blow it up or not). Image bit depth describes the number of shades in each color channel, not resolution(number of pixels) of the image.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Shinew - Not quite, upsampling the bit depth from 16bit to 24bit will make no difference whatsoever. There is no difference between a 16bit file and a 24bit file with the last 8 LSBs set to zero. Upsampling the sample frequency from 44.1 to 96k may have a positive or negative perceivable effect depending on the filters in your DAC. I don't really know enough about the digitisation of visual images to comment much on specific analogies with digital audio, except to mention my own observation that increasing the bit depth in images does make them look better._

 

I see, move on to read up more on frenquency upsampling


----------



## jenghanHsieh

This thread is an interesting read !!

 Here is my 1 cent.

  Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_Assuming we're still talking about bit depth. It's true that higher bit in image does not always mean higher perceivable image quality, but foundamentally it has nothing to do with dpi or ppi(doesn't matter if you blow it up or not). Image bit depth describes the number of shades in each color channel, not resolution(number of pixels) of the image._

 

For image, a still picture is sliced into DISCRETE 'pixels' in spatial domain. 
 And we use DISCRETE values to describe the color of a pixel.
 For video, DISCRETE frames are use to keep track of each pixel at a pre-defined framerate. 

 For digital audio, DISCRETE samples are use to represent CONTINUOUS variation of air pressure, and is reconstructed as CONTINUOUS variation of electric potential for playback.

 I'm not a TV/LCD expert, but the representation/reconstruction of image
 and audio elements seems to be quite different and seems to be NOT comparable using this over-simplified analogy.


----------



## shinew

Quote:


  Originally Posted by *jenghanHsieh* /img/forum/go_quote.gif 
_This thread is an interesting read !!
  Quote:


 Assuming we're still talking about bit depth. It's true that higher bit in image does not always mean higher perceivable image quality, but foundamentally it has nothing to do with dpi or ppi(doesn't matter if you blow it up or not). Image bit depth describes the number of shades in each color channel, not resolution(number of pixels) of the image. 
 

I'm not a TV/LCD expert, but the representation/reconstruction of image
 and audio elements seems to be quite different and seems to be NOT comparable using this over-simplified analogy._

 

1) I don't believe I've used any analogy for audio in the post you quoted.
 2) they do seem to be different, what do you think confused me in the first place?


----------



## jenghanHsieh

shinew,

  Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_1) I don't believe I've used any analogy for audio in the post you quoted.
 2) they do seem to be different, what do you think confused me in the first place? 
	

	
	
		
		

		
		
	


	


_

 

Sorry for I've quote the inapproprieate message. 

 The 'Analogy' is in response to the two flower pictures that Crowbar posted earlier in this thread. For it may be invalid to compare 'bit-depth' in such manner for the D->A process ( or the whole 'recreation process' ) in video seems to be very different from what is used in audio.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_higher resolution in visual images makes them "look" better, the same of digital audio is not true._

 

You're confused. Resolution in images is pixel density. The corresponding characteristic in audio is sampling frequency. The difference is, one is in the spatial domain, one in the time domain. Bit depth is something completely different; this is quantization of each sample and is the same in both images and audio. Here's what 8-bit music sounds like: YouTube - 8-Bit Rick Roll
 How is this not less accurate than the 16 bit source? That's ludicrous and this trivial example destroys your claim that accuracy is not better with finer quantization.

  Quote:


 b0hdi - Just because you are unable or unwilling to understand basic digital audio theory, doesn't mean that other people reading this thead feel the same way. 
 

It's clear that b0hdi understands signal processing theory. Math is math. Be it images or sound, the fundamentals are the same.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_From a layman's point of view, if I have a digital image which is encoded using a certain number of bits and then blow it up it will look less pixellated than the same image encoded using fewer bits. So at least in theory the added resolution exists, even if you can't see it. Is that correct?_

 

Again, this has nothing to do with bith depth.
 Images are layed out in space. Audio is laid out in time. Image resolution is the number of samples in the image. Same thing goes for audio, though there the practice is to normalize it per unit time and call it sampling rate.
 My example was addressing the number of bits per sample. Audio and image samples are both quantized based on the level of a given sample. Taking an image and decreasing the quantization results in the first picture in the example I showed in a previous post. In the last post I gave an audio example of the same thing. Now, 8-bit is pretty extreme, but it illustrates the point that finer quantization (higher bit depth per sample) allows for more accuracy.


 Of course, there's a complication here in that by using dither you can make up for a lower quantization with higher sampling rate--that's why DSD and 1-bit sigma delta DACs work. In theory, the total information matters only. In practice, the tradeoffs cannot really be made to be perfect.
 Dither is used to the same effect in images.


----------



## b0dhi

Quote:


  Originally Posted by *jenghanHsieh* /img/forum/go_quote.gif 
_For image, a still picture is sliced into DISCRETE 'pixels' in spatial domain. 
 And we use DISCRETE values to describe the color of a pixel.
 For video, DISCRETE frames are use to keep track of each pixel at a pre-defined framerate. 

 For digital audio, DISCRETE samples are use to represent CONTINUOUS variation of air pressure, and is reconstructed as CONTINUOUS variation of electric potential for playback.

 I'm not a TV/LCD expert, but the representation/reconstruction of image
 and audio elements seems to be quite different and seems to be NOT comparable using this over-simplified analogy._

 

Actually, in both cases, discrete samples are used to represent continuous variation of air pressure/colour. Your video card for example, if connected to an analogue CRT, also produces continuous output from discrete digital data. Conversely, there are also audio DACs that produce "discrete" (in the context you used the word) output. These are non-oversampling (NOS) DACs and they just convert each sample of digital audio into the corresponding voltage, and do nothing else. The output you get from common DACs is oversampled, giving it its "continuous" appearance.

 As Crowbar said, dithering is used in both audio and video in the same way. In both audio and video, bit-depth is directly related to quantisation resolution. You can't simply take a low bit-depth recording or image, and dither it into perfection like gregorio seems to think. The same applies to audio. Here is a visual example of what dithering does and does not:

*Pseudo low bit-depth image:*






*Pseudo low bit-depth image with dithered quantisation:*





*High bit-depth image:*





 Before we start mincing words, I'm aware that the image resolution is the same for all three. The quantisation resolution, however, is not the same, and is obviously the best in the high bit-depth image. A similar process applies to audio. Although you can dither low bit-depth audio to improve it, it isn't, as gregorio said, as "perfect" as high bit-depth audio. Whether raising the bit-depth of audio beyond 16 bits results in a perceptual improvement, I don't know, but bit-depth certainly _does_ affect quantisation resolution (and time-domain resolution, but this is already in the picosecond range for most of the audible spectrum in 16bit audio anyway).


----------



## grawk

We're not talking about comparing 8bit audio wtih 24 bit audio. We're talking about comparing 16 bit audio with 24 bit audio. 16 bit audio goes from silent to being directly underneath an airplane taking off. The only benefit from 24 bit is more headroom when recording. Any engineer with a brain can easily master audio into 16 bits without resorting to compression and lose no data at all.


----------



## b0dhi

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_We're not talking about comparing 8bit audio wtih 24 bit audio. We're talking about comparing 16 bit audio with 24 bit audio. 16 bit audio goes from silent to being directly underneath an airplane taking off. The only benefit from 24 bit is more headroom when recording. Any engineer with a brain can easily master audio into 16 bits without resorting to compression and lose no data at all._

 

1-bit audio also goes from being silent to being underneath an airplane. My point ofcourse is that bit-depth doesnt have anything to do with loudness, you are confusing two different measurements. I don't know if 24bit sounds any better than 16bit, but the human hearing mechanism has a dynamic range of about 120dB (130dB if you count the threshold of pain). It wouldn't hurt to be able to cover this range in audio.


----------



## Crowbar

My point with the images was narrow: I was refuting the specific claim that higher bit depth doesn't allow for higher accuracy.


----------



## NapalmV5

16bit




 24bit




 32bit




 OscilloMeter

 24bit better? 
	

	
	
		
		

		
		
	


	




 gimmie 32bit!


----------



## gregorio

The assumptions being made are entirely accurate, providing you are talking about a non-dithering quantizer. A dithering quantizer does not provide a stepped output but a linear output. All the quantization errors are converted into uncorrelated white noise. So the only difference between a 24bit output and say an 8bit output is increased noise. The output waveform at 24bit is going to look different to the output waveform at 8bit, if measured from the analogue output of a DAC, because the 8bit output will have significantly more noise. The same must be true of Crowbar's 24bit vs. 8bit, the difference between them is the amount of noise. The noise floor of 8bit is magnitudes higher than at 24bit, each additional bit of data moving the noise floor down by roughly 6dB. Providing a waveform exceeds the noise floor of the system though, the output waveform will be identical (linear) from both a 24bit system and an 8bit one. The difficulty of course with 8bit is that the noise floor is so high that it's impossible to capture a waveform without incurring significant noise. However, this is not the case when comparing 16bit to 24bit because the theoretical noise floor at 16bit is lower than the noise floor caused by other factors, the recording and playback equipment and environment for example. At the 24bit level the noise floor is -144dB, which is approaching the noise level created when two hydrogen atoms collide. Your playback system (or any playback system) is going to be orders of magnitude away from being able to respresent this level. If you could in theory crank your amp up high enough so you can hear individual atoms colliding, then the loud parts in your audio file would severely injure or kill you. So whoever thinks that 32bit audio is worthwhile as a consumer format obviosly doesn't understand the issues. Bare in mind that even the very best professional DACs have an output dynamic range of about 108dB, which is effectively 18bits and is still way beyond what is practical, even with the extremely quiet environment of a world calss recording studio.


----------



## jenghanHsieh

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_Actually, in both cases, discrete samples are used to represent continuous variation of air pressure/colour. Your video card for example, if connected to an analogue CRT, also produces continuous output from discrete digital data. Conversely, there are also audio DACs that produce "discrete" (in the context you used the word) output. These are non-oversampling (NOS) DACs and they just convert each sample of digital audio into the corresponding voltage, and do nothing else. The output you get from common DACs is oversampled, giving it its "continuous" appearance.

 ...

 Whether raising the bit-depth of audio beyond 16 bits results in a perceptual improvement, I don't know, but bit-depth certainly does affect quantisation resolution (and time-domain resolution, but this is already in the picosecond range for most of the audible spectrum in 16bit audio anyway)._

 

I see, thanks for your clarification. 

 I've found some interesting explanation on Dynamic Range in digital photography ( see Understanding Dynamic Range in Digital Photography ). Perhaps someone with enough time and expertise can give us a similar explanation on digital audio... 
	

	
	
		
		

		
		
	


	




 FYI. the last two paragraphs quoted from above web page : 
  Quote:


 "We first need to distinguish between whether we are speaking of recordable dynamic range, or displayable dynamic range. Even an ordinary 8-bit JPEG image file can conceivably record an infinite dynamic range-- assuming that the right tonal curve is applied during RAW conversion (see tutorial on curves, under motivation: dynamic range), and that the A/D converter has the required bit precision. The problem lies in the usability of this dynamic range; if too few bits are spread over too great of a tonal range, then this can lead to image posterization.

 On the other hand, displayable dynamic range depends on the gamma correction or tonal curve implied by the image file, or used by the video card and display device. Using a gamma of 2.2 (standard for PC's), it would be theoretically possible to encode a dynamic range of nearly 18 f-stops (see tutorial on gamma correction, to be added). Again though, this would suffer from severe posterization. The only current standard solution for encoding a nearly infinite dynamic range (with no visible posterization) is to use high dynamic range (HDR) image files in Photoshop CS2 (or other supporting program)." 
 

To me, one major difference between Audio and Video reproduction seems to be the relative capabilities of playback devices. The best video card + LCD/CRT monitors in the market can only put out a fraction of dynamic range that our eyes can handle, while the best DAC+Amplifier+Speaker in the market can render almost the 'full spectrum' of what we can practically hear (with respect to dynamic range).

 And for digital audio, 0db is the absolute maximum for we are very sensitive to clipping, ( extra bit depth push the noise floor down to -96 or -100+ db ) but while taking / viewing digital photos on commercially available devices, most people simply won't notice a large area of white pixels.


----------



## gregorio

jenghanHsieh - Unfortunately, b0hdi has not clarified the situation but confused it. b0hdi would have been correct 20 years ago but not today when we use dithering quantizers. Oversampling does not give a continous output wave, unless a dithering quantizer is involved in the output path. A dithering quantizer creates a continuos linear (non quantized) output and therefore bit depth is irrelevant because we are not dealing with quanta. Dithering audio doesn't improve it, it converts all quantization errors into noise. So a low bit depth waveform will appear to have more noise than a high bit depth waveform but in every other respect the waveforms at both bit depths will be entirely linear.


----------



## jenghanHsieh

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_jenghanHsieh - Unfortunately, b0hdi has not clarified the situation but confused it. ... a low bit depth waveform will appear to have more noise than a high bit depth waveform but in every other respect the waveforms at both bit depths will be entirely linear._

 






 thanks gregorio, I think I've got yor point. For me you guys are talking about the same thing with different names ( and under different model ).

 I've only took an introductory course on video coding technology at school 5 years ago, this thread is really interesting to read and I'm now happy reading about both digital audio / visual processing technologies. 

 The 'uncorrelated white noise' is clear and good enough for me.


----------



## maarek99

Bit depth does define resolution. An 8-bit file definitely has less resolution than a 16-bit, I don't care how you guys minge around it.

http://img3.musiciansfriend.com/dbas...p/bitdepth.gif

http://img3.musiciansfriend.com/dbas...p/bitdepth.gif


----------



## maarek99

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Higher numbers don't mean a better image in pictures either. Assume you have a photo which is 600 dpi. Display that on a 72 dpi monitor at the same size on the screen as an image that is 72 dpi. They are going to look exactly the same, because you can't view the added resolution without blowing the image up to see._

 

That is so wrong it's not even funny. How come so many people completely miss how dpi is supposed to work?

Dots per inch - Wikipedia, the free encyclopedia

 "Some digital file formats record a DPI value, which is to be used when printing the image."

 It is nothing else. It does not define the resolution of the image. That is defined by the...well, pixel resolution. DPI just affects how it will be printed out. A file with a dpi of 72 will look exactly the same as a file with a dpi of 1000 on your computer monitor


----------



## jcx

you could ignore the video analogies and actually listen to audio at differing bit depths:

Homepage of Alexey Lukin

 8,12 and 16 bit audio samples as well as more dither theory on that site

 also if you want to see how various recoding format map over human perceptual limits:

http://www.meridian-audio.com/w_paper/Coding2.PDF

 the paper is very technical (by head-fi standards at least, it is a “popularization” of a JAES convention paper)
 in support of his argument for a particular coding scheme Stuart summarizes some of “conventional” audio engineering/psychoacoustic understanding of human perception limitations and their relation to reproduced audio – you might want to jump to the figures/graphs at the end of the paper and then search back into the text for the explanatory context


----------



## eruditass

Quote:


  Originally Posted by *jcx* /img/forum/go_quote.gif 
_you could ignore the video analogies and actually listen to audio at differing bit depths:

Homepage of Alexey Lukin

 8,12 and 16 bit audio samples as well as more dither theory on that site

 also if you want to see how various recoding format map over human perceptual limits:

http://www.meridian-audio.com/w_paper/Coding2.PDF

 the paper is very technical (by head-fi standards at least, it is a “popularization” of a JAES convention paper)
 in support of his argument for a particular coding scheme Stuart summarizes some of “conventional” audio engineering/psychoacoustic understanding of human perception limitations and their relation to reproduced audio – you might want to jump to the figures/graphs at the end of the paper and then search back into the text for the explanatory context_

 

just looking at figure 7, it is clear that it makes a difference if we are using files that want to reach 120 dB in the most silent of parts. I sincerely hope no one is going to turn up their music that loud, so by lowering the maximum noise level, 16-20 bits becomes indeed indiscriminable. By all means some data would be lost, but no one would be able to hear it unless dynamic compression / volume control is used afterwards. i'd read more but i have an exam in a couple hours.


----------



## bigshot

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_If so, this is not the case with digital audio. The waveform will be perfectly linear at any bit depth. In other words, in digital audio the resolution will be identical (perfect) regardless of the bit depth, provided the amplitude of the waveform exceeds the noise floor._

 

My generalized point was that the equivalent of blowing up the picture is turning the volume up to hear the added resolution below. You can't appreciate a photo of an elephant blown up so big you can only see his big toe. And you can appreciate music with the volume turned up to hear the heartbeat of the guy playing the piano.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_Image bit depth describes the number of shades in each color channel, not resolution(number of pixels) of the image._

 

Yes, but on a monitor, it's still only three colors... RGB and printed out, it's four CMYK.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_In the last post I gave an audio example of the same thing. Now, 8-bit is pretty extreme, but it illustrates the point that finer quantization (higher bit depth per sample) allows for more accuracy._

 

If 8 bit sounds half as good as 16 bit, then 16 bit must sound half as good as 32 bit, right?

 Wrong. Once you reach a level where the perceptible range of human hearing is matched, more numbers don't mean better sound. Unless of course, if you are a bat or a dog.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_the human hearing mechanism has a dynamic range of about 120dB (130dB if you count the threshold of pain). It wouldn't hurt to be able to cover this range in audio._

 

Anything above 85dB is considered in the danger range for causing hearing loss. The 96dB dynamic range on CDs is perfectly capable of making you quite deaf in short order. When you talk about numbers, you need to know what they sound like. 96dB is not a volume level you want in your home, even for short periods of time.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *maarek99* /img/forum/go_quote.gif 
_A file with a dpi of 72 will look exactly the same as a file with a dpi of 1000 on your computer monitor_

 

Uh... that was my point.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *jcx* /img/forum/go_quote.gif 
_you could ignore the video analogies and actually listen to audio at differing bit depths:

Homepage of Alexey Lukin

 8,12 and 16 bit audio samples as well as more dither theory on that site

 also if you want to see how various recoding format map over human perceptual limits:

http://www.meridian-audio.com/w_paper/Coding2.PDF

 the paper is very technical (by head-fi standards at least, it is a “popularization” of a JAES convention paper)
 in support of his argument for a particular coding scheme Stuart summarizes some of “conventional” audio engineering/psychoacoustic understanding of human perception limitations and their relation to reproduced audio – you might want to jump to the figures/graphs at the end of the paper and then search back into the text for the explanatory context_

 

Stuart's arguments rest on a few slightly iffy assumptions.

 1) Playback at 120db - of course this will expose 16 bit audio's noise limitations, but this is an absurd level to listen at by any rational criteria and assumes a zero noise floor which is not attainable. By this I mean the CD noise may be theoretically audible at 120db but a noise floor of 25db will hide the CD noise.

 2) Oohashi again, this paper must be considered suspect, certainly it has never been replicated and a more recent paper has questioned that their effect may be due to IMD. Also he places much reliance on anecdotal evidence.

 3) The ability to render 26K signals, which is (one) overkill due to the limits of human hearing and (two) even signals that have energy at that level have it at very low levels and these are frequencies that even those that can hear even close to them need enormous energy output to do so.

 4) He keeps talking about distortion in undithered signals, no engineer in their right mind does not apply dither, it is a straw man argument

 5) His test case is a -90db 1K signal, talk about stacking the odds, this is just about the worst case you can imagine for CD, but even here it doesn't matter until you crank up the volume to insane levels. A -90db signal is basically 1 bit !, any signal at less than - 72db (4 bits) will be below the noise floor for anything other than an anechoic chamber !


----------



## Crowbar

Quote:


  Originally Posted by *jcx* /img/forum/go_quote.gif 
_8,12 and 16 bit audio samples as well as more dither theory on that site_

 

Make sure you use a blind methodology to compare. There's a program that makes it very easy on a computer: ABC/HR Audio Comparison Utility


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Make sure you use a blind methodology to compare. There's a program that makes it very easy on a computer: ABC/HR Audio Comparison Utility_

 

FooBar2000 also has a nice ABX plug-in , ABC/HR will not play 24 bit files in my setup, but both FooBar and Winamp will


----------



## shinew

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Yes, but on a monitor, it's still only three colors... RGB and printed out, it's four CMYK.

 See ya
 Steve_

 

For photo editing, no one uses CMYK. You'll edit in RGB, preview in RGB & send it to the printer in RGB format. The printer's driver also receives RGB information and (some) internally converts it to CMYK info. So you really don't have to think about it in CMYK when you do photo editing/printing.

 But I really don't think "photographic bit depth" is a good analogy for audio. It'll do more to confuse people than to clear things up.

 One of the main reason I think people who think the 24bit(144db) will sound superior in a playback environment than a 16bit(96db) is when they compare them to people's hearing limit(120db). Even disregarding the hearing damage it'll cause, we have to remember that the CD we listen to is not the source of music, it is just a _recording_ of the music source. As some have pointed out(as I've learned), in the recording process there are so many factors such as the environment & equipments(even with the highest end equipment the current technology allows) used have tremendously raised the noise floor & compressed the dynamic range to make the 16bit more than capable for playing back the recorded music.

 EDIT:edited to keep it on topic 
	

	
	
		
		

		
		
	


	




 .


----------



## grawk

topic drift ftl


----------



## gregorio

Thanks for that article jcx, haven't seen that one before. Some interesting points:

 "Even more important was the realisation that there is a right sort of random noise to add, and that when the right dither is used, the resolution of the digital system becomes infinite.... Regarding temporal accuracy, (ii), if the signal is processed incorrectly (i.e. truncated) it is true that the time resolution is limited to the sampling period divided by the number of digital levels. However, when the correct dither is used the time resolution also becomes effectively infinite.... This is perhaps the most fundamental point of all – if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white, uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"

 I also found interesting the conclusions of maximum perceivable bit depth and sample frequency. Bare in mind that these conclusions are based on the possible theoretical extremes of human hearing and not what is practical or possible with current recording and playback technology. Even theoretically, 24/96k is far higher than the limitations of human hearing, the conclusion of the paper being 18.5bit/58kFs/s would be theoretically optimal. However, when we factor in the practicalities of recording and playback, 16bit/44.1k is easily sufficient. Like Nick_Charles, I would question the basis for needing to encode up to 26kHz.

 maarek - your graphs are an accurate representation of the digital data stored in it's raw format but these graphs give no indication of the output waveform which will be generated once this digital data is converted to analogue. The only difference when your two digital representations are converted back to analogue will be the amount of noise.


----------



## bigshot

Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_For photo editing, no one uses CMYK._

 

If you are going to press, you sure do!

  Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_But I really don't think "photographic bit depth" is a good analogy for audio._

 

The point was that bigger numbers don't necessarily mean better quality. You reach a point where it doesn't matter any more. This whole argument reminds me of Ken Rockwell's The Megapixel Myth. You have to know what those numbers mean in practical application. You can't just blindly take the "more is better" approach.

 See ya
 Steve


----------



## shinew

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_If you are going to press, you sure do!

 The point was that bigger numbers don't necessarily mean better quality. You reach a point where it doesn't matter any more. This whole argument reminds me of Ken Rockwell's The Megapixel Myth. You have to know what those numbers mean in practical application. You can't just blindly take the "more is better" approach.

 See ya
 Steve_

 

I have plenty to say about color, profiles, conversion & pixel quality. but this is not a photographic discussion. so i'll stop it here.
 But your basic direction is correct - more is not better.


----------



## sejarzo

I certainly agree that each added bit adds 6 dB dynamic range, but please explain why my dumb chemical engineering logic on resolution is wrong.......

 In the 16 bit system, let's say that 2^16 represents an instantaneous amplitude of 2.0 V. That means 2^15 represents an amplitude of 1.0 V. That 1.0 V difference must be resolved over (2^16 - 2^15) steps......32,768 steps.

 In a 24 bit system designed such that 2^24 also represents an instantaneous amplitude of 2.0 V, then 2^23 must represent an amplitude of 1.0 V. That 1.0 V difference is then resolved over (2^24 - 2^23) or 8,388,608 steps, or 256 times more steps than in the 16 bit case.

 If we look at the problem "from the top down", from 0 dBFS down to -60 dBFS (assuming a decently recorded symphonic performance such that the volume peaks just under 0 dBFS, and the lowest signals are 60 dB down), with 24 bit resolution, there are (2^24 - 2^14) or 16.8 million steps of amplitude resolution available to quantify the amplitude over a musically significant range.

 The same analysis for a 16 bit system implies to me that there would be only (2^16 - 2^6) or only 65,472 steps to resolve that same range of signals from 0 dBFS down to -60 dBFS. 

 Over any given difference in signal level *down from 0 dBFS*, the 24 bit system provides a capability of 256 times finer resolution for that given signal level range......assuming that the signal voltage is the same at 0 dBFS in either system.

 Thus, wouldn't the appropriate way to analyze this be from "the top down" (meaning signal levels referenced to 0 dBFS) rather than analyzing it from the theoretical overall dynamic range for either system? Either 16 bit or 24 bit consumer DAC's tend to standardize on 2.0 Vrms for a 0 dBFS output, right?

 Maybe that point is moot in the case of dithered quantizers/quantization, but I've not found a reference that explains it well enough for my limited knowledge to grasp how that would impact the above analysis.


----------



## HFat

I'm not a recording engineer so I'll just assume the following is true:
  Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_In the 16 bit system, let's say that 2^16 represents an instantaneous amplitude of 2.0 V. That means 2^15 represents an amplitude of 1.0 V. That 1.0 V difference must be resolved over (2^16 - 2^15) steps......32,768 steps.

 In a 24 bit system designed such that 2^24 also represents an instantaneous amplitude of 2.0 V, then 2^23 must represent an amplitude of 1.0 V. That 1.0 V difference is then resolved over (2^24 - 2^23) or 8,388,608 steps, or 256 times more steps than in the 16 bit case._

 

This would mean that 0.5 V would be represented by 2^14 and 2^22 respectively, that 0.25V would be represented by 2^13 and 2^21 respectively and so on.
 So the "steps" would have exactly the same "size" until the 16 bits are exhausted, around the microvolt range. Therefore, the added "resolution" of 24 bit would be below the noise floor in most cases.


 I think that some of the misunderstanding here follows from people having different definitions of resolution.
 The mud-slinging is painful to watch... could you zealots (you know who you are) tone it down a bit?


----------



## sejarzo

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_ The 96dB dynamic range on CDs is perfectly capable of making you quite deaf in short order. When you talk about numbers, you need to know what they sound like. 96dB is not a volume level you want in your home, even for short periods of time._

 

Directly equating dB of dynamic range of signal level on a CD to dB SPL isn't reasonable.

 OSHA allows workplace exposure to 97 dB SPL (A weighting, slow response) for 3 hours per day before hearing protection is required.


----------



## HFat

Is OSHA in the business of preserving your enjoyment of music until old age?

 I don't want to hear peaks at 97dB or above except when the surroundings are very noisy and even then it's something I regret having done but I agree that this is a personal choice.


----------



## sejarzo

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_This would mean that 0.5 V would be represented by 2^14 and 2^22 respectively, that 0.25V would be represented by 2^13 and 2^21 respectively and so on.

 So the "steps" would have exactly the same "size" until the 16 bits are exhausted, around the microvolt range. Therefore, the added "resolution" of 24 bit would be below the noise floor in most cases._

 

Not really, seems to me that you are equating the resolution of a 0.25V change quantified from 2^22 to 2^21 as being equivalent to it being quantified between 2^14 to 2^13, and that definitely isn't so. I might be misunderstanding you, or it just might be a semantic difference, though. 

 Punch it into a calculator and work it out for yourself. A 24 bit system does indeed provide much greater resolution of amplitude, and it really cannot be any other way.

 As you note, the steps are indeed a constant factor of 2--they have to be, otherwise the system wouldn't be linear! But there are more bits available to resolve each 6 dB step in amplitude in the 24 bit system, so the added resolution is available throughout the entire dynamic range....not just at the lower end!

 It is important to note that the Meridian paper does not specifically claim, as far as I could tell, that dithered quantization provides strictly infinite resolution--what I saw was a statement that it provides "_effectively _infinite resolution". 

 Now the question becomes whether it is an audible difference or not--but it is certain that the increased resolution of 24 bit exists throughout the whole dynamic range.


----------



## sejarzo

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_I don't want to hear peaks at 97dB or above except when the surroundings are very noisy and even then it's something I regret having done but I agree that this is a personal choice._

 

I don't know what sort of music you prefer, but much live orchestral music peaks well above 100 dB out in the house and is louder on stage. Such peaks aren't going to hurt your ears by any means.


----------



## nick_charles

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_I don't know what sort of music you prefer, but much live orchestral music peaks well above 100 dB out in the house and is louder on stage. Such peaks aren't going to hurt your ears by any means._

 

But the noise floor in your home is unlikely to be below 25db so to render a peak at 97db above the noise floor the actual overall sound level peak will be 122db, which is loud by any standards... Your orchestra will be operating in a hall which probably is just as noisy , full of noisy humans breathing, fans, and what have you.


----------



## HFat

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Not really, seems to me that you are equating the resolution of a 0.25V change quantified from 2^22 to 2^21 as being equivalent to it being quantified between 2^14 to 2^13 ..._

 

Yeah, I was quite sloppy. I should not have answered your post carelessly but now that the deed is done...
 I was actually thinking in terms of individual bits and their positions rather than in terms of values. With all the talk of padding 16 bits data with zeros to get 24 bit data, I assumed the most significant bits to code much bigger changes in amplitude than the extra eight so that the first bits code the same amplitude change at every bit depth (not that you're saying anything else). I was trying to say that the extra "steps" you get would all have amplitudes which are (way) below the noise floor in most cases... which doesn't address your point really: I get the impression that you're actually saying that it may be possible to hear such "steps" in spite of the noise. I don't see how that would be possible if the noise was present at all frequencies but that's a different issue that the one you brought up I think. In other words, it's my turn to wonder exactly what your point was. 

 And you guessed it right: I don't listen to much (if at all) orchestral music. I lean towards chamber and away from the loudest instruments as far as classical is concerned. I don't like a bombastic dynamic range in music although I agree it fits movies well.


----------



## HFat

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_But the noise floor in your home is unlikely to be below 25db so to render a peak at 97db above the noise floor..._

 

We were talking about 97dB to zero (so to speak, what's the proper way to say that?), not relative to the noise.

 There may be some confusion about what noise is or what noise does exactly among the least knowledgeable posters in this thread (that would include me).


----------



## Tarkovsky

It's been said that the volume levels would have to be deafening for 24 bit to make a difference, but this assumes you're not wearing hearing protection.
 But wait that's silly you say, you've just lost you're noise floor by effectively lowering the whole outputs volume relative to your ears. Yes. But your ears aren't the only way you perceive sound.
 If you're into D n B you'll know that a large part of the music is infrasound. So, in short I suppose we do have a use for 24 bit playback.
 Some might not strictly describe this as music though. And I think you'd have a hard time ensuring everyone was wearing ear protection. It would incredible though.


----------



## bigshot

Just for perspective, 96dB is somewhere around the volume of a leaf blower. 120dB is the level of a jackhammer. You aren't going to have this kind of volume in your living room, even for short bursts.

 See ya
 Steve


----------



## jenghanHsieh

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Punch it into a calculator and work it out for yourself. A 24 bit system does indeed provide much greater resolution of amplitude, and it really cannot be any other way...

 Now the question becomes whether it is an audible difference or not--but it is certain that the increased resolution of 24 bit exists throughout the whole dynamic range._

 

After reading this interesting thread and some related articles.

 I think the 'perfect' means :

 "A system with perfect *precision*, and good enough *accuracy* for the design purpose"

 ( see Error - Wikipedia for the definition of precision and accuracy ).

 For Digital Audio, 

 [1]
 In theory, the resulting amplitude in every given time slice can be 100% precise, independent of the bit-depth we choose. (with proper dithering ?)

 [2]
 The accuracy may vary according to the bit-depth we used, but it does not matter once we have high-enough accuracy for our purpose. 

 [3]
 Less accuracy = higher noise floor, and, for playback, 16bits / 24bits is all about pushing down the noise floor.


 If a system can hit the bull's-eye every single time, it is a perfect system. 
 It does not need to hit "EXACT THE SAME POINT EVERYTIME" to be perfect.


----------



## Tarkovsky

Yes but in a rave the size of a small airport it could come in handy!


----------



## gregorio

sejarzo - have a look at the graphs on the link below:

Quantization Error - DiracDelta Science & Engineering Encyclopedia

 These graphs show exactly what you are describing and 20 years ago would represent what you would hear. However today, the dithering quantizer takes all those quantization errors and converts them into noise. So you don't get a stepped output, as displayed by the blue lines on the graphs, you get a perfectly linear output (at any bit depth), as displayed by the grey lines on the graphs. Look at the 4 graphs, what is the difference between the higher bit representations and the lower bit ones? the difference is that the higher bit depth has fewer quantization errors, and to a dithering quantizer this does not mean a more linear output but a lower noise floor as there are fewer errors requiring conversion into noise.

 Remember, the noise floor isn't the end of audio, we can commonly hear signals substantially below the noise floor. Also remember that the dB scale is not linear but logarithmic. Double the dynamic range of 96dB is not 192dB, it's 99dB! So the difference between 16bit (96dB) and 24bit (144dB) is absolutely massive, thousands of times greater.

 When using 24bit we are not talking about having to crank up your amp a little to hear the lower noise floor. We are talking about an incomprehensible dynamic range. In theory, 24bit is capable of capturing (above the noise floor) the sound pressure level created when two hydrogen atoms collide. If you crank your amp up high enough so that you could hear two hydrogen atoms colliding, when you get a fortissimo chord from the orchestra, you are going to wake up in hospital (or not at all!). Of course in practice, no microphone, amp or any other part of the signal chain is capable of anywhere near being able to capture or playback this dynamic range, which is why 24bit it totally superfluous as a playback medium.

 The problem of high SPLs and hearing damage for orchestral musicians has been recognised for a long time. Particularly for viola and cello players who may be sitting directly in front of the trumpets and trombones. Ear plugs which reduce SPLs linearly are now routinely used by many orchestral musicians, at least during rehearsal if not for performance.


----------



## HFat

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_These graphs show exactly what you are describing and 20 years ago would represent what you would hear. However today, the dithering quantizer takes all those quantization errors and converts them into noise._

 

I don't remember hearing "steps" back then but then again I can't hear the 16 bit quantization noise in normal conditions now (by a long shot).
 I thought that the fact that I can't hear the noise implies that I couldn't hear the errors if they were left alone anyway... right or wrong?

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Remember, the noise floor isn't the end of audio, we can commonly hear signals substantially below the noise floor._

 

Even if the noise is white? Wouldn't it interfere with the faint signal, drowning it in randomness?


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_
 Remember, the noise floor isn't the end of audio, we can commonly hear signals substantially below the noise floor. Also remember that the dB scale is not linear but logarithmic. Double the dynamic range of 96dB is not 192dB, it's 99dB! So the difference between 16bit (96dB) and 24bit (144dB) is absolutely massive, thousands of times greater.
_

 

Hang on, there is the doubling with 3db and doubling with 6db thing that are context dependent. Each bit in a digital PCM system doubles the dynamic range i.e a 16 bit system has 65536 levels and a 17 bit system has 131072 levels. And each bit gives you approximately 6.02db extra dynamic range. So a 24 bit system that has 144db dynamic range will have 16777216 levels which is 256 times more than 16 bits. It is still massively bigger, but not by thousands.


----------



## sejarzo

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_ sejarzo - have a look at the graphs on the link below:

Quantization Error - DiracDelta Science & Engineering Encyclopedia

 These graphs show exactly what you are describing and 20 years ago would represent what you would hear. However today, the dithering quantizer takes all those quantization errors and converts them into noise. So you don't get a stepped output, as displayed by the blue lines on the graphs, you get a perfectly linear output (at any bit depth), as displayed by the grey lines on the graphs._

 

Nice graphs, but they represent *ADC *performance, not *DAC* performance. All of the X axes are labled as "Analogue _input_". So they don't show me "exactly" what I am hearing.

 Please direct me to a similar reference that describes the difference in DAC operation, if you would, that shows the difference in analog *output* between 16 bit and 24 bit DAC operation when fed true 16 bit and 24 bit streams, respectively. I'm not a recording engineer, just a listener....and I have a collection of mostly 16 bit music, but also a number of 24 bit files.


  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_ When using 24bit we are not talking about having to crank up your amp a little to hear the lower noise floor. We are talking about an incomprehensible dynamic range. In theory, 24bit is capable of capturing (above the noise floor) the sound pressure level created when two hydrogen atoms collide. If you crank your amp up high enough so that you could hear two hydrogen atoms colliding, when you get a fortissimo chord from the orchestra, you are going to wake up in hospital (or not at all!). Of course in practice, no microphone, amp or any other part of the signal chain is capable of anywhere near being able to capture or playback this dynamic range, which is why 24bit it totally superfluous as a playback medium._

 

So you are saying that there is no audible benefit at all to more accurate quanitization of signal levels in the say, -10 dBFS down to -40 dBFS range? 

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_ The problem of high SPLs and hearing damage for orchestral musicians has been recognised for a long time. Particularly for viola and cello players who may be sitting directly in front of the trumpets and trombones. Ear plugs which reduce SPLs linearly are now routinely used by many orchestral musicians, at least during rehearsal if not for performance._

 

I certainly agree with that statement. 

 My point is that too many people on this forum are utterly confused re the difference between dynamic range of a recording medium and resulting SPL on playback. There is no strict connection; people seem to forget that dB is always a relative measurement rather than an absolute one.

 Because a CD has a 96 dB range doesn't mean that a 0 dBFS peak will be harmful to your hearing, it all depends on how one sets playback volume.

 A sine wave recorded at only 8 bit resolution at 0 dBFS and played back at 110 dB SPL would harm your hearing just as much as one recorded at 16 bit resolution, 0 dBFS, at the same SPL.

 To me, 85 dB SPL (C weighted) pink noise sounds pretty loud during setting levels on a home theater system. But listening to wind ensemble music at an average level of 75 db SPL, at a lower volume setting on the processor than required to get the 85 dB SPL calibration during set up, includes peaks around 100 dB.....and I don't worry about that being harmful at all.


----------



## grawk

The only way you'll be able to hear a benefit to 24 bit is if you turn the volume up loud enough to be beyond the range of 16 bit. That means 96dB at a minimum, and 96dB above noise floor ideally. You definitely won't see a benefit between -10 and -40. Those bits are the same. 

 24 bit has and probably will remain primarily beneficial at record time, because it allows levels to be set high enough to capture dynamics while allowing enough headroom for unexpected or transient peaks. That way compression, if used at all, won't need to be applied at capture, but can be applied as needed or desired in post.

 16bit exceeds the SNR of most gear, and certainly exceeds the distance between noise floor and the pain threshhold of human hearing.


----------



## bigshot

As I've been told, there is some added resolution in very low volume areas that overlap the edge of coverage between 16 and 24 bit. In normal listening, this added resolution would be below your ability to hear the difference, but it can make a difference in a mix where you might be boosting the level of something quiet to make it audible in the mix. I don't know all the gobbledegook principles behind why that is, just the practical application.

 See ya
 Steve


----------



## sejarzo

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_ The only way you'll be able to hear a benefit to 24 bit is if you turn the volume up loud enough to be beyond the range of 16 bit. That means 96dB at a minimum, and 96dB above noise floor ideally. You definitely won't see a benefit between -10 and -40. Those bits are the same._

 

You do mean that the most significant 10 to 12 bits are the same, right?

 Personally, I think well done CD's sound virtually as good as well done SACD's or DVD-A's, so I tend to agree that 16 bit playback is indeed sufficient.

  Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_24 bit has and probably will remain primarily beneficial at record time, because it allows levels to be set high enough to capture dynamics while allowing enough headroom for unexpected or transient peaks. That way compression, if used at all, won't need to be applied at capture, but can be applied as needed or desired in post.

 16bit exceeds the SNR of most gear, and certainly exceeds the distance between noise floor and the pain threshhold of human hearing._

 

I don't disagree with any of those points.

 My main reason for asking is that from reading through similar discussions on recording forums, there appear to be a number of (for lack of a better phrase) "non-techie" musician-recordists who are certain that their own instrument's timbre is reproduced more accurately when recorded at 24 bit versus 16 bit. Several claim that 24/44.1 is more accurate than 16/96. To them, the bit depth makes more difference than sample rate--a few were quite adamant about that!

 Were they "too conservative" in setting levels when tracking at 16 bit to avoid clipping, maybe only using 12 bits.....but pushed the levels more than proportionally when going to 24 bit, thereby truly gaining an audible difference in resolution, perhaps?


----------



## gyrodec

sejarzo

 "Were they "too conservative" in setting levels when tracking at 16 bit to avoid clipping, maybe only using 12 bits.....but pushed the levels more than proportionally when going to 24 bit, thereby truly gaining an audible difference in resolution, perhaps?"

 That sounds bar far to most plausible explanation. Anything else I can think of ends up just being variations on placeabo effect at the end of the day, that or feats of the impossible.


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Hang on, there is the doubling with 3db and doubling with 6db thing that are context dependent. Each bit in a digital PCM system doubles the dynamic range i.e a 16 bit system has 65536 levels and a 17 bit system has 131072 levels. And each bit gives you approximately 6.02db extra dynamic range. So a 24 bit system that has 144db dynamic range will have 16777216 levels which is 256 times more than 16 bits. It is still massively bigger, but not by thousands._

 

OK, now you've done my head in! I was working on there being a 48dB difference between the noise floor of 16bit and 24bit. A 48dB difference relates to about 64,000 times the audio power. I don't think I can mix and match my dB scales this way though, so you could well be correct, on the other hand we could both be correct if you are talking about dBSPL and I'm talking about dBW. It's academic though, 256 times or 64,000 times, it's cookoo land either way! Mind you, it might be interesting to hear two hydrogen atoms collide! lol

  Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Nice graphs, but they represent *ADC *performance, not *DAC* performance. All of the X axes are labled as "Analogue input". So they don't show me "exactly" what I am hearing.

 Please direct me to a similar reference that describes the difference in DAC operation, if you would, that shows the difference in analog *output* between 16 bit and 24 bit DAC operation when fed true 16 bit and 24 bit streams, respectively. I'm not a recording engineer, just a listener....and I have a collection of mostly 16 bit music, but also a number of 24 bit files._

 

In theory, there's not much point to a graph because it should be entirely linear until you get to the noise floor. In practice, the precise analogue output, exactly where the noise floor is and what it looks like on a graph, will depend on the circuitry of the individual DAC and not many companies provide these graphs. I did find one though, scroll down the page to figure two in the link below. I have to say, it's very poorly labeled.

Stereophile: Nagra DAC D/A processor

 "So you are saying that there is no audible benefit at all to more accurate quanitization of signal levels in the say, -10 dBFS down to -40 dBFS range?"

 Not only would there be no audible benefit but in theory there should be no difference whatsoever, audible or not.

 I completely agree with your comments regarding the confusion in relative levels. The volume that you hear out of your speakers at 0dbFS (digital dB scale) is entirely dependant on where you set your amp.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_FooBar2000 also has a nice ABX plug-in , ABC/HR will not play 24 bit files in my setup, but both FooBar and Winamp will 
	

	
	
		
		

		
			



_

 

It should be noted that ABC+hidden reference is recommended over ABX testing by the ITU.

 BTW, some people here seem to be confusing dB scales for voltage and for power/intensity. One is dB_power = 10 * log10(power_ratio) and the other is dB_voltage = 20 * log10(voltage_ratio). Thus, doubling of power is an increase of about 3 dB, whereas voltage is about 6 dB.
 The 120 dB range of hearing is about a trillion as a power ratio, but only a million as voltage ratio (and log2(1000000) gives about 20 bits). Since PCM is linear with voltage, power is actually quadratic with PCM code word. From 16 to 24 bits voltage increases 256 times, but power increases 65536 times.


----------



## gregorio

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Were they "too conservative" in setting levels when tracking at 16 bit to avoid clipping, maybe only using 12 bits.....but pushed the levels more than proportionally when going to 24 bit, thereby truly gaining an audible difference in resolution, perhaps?_

 

Generally you need to leave roughly 6-8bits of headroom when recording. At 16bit that will mean that most of the signal recorded will be around the 8-10bit level and noise is likely to be an issue when mixing. 8bits of headroom in a 24bit system is still 16bit, which is why 24bit was invented.


----------



## gregorio

Quote:


  Originally Posted by *HFat* /img/forum/go_quote.gif 
_I thought that the fact that I can't hear the noise implies that I couldn't hear the errors if they were left alone anyway... right or wrong?

 Even if the noise is white? Wouldn't it interfere with the faint signal, drowning it in randomness?_

 

Not neccesarily, in reality the process is much more complex than just adding some white noise. Not dithering is going to mean that some of your quantization errors correlate (are not converted to white noise), correlated noise makes tones! Just in case anyone is tempted, don't truncate from 24bit to 16bit, you'll hear less white noise but a proportional amount of audio distortion, this can take many forms but ring modulation is the most likely, yuk! 

 So what about the white noise drowning the signal? To answer both this and the previous question really requires a better understanding of dither. Oh no, please don't start me on dither! 
	

	
	
		
		

		
		
	


	




 Oh well, but I'm only going to do this once!! If we think about the dithering process in very simplistic terms then your statement is entirely logical and correct. However, this thread has moved through the history of digital audio, from quantization errors, the dithering quantiser, noise shaped dither and now, the current gob full - the multi-level delta-sigma modulator with a full noise-shaped dither feedback loop or just for fun, the ninth order noise-shaped dither loop!!! 
	

	
	
		
		

		
		
	


	




 The simplest way I can think of explaining this is that the end result of some clever maths is that the white noise isn't really white anymore, it's white-ish, technically it's broadband noise. 
	

	
	
		
		

		
		
	


	




 It's been filtered and it's energy has been re-distrubuted away from the 1-4kHz range where our hearing is most sensitive. All the energy of the "white" noise is still there as is exactly the same probabilty of converting all the quantization errors, but now we can't hear it anymore. Interestingly, Bob Katz (one of the world's mastering engineering gurus) recons that 16bit is capable of resolving signals down to the -120dB level, way below the theoretical -96dB noise floor. This sounds plausable from what I know of current technology but I'm nearing the end of my knowledge. 
	

	
	
		
		

		
		
	


	




 If I've lost anyone, sorry! 
	

	
	
		
		

		
		
	


	




 Here's the quick answer: If you could transparently whack your amp up to well over 100dB above the noise floor, just a few seconds before your brains started dripping out of your ears, you would notice that you could hear the opening of the Alpine Symphony well below the theoretical noise floor of 96dB in 16bit audio. 
	

	
	
		
		

		
		
	


	




 If I've still lost anyone, here's the explanation I use for rock drummers: Hey, it's magic!! 
	

	
	
		
		

		
		
	


	




 I hope after all this that at least one person is now convinced of the futility of using 24bit audio for anything other than recording!


----------



## b0dhi

Ok, so I'm not 100% about these calculations, but assuming 1W/m^2 sound intensity at the eardrum = 20 Pa = 120dB SPL (100dB sound and 20dB amplification by the ear itself), this corresponds to roughly 7670 nanometers air displacement at the ear, using displacement = p/2pi.f.Z, where Z is the acoustic impedance (415 Pa.s/m usually), and f = 1Khz. 

  Quote:


 we show that a hair cell’s transducer current carries information that allows the detection of vibrational amplitudes with an accuracy on the order of nanometers. 
 

 (http://www.pnas.org/cgi/reprint/2632626100v1.pdf)

 Assuming the ear can detect displacement with an accuracy of 1nm, this means that only about 13 bits (8192) are required to encode the maximum displacement precision of the ear?


----------



## Crowbar

One centimeter is too large. A cilium in the inner ear moves in response to sound far less even for the loudest sounds. For the quietest sounds, the motion is far smaller than even you suggested.

 From Sensitivity of Human Ear:

 "The threshold of hearing corresponds to air vibrations on the order of a *tenth of an atomic diameter*."
 (specifically, 20 millionths of a Newton per square meter)


----------



## gregorio

b0hdi - I'm not aux fait with the calculations either. I have read somewhere that the maximum dynamic range of the human ear in theory, would correspond to 18.5bits. However as mentioned above, with properly noise shaped dither, a 16bit system can in theory perform down to -120dB, or the equivalent of a 20bit un-noise shaped system.

 The ear is a complex bit of gear though. For example, although we talk about the ear having a dynamic range up to 120dB, this is not quite accurate. It has a dynamic range quite a bit smaller than that. The ear is capable of moving is dynamic range window within a larger range. This is called TTS (Temporary Transient Shift). If you've ever experienced going to a club or concert and then outside, it sounds like everyone is speaking quieter or more muffled than normal, you've got TTS. The ear recovers from this after an hour or so. Pushing the ear towards the pain threshold can easily cause PTS (Permanent Threshold Shift) and as far as I know there is no treatment for this condition, the damage is permanent. It is thought that even TTS is causing some minor permanent damage, which is cumulative.


----------



## regal

This is why you NEED a 24 bit DAC with computer audio:

 The biggest advantage that computer audio has over CDP's is that you the user can do professional on the fly mastering with VST's like Ozone. If you don't have a 24 bit DAC you have no room left to add anything. This is because most CD's were compressed to use the upper 4 bits.

 Nearly every CD in my collection released in the last 10 years has about 4 bits or less of dynamic range.


----------



## b0dhi

I just want to point out that the measurements in those calculations relate to the ear's precision of measurement of displacement, not dynamic range. In this case we aren't talking about how loud or how soft the ear can hear, but how _precisely_ it can measure air displacement (at various SPLs). Forgot to mention that.


----------



## tschanrm

Great thread, it provides lots of good technical information and informed viewpoints. 

 Through reading this thread, ignoring the banter and other analogies, I agree with Crowbar's underlying opinion of reaching beyond mediocrity. This type of mindset can help break glass ceilings in a given sector, be it economics, sociology, mathematical research, etc. - you just need to be collaborative and diplomatic with the approach.


----------



## bigshot

I believe in reaching beyond mediocrity too, but I realize that human error is going to be a much bigger problem than technical error. The best recording I have ever heard- most natural sounding and clean- was recorded in 1954. Why? Not because analogue sounded better. It was because of absolute top notch engineering- mike placement, mixing, mastering.

 See ya
 Steve


----------



## monolith

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I believe in reaching beyond mediocrity too, but I realize that human error is going to be a much bigger problem than technical error. The best recording I have ever heard- most natural sounding and clean- was recorded in 1954. Why? Not because analogue sounded better. It was because of absolute top notch engineering- mike placement, mixing, mastering.

 See ya
 Steve_

 

What recording was that? I'm curious.


----------



## gregorio

Regal - Remember that in reality there is no such thing as a 24bit DAC! I know that most DACs advertise they are 24bit but that's because they can mathemetically convert 24bit but the reality is, they can't actually output anywhere near the dynamic range that a 24bit audio file represents. Have a look at the specs for your DAC, does it give a dynamic range? If so, I guarantee you that it's way below the 144dB range of the 24bit format. One of the finest professional DACs on the market (Prism ADA-8XR) has an output dynamic range of 108dB. That range is theoretically possible to encode on a 16bit CD. I'm not saying that you shouldn't be using a 24bit DAC when you're recording or processing audio, just that a 24bit DAC is not in practice quite what you think it is! In practice, even the very best 24bit DACs are actually less than 20bit DACs.

 20 years ago you needed tens or hundreds of thousands to set yourself up as a recording engineer, producer, mastering engineer or studio. Anyone with one of these job titles usually knew what they were on about. These days anyone with a PC and a sound card can (and do) call themselves a producer. The gap between the best and the worst practitioners has grown massively. A producer listed on a recording could be a top class professional or a spotty kid in his bedroom with a cracked version of Cubase and Reason. The quality you perceive on a recording is going to be far more affected by the engineer, the gear used, the producer and the mastering engineer than the potential resolution of digital audio format. Take Mike Skinner of The Streets, he's written some good songs but he unfortunately thinks he's a producer. There are some basic errors on some of The Streets albums which I wouldn't expect to hear from any of my first year students and Mike Skinner is not uncommon is this regard. In conclusion, I couldn't agree more with Bigshot's last post!

 For mixing or processing audio I agree that you need much higher bit depths to compensate for SNR issues and the cumulative effect of mathematical processing errors. My contention, and the contention of the theoriststs and manufacturers of pro audio gear, is that 24bit is not sufficient for this purpose. 32bit or preferably 48bit internal processing is required. You need to be careful with programs or plugins like Ozone, which is essentially a psuedo-professional mastering tool. The biggest growth area by far in pro-audio gear is in the budget end of the market. Ozone (or pretty much any VST), is a tool aimed at the spotty kid mentioned above, rather than at a top class professional mastering engineer. 

 I don't entirely agree with tschanrm, that we should be aiming for higher than mediocracy. Bet you didn't expect me to say that! I'm saying it because there's a growing percentage of so called engineers and producers for whom mediocracy still appears to be a distant dream! They often don't have the experience, practical skills or technical knowledge to even realise they don't know what they're doing. It's the same in most fields, you have the mechanic who works out of his back garden and the mechanic who works for the Ferrari grand prix team. I personally am always striving for top quality and so would any decent engineer or producer. You have to bare in mind also that even good engineers and producers are often ham-strung by commercial or contractual issues.


----------



## regal

Gregorio,

 You are correct that the noise floor is too high to have a true 24bit DAC, but losing some resolution to hiss is much better than clipping in a 16 bit DAC. 
 Do you understand that if you boost bass frequencies with a digital equalizer you will clip 99% of all 16 bit recordings on a 16 bit dac? And if you had 24 bit with headroom (with the right VST upscaling) you wouldn't ?

 Maybe your argument is the same end result could be attained by lowering the digital preamp volume, but then you are down 1 bit of resolution for every 6 dB. I just don't understand where you are coming from. I have 1000's of hrs of raw SBD's and VST's like Ozone + a 24bit DAC are indispensable. If you only listen to commercial RBCD's and like the mastering a 16 bit DAC is all you need.


----------



## bigshot

6dB should be enough for allowing headroom for EQ, shouldn't it? It would be better to EQ subtractively though.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *monolith* /img/forum/go_quote.gif 
_What recording was that? I'm curious._

 

Fiedler's first stereo Gaite Parisienne. It's out on SACD. My copy is from the old Franklin Mint 100 Greatest Recordings set. The balances are absolutely perfectly judged.

 See ya
 Steve


----------



## regal

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_6dB should be enough for allowing headroom for EQ, shouldn't it? It would be better to EQ subtractively though.

 See ya
 Steve_

 

Should be but not for a raw SBD where your adding plate reverb, compressing bass, etc. I agree that for most commercial CD's 6db is enough.


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## grawk

if you're doing all that every time you listen, you should do it once, and save a copy for listening.


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## gregorio

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_ Do you understand that if you boost bass frequencies with a digital equalizer you will clip 99% of all 16 bit recordings on a 16 bit dac? And if you had 24 bit with headroom (with the right VST upscaling) you wouldn't ?

 Maybe your argument is the same end result could be attained by lowering the digital preamp volume, but then you are down 1 bit of resolution for every 6 dB. I just don't understand where you are coming from. I have 1000's of hrs of raw SBD's and VST's like Ozone + a 24bit DAC are indispensable. If you only listen to commercial RBCD's and like the mastering a 16 bit DAC is all you need._

 

Whether you clip the output would depend on how much you boosted the bass by, how the CD was mastered and of course the playback material.

 If you've got a 16bit wav and you want to "remaster" it in say Ozone, I contend that depending on the processing you're doing, 24bit is not enough (let alone 16bit), you stand a chance of causing unwanted audible artefacts. Presumably your computer and plugins are operating at 32bit. However when it comes time to outputting the result of your 32bit audio to your DAC or to a new file for storage, I contend that 16bit is sufficient. The only proviso is of course that an appropriate noise-shaped dither algorithm is used to get your 32bit audio down to 16bit.

 EQ: Try this, put an audio file containing white noise into an audio program; Ozone, ProTools or whatever and read the exact output level. Now put in an EQ filter, let's say @ 18kHz. OK, so you should now notice that your output level has gone up, not down like you'd expect. Make the filter steeper, let's say 24 or 36dB/8ve, the level goes up even higher! So it's not just boosting the bass freqs we have to worry about, it's filtering too, at any freq. The filters in your DAC will likely produce a siminlar effect, that's why as a general rule, you should leave 3dB or more headroom when bouncing your output to a CD or wav file. To be honest I'm not going to be worrying about 3dB or even 6dB headroom on a CD, it's still going to have a noise floor which couldn't be heard.


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## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_In practice, even the very best 24bit DACs are actually less than 20bit DACs._

 

As Anagram Technologies has shown, combining multiple 24 bit DACs and not naively simply sending the same signal to each allows linearity to be reached down to -140 dB. The new ESS Sabre chip does something like this internally to get comparable performance--it's better than 132 dB and is a chip that *directly contradicts* gregorio's assertion. You need to keep up with the tech better.
 On the analog follow-up, circuits exist that have distortion of a few dozen parts per billion, so the electronics is simply not an issue--the tech has been around for some time already. It's the headphones/speakers that really need more work.


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## grawk

with 140db of SNR, it'd be resolving things at 65dB below absolute silence at my normal listening levels.


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## regal

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_if you're doing all that every time you listen, you should do it once, and save a copy for listening._

 


 I have 2 TB's of raw unmasters flac music. There is no way I want to master all of them, I do it on the fly as I listen/enjoy them. Plus I would want to keep the raw masters so I would need another 2 hard drives to do this.

 I honestly would prefer a 16 or 20 bit DAC, because I think DAC chips like the AD1862 sound better than anything. I guess because they are R2R instead of Sigma-Delta.


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## Crowbar

AD1952 is significantly better than the 1862, and the ESS Sabre DAC is another big improvement yet again (and this far, the latter is the only one that crosses any perceptual threshold to transparency--or it would, anyway, if one could use an external filter with it--guess I'll have to wait till the next revision).


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## gregorio

crowbar - The latest technology allows up to 150dB of headroom to be captured in 24bit audio but why would you want to? You won't be able to hear it. Unless you can wire your DAC straight into your brain, your audio is going to have to travel down a wire, to an amp, down another wire, out of your speaker or cans. By the time the audio gets to your ears you'll be lucky to achieve an 80dB range even with decent quality gear. Top quality gear might give you 90 or 100dB, why do you need an output resolution of 150dB? Unless you want to hear the real difference between a pianissimo violin and a sun going supernova!

 It's like fitting the jet engine from a euro-fighter to a bicycle. The bicycle just doesn't have the ability to use or control even a fraction of what the jet engine is capable of producing. Likewise, you've got an audio resolution which no piece of gear in the recording or playback chain can get anywhere near. Even if you could output that dynamic range it would probably kill you so what's the point? You get to enjoy fantastic resolution for a nanosecond and then never hear anything ever again!

 Think about it, if 24bit can in theory represent two hydrogen atoms colliding, how many air molecules are colliding in your sitting room? Ultimately, no matter how good your playback system, you are always going to have to consider the noise floor of the listening environment. World class studios achieve 25-30dB noise floor, a sitting room probably 40 - 50db. No matter which way we look at it or how good the gear, 24bit resolution for playback cannot be justified.


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## gregorio

My advice Regal is to add your processing, EQ and whatever, and then properly dither it down and store it as a 16bit file. Outputting from the file rather than processing on the fly is going to be less strain on your DAC and CPU and is likely to result in fewer errors and therefore better quality.

 Looking for more bits from an output file is not the way to improve your audio quality. Buy a better quality EQ, learn more about mastering and EQ techniques, buy a better amp or speakers/cans, buy a better DAC, improve the acoustics of your listening environment, use balanced analogue audio transfers, buy high quality well mastered albums. Every single one of these items (let alone all together) is going to improve the perceived sound quality astronomically, compared to increasing bit depth to 24bit.

 Very high sampling rates and bit depths beyond 16bit may have applications for recording but the marketing wheels of the consumer marketplace are taking you for a ride. I mean, who coined the phrase hi-res audio? It's no more hi-res than 16bit. What about the ridiculous 192kFs/s, it's a joke. The worst part is some people thinks it sounds better ... No! The DAC maker has deliberately put in a good filter at 192k and rubbish ones at lower rates. Instead of the consumer thinking and telling all his mates how great his DAC is and that everyone should be using 24/192, he should be telling his mates that his DAC is cr*p and only sounds any good when it runs at 192k.


----------



## regal

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_My advice Regal is to add your processing, EQ and whatever, and then properly dither it down and store it as a 16bit file. ._

 


 I agree this would be best but do you have time to master and dither 2 TerraBytes of flac files? I sure as heck don't. I'll do it on the fly and use my PCM1704 DAC (24 bit)which Erno Borbely says is the finest ever made. Surely he's wrong but its good enough for me and it saves me a lot of time. I can change masterings on the fly rather than have it fixed. 

 If I could find a VST that would up-convert from 16 bit to 24 bit, apply DSP, and then dither back to 16 bit , I could use a 16 bit DAC no problem. Do you know of any?

 I agree 100% that increasing bit depth of a 16 bit file on its own does nothing to improve SQ, but a 24 bit DAC adds a lot of flexibility. This is why no engineers have made a 16 bit DAC in the last decade. You are just listening to the marketing divisions too much, they don't even believe themselves.


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## gregorio

Hi Regal,

 I see your problem. As you're listening can't you also record the file down? Next time you can just output the 16bit file you recorded rather than processing again and you can gradually work you way through your collection as you listen to them. If you're using Ozone 3, it has various noise shaped dither algorithms built in, which are user selectable. You need to have a look at how it's set as Ozone does much of it's processing at 64bit.

 Also have a look at how you're EQ'ing. As a general rule, it's usually better to cut rather than boost. In other words, if you want more bass, cut the mid and treble rather than boosting the bass. You can also use a multi-band compressor to increase the perceived volume of the bass without raising it's output level. AFAIK there is a multi-band compressor in Ozone but I can't say how good it is.

 I don't know any decent professional who uses Ozone and my personal choice of dither is usually Pow-R (or Waves) but I don't know if it's available as a VST. Nevertheless, Ozone 3 should give you decent results. Have a look at this:

http://www.izotope.com/products/audi...eringGuide.pdf


----------



## bigshot

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Next time you can just output the 16bit file you recorded rather than processing again and you can gradually work you way through your collection as you listen to them._

 

That's what I do with my LPs. In about eight hundred years, I should have all of them transcribed!

 See ya
 Steve


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## jiiteepee

Quote:


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

*If I could find a VST that would up-convert from 16 bit to 24 bit, apply DSP, and then dither back to 16 bit , I could use a 16 bit DAC no problem. Do you know of any?*

 ..._

 

 Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Hi Regal,

 I see your problem. *As you're listening can't you also record the file down?* Next time you can just output the 16bit file you recorded rather than processing again and you can gradually work you way through your collection as you listen to them. ...

 [/url]_

 

Hmm... aren't everything converted to 32- or 64 bit data nowadays when read into software/plug-in and then finally dithered by the settings you make? Here's what you can try ... get Acon Digital's EffectChainer VST, (Sonalksis FreeG VST), Voxengo Recorder VST and Voxengo SPAN VST (all freeware). 
 - load the EffectChainer into your playback software (Foobar/Winamp/etc.) VST slot, and load other plug-ins into its slots:
 (- a level control VST), 
 - your favorite EQ 
 - Voxengo SPAN for analys and finally 
 - Voxengo Recorder for to record the processed audio (i.e. EQ becomes saved as in selected resolution)

 This is just an example (a very slow process since you need to cut the resulting wave if you play whole album at once as for an example).

 Your playback software should be set for 24- or 32-bit output (or even 64-bit, if all other components supports this resolution).

 If there's clipping on your source audio, iZotope released just a plug-in version for their RX. iZotope, Inc - Audio Signal Processing Hardware, Software, Plug-ins, Technology Licensing


 Here's a screen shot of an example from Foobar 0.9.x:

http://img134.imageshack.us/img134/6668/0expfxgc0.jpg


 jiitee


----------



## gregorio

Jiitee - "Hmm... aren't everything converted to 32- or 64 bit data nowadays when read into software/plug-in and then finally dithered by the settings you make?"

 It depends, it might be or it maybe that the end result of calculations is just a 32 or 64bit result. It's likely that your plugin effects take the 32bit datastream, calculate a result at 64bit and then truncates (or dithers) back to 32bit. Professional software like ProTools works with 24bit files in a 48bit environment and all datapaths are dithered back to 24bit.

 "Your playback software should be set for 24- or 32-bit output (or even 64-bit, if all other components supports this resolution)."

 Your playback software almost certainly should not be set for 32 or 64bit output! You should have a plugin (or inbuilt process) at the end of your processing chain that dithers back down to 16bit. You output a 32bit or 64bit file to a DAC and it's going to truncate all the least significant bits (LSBs) that it can't process, anything above 24bits with a 24bit DAC. Truncation is about the worst thing you can do to digital audio and you are going to get higher quality audio with fewer nasty artefacts dithering to 16bit than truncating to 24bit. The errors created by truncation are correllated to the audio, by definition you are going to get artefacts, probably ring modulation, that's bad!!

 The only time you want to output a file of more than 16bit resolution is if your software doesn't have a decent algorithm for noise shaped dithering and you want the raw data to pass through a different processor or if you are passing the data files on to a mastering engineer.


----------



## sejarzo

Interesting point, gregorio......typically, it's recommended by most set up guides for Foobar to use "24 bit padded to 32 bit" or "32 bit fixed point", I think, for the playback output format.

 Now I'm unsure if that's the best option.

 EDIT: After doing some experimenting, the *only *way the 0404 USB can apparently operate over ASIO is to use the "24 bit fixed padded to 32 bit" or "32 bit fixed point" options. Seems as if the onboard Realtek on my desktop can tolerate 16, 24, or 32 bit format options when run via DirectSound or Kernel Streaming, though.

 Somewhere I read what the difference was between the two options I listed above for the 0404 USB, but I can't recall the specifics at this point.

 I don't run any plug-ins in Foobar, no replaygain, no volume control, etc. etc. Could it be that for playback-only applications (of commercially available music) that the output format is simply that, a format to make it compatible with a specific device driver?


----------



## HFat

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_typically, it's recommended by most set up guides for Foobar to use "24 bit padded to 32 bit" or "32 bit fixed point", I think, for the playback output format._

 

Foobar's UI says that one shouldn't set a higher bit depth than what the DAC supports (or SQ might suffer). Surely it's more trustworthy than some guide.

 EDIT: Foobar also has a dithering option... I take it to mean that truncation is what happens if you don't check the box.


----------



## jiiteepee

*WARNING* for Foobar/Winamp users who streams audio through the older VST addon which needs Winamp DSP to work ... by the TobyBear Bitviewer VST plug-in, data coming from player into vst addon is always 16-bit for Foobar (not by the player settings nor by the source) and for Winamp it's by the source (if you change the output format w/ or w/o dithering it does not have effect) ... when some processing plug-in is enabled data becomes 32-bit (normally). I did this test by inserting the bitviewer only and by inserting the effectchainer into foobar and then added Bitviewer VST -> EQ VST -> bitviewer VST. 

 EDIT: The newer VST addons (both, Cannibal Zerg's and George Yohng's) for Foobar looks to work by the source ... so, looks like the output bit-resolution setting found on Foobar does not have effect in processing when DSP VST is activated in Foobar DSP Manager.

 jiitee


----------



## b0dhi

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_crowbar - The latest technology allows up to 150dB of headroom to be captured in 24bit audio but why would you want to? You won't be able to hear it. Unless you can wire your DAC straight into your brain, your audio is going to have to travel down a wire, to an amp, down another wire, out of your speaker or cans. By the time the audio gets to your ears you'll be lucky to achieve an 80dB range even with decent quality gear. Top quality gear might give you 90 or 100dB, why do you need an output resolution of 150dB? Unless you want to hear the real difference between a pianissimo violin and a sun going supernova!_

 

You seem be be confused about SPL and dynamic range of hearing. The ear _can hear_ a dynamic range of 130dB (between threshold of audibility and threshold of pain). This nonsense you're spewing about hydrogen atoms colliding and supernovae is unsubstatiated hyperbole.

 If the electronic hardware doesn't have the ability to reproduce a dynamic range equal to that of the ear (at the equivalent SPL), then that's a technological limitation that should be improved. The _ear_, however, has no such limitation, and _can_ hear 130dB. It's up to the audio chain to rise to meet the abilities of the ear. This will only be impeded when people like you keep hitting glass ceilings and encouraging others to do the same.


----------



## regal

Quote:


  Originally Posted by *jiiteepee* /img/forum/go_quote.gif 
_*WARNING* for Foobar/Winamp users who streams audio through VST addon ... by the TobyBear Bitviewer VST plug-in, data coming from player into vst addon is always 16-bit for Foobar (not by the player settings nor by the source) and for Winamp it's by the source (if you use dithering it does not have effect) 
 jiitee_

 

Thanks for this info. I tried the Bitviewer and it is showing that foobar + emu 4040 + ASIO is outputing 32 bit as setup under foobar output. So now I am setting the Ozone to dither to 24 bits. This gives me the headroom I need to add DSP. If I had known all this earlier I would have stuck with a 16 or 20 bit DAC and just use dithering.

 I think that's Gregorio's argument. Dither 24 bit to your 16 bit DAC instead of using a 24 bit DAC and you will never hear the difference with today's analog technology. But when he talks about atoms colliding and what not he's losing a little focus.


----------



## jiiteepee

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_Thanks for this info. *I tried the Bitviewer and it is showing that foobar + emu 4040 + ASIO is outputing 32 bit as setup under foobar output.* So now I am setting the Ozone to dither to 24 bits. This gives me the headroom I need to add DSP. If I had known all this earlier I would have stuck with a 16 or 20 bit DAC and just use dithering.

 I think that's Gregorio's argument. Dither 24 bit to your 16 bit DAC instead of using a 24 bit DAC and you will never hear the difference . But when he talks about atoms colliding and what not he's losing a little focus._

 

The newer version of VST addon for Foobar looks work properly by the source but the old seem to have some issues (the one which uses the Winamp bridge). (I added this info into my previous post) 
 Maybe the driver model has something to do w/ this as well. I didn't get the fixed bit-resolution from Foobar work w/ any "native" audio material (WAV). Mp3 showed 32-bit. This test was made w/ SB card ... I'll look if the 0404 shows something different.

 jiitee


----------



## regal

Quote:


  Originally Posted by *jiiteepee* /img/forum/go_quote.gif 
_The newer version of VST addon for Foobar looks work properly by the source but the old seem to have some issues (the one which uses the Winamp bridge). (I added this info into my previous post) 
 Maybe the driver model has something to do w/ this as well. I didn't get the fixed bit-resolution from Foobar work w/ any "native" audio material (WAV). Mp3 showed 32-bit. This test was made w/ SB card ... I'll look if the 0404 shows something different.

 jiitee_

 


 The only DSP I was using was the Yohng VST wrapper. But these are all flac.


----------



## jiiteepee

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_The only DSP I was using was the Yohng VST wrapper._

 

In Foobar, if you change the output resolution to 8-bit (dither or not), do you get the data showing up in bitviewer as 8-bit data? Meaning, where from/in which point is the bit-depth read ... is it from source file or from Foobar/addon plug-in settings?

 jiitee


----------



## regal

Quote:


  Originally Posted by *jiiteepee* /img/forum/go_quote.gif 
_In Foobar, if you change the output resolution to 8-bit (dither or not), do you get the data showing up in bitviewer as 8-bit data? Meaning, where is the bit-depth taken ... from source file or from Foobar or addon plug-in settings?

 jiitee_

 


 Well I can't change the output resolution (its greyed out) when using the EMU ASIO driver. If I go to ASIO virtual devices/edit I can see that the ASIO driver is set to 32 bit. Also I am not dithering in foobar. I meant when using Ozone 3 I would dither down to 24 bit there.


----------



## Febs

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_The ear, however, has no such limitation, and can hear 130dB._

 

Not for very long. SPLs at that level can cause permanent hearing damage almost instantaneously.


----------



## gregorio

b0hdi - Of course what I'm refering to is throetical, or do you actually think that a microphone is capable of recording two hydrogen atoms colliding? Also of course a supernova is silent, space is a vacuum and therefore unable to transmit sound. 24bit is capable of recording the SPL which would in theory be created by two hydrogen atoms colliding, if full scale was the upper limit of hearing which is about 120dBSPL.

 There is of course no point to a dynamic range of 130dB as this is beyond the point of causing pain and definitely beyond the point which would cause irrepairable damage. Bare in mind we've also the noise floor to consider, so that's a significantly smaller dynamic range again. In fact an output of higher than 85dBSPL is not recommended by government health studies and this includes the noise floor.

 There is not much difference between dynamic range and dBSPL except that the maximum level of hearing (dBSPL), let's say 125dBSPL is actually higher than the ear's dynamic range. The ear has a smaller dynamic range than this but is capable of moving it's window within the larger 125dB range. When listening to high SPLs roughly +90dBSPL, the ear shifts it's window (TTS) to desensitize itself to help avoid permanent damage. Of course when in a TTS condition the ear is insensitive to low SPLs. So again, the realistic usable dynamic range of the ear without causing damage is very roughly about 50-60dB above the noise floor of a quiet sitting room. 16bit and modern equipment is capable of recreating this range easily, however most CDs use a range significantly lower than this. No sensible mastering engineer or record company will create a product with more than a 60dB dynamic range, because we don't want to damage the hearing of our end customers.


----------



## sejarzo

gregorio--re the bit depth issue:

 What do CD players with 24 bit DAC's do with the 16 bit data stream from the transport? I would presume they don't re-dither it, right--presuming that the original source was > 16 bit and dithered down in the production process?

 If one isn't using any DSP in a player, is there any problem with simply zero-padding the LSB's?

 I did find that the 0404 USB also works in Foobar via ASIO with the 16 bit fixed padded to 32 bit option. The only way the 0404 USB ASIO driver will accept the data stream is if it's in 32 bit format, as far as I can tell.


----------



## gregorio

Regal, et al - It starts getting complicated when we start to look at what individual pieces of software actually do to the digital data. It is quite common for bit depths to be changed several times within the processing chain as the data is routed in and out of various plugins. There is nothing intrinsically wrong with this providing the changes are happening at very high bit depths (32bit or greater) with dithering going on between the changes. When it comes to output it can again be tricky to work out what is going on. It's important to find out the output dit depth of your audio program and make sure it's not higher than the maximum input bit depth of your DAC. If it is, your DAC will just ignore the lower bits outside it's range, effectively truncating these bits, causing not additional noise but actual tones which are going to interact with your music and are going to cause unwanted effects. The amount of damage (quantisation errors) caused by truncating are going to be proportional to the bit depth, so the higher the bit depth the less audible these errors are going to be. However, who wants to have any errors when they can relatively easily be totally avoided by noise-shaped dithering down to a bit depth supported by your DAC.

 AFAIK, there are no 32bit DACs on the market, so if your DAC accepts a 32bit signal it could be worth finding out how the DAC gets the data back to 24 or 16bit. I wouldn't be too happy to discover that it's truncating, although truncating 32bit to 24bit probably wouldn't cause too many audible problems.


----------



## b0dhi

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Not for very long. SPLs at that level can cause permanent hearing damage almost instantaneously._

 

It would take longer than "instantaneously", but yes, extended listening at that level would be damaging. But there's no reason why a crescendo couldn't hit 120-130dB and then die down, as it does in a real concert. Just because it's possible to produce 130dB SPL doesn't mean the music must do so constantly, as it does in today's music.


----------



## Kaviar

Soooo.... I could just as well get a 16 bit dac then? 
	

	
	
		
		

		
			





 What is the conclusion?


----------



## b0dhi

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_b0hdi - Of course what I'm refering to is throetical, or do you actually think that a microphone is capable of recording two hydrogen atoms colliding? Also of course a supernova is silent, space is a vacuum and therefore unable to transmit sound. 24bit is capable of recording the SPL which would in theory be created by two hydrogen atoms colliding, if full scale was the upper limit of hearing which is about 120dBSPL._

 

I hope you understand that dBSPL is a measure of absolute sound intensity. It's used to indicate how loud things are. Two atoms colliding is well below the the lower bound of hearing (quieter than 0dB SPL), and therefore it has no relevance to this discussion.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_There is of course no point to a dynamic range of 130dB as this is beyond the point of causing pain and definitely beyond the point which would cause irrepairable damage._

 

I don't know if you've ever been to a concert, or out of your basement before, but sound at that level does not cause instant damage. Only _sustained_ listening does, which is why it's called the threshold of pain and not the threshold of hearing damage (which is typically beyond 130dB). So yes, the ear can hear it, and it's safe to do so for _short_ durations, so there _is_ a point to being able to reproduce it.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_There is not much difference between dynamic range and dBSPL_

 

One is an absolute measure of sound intensity, the other is simply a ratio.


----------



## sejarzo

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_AFAIK, there are no 32bit DACs on the market, so if your DAC accepts a 32bit signal it could be worth finding out how the DAC gets the data back to 24 or 16bit. I wouldn't be too happy to discover that it's truncating, although truncating 32bit to 24bit probably wouldn't cause too many audible problems._

 

My less-than-completely educated guess is that for a USB device such as the 0404 USB that uses its own drivers, it somehow made sense to use a 32 bit word to represent each sample, either in or out, considering that it is first and foremost a 24 bit recording device.

 Seems to me that a 16 bit sample would be padded with 8 leading zeroes and 8 trailing zeroes to make the 32 bit word. The driver essentially tells the external box what bit depth and sample rate is coming over with each "bunch" of data (unsure if the right word is "packet", "frame", or something else, but I think you can understand what I am driving at.)

 The DAC itself is 24 bit, so there shouldn't be any dithering necessary when feeding it 16 bit material. Truncating zeroes shouldn't make any difference.


----------



## gregorio

Serarzo - what you're suggesting is absolutely true, provided that there is absolutely no processing going on while the signal is at 32bit. If you're converting from 16bit to 32bit and back to 16bit again, in theory there shouldn't be any difference and truncating won't do any harm but if there is some processing going on then the last 16bits of the 32bit file may contain data which will be truncated. If all you're doing is truncating zeros there shouldn't be any artefacts.

 Kaviar - Unless you are recording and manipulating audio, in other words just using your DAC to playback CDs or 16bit wavs, MP3s, Flacs, etc., then no, there is absolutely no need to get a 24bit DAC. There could even be an advantage to buying a 16bit DAC, as a high quality 16bit DAC should in theory cost slightly less than the same quality 24bit DAC as less processing is required.

 b0hdi - Even a very loud rock concert will not exceed 110dB. 120-130dB will not be experienced by an audience in a concert venue. If you wish to believe that it's perfectly safe to to listen to 120-130dB levels then you go right ahead. I need my hearing for my professional work and there is no way I'd expose my ears to anywhere near those levels. The legal limit in this country currently stands at prolonged exposure to levels of 85dB and there is growing evidence to suggest this figure it too high and needs to be lowered to 75dB. Bare in mind that if you're listening to music around the 120dB level then you are not going to hear what is going on down at the 40dB level as your ears will temporarily transient shift (TTS). If you do this with any regularity it will become a permanent transient shift!! If you don't want to permanently damage your hearing, keep your listening below 90dB and better still below 80dB.

 If no one wants to believe anything else I've said in this thread that's fine but this is really serious, we are not talking about your listening enjoyment but about your ability to function in society. Please do not exceed the figures I've suggested. *You have been warned*!!
 .


----------



## Febs

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_It would take longer than "instantaneously", but yes, extended listening at that level would be damaging. But there's no reason why a crescendo couldn't hit 120-130dB and then die down, as it does in a real concert. Just because it's possible to produce 130dB SPL doesn't mean the music must do so constantly, as it does in today's music._

 

According to the Dangerous Decibels site, exposure to 120 dB can cause immediate damage. The NIOSH recommend exposure for 115 dB is *30 seconds*, and the recommend exposure duration is halved for each additional 3 dB of exposure. OSHA considers exposure above 105 dB to be an "extreme risk." You're talking about SPLs several orders of magnitude higher than that.

 Moreover, I'm not aware of any musical instrument that is capable of producing SPLs of 130 dB. You're talking about volume levels that would rival the sound of a jet airplane taking off.


----------



## bigshot

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_It would take longer than "instantaneously", but yes, extended listening at that level would be damaging. But there's no reason why a crescendo couldn't hit 120-130dB and then die down, as it does in a real concert.._

 

Have you checked your neighborhood at RottenNeighbor.com? If you're cranking out peaks of 130dB on your rig, I bet your neighbors have something to say about it. What kind of speakers are you using?

 See ya
 Steve

 P.S. It would be interesting to find out how many people here have reports at Rotten Neighbor about them!


----------



## b0dhi

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_According to the Dangerous Decibels site, exposure to 120 dB can cause immediate damage. The NIOSH recommend exposure for 115 dB is *30 seconds*, and the recommend exposure duration is halved for each additional 3 dB of exposure. OSHA considers exposure above 105 dB to be an "extreme risk." You're talking about SPLs several orders of magnitude higher than that.

 Moreover, I'm not aware of any musical instrument that is capable of producing SPLs of 130 dB. You're talking about volume levels that would rival the sound of a jet airplane taking off._

 

Not to pick nits in the numbers, but clearly levels of around 115dB are permissible for short durations, which is exactly how I was suggesting they be used in a high dynamic range audio system. A solid drum hit, for example, would hover around 125dB, but because it's a very short percussive sound, it normally won't do damage unless you're listening at that level for an extended duration. My point is that having that dynamic range available (or at least something around 115-120dB) will be both safe and useful so long as it's used responsibly by the audio engineers and the listener.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_b0hdi - Even a very loud rock concert will not exceed 110dB. 120-130dB will not be experienced by an audience in a concert venue._

 

 Quote:


 The power metal band Manowar is one claimant of the title of "loudest band in the world", citing a measurement of 129.5 decibels in 1994 in Hanover. The Guinness Book of World Records once listed it as the record holder for the loudest musical performance for an earlier performance in 1984. However, Guinness does not recognize Manowar's later claim, because it no longer includes a category of loudest band, reportedly because it does not want to encourage ear damage.[4][5][6]

 The Who were the last band listed as the record holder, at 126 decibels, measured at a distance of 32 metres from the speakers at a concert at Charlton Athletic Football Ground on 1976-05-31. Other previous record holders include Deep Purple (117 decibels), The Rolling Stones , and KISS.[7][8][9]


----------



## gregorio

b0hdi - The problem with all the figures is quite simply no one knows for sure the exact point when damage starts and it appears to vary on an individual basis.

 What is obvious is that just through normal usage our ears naturally deteriorate. The question is, will subjecting your ears, even for short durations, at levels exceeding 90dB damage your hearing. My guess, and before anyone asks I have no definitive proof here, is that it will contribute to the natural deterioration of your ears and bring on loss of level and loss of high freq sensitivity earlier and more severely than would have occurred otherwise. I always try to keep my peak levels down to <85dB, my job depends on my hearing and I don't want to risk it. Also, the ear needs time to transient shift (to get into the TTS state) so it's easily possible to damage your hearing with just occasional high energy peaks. Every composer and musician should know that it's the relative difference in levels which make for loudness perception.

 Interestingly some of the levels of gigs listed in the Guinness Book are unlikely to be beaten as the legal permissible SPLs have, over the years, been reduced as better understanding of hearing damage filters it's way through to the legal system. That's why I said that it's unlikely (currently) for you to experience levels higher than 110dB even at a rock concert.


----------



## regal

I don't understand this push for 115db, right now any CD you buy is luck to have 20 dB dynamic range


----------



## gregorio

I agree Regal, in fact some have a dynamic range of less than 10dB! However, for classical music the dynamic range is much larger and listening to a live orchestra is likely to give you a bigger dynamic range than you are likely to hear on a recording. So the idea behind some of the suggestions in this thread is to match the dynamic range of a recording with that of a live performance. In reality though a 16bit recording can encode the dynamic range of a live orchestral performance but this range is then limited by engineers to take account of the fact that the vast majority of listeners do not have systems capable of recreating the full dynamic range that a CD can record. The main tool we use for this is compression, which works well at providing the illusion of more dynamic range than is actually on the recording. The problem comes when compression is overused but that's another thread!!


----------



## nick_charles

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_I don't understand this push for 115db, right now any CD you buy is luck to have 20 dB dynamic range_

 

Large scale symphonic works can have much bigger dynamic ranges. Mahler's first, the very first Classical CD I bought back in 1984 (Solti, CSO), starts with a barely audible theme that you need a really quiet room to hear, and gets very loud in parts...actually if you want Music to exploit the dynamic range of your rig Mahler is a pretty good composer...


----------



## gregorio

Hi Nick, Mahler is good for the large scale symphonic works and large dynamic ranges, as are most of the big post-romantics and neo-classicists: R. Strauss, Stravinsky, Shostakovich, Holst, et al.


----------



## jiiteepee

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_Well I can't change the output resolution (its greyed out) when using the EMU ASIO driver. If I go to ASIO virtual devices/edit I can see that the ASIO driver is set to 32 bit. Also I am not dithering in foobar. I meant when using Ozone 3 I would dither down to 24 bit there._

 

By the BitViewer VST, when inserted in Foobar 0.9.5.1 VST Host DSP (links in my sig), data from various format source files comes into DSP slot as following resolutions: 

 Lossy:
 16-bit MP3 --> 32-bit
 24-bit WMA /440kbps) --> 16-bit (is there some issue in en-/decoder?)
 16-bit AAC --> 32-bit
 16-bit OGG --> 32-bit
 Lossless:
 FLAC --> file bit depth
 WAV --> file bit depth

 I checked all above by recording the output from Foobar->VST->(no processing plug-ins) using Voxengo Recorder to 32-bit WAV files and then looked the data in Hex Editor ... :

 - 24-bit WAV and FLAC to -> 32-bit audio become padded to 3 byte values (00 0f 0f 0f) ... OK
 - 24-bit WMA padded to 2 byte values (00 00 0f 0f) ... ??? 
 - mp3, ogg and aac ... 4 byte values, no padding (0f 0f 0f 0f) ... OK
 - 16-bit FLAC -> padded to 2 byte values (00 00 0f 0f) ... OK

 As the processing by plug-ins (EQ, etc.) added into VST DSP is done either using 32- or 64-bit resolution, which would be the best playback 'source' format and should there be 32 to 24-bit dithering as you suggest?

 EDIT: To connect this matter to the OP question, wouldn't it be better to have 24-bit or even 32-bit source files against 16-bit when you like to add processing into signal path (though, this isn't the case w/ CD audio rips but for vinyl/tape ripping only)? 


 jiitee


----------



## shinew

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_ Every composer and musician should know that it's the relative difference in levels which make for loudness perception._

 

as a musician, I cannot agree more.

  Quote:


  Originally Posted by *regal* 
_I don't understand this push for 115db, right now any CD you buy is luck to have 20 dB dynamic range_

 

This is probably true for non-classical music. I was just listening to Mahler Symphony No. 1 & 5 conducted by Berstein last night from this collection(highly recommended if you like mahler btw) -> Amazon.com: Mahler: The Complete Symphonies & Orchestral Songs / Bernstein: Gustav Mahler,Leonard Bernstein,Barbara Hendricks,Dietrich Fischer-Dieskau,Wiener Philharmoniker,Christa Ludwig,Philip [1] Smith,Joseph Alessi,Helmut Wittek,Jaap Van Zweden,L, I found myself having to adjust the volume between the loudest & softest passage because otherwise I either could not hear all the details or it was too loud for my ears to handle(painful to listen to). I wonder what's the dynamic range would be on that. I bet it's not even close to 80db, let alone 115db.

 EDIT:didn't see that people have commented on Mahler already


----------



## shinew

Ok I just did a quick check on the dynamic range of Mahler 5 4th mov Adagietto(Berstein). It has the dynamic range roughly around 57 dB between the softest & loudest section.

 EDIT:I meant 4th mov, not 3rd.


----------



## fault151

Hi guys i have a question. I have a 24k music fidelity dac. I have set my output on my mac to 96khz. Will my dac try and play every song in 96khz by up sampling it, or, does i play the music at the highest rate it was recorded in at? Ie. 44.1khz?


----------



## nick_charles

Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_Ok I just did a quick check on the dynamic range of Mahler 5 3rd mov(Berstein). It has the dynamic range roughly around 57 dB between the softest & loudest section._

 

Hmmm, that is less than I would have expected.

 How about the first movement of the first ? - I am at work so I cannot check but on my version the opening is so quiet is is barely audible...

 But I do have it on MP3...

 According to CoolEdit the left channel has a dynamic range of about 68db and the right channel has a dynamic range of about 56db 
	

	
	
		
		

		
			





 On my Mahler 5 mvt 3 Cooledit clocks it in at about 68db , but a wav rip is probably a more reliable test...

 In any case well within the capabilities of 16 bits


----------



## shinew

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Hmmm, that is less than I would have expected.

 How about the first movement of the first ? - I am at work so I cannot check but on my version the opening is so quiet is is barely audible...

 But I do have it on MP3...

 According to CoolEdit the left channel has a dynamic range of about 68db and the right channel has a dynamic range of about 56db 
	

	
	
		
		

		
		
	


	


_

 

At the 1st second of the very begining(the softest part) of Mahler 1 mov 1, it's about -50db.
 i don't see such big variations between channels, but i think some variations in the l/r channles are to be expected since if both channel sound equally loud all the time, we won't be able to pinpoint where the instrument's sound is coming from,isn't it?


----------



## nick_charles

Quote:


  Originally Posted by *shinew* /img/forum/go_quote.gif 
_At the 1st second of the very begining(the softest part) of Mahler 1 mov 1, it's about -50db.
 i don't see such big variations between channels, but i think some variations in the l/r channles are to be expected since if both channel sound equally loud all the time, we won't be able to pinpoint where the instrument's sound is coming from,isn't it? 
	

	
	
		
		

		
		
	


	


_

 

I will bring in my Sinopoli CDs tomorrow and try again. I am not sure I trust my results here...


----------



## bigshot

Check Karajan's digital Parsifal. I could never listen to that because the dynamics were unlistenably wide. I bet it was over 60dB- too much!

 See ya
 Steve


----------



## HFat

This is off-topic but this thread is already big enough that no one will care... hopefully.

  Quote:


  Originally Posted by *fault151* /img/forum/go_quote.gif 
_I have set my output on my mac to 96khz. Will my dac try and play every song in 96khz by up sampling it, or, does i play the music at the highest rate it was recorded in at? Ie. 44.1khz?_

 

Your DAC will never know that the music was originally recorded at anything else than 96k.
 In theory, it would be better not to upsample to 96k but you won't hear the difference if your Mac does it well.
 Some people say that upsampling can work around problems in some systems. I'm not saying it does in your case or even that it's likely but it's possible.


----------



## gregorio

Hi Nick - 68dB is a pretty big dynamic range for a recording. 50-60dB is more usual for something like Mahler.

 The dynamic range is normally limited to this otherwise listeners are likely to run into the types of problems experienced by Shinew. He found himself having to adjust his levels and that was with a 57dB range. Hopefully people here can now see why even 16bit CD provides for a dynamic range far in excess of what is required even by a wealthy audiophile.

 It's not unusual to find different dynamic ranges on each channel, particularly with orchestral recordings where it's often neccesary to compress the low end of the orchestra because of the additional energy in the low frequencies. Much of the low frequency in an orchestra is biased towards the right hand side, due to the placement of the tuba and double basses.

 Fault151 - "I have set my output on my mac to 96khz. Will my dac try and play every song in 96khz by up sampling it, or, does i play the music at the highest rate it was recorded in at? Ie. 44.1khz?"

 It depends on the DAC. My guess is that it will be trying to upsample, which is the latest fashion in DACs. Depending on the DAC this may result in a perceived improvement, even though upsampling is likely to cause additional errors.


----------



## shinew

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_The dynamic range is normally limited to this otherwise listeners are likely to run into the types of problems experienced by Shinew. He found himself having to adjust his levels and that was with a 57dB range. Hopefully people here can now see why even 16bit CD provides for a dynamic range far in excess of what is required even by a wealthy audiophile._

 

Just to clarify, if i were to really fine tune the volume pot in the begining of the movement with the dynamic range in mind, I could still get a enjoyable listening experience without adjusting the volume during the movement. I was a bit "loose" with the volume in the begining since the majority of the 4th mov are very soft w/ strong sonority. I would say my top limit for the dynamic range is about 60-63dB.
 I was using ER-4S in a quite home environment btw.


----------



## grawk

If you set IEMs so you can hear the lowest part of a 60db range, when it hits it's peak, you will receive hearing damage.


----------



## shinew

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_If you set IEMs so you can hear the lowest part of a 60db range, when it hits it's peak, you will receive hearing damage._

 

why?


----------



## bigshot

Because you have to raise the lowest level above the ambient level of the room. That takes it up to extremely loud ranges. The comfort zone for normal orchestral music is 40-50dB. Anything beyond that and you're into the neighborhood where your neighbors will hate your guts!

 See ya
 Steve


----------



## shinew

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Because you have to raise the lowest level above the ambient level of the room. That takes it up to extremely loud ranges. The comfort zone for normal orchestral music is 40-50dB. Anything beyond that and you're into the neighborhood where your neighbors will hate your guts!

 See ya
 Steve_

 

I was in a quite room, using my IEM which are capable of blocking more than 30dB of outside noise, so there really isn't any ambient noise to speak of, or neighbors to annoy.


----------



## gregorio

shinew - I would say in this particular case you are probably going to be OK with a 60dB dynamic range. You are still going to have a noise floor, even with IEMs. If the noise floor is say 20dB then another 60dB above that will be very loud but shouldn't cause damage. If your noise floor is higher, then you could be running a risk. Bare in mind that the noise floor of a world class recording studio is likely to be around 30dB, in an average sitting room you are probably looking at 40-50dB noise floor.


----------



## bigshot

With big contrasts in volume, you're still going to contend with your ears' natural dynamic shift as mentioned above. I find dynamics become unpleasant to listen to above 50dB. It sounds like I'm sitting on top of the orchestra instead of sitting in a seat in the audience. It doesn't sound natural.

 See ya
 Steve


----------



## gregorio

I agree with Bigshot here, 50dB range is normally fine for me, too much more than that and I'm reaching for the volume control. There are some instances where I'm happy with a bigger dynamic range and the Adagietto from Mahler's 5th is one of them. The reason being that the low frequencies need more power to subjectively appear the same volume. So when you've got the huge bass note joining the rest of the strings a couple of mins from the end, you need a lot of energy (and therefore dynamic range) to represent the climax.


----------



## OblivionLord

I think that it's all totally subjective. Not everyone's ears have the the same capabilities but we can surly go off of a standard.


----------



## bigshot

I think that the people who want more dynamics than the 90dB or so of redbook just don't know what 90dB sounds like. It's not just abstract numbers. Those numbers represent sound. I learned early on to try to equate specs to what they sound like in the real world. I discovered that the fraction of a percentage of THD that I was sweating didn't mean a damn.

 See ya
 Steve


----------



## gregorio

Quote:


  Originally Posted by *OblivionLord* /img/forum/go_quote.gif 
_I think that it's all totally subjective. Not everyone's ears have the the same capabilities but we can surly go off of a standard._

 

The difficulty is worse than that because it's not just people's hearing but also the systems and environment in which they're listening. Producers and engineers have to put out a product which can be appreciated by as many people as possible. So large dynamic ranges on CDs are off the menu, unless there is a world wide general improvement in the systems which most people use to listen to their music. As a general rule, people's equipment and listening environment are currently the most limiting factors. There's no point in us putting out a product that only audiophiles can appreciate, at the expense of the vast majority of the buying public.


----------



## bigshot

Wider dynamics would not make the mix any better for an audiophile. If you have to jump up and change the volume every few minutes for a normal stereo, you'd have to do the same on an audiophile system. The reason most mixes sit around 40dB is because that is what sounds good to human ears.

 See ya
 Steve


----------



## grawk

Well, most mixes lately are more like 12db. Most GOOD mixes are around 40.


----------



## frankR

I'm surprised this discussion raged on as long as it did after I gave up.

 A few weeks ago I got the chance to listen to a comparison of SACD and redbook from my EMU-0404 on VMPS RM40 speakers.

 Bottom line. You are near as can’t hear if you can not discern the difference between high resolution audio and redbook.

 I also did a more careful analysis of the frequency spectrum of the high resolution audio tracks using Matlab. The results were the same. The frequency content is obviously distorted in the 16/44 audio. It is unable to resolve details in the audible spectrum that 24/96 audio can.

 The science confirms what my ears hear.

 I don’t know how to make this anymore clear. As far as I’ve seen, I’m the only who has posted solid data which backs-up my assertions.


----------



## regal

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_A few weeks ago I got the chance to listen to a comparison of SACD and redbook from my EMU-0404 on VMPS RM40 speakers.

 ._

 


 You are hearing mastering differences.

 Ever compare 16/44.1 to DSD cut from the same master?


----------



## maarek99

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_You are hearing mastering differences.

 Ever compare 16/44.1 to DSD cut from the same master?_

 

Exactly. Most of the differences between dts and dd tracks come from completely different masters.


----------



## nick_charles

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_I'm surprised this discussion raged on as long as it did after I gave up.

 A few weeks ago I got the chance to listen to a comparison of SACD and redbook from my EMU-0404 on VMPS RM40 speakers.

 Bottom line. You are near as can’t hear if you can not discern the difference between high resolution audio and redbook._

 

Interesting, your results simply do not agree with a large scale carefully controlled study in a peer reviewed journal. 

 Can I ask you for some more details ?

 How did you downsample the SACD to 16/44.1 for comparison ?


  Quote:


 I also did a more careful analysis of the frequency spectrum of the high resolution audio tracks using Matlab. The results were the same. The frequency content is obviously distorted in the 16/44 audio. It is unable to resolve details in the audible spectrum that 24/96 audio can. 
 

Can you describe how you did this ? - what procedure did you use to get the two samples into Matlab - what were the measured distortion figures you found ? Matlab is just a computational environment and language in't it ? what audio analysing functions does it support ?

 More to the point can you posts the results including waveforms and spectral analyses - what level of zoom did you have to use to see visible differences in the wave forms ?. When I compared 24/96 and 16/44.1 versions of the same track they were indistinguishable until you zoomed to sample level exactly as you would expect, zooming more made both into straight lines.

  Quote:


 The science confirms what my ears hear. 
 

Please can you provide the data to support your assertions , so far you have just spoken in broad terms, thanks.

  Quote:


 I don’t know how to make this anymore clear. As far as I’ve seen, I’m the only who has posted solid data which backs-up my assertions. 
 

I do not see any real data in your current post , if you are going to argue you are using data to support your case you need to provide the data.

 I assume you are not referring to the stuff you posted previously.
 Thanks.

 Now if it is data you are interested in. In 2007 Meyer and Moran of the BAS ran a large number of blind listening tests using a variety of systems and material and listeners. Their paper is avalable from the AES at $5 (for members, student membership is $30) it is very interesting reading. 

 Key findings, not one subject (n = 60) was capable of detecting the difference between high res and 16/44.1 at a sufficient level of confidence (95%), the total # of trials was 554 and correct answers 276 (49.819%). Subjects did up to 10 trials. One subject scored 8/10, two scored 7/10, no others scored even 7/10, even these few high performers are results what you would expect by chance with a large enough sample. Engineers and "Audiophiles" did slightly better at 52.7%. Females did worse at 38%, those with better high frequency hearing did worse(45.3%) as did younger listeners. There was no effect due to different systems.

 Now here is the kicker, Meyer and Moran did not even use noise-shaping on the A/D/A stage - this is handicapping the A/D/A stage big time, giving the High-res the best possible hand (metaphorically speaking).


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_I don’t know how to make this anymore clear. As far as I’ve seen, I’m the only who has posted solid data which backs-up my assertions._

 

If you are interested, you can search the archive for my posts about a comparison between SACD and redbook that I did. When I finally located an SACD that was both DSD and had a redbook layer with no mastering differences, I couldn't hear a difference between layers.

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_If you are interested, you can search the archive for my posts about a comparison between SACD and redbook that I did. When I finally located an SACD that was both DSD and had a redbook layer with no mastering differences, I couldn't hear a difference between layers.

 See ya
 Steve_

 

This was of course a carefully proctored blind test, yes ?


----------



## b0dhi

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I learned early on to try to equate specs to what they sound like in the real world. I discovered that the fraction of a percentage of THD that I was sweating didn't mean a damn._

 

So did I. I discovered that IMD levels in even the best dynamic, electrostatic, or piezo drivers are entirely unacceptable at medium-high volume levels, and that numbers given for THD and IMD (both of which typically tend to be about the same level in any given driver) are always measured in "best case scenarios" to give low values. Worst case scenarios (which happen to be the most _common_ scenario for the type of music I listen to - during a bass drum kick while there's activity around 2-3khz) are much, much worse, and are easily audible, particularly for IMD.


----------



## frankR

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_You are hearing mastering differences._

 

No I'm not.

 It's a hybrid SACD. Analog recording. Same remastered audio. Separate layers have Redbook and DSD.

Amazon.com: Gerry Mulligan Meets Scott Hamilton: Soft Lights & Sweet Music: Gerry Mulligan,Scott Hamilton: Music

  Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_Ever compare 16/44.1 to DSD cut from the same master?_

 

Yes. The difference is night and day. It's not at all subtle.


----------



## nick_charles

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_No I'm not.

 It's a hybrid SACD. Analog recording. Same remastered audio. Separate layers have Redbook and DSD.
_

 

 Quote:


 Yes. The difference is night and day. It's not at all subtle. 
 

So, you ran some blind tests to prove this ?

 I am puzzled, your PC CD/DVD drive can read SACD ? - I didnt think that was possible ?, also I did not think the 0404 supported DSD anyway 
	

	
	
		
		

		
		
	


	




  Quote:


 A few weeks ago I got the chance to listen to a comparison of SACD and redbook from my EMU-0404 on VMPS RM40 speakers.


----------



## frankR

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Interesting, your results simply do not agree with a large scale carefully controlled study in a peer reviewed journal. 

 Can I ask you for some more details ?_

 

I’ll refer back to my earlier analysis posted last month.
http://www.head-fi.org/forums/f46/24...ml#post4017905

  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Can you describe how you did this ? - what procedure did you use to get the two samples into Matlab - what were the measured distortion figures you found ? Matlab is just a computational environment and language in't it ? what audio analyzing functions does it support ?_

 

Matlab is high-level scientific mathematics analysis package. It can import PCM audio. I analyzed short samples of various lengths (millisecond to several seconds) and of different instruments from the high-res-low-res comparison audio clips. I performed power frequency spectrum analysis of these samples using fourier transform. The fourier transform gives a time-independent analysis of the audio frequency spectrum.

 What I found was consistent with my earlier crudely performed analysis. There were significant differences between their respective power frequency spectra in the audible frequency range (20 hz to 20 khz).


  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_
 Key findings, not one subject (n = 60) was capable of detecting the difference between high res and 16/44.1 at a sufficient level of confidence (95%), the total # of trials was 554 and correct answers 276 (49.819%)._

 

I’ve read similar studies on the subject. There is a belief that these types of double blind audio tests are inherently flawed because, unlike images, the human brain has difficulty retaining fidelity differences, though differences are perhaps perceived.

 Nevertheless, these findings do not nullify what my own ears and analysis makes me believe.

 An interesting experiment at the upcoming CES in Las Vegas will compare live play-back to recorded material using suburb equipment.
Ampzilla, VMPS and Live Music at January CES


----------



## frankR

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_So, you ran some blind tests to prove this ?_

 

Yes.

 The test was performed using the Gerry Mulligan SACD, as well as others.

 There were three playback devices used: Marantz SA8260 for DSD track and Redbook from its own PCM DAC and the EMU-0404 using the optical S/PDIF output, playing two seperate encoded audio tracks from the same master (DSD & Redbook). note: the SA8260 is not the state-of-the-art for DSD playback.

 It wasn't even fair. I consistently spotted the DSD material within seconds of playback on multiple song, with similar results on other recordings.

  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_I am puzzled, your PC CD/DVD drive can read SACD ? - I didnt think that was possible ?, also I did not think the 0404 supported DSD anyway 
	

	
	
		
		

		
			



_

 


 I appologize for being confusing. The numerical analysis is completely seperate. Although I can hear differences in that matieral using my own head-phone equipment. However, it's not as dramatic as the SACD/Redbook on RM40 loudspeakers. I believe headphones may not be the ideal choice for making a comparision.


----------



## frankR

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_If you are interested, you can search the archive for my posts about a comparison between SACD and redbook that I did. When I finally located an SACD that was both DSD and had a redbook layer with no mastering differences, I couldn't hear a difference between layers.

 See ya
 Steve_

 

That maybe truthful in your case. But there is no question I can hear the difference in my own tests.

 Would you mind telling us your playback equipment used?

 What I don't get is, the implication those who believe high-res audio is indistinguishable from Redbook is: you must also believe the millions of audiophiles out there who spend millions of dollars per year (I'm just making up numbers here), are wasting their money. Throw in those who say vinyl/analog recordings sound between then digital as well.


----------



## nick_charles

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Yes.

 The test was performed using the Gerry Mulligan SACD, as well as others.

 There were three playback devices used: Marantz SA8260 for DSD track and Redbook from its own PCM DAC and the EMU-0404 using the optical S/PDIF output, playing two seperate encoded audio tracks from the same master (DSD & Redbook). note: the SA8260 is not the state-of-the-art for DSD playback.
_

 

Let me make sue I have this right.

 You did level matched blind tests, i.e tests where somebody else controlled the playback order . Using the SACD and redbook layers playing back through the Marantz ?

 You played back some samples from a PC via the EMU , this still doesnt make sense as the EMU cannot play back DSD


----------



## bigshot

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_This was of course a carefully proctored blind test, yes ? 
	

	
	
		
		

		
		
	


	


_

 

Naw. I only test for my own purposes. I do comparison tests and experiment with just about every piece of equipment I use.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_It's a hybrid SACD. Analog recording. Same remastered audio. Separate layers have Redbook and DSD._

 

The redbook layer on many analogue reissue SACDs I compared was different than the SACD layer. Try comparing a native DSD recording on an audiophile SACD hybrid label like Pentatone. You'll be surprised at the sound of the redbook layer when it isn't deliberately hobbled to make the SACD layer sound better. I used Jaarvi's Stravinsky chamber works. It's one of the most natural and dynamic recordings I've ever heard, and it sounds EXACTLY the same on redbook as it does on SACD.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Would you mind telling us your playback equipment used?_

 

We used a Philips 963sa SACD player, Yamaha and Sony CD players and two different rigs... my own and an experimental system being designed by an engineer friend of mine. It's all in the archive.

 See ya
 Steve


----------



## frankR

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_The redbook layer on many reissue SACDs I compared was different than the SACD layer. Try comparing a native DSD recording on an audiophile SACD hybrid label like Pentatone. You'll be surprised at the sound of the redbook layer when it isn't deliberately hobbled to make the SACD layer sound better.

 See ya
 Steve_

 

You're probably right.

 However, there is difference in the SACD that I hear that outright distinguishes itself.

 The dynamics have a lot more punch. The instruments sound more lifelike. I can throw out adjuctives all day, and I may not convince you. What I hear from the SACD I've never heard, by a mile on redbook, on any playback system, and I've heard some good ones. It sounds as good as the best vinyl, or better, more dynamic range, no pops, ect.


----------



## frankR

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_We used a Philips 963sa SACD player, Yamaha and Sony CD players and two different rigs... my own and an experimental system being designed by an engineer friend of mine. It's all in the archive.

 See ya
 Steve_

 

Speakers?


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_The dynamics have a lot more punch. The instruments sound more lifelike. I can throw out adjuctives all day, and I may not convince you. What I hear from the SACD I've never heard, by a mile on redbook, on any playback system, and I've heard some good ones. It sounds as good as the best vinyl._

 

I don't doubt at all that you heard those differences. But they're are all due to mastering. Compare a native DSD on a label that only releases SACD hybrids like Pentatone. That will give you a perfect comparison of high resolution digital and redbook with no monkeying around with the mastering.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_Speakers?_

 

Custom made speakers that had just been calibrated to provide perfectly flat response from 25Hz to 20kHz.

 See ya
 Steve


----------



## b0dhi

I'm not sure whether the reason for it is filters or what, and actually it doesn't really matter, because each of these sounds _clearly_ progressively better than the last: -

http://members.iinet.net.au/~hararghost/1377_441.wav

http://members.iinet.net.au/~hararghost/1377_48.wav

http://members.iinet.net.au/~hararghost/1377_96.wav

http://members.iinet.net.au/~hararghost/1377_192.wav


----------



## frankR

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_I'm not sure whether the reason for it is filters or what, and actually it doesn't really matter, because each of these sounds clearly progressively better than the last: -

http://members.iinet.net.au/~hararghost/1377_441.wav

http://members.iinet.net.au/~hararghost/1377_48.wav

http://members.iinet.net.au/~hararghost/1377_96.wav

http://members.iinet.net.au/~hararghost/1377_192.wav_

 


 What were those nasty clips that almost caused me to lose me hearing.

 You have me interested.


----------



## b0dhi

^ Sorry I forgot to indicate that in my post. They're 1377Hz (chosen randomly) saw waves at 44.1, 48, 96 and 192Khz respectively. There's an easily audible difference between them, even in my modest setup (Juli@ > PK1). The quality progressively improves going from 44.1 to 192. Actually, I was expecting there to be no audible change from 96Khz upwards, but I guess not. I can only imagine what 1000Khz or something would sound like.


----------



## gregorio

There seems to be a little confusion with regards to the format vs. the equipment. Some of you hear differences between CD and SACD and then conclude that SACD is better than CD. In theory, both formats exceed what your ears are capable of hearing. So what are the differences you're hearing? You are hearing the implementation of the DAC process being handled differently as well as the different processing which occurs during mastering. Maybe your DAC has particularly good filters at 192kFs/s maybe the DSD reconstruction is better implemented than the standard PCM reconstruction. Maybe the decimation from DSD to PCM is not well implemented. There are a whole host of potential differences in the implementations between different DACs and even between different processes within the same DAC.

 My advice: Don't jump to conclusions about what is causing strengths and weaknesses between formats, generally it's not an intrinsic problem but purely a manufacturing one. Specifications on DACs (and ADCs) are generally next to useless, use your ears. Remember, we all hear slightly differently anyway, so what might sound good to one person might not sound quite so good to another.


----------



## b0dhi

Whether they're problems intrinsic to the format or manufacturing problems associated with the format is a moot point. What matters is the end result to us music listeners.

 The best 44.1khz PCM reproduction will never be as good as the best 192khz PCM or DSD reproduction, simply because there are theoretical limits to how good a brick-wall filter used in 44.1 can be, whereas the theoretical limits on how good a 192khz filter can be are much higher in terms of fidelity. It's that simple.


----------



## gregorio

Unfortunately, it isn't that simple b0hdi. 192kFs/s can cause significant problems relative to 44.1. Sure a nice smooth filter can be applied at 192 but the filter is not the entire story. There's been a fair amount of research to suggest that around 60kFs/s is the optimal sample freq and the further away from this we go the more artefacts we are likely to encounter.

 What I'm trying to say is that the end result will vary according to your DAC. One DAC might sound better at 192 than at 96, with another DAC it may well be the other way around. But to say that 192 will always sound better than 96 is not true. If we are looking for a general rule, then 96k should sound better than 192. 192 produces alias images closer to the hearing spectrum and is likely to cause more errors due to the very high data rate. However, what is true in theory may or may not be true in practice, depending on the implementation of the various filters and the decimation, reconstruction and other processes going on inside a particular DAC.


----------



## b0dhi

It's pretty clear by listening to those example waveforms I provided earlier that, in reality, 192khz has far less artifacts than 44.1.

 But please, enlighten me. Provide evidence of this research that shows that 192khz has theoretically higher artifacts than 44.1khz.


----------



## nick_charles

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_It's pretty clear by listening to those example waveforms I provided earlier that, in reality, 192khz has far less artifacts than 44.1.

 But please, enlighten me. Provide evidence of this research that shows that 192khz has theoretically higher artifacts than 44.1khz._

 

Dan Lavry for one is skeptical about the value for 192K sampling which he maintains has lower accuracy...

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

 I cannot say whether he has a strong case or not but he chooses not to make a 192k DAC


----------



## b0dhi

On Page 25 he has an approximation of an analog 1KHz square wave which he says is sampled at ~700Khz. Then he makes a miraculous leap by saying that "The wave is not "perfectly square" because it is bandwidth limited to 20KHz.". I nearly choked. He's trying to compare high sampling bandwidth with low sampling bandwidth by providing a conveniently low bandwidth limited input waveform. It's laughable.

 But that's beside the point - accuracy in the context that he's talking about isn't really an issue these days. Even if accuracy is lower at 192khz compared to slower rates, it's still high enough that it's beyond hearing accuracy of the ear. Far more significant distortions come from the brick wall filters used in 44.1khz, and those distortions are unavoidable, no matter how good the filters are.

 For the sake of argument, let's say he's right. Even if, as Lavry suggests, there exist significant limitations in accuracy in 192Khz sampling as a result of hardware limitations, that's something that will no doubt change very soon as technology improves. Soon enough (if it hasn't already), the technology will be able to surpass the accuracy of the ear, and there will be no need for brick wall filters, and 44.1khz sampling will be inferior in every way.

 So, according to him:
  Quote:


 So we have the pros and cons to increased sampling: 

 Pro: Easier filter 
 Overcome Sinc problem 

 Con: Reduced accuracy 
 Significant increase in data files size 
 Significant increase in processing power required 
 

How nice of him to include data size and processing power, even though it has nothing to do with the sound quality. 

 Anyway, take away the "reduced accuracy" in the Cons, as will inevitably happen with improvements in technology (if it hasn't already), and all you have is better sound.

 All this waxing lyrical about theory is fun, but I ultimately believe what my ears tell me, and there's a _clear_ and proportional improvement going from 44.1, 48, 96, to 192khz sampling rates. And that's the bottom line.

 Thanks for the link, though. I'll be sure to avoid purchasing his DAC in the future.


----------



## nick_charles

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_The best 44.1khz PCM reproduction will never be as good as the best 192khz PCM or DSD reproduction, simply because there are *theoretical limits* to how good a brick-wall filter used in 44.1 can be, whereas the theoretical limits on how good a 192khz filter can be are much higher in terms of fidelity. It's that simple._

 

Yet, empirical tests have seemed to show decent evidence that even simple 44.1 A/D/A loops can be made to be wholly transparent and 16/44.1 downsampling/transcoding from DSD and 24/96 PCM can be transparent.


  Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_All this waxing lyrical about theory is fun, but I ultimately believe what my ears tell me, and there's a clear and proportional improvement going from 44.1, 48, 96, to 192khz sampling rates. And that's the bottom line._

 

So you are happy to use theory when it supports your arguments but discount it when it does not, have you ever considered a career in politics


----------



## bigshot

Processing power definitely impacts the sound. When a computer has to chug through a file in realtime, it increases the possibility of error. The truth of the matter, though, is that all of this is theoretical. You can make files the size of Mount Everest and computers capable of processing a gazillion samples, and it won't buy you any more sound quality. You just can't hear the difference. There are MUCH better ways to achieve better sound than constantly upping the numbers.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_When a computer has to chug through a file in realtime, it increases the possibility of error._

 

This statement shows extreme ignorance about how digital computers work. The computer is a deterministic, not stochastic, machine. Very rarely errors might happen, such as when a cosmic ray flips a bit in RAM, but it has nothing to do with processing power (I'm not counting programming bugs as they still result in a deterministic program). If the filter program cannot cope with the stream in real time, you'll get a buffer underrun and the stream will interrupt with pauses. One could construct a filter program that changes its filtering, say by going to less taps if it's a FIR filter, as load increases to the point where it might become non-realtime, but I have never seen anyone go to that trouble. The same goes with digital transmission lines, where the hardware can get errors more often, but it uses error correction coding to deal with them (the one glaring omission is the USB Audio specification, which does not use error correction and system configuration and load could potentially affect the error rate).


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_This statement shows extreme ignorance about how digital computers work._

 

Yeah... yeah... yeah... Try stacking up too many RT plugins in ProTools on a nice high bit/sample rate file and see what happens. You get loads of stutters, some delightful latency and an occasional rude surprise when the comp crashes on you.

 When it comes to just listening to music, there is absolutely no benefit from high bitrates and sampling rates. It can have a benefit in mixing and mastering, but that has to be balanced against how much real time processing you need to be doing, and what the added range of nice big files is going to buy you in mixing flexibility.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Try stacking up too many RT plugins in ProTools on a nice high bit/sample rate file and see what happens. You get loads of stutters, some delightful latency_

 

These are not errors--the bits of the music are the same. It's simply a slowdown. The two things couldn't be more different. An error is something which produces incorrect output, not just delaying the output and leaving real-time.


----------



## Publius

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_I'm not sure whether the reason for it is filters or what, and actually it doesn't really matter, because each of these sounds clearly progressively better than the last: -

http://members.iinet.net.au/~hararghost/1377_441.wav

http://members.iinet.net.au/~hararghost/1377_48.wav

http://members.iinet.net.au/~hararghost/1377_96.wav

http://members.iinet.net.au/~hararghost/1377_192.wav_

 

ZOMG. That is quite possibly the poorest sampling rate comparison I've ever seen in my life. 

 Why do I say this? Well, the 1413hz peak that is *60db* higher at 192khz is pretty damn convincing. Or the 1449hz peak that's 55db higher, or the bajillion other peaks that are different by <40db between the two clips. 

 You'd have to be deaf not to tell a difference, because the signals are only superficially similar - the resampling involved has to be comically bad to generate wavs like that.

 So congrats! You're not deaf. However, you certainly can't hear the difference between 192khz and 44khz in a valid comparison.


----------



## Publius

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_These are not errors--the bits of the music are the same. It's simply a slowdown. The two things couldn't be more different. An error is something which produces incorrect output, not just delaying the output and leaving real-time._

 

I think what bigshot is trying to say is that despite what you and b0dhi believe, high res support still costs a lot of money. Good playback devices for high res cost incrementally more than straight 16/44 DACs, but upgrading a studio to 24/96 is, to the best of my knowledge, still not cheap. And by spending additional money on high res support for recording and playback, you will wind up with an inferior sound quality, compared to if you had spent that money for something more productive.


----------



## b0dhi

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_ZOMG. That is quite possibly the poorest sampling rate comparison I've ever seen in my life. 

 Why do I say this? Well, the 1413hz peak that is *60db* higher at 192khz is pretty damn convincing. Or the 1449hz peak that's 55db higher, or the bajillion other peaks that are different by <40db between the two clips. 

 You'd have to be deaf not to tell a difference, because the signals are only superficially similar - the resampling involved has to be comically bad to generate wavs like that.

 So congrats! You're not deaf. However, you certainly can't hear the difference between 192khz and 44khz in a valid comparison._

 

There is no resampling involved at all. They were generated natively at the indicated sampling rate.


----------



## scompton

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_These are not errors--the bits of the music are the same. It's simply a slowdown. The two things couldn't be more different. An error is something which produces incorrect output, not just delaying the output and leaving real-time._

 

I've worked in real time computing and, in the project I've worked on, it was considered an error if the computer couldn't keep up with the data. We spent a lot of time changing code to keep up with the data. 

 You're parsing the word error like a politician. To the world as a whole stuttering music coming out of a computer is an error.


----------



## regal

and the error (pause) can be in the millisecond range. It doesn't always sound like a click.


----------



## nick_charles

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_There is no resampling involved at all. They were generated natively at the indicated sampling rate._

 

I do not know what this was meant to prove. 

 However I did a similar experiment but I used 44.1 and 88.2 and ran the FFT analysis at different rates for the different samples to make them comparable both had measurements at the same frequencies up to 21360. 

 Then I looked at the differences across the 20 - 20K range. The 88.2 sample was 3.6db louder across the audible spectrum, the biggest difference was 8db.

 Ffor the 17 peaks that made up the vast majority of energy
 (average 1.58db (88.1) - 1.22db (44.1) ) the 88.2 was never less than 2.1db louder, the average difference was 2.79db louder. 

 Conclusion these waveforms are just different. So generating sawtooth waves at different sampling rates gives rise to different waveforms.


----------



## Nocturnal310

On my laptop i changed from 16 bit to 24 bit.

 Didnt notice any Perceivable or even apparent Difference.

 its just psychological... u force ur ears to perceive it so they ll perceive something virtual.

 It depends on the Player u are using and not even the headphone.

 IF u connect your Headphones to an Amp maybe then and if u use a non-compressed Format 

 and after that also u put the volume at high level.

 and finally u may perceive some difference but what difference does that difference make?


 Look, i ve wasted so many hours and ideas to just feel the difference but couldnt perceive anything.

 i am into freelance DJing and i know when the Music sounds different.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_These are not errors--the bits of the music are the same. It's simply a slowdown._

 

My boy, you could have a great career as a poltician... Have a cigar!

 See ya
 Steve


----------



## Publius

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_There is no resampling involved at all. They were generated natively at the indicated sampling rate._

 

Well, there's your problem! Don't do that. It's wrong.

 Triangle waves intrinsically have infinite bandwidth. Naively generating one at the output sample rate will create aliasing for this reason - and varying amounts of aliasing at different sample rates.

 Alias-free triangle wave generation is actually a surprisingly hard problem. If you don't use some pre-rolled algorithm (and apparently some get it wrong) then you may need at least 32x oversampling over your output sample rate to get the aliasing artifacts reasonably small.


----------



## b0dhi

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_I do not know what this was meant to prove. 

 However I did a similar experiment but I used 44.1 and 88.2 and ran the FFT analysis at different rates for the different samples to make them comparable both had measurements at the same frequencies up to 21360. 

 Then I looked at the differences across the 20 - 20K range. The 88.2 sample was 3.6db louder across the audible spectrum, the biggest difference was 8db.

 Ffor the 17 peaks that made up the vast majority of energy
 (average 1.58db (88.1) - 1.22db (44.1) ) the 88.2 was never less than 2.1db louder, the average difference was 2.79db louder. 

 Conclusion these waveforms are just different. So generating sawtooth waves at different sampling rates gives rise to different waveforms._

 

Ofcourse they're different. The higher sampling rates allow encoding of the waveform to a higher precision. But more importantly, the ear is not an FFT. It doesn't hear things only in terms of frequency co-efficients. It detects subtle differences in phase, subtle changes in the shape of the leading edge, and other nuances that aren't represented by the broad and simplistic statement "the ear can hear between 20-20khz". Hearing is far more complicated than just that. Also, there are other differences in the output of your DAC that will not be represented by a digital domain analysis of the wav files. The filter, for example.


----------



## Crowbar

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_You're parsing the word error like a politician._

 

I'm parsing the error like a computer scientist, which is what I am.


----------



## b0dhi

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Well, there's your problem! Don't do that. It's wrong.

 Triangle waves intrinsically have infinite bandwidth. Naively generating one at the output sample rate will create aliasing for this reason - and varying amounts of aliasing at different sample rates.

 Alias-free triangle wave generation is actually a surprisingly hard problem. If you don't use some pre-rolled algorithm (and apparently some get it wrong) then you may need at least 32x oversampling over your output sample rate to get the aliasing artifacts reasonably small._

 

The reason I chose a ramp (a square would've worked too, it's just harder to hear the difference) was because of the infinite bandwidth. Listening to the samples, it becomes obvious that the higher the bandwidth, the less aliasing is produced - or rather - aliasing is moved further _away_ from the hearing range of the ear. Thus, using a higher sampling rate would reduce aliasing in the audible spectrum, enabling use of less distorting filters both at the recording and reconstruction stages. The point is to avoid using those brick wall filters, because they introduce their own distortions which are almost as bad as aliasing.

 There's a legion of people that stand by non-oversampling DACs mainly because there's no need to use a brick wall filter. Even though they measure abysmally, many people still prefer them because the design focuses on what the ear is sensitive to, not what measures well.


----------



## Crowbar

So did you lowpass the files appropriately for each rate before converting? Even if you didn't use a brick-wall filter, there should still have been a filter used that puts anything above fs/2 to a level corresponding to < 1 LSB.
 It is correct that a brickwall (in Fourier domain) digital filter is not optimal, because the ear doesn't only do frequency domain analysis. But you still need to lowpass it otherwise your test is invalid.


----------



## Publius

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_The reason I chose a ramp (a square would've worked too, it's just harder to hear the difference) was because of the infinite bandwidth. Listening to the samples, it becomes obvious that the higher the bandwidth, the less aliasing is produced - or rather - aliasing is moved further away from the hearing range of the ear. Thus, using a higher sampling rate would reduce aliasing in the audible spectrum, enabling use of less distorting filters both at the recording and reconstruction stages. The point is to avoid using those brick wall filters, because they introduce their own distortions which are almost as bad as aliasing._

 

But that aliasing is entirely the fault of the algorithm you chose. The fact that the aliasing gets better with increased sample rate doesn't mean that increasing the sample rate always yields higher quality - it merely means that your synthesis algorithm is sensitive to sample rate.

 It's important to note that your 192khz sawtooth wave (sorry, I'm getting triangle and sawtooth mixed up) still has *lots* of aliasing. It's blindingly obvious on a sufficiently resolving amplitude spectrum plot. One could argue that 192khz is still not enough to enable alias-free generation of a triangle wave. Or even that, for a signal like a pulse train whose spectral power does not degrade with higher frequency, _no_ sampling rate is accurate enough with the synthesis style you are using.

 Again, that proves nothing beyond the fact that your synthesis algorithm sucks.

 A much better algorithm would be to calculate the signal components in the frequency domain and compute an inverse FFT. Frequency synthesis is obviously alias-free, and it does not suffer from any frequency resolution or bit depth issues. It's considerably harder, but tough luck - you chose a hard problem.


----------



## gregorio

"The higher sampling rates allow encoding of the waveform to a higher precision."

 No they don't!!! Oh b0dhi, I thought we'd gone through this once already. I'm sorry you still don't get it but there has been plenty of proof provided in this thread. You need just two samples per waveform, you're not going to get any percievable benefit than if you used 5 million samples per waveform. If you still disagree with this statement then take it up with Nyquist or Shannon, it's their theory!

 Adding more bits or more samples does not increase resolution, it just extends the ranges of dynamics or frequencies which can be captured. The good 'ol redbook CD specification already allows for the "resolution" of digital audio to exceed that of the human ear in both the dynamic and the frequency ranges. It was a deliberate attempt to future-proof the format from the outset! It would have worked but for the manufacturer's marketting strategies and the ignorance they engender.

 If you are able to create audio files which are identical in every respect except for the bit depth and sample frequency, and can still tell a difference, then you seriously need a new DAC or you need to check your family tree for any dogs, bats or dolphins!

 If you do need a new DAC then you really stuck your foot in your mouth with your comments about Lavry. Lavry are one of the world leaders in pro-audio ADCs and DACs. If you've never heard a high end Lavry or even a Prism (in a top class environment), then you have no idea of the sound quality achievable outputting at 44.1/16bit. It's difficult to explain if you've never heard it, it's truely stunning. What is possible in Air Studios and Abbey Road today exceeds anything I've ever heard, even in the top analogue studios like the old Hit Factory.

 What needs to catch up isn't the format but the consumer listening environment. There are experts who can perceive differences in almost every single component which comprises a professional or a home listening enviroment but we are arguing about the theoretical limitations of 44/16, which lay on or well outside of what the perfect human ear in the perfect listening environment could perceive!


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_the theoretical limitations of 44/16, which lay on or well outside of what the perfect human ear in the perfect listening environment could perceive_

 

Uh, you're changing your story now. Before it was that one couldn't achieve the perfect environment and that's why 44/16 was enough. Again, the ear is 120 dB to threshold of pain and 16 bit is not 120 dB, so no, in a perfect or near perfect environment 16 bit is not even close.


----------



## gregorio

Crowbar - Read this thread more carefully. With modern noise-shaped dither a CD is capable of a dynamic range approaching 120dB and of course this is 120dB above the noise floor of the enviroment, which is likely to be roughly 40-50dB. So we are orders of magnitude beyond the point at which the ear can comfortably listen in a realistic environment. Even if a perfect evironment with no noise floor were possible (which of course it isn't), we are still many times beyond the dynamic range our ears are comfortable with. No one is ever going to put out a product with a dymanic range anywhere near 100dB let alone 120dB.

 The practicalities with the sample frequency is not quite so obvious as the case with bit depth. This is due to the implementation of the filters and other processes in different DACs. Using very high quality DACs in a top quality listening environment is usually enough to fool most experienced audio professionals. So for the average listener and almost certainly even for the wealthy audiophile, there is no point to higher than 44.1kFs/s.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Read this thread more carefully. With modern noise-shaped dither a CD is capable of a dynamic range approaching 120dB_

 

Not throughout the audio band. Nothing's for free. Noise shaping lowers noise in one part of the band and raises it in another. 44 kHz means that to really get 120 dB in the 0-20 kHz, you'd have to squeeze an amount of noise in the top 2 kHz that is impossible with any type of dithering. In real life noise shaping of 44.1/16 you only get significant increase of dynamic range in a small portion of the band.


----------



## b0dhi

^ Not only that, but pushing all that noise into the ultrasonic means that the speaker, especially if it's a single or two-way driver system, will exhibit a lot of IM distortion between the noise band and the lower frequencies. Although the ultrasonic wont be audible (to most), the lower frequency IM products it causes will be.


----------



## Crowbar

Not just the speaker. High frequency garbage affects the analog electronics.


----------



## bigshot

I would like to know who listens to music anywhere near the threshold of pain. I seriously doubt even Norwegian Death Metal fans come close to exhausting the dynamics of redbook.

 The problem here is you are thinking in terms of numbers on a page. You just don't know what those numbers you're citing sound like. It's all well and good to have an armchair theory like "all specs should meet or exceed the range of human hearing", but you need to temper that with the knowledge that we're reproducing music here, not supersonic jets taking off and industrial metal stampers punching out auto parts.

 I don't know if you realize it, but it's clear to a lot of us reading this thread that gregorio knows his stuff. He's talking about the technical aspects and the science behind digital audio as well as the practical application of the technology. You two seem to be arguing for argument's sake from a purely theoretical standpoint. You should be picking his brain for real world experience and learning from what he says, regardless if you agree with him or not. But I don't think you're even listening.

 See ya
 Steve


----------



## b0dhi

While I'm sure he has some hands-on experience in a studio, he has no technical knowledge beyond that. I know this because when asked for a simple reference to back up one of his more unrealistic claims, he not only couldn't provide one, and not only didn't understand what a reference _meant_ (while claiming to be a university lecturer - that was a good laugh), but in the end resorted to making up false quotes and immature ad hominems. He also doesn't understand that "asK Nyquist and sHannon!!1" isn't evidence. I personally would be more inclined to hear the experiences of someone I suspect is beyond their teenage years. He also does other strange things like making irrelevant quotes of Wikipedia, then when a valid counter-quote is made from the same source, claims moral high ground by the fact that the counter-quote was from _Wikipedia_ of all places, and not the authoritative sources he uses. I'm not complaining though, it's amusing to watch


----------



## bigshot

You don't really think Gregorio doesn't know what a "reference" is, do you? "Nyquist's Theory" is the answer to why you only need two points to graph a waveform. He gave you your answer.

 Gregorio has provided the lion's share of the solid information in this thread. if you're interested in engaging in a give and take dialogue, go back and review what he's said and come up with questions that take into account everything he's said, not just a single post at the end of the string of posts. He got impatient because you were asking him questions that he'd already answered.

 See ya
 Steve


----------



## Publius

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_You don't really think Gregorio doesn't know what a "reference" is, do you? "Nyquist's Theory" is the answer to why you only need two points to graph a waveform. He gave you your answer.

 Gregorio has provided the lion's share of the solid information in this thread. if you're interested in engaging in a give and take dialogue, go back and review what he's said and come up with questions that take into account everything he's said, not just a single post at the end of the string of posts. He got impatient because you were asking him questions that he'd already answered_

 

Quoted for truth.

 Between the two, b0dhi and Crowbar have pretty much spouted nothing but misinformation and insults for all of the thread that I've cared to read.


----------



## b0dhi

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_You don't really think Gregorio doesn't know what a "reference" is, do you?_

 

When I challenged him to provide a reference, he responded by stating that he _knew what the Harvard referencing system was_. hahahaha, i'm still laughing at it. Oh, he never actually provided a reference, either.

  Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_"Nyquist's Theory" is the answer to why you only need two points to graph a waveform. He gave you your answer._

 

If this were an adjudicated debate and you attempted to justify an assertion by just saying "Nyquist's Theory is why", you would be laughed at. Just for the record, this is the Nyquist theorem:

  Quote:


 If a function f(t) contains no frequencies higher than W cps, it is completely determined by giving its ordinates at a series of points spaced 1/(2W) seconds apart. 
 

Note that it doesn't state that 2 points are needed to represent any particular waveform, which is what gregorio was implying. What gregorio was probably saying in his "I'm a university lecturer" kind of way was that 2 points are all that are needed to represent the frequency and amplitude of a particular periodic function when it's waveform is _already known_.

 There are waveforms:






 You may notice, using your eyes, that they look different from one another. This is because they are "different" waveforms. Because each of these waveforms are periodic, you could represent their frequency and amplitude by only 2 points - if and only if you knew which waveform it was whose frequency and amplitude you were representing.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I would like to know who listens to music anywhere near the threshold of pain._

 

This was already answered multiple times. I'm not concerned with the average level of loud passages, but individual large transients.


----------



## Crowbar

b0dhi, note that the theorem you quoted refers to "frequencies". Frequencies are sines. All other waveforms are composed of sines, so your argument is incorrect.

 In your drawing, the period of these waveforms doesn't determine their frequency content. That is only the case for the sine wave. The other waveforms shown are still composed to sine waves: one of period as the first one, and a bunch of harmonics. For example, here's how a square wave is composed of sines (the animation shows increasing bandlimit, and the wave aproaches a perfect square wave as the gandlimit goes towards infinity):





 The waveforms you've drawn are perfectly sharp and thus infinite in bandwith. Such waves cannot exist in reality, and most certainly you cannot hear them. Your ear is a lowpass filter at around 20 kHz and removes all harmonics higher than that. Thus, if the sine you drew is at 20 kHz, all other waves would be filtered to leave the base harmonic, i.e. they would become identical after they go through the ear. If the 'square' wave was with base period corresponding to a 10 kHz sine, then after the ear lowpasses it it would be received as if it were the version with "harmonics: 2" above. And so on. The point is that the highest frequency component is what is limited, and thus the needed Nyquist rate is the same regardless of the actual shape of the waveform.

 Note that any sound, or indeed any analog signal, whether audio, image, whatever, can be represented by a combination of sine waves (in the case of images and other 2D signals they're planar waves, etc.). You may wonder about representing signals that are not periodic, but that doesn't matter because trivially you take the period to be the length of the signal and ignore the repetitions, just choosing one image.


----------



## regal

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Not just the speaker. High frequency garbage affects the analog electronics._

 

This is exactly my experience with these NOS filterless DAC's, all of them I have heard have obvious clipping on the output. They also all failed the udial clipping test.


 The Lavry Gregorio mentioned is a different animal (has both a DF and an analog one.) But really for that kind of money he


----------



## nick_charles

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_you could represent their frequency and amplitude by only 2 points - if and only if you knew which waveform it was whose frequency and amplitude you were representing._

 

You are creating an imaginary problem, resolving a square wave to be a perfect representation is impossible at 44, 88, 96, 192, 384, 768 or any arbitrary frequency you choose. A square wave is a fundamental plus an infinite set of harmonics at 3f 5f 7f and so on.

 But once you bandwidth limit the signal to frequencies you can actually hear the problem becomes moot as you are left with a finite set of sine waves at actually audible frequencies rendeerd accurately.

 So a sampling rate of 44.1khz cannot render a square wave **exactly**, big deal, you cannot hear 99.99999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 9999% of the harmonics anyway.


----------



## regal

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_
 So a sampling rate of 44.1khz cannot render a square wave **exactly**, big deal, you cannot hear 99.99999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 9999% of the harmonics anyway._

 


 But I think you guys are missing the electronics required to get your 44.1k to work. You need a filter and a well designed filter for a 1X OS DAC will always sound worse than a well designed filter for a 8X OS DAC. 

 Your argument would be correct if we didn't have to deal with things like inductors and capacitors, but we live in a mechanical world. Your theories have to include the mechanics.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_This was already answered multiple times. I'm not concerned with the average level of loud passages, but individual large transients._

 

And it's already been pointed out that there are no loud transients beyond 50dB or so in recorded music.

 See ya
 Steve


----------



## gregorio

Thanks BigShot, Publius and others. Bigshot - your observation was spot on the money. My understanding of digital theory must, by necessity, be tempered by the practical realities of recording equipment and the capabilities of the human ear.

 I'm not exactly sure what b0dhi's problem is. Maybe he just feels aggrieved because digital audio theory is sometimes counter intiutive, or because he just doesn't like me addressing his questionable understanding of the theory.

 b0hdi's different waveform types question has been well answered by others, except to state perhaps the obvious, which is that the Nyquist Theorum pre-supposes the sine wave type waveform. Which is all we need to worry about in the real world.

 Crowbar - "In real life noise shaping of 44.1/16 you only get significant increase of dynamic range in a small portion of the band."

 True. However, the portion of the freq band we're talking about is the area where our hearing is at it's most sensitive. That's the whole point of noise-shaping in the first place! Depending on the precise noise shaping alogrithm, the noise is usually re-distributed to above 12kHz and below 60Hz, leaving the 1kHz-4kHz range pretty much clear of any dither noise. Above 12kHz and below 60Hz were chosen because at these freqs our sensitivity to amplitude rapidly deteriorates. This extends the perceived noise floor by about 20dB or so in a 16bit system. Bare in mind also that we're not talking about large amounts of energy being redistributed, in total the white noise content (introduced by the dither process) is roughly equivalent to the least significant bit, which even at 16bit is going to be insignificant relative to the noise floor of the audio equipment used and of the recording and playback environments. Lastly, most noise shaping programs give the option of several different noise re-distribution types, this is to allow for the further processing of audio where the summing of the re-distributed noise may become problematic.

 Regal - What you said in your last post is true. Oversampling usually gives better results. Remember though that the digital audio you are playing back already contains the artefacts of a brickwall filter, from the ADC. In fact, pretty much all modern ADCs use oversampling and the vast majority of the filtering occurs in the digital domain. This is the most usable explanation of filtering that I've seen:

http://www.users.qwest.net/%7Evolt42...ng/Filters.pdf


----------



## Crowbar

regal, you're talking about oversampling. The 44.1 kHz being mentioned here is the starting sample rate.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_You are creating an imaginary problem, resolving a square wave to be a perfect representation is impossible at 44, 88, 96, 192, 384, 768 or any arbitrary frequency you choose. A square wave is a fundamental plus an infinite set of harmonics at 3f 5f 7f and so on.

 But once you bandwidth limit the signal to frequencies you can actually hear the problem becomes moot as you are left with a finite set of sine waves at actually audible frequencies rendeerd accurately.

 So a sampling rate of 44.1khz cannot render a square wave **exactly**, big deal, you cannot hear 99.99999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 99999999999999999999999999999999999999999999999999 9999% of the harmonics anyway._

 

Way to repeat what I already posted


----------



## grawk

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Way to repeat what I already posted 
	

	
	
		
		

		
		
	


	


_

 

Way to ignore the point of what he was saying


----------



## Crowbar

There is no point contained therein that was not already made in my post.


----------



## grawk

Yes, there was.


----------



## b0dhi

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_b0dhi, note that the theorem you quoted refers to "frequencies". Frequencies are sines. All other waveforms are composed of sines, so your argument is incorrect._

 

You misunderstand my argument, which was that gregorio didn't refer to _sine waves_ or _frequencies_, he referred to waveforms. I take it I won't have to explain that although a sine is a waveform, a waveform is not a sine. He was implying (whether intentionally or through lazy prose) that any waveform could be represented as well by 2 points/samples as by 5 million, which is patently untrue. The fact that any waveform can be constructed by a number (possibly infinite) of sine waves should not be confused with the concept of waveforms itself.

 (snip true but irrelevant digression)


----------



## Publius

What exactly is your point? So gregorio got his terminology wrong in places. Big deal. I think everybody can look past that.


----------



## b0dhi

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_What exactly is your point? So gregorio got his terminology wrong in places. Big deal. I think everybody can look past that._

 

But he's a university lecturer that knows the Harvard referencing system? Surely he understands the importance of precision in language, even if his lap-dogs don't? I wonder - which university do you lecture in, gregorio, and in which faculty?


----------



## grawk

How are academic credentials relevent?


----------



## gregorio

b0hdi - I used the term "waveforms" meaning any simple or complex sine wave. Sorry, I mistakenly took it for granted that you knew we were talking about sound waves which actually exist in the real world rather than sawtooths, square waves or pink elephants! I also mistakenly assumed that you knew that the whole theory behind digital audio is only true for sine waves.

 So what now? You're starting to sound desparate, to be picking up on such irrelevant details as my marginally vague use of the term waveforms. BTW, there's no way I'm telling you where I lecture, you sound far too much like an adolescent nutter!


----------



## regal

I think credentials are revelant. Gregorio's background is obviously the science of sound, others come from the EE side. The problem that we are dealing with is a combination of these two disciplines, both have to be considered in this argument. Scientists and engineers never agree.

 I appreciate Gregorio sharing his knowledge on the subject, both sides can learn from each other.


----------



## gregorio

Hi Regal - My background was as an orchestral musician and then as a composer, engineer and producer and more recently as a lecturer. I'm not a scientist and don't fully understand much of the math behind digital audio. I have a reasonably good understanding of digital audio theory (for a layman) because it so impacted on my work.


----------



## bigshot

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_Surely he understands the importance of precision in language, even if his lap-dogs don't?_

 

Ad hominem attacks and irrelevant nit picking isn't going to make your case. Behave like an adult.

 See ya
 Steve


----------



## b0dhi

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_b0hdi - I used the term "waveforms" meaning any simple or complex sine wave._

 

That isn't what the word means. Instead of making excuses, use the correct word. You've done this more than once.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_So what now? You're starting to sound desparate, to be picking up on such irrelevant details as my marginally vague use of the term waveforms. BTW, there's no way I'm telling you where I lecture, you sound far too much like an adolescent nutter!_

 

The reason I highlighted your incorrect use of terminology is because as soon as the discussion deviates beyond your area of expertise - which is limited to audio mastering - you stop having a clue what you're talking about and start mincing words and concepts. This has happened far more often than once.

 I don't have any problem with you bringing practical audio mastering experience to the discussion, but I do have a problem when you use the credibility earned with that knowledge in other areas where your knowledge is, as you admitted, at the level of a layman.

 And believe me, the only reason I asked for your credentials is to demonstrate that you wouldn't provide any. No respectable university would offer tenure to someone who can't spell, or someone who can't provide references in support of his assertions, or someone who doesn't understand what is _meant_ when a reference is asked for.


----------



## regal

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_. No respectable university would offer tenure to someone who can't spell, or someone who can't provide references in support of his assertions, or someone who doesn't understand what is meant when a reference is asked for._

 

Some of the brightest university research professors I have worked with couldn't spell. This has no relevance.

 Even though I disagree with Gregorio about NOS vs. OS in D-A conversion and A-D conversion, I think your issue is one of a language barrier.


----------



## hciman77

oops wrong forum


----------



## hciman77

oops wrong forum again


----------



## bigshot

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_The reason I highlighted your incorrect use of terminology is because as soon as the discussion deviates beyond your area of expertise - which is limited to audio mastering - you stop having a clue what you're talking about and start mincing words and concepts. This has happened far more often than once._

 

If you are going to build your argument on logical fallacies based on authority, you aren't going to get anywhere. Argue on point, don't attack your opponent. If you don't get back on track soon, we're all going to just write you off as a jerk and move on.

 See ya
 Steve


----------



## lexnasa

This debate is stupid... I suggest we all leave it to rust. 24 bit is inherently superior to 16 bit at the same sampling rate, but quality of playback depends on the playback medium and decoding hardware and software. A high end 16 bit system may outperform a low end 24 bit system, so what?


----------



## bigshot

I'd agree with you if you change the word "inherently" to "theoretically". Because in playback of music there is no audible difference between the two formats. High end or low end- all of the superiority of 24 bit is beyond the range of human hearing.

 See ya
 Steve


----------



## lexnasa

Sorry, rubbish. I write and produce music for a living... you're being duped by the scientific studies that pretend to understand human hearing.

 I can play you a native 24 bit 44.1khz recording vs a native 16 bit 44.1khz recording of the same sound and there is a difference. By native I mean that no dithering has been used.

 To address your human hearing point, what you may have been duped by is the empirical limit of human hearing, on which CDs were based. However, this empirical limit ignores the physicality of playback. It ignores harmonic interraction. So by removing what you can't hear, it also removes the interraction of what you can't hear with what you can hear.

 As you're ignoring sampling frequency, and only addressing bitrate, your argument is fundamentally flawed. If you imagine analogue as being a continous flow of data, and digital as a series of samples that are taken and reconstructed, which do you think would be better? And taking more detailed samples should be good? By your argument 4 bit should sound as good as 16 bit... it doesn't! 4 bit or 16 bit or 24 bit at the same sampling frequency are addressing exactly the same range of human hearing.

 I can't believe I'm defending myself against someone that thinks 16 bit audio is the endgame... wait, I have to pinch myself!


----------



## shinew

Quote:


  Originally Posted by *lexnasa* /img/forum/go_quote.gif 
_To address your human hearing point, what you may have been duped by is the empirical limit of human hearing, on which CDs were based. However, this empirical limit ignores the physicality of playback. It ignores harmonic interraction. So by removing what you can't hear, it also removes the interraction of what you can't hear with what you can hear._

 

but isn't it frequncy's job, not bit rate?


----------



## nick_charles

Quote:


  Originally Posted by *lexnasa* /img/forum/go_quote.gif 
_Sorry, rubbish. I write and produce music for a living... you're being duped by the scientific studies that pretend to understand human hearing.

 I can play you a native 24 bit 44.1khz recording vs a native 16 bit 44.1khz recording of the same sound and there is a difference. By native I mean that no dithering has been used.

 What you may have been duped by is the empirical limit of human hearing, on which CDs were based. However, this empirical limit ignores the physicality of playback. It ignores harmonic interraction. So by removing what you can't hear, it also removes the interraction of what you can't hear with what you can hear.

 I can't believe I'm defending myself against someone that thinks 16 bit audio is the endgame... wait, I have to pinch myself!_

 

It really isnt *just* a matter of theory. There has been a rather well-publicised study examining how well listeners can differentiate between high res and 16/44.1 music. With 60 Audiophile, Engineer and Sound Engineering students as subjects the level of differentiation was 50% i.e complete chance.

 If you would like to post some 16/44.1 and 24/44.1 samples taken under otherwise identical conditions i.e recorded simultaneously with identical microphone placement and identical mastering and identical average volume levels I would be happy to audition them under DBT conditions as I guess would several folks here.

 And of course you could support your case better by doing so yourself.

 There is a plethora of "I can hear a difference" stories out there but very few have been supported by unbiased i.e carefully controlled unsighted listening tests, so far the vast bulk of empirical evidence suggests that we just cannot hear the difference.

 Seriously, if you can provide us with some new comparable 16 and 24 bit samples for testing it would be interesting, please not the ones we have all seen from Linn and others, they just are not identical between formats.

 And in any case nobody is arguing against 24 bits for recording and mastering, the issue is whether we need 24 bits for playback.


----------



## CyberTheo

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Seriously, if you can provide us with some new comparable 16 and 24 bit samples for testing it would be interesting, please not the ones we have all seen from Linn and others, they just are not identical between formats._

 

I've been following this thread with interest. Let me throw a wrench. Why not Linn samples? I feel they are good base for comparison, unless you're proposing another theoretical study here that we've seen beaten to death already. All commercial CDs were recorded in much higher bit and sampling rates than 16/44; they were all down sampled to comply to CD (or whatever) format. Even if someone is successful in proving the theoretical aspect to human hearing audibility between 16/44 and higher rate samples, it's of no value to the end user - who are always getting down sampled digital format from higher rate originals.


----------



## gyrodec

lexnasa, you don't seem to know where the extra 8 bits of 24 bit go. They just tack on the bottom and define detail that is already below any meaingful noise floor (this is for delivery only, for recording the extra bits have obvious advantages in level setting on recording, but we are talking normalized at playback). Unless you think you can hear signal at -90db in your listening room, 16 and 24 bit will be identical.

 "As you're ignoring sampling frequency, and only addressing bitrate, your argument is fundamentally flawed. If you imagine analogue as being a continous flow of data, and digital as a series of samples that are taken and reconstructed, which do you think would be better? And taking more detailed samples should be good? By your argument 4 bit should sound as good as 16 bit... it doesn't! 4 bit or 16 bit or 24 bit at the same sampling frequency are addressing exactly the same range of human hearing."

 You don't take the range of human hearing, in volume terms, and split it into as many samples as you have. You start at pegging the meters and then each bit defines a sound half as loud. The extra 8 bits define the sounds as quite as nose-hair waving and the self-noise of the microphones. Sampling should make a difference from various secondry isses (anti-alasing filter slopes etc), but bit-depth does nothing but give you more signal-to-noise. And as 16 bit is 96db, and there is only about 70db in even the most dynamic recordings, 16 would seem to cover it.


----------



## nick_charles

Quote:


  Originally Posted by *CyberTheo* /img/forum/go_quote.gif 
_I've been following this thread with interest. Let me throw a wrench. Why not Linn samples? I feel they are good base for comparison, unless you're proposing another theoretical study here that we've seen beaten to death already. All commercial CDs were recorded in much higher bit and sampling rates than 16/44; they were all down sampled to comply to CD (or whatever) format. Even if someone is successful in proving the theoretical aspect to human hearing audibility between 16/44 and higher rate samples, it's of no value to the end user - who are always getting down sampled digital format from higher rate originals._

 

The problem with the Linn samples is that the high res and red book versions are just not quite the same, they are slightly but definitely different, even within the 20 - 20k range. I have run these through a spectrum analyser and the high res sample is just louder across the whole audible spectrum, about 3db louder on average - I did this at the highest res available where the level was sampled every 5hz. 

 This subtle difference is enough to render comparisons misleading.

 However you can take Linn's high res samples and then downsample them with suitable dither, that would be a valid comparison, if 24 bits downsampled to 16 is audibly different from 24 bits then that would be an interesting finding.


----------



## gyrodec

I'b in ineterested in the results of that test. It shouldn't be audible, but any piece of science is always one experiment away from being found wanting. (Then the real fun starts of trying to work out what really is going on.)


----------



## VeipaCray

The difference between 24 and 16 bit is well.. 8 bit. A third grader could have told you that. 
	

	
	
		
		

		
			





 What that equates to in real life...well I'll leave that to the more knowledgeable people who have previously posted on this thread.


----------



## nick_charles

Quote:


  Originally Posted by *gyrodec* /img/forum/go_quote.gif 
_I'b in ineterested in the results of that test. It shouldn't be audible, but any piece of science is always one experiment away from being found wanting. (Then the real fun starts of trying to work out what really is going on.)_

 

Well since I could not tell the difference between 88.2K and 44.1K samples anyway in a blind test I will leave it to younger ears to prove me wrong, or not as the case may be 
	

	
	
		
		

		
		
	


	




 I may try again when my new super-whizzo 24/96 DAC and headphone amp arrives from China.


 For now Linns free samples can be found here

Download our testfiles

 NB the 44.1 bit and 88.2 khz samples are not exactly aligned or the same length so some judicious trimming is required.


----------



## frankR

Quote:


  Originally Posted by *lexnasa* /img/forum/go_quote.gif 
_To address your human hearing point, what you may have been duped by is the empirical limit of human hearing, on which CDs were based. However, this empirical limit ignores the physicality of playback. It ignores harmonic interaction. So by removing what you can't hear, it also removes the interaction of what you can't hear with what you can hear._

 

My advice, you're wasting your time with these guys.

 Forty pages ago or so, I posted analysis of 24/96 and 16/44 audio by comparing power frequency spectra which demonstrates exactly what your saving. Namely, 16/44 Redbook audio does not render the audible frequency range as well as 24/96. They continually revert back to a sophomoric half-sampling theorem argument.

 Using the Nyquist sampling theorem to claim 16/44 is adequate is a weak argument to begin with. It merely describes the maximum resolved frequency. It is far too inadequate to describe how well sampled a signal is.

 I used an imaging analogy to show details throughout the frequency domain are better rendered with greater resolution. I don’t need 1080i HD to watch a baseball game, but it sure is more enjoyable.

 It’s also frustrating to hear them say Redbook is all you need when I know SACD sounds unmistakably superior to Redbook.


----------



## frankR

Quote:


  Originally Posted by *VeipaCray* /img/forum/go_quote.gif 
_The difference between 24 and 16 bit is well.. 8 bit._

 

It's actually 16,711,680.

 Ever heard of a logarithm?

 That's like saying the difference between a magnitude 6 earth quake and a magnitude 8 is 2.


----------



## gyrodec

frankr - this is the 16/24 bit thread, not the sample rate thread. Nyquist sampling theory has no place here, just db. 
	

	
	
		
		

		
		
	


	




 Technically, as VC said 8 bit and not 8, and as 8 bit is 16,711,680 (as you stated) he is completely correct. (May be missleading for non-technoical readers, but correct all the same.)


----------



## frankR

Quote:


  Originally Posted by *gyrodec* /img/forum/go_quote.gif 
_frankr - this is the 16/24 bit thread, not the sample rate thread. Nyquist sampling theory has no place here, just db. 
	

	
	
		
		

		
		
	


	


_

 

We're discussing whether there is a difference between high resolution audio and rebook.

 DSD is 1-bit audio.

  Quote:


  Originally Posted by *gyrodec* /img/forum/go_quote.gif 
_Technically, as VC said 8 bit and not 8, and as 8 bit is 16,711,680 (as you stated) he is completely correct. (May be missleading for non-technoical readers, but correct all the same.)_

 

No, 8-bit is 256.


----------



## VeipaCray

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_It's actually 16,711,680.

 Ever heard of a logarithm?_

 

Ever hear of a joke? 

  Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_No, 8-bit is 256._

 

A bit is a single binary digit, so actually the maximum value for 8 binary digits 11111111 = 255 NOT 256 as you stated. 256 would be 100000000 which takes 9 bits to represent in binary. 

 Stick whatever unit label you want after the number, the difference in base 10 between the two numbers is still 8. Call it 8 cucumbers it doesn't matter.


----------



## flibottf

Quote:


  Originally Posted by *VeipaCray* /img/forum/go_quote.gif 
_Ever hear of a joke? 

 A bit is a single binary digit, so actually the maximum value for 8 binary digits 11111111 = 255 and NOT 256 as you stated. 256 would be 100000000 which takes 9 bits to represent in binary. 

 Stick whatever unit label you want after the number, the difference in base 10 between the two numbers (24 - 16) is still 8. Call it 8 cucumbers it doesn't matter._

 

He meant 256 levels.... 0 counts as 8 bits goes from 00000000 to 11111111


----------



## gyrodec

frankR - quite right, 8-bit is 256 levels, thats what for posting in the middle of something else and not thinking clearly. So, now the real answer, yes 24 bit gives you 16,711,680 more encodeables values than 16 bit. True, but almost completely meaningless, because the aubible volume spectrum isn't neatly devided by those nice new steps. Everythime you add bit to a recording they just give you more resolution at the bottom of the volume scale, where there is almost nothing but noise (in an accoustic recording anyway). Bit depth defines dynamic range, it deosn't give you anything meaningul in terms of resolution at higher volumes. Yes there will be more steps there, but the reconstruction filter will turn a 16 or 24 bit , say, -20db, signal into exactly the same output wave anyway. (Again, this is a play-back only argument, for recording the extra bit are very useful for other reasons.)

 I think higher sampling rates do produce better sound, and argued this a long time with other on another thread. But for playback, if the levels are normalized, 16/24 is meaningless as all it does is define sounds so quite the can't be heard in any real listenming room at any volume music could be played at.

 Lets face it 16/44 was the engineering compromise forced on Philips/Sony by the storage capacity of a CD. I think people trying to post-justify it now as some great standard are wasting their time and are a bit silly, but red books sound quality issues are more about the engineering limitations of its delivery than they are about its theoretical short commings.


----------



## bigshot

Quote:


  Originally Posted by *lexnasa* /img/forum/go_quote.gif 
_I can play you a native 24 bit 44.1khz recording vs a native 16 bit 44.1khz recording of the same sound and there is a difference._

 

I've done that comparison on a ProTools workstation. No difference in normal playback.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *frankR* /img/forum/go_quote.gif 
_It’s also frustrating to hear them say Redbook is all you need when I know SACD sounds unmistakably superior to Redbook._

 

Go get Pentatone's Jaarvi Stravinsky disk. (A DSD digital recording.) Rip the redbook layer or get two copies and line it up in a CD player. Line up the SACD in an SACD player. Balance the line levels carefully with preamps. Hit play. See if you can hear a difference. I did this comparison. Both sounded just as good. Hard as I tried, I couldn't hear a difference. It certainly wasn't an unmistakable superiority.

 Comparing analogue reissues on hybrid CDs isn't a good comparison, because the mastering is often different on the different layers. Using one machine to compare doesn't work because there is too much of a lag as you switch layers to be able to get a direct A/B comparison.

 See ya
 Steve


----------



## gregorio

I think lexnasa was just trying to throw a spanner (wrench) into the works. The posts must have been put up there as a joke. If I had to write a few paragraphs deliberately trying to misunderstand digital audio I couldn't have done it better myself. If he/she was being serious, that's a serious concern. I have to say that even among people earning a lot of money and calling themselves professionals, it's by no means uncommon to find a lot of ignorance and mis-understanding.


----------



## Publius

What is "physicality of playback", anyway? Do 24 bits fill my ear up more than 16 bits? Do bits have a caloric value when eaten?

 Moreover: 4 bits _does_ sound as good as 16 bits. Actually, heh, it sounds better. I'd bet that most sigma-delta DACs nowadays run at 4 bits or less internally.


----------



## CyberTheo

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I've done that comparison on a ProTools workstation. No difference in normal playback.

 See ya
 Steve_

 

Lets conclude that the average listener can't hear any difference, though an average listener probably doesn't have a high resolution system either, nor golden ears that so many self-proclaimed audiophiles have. Being a self-proclaimed audiophile like myself, I would like to assume I have good ears. 
	

	
	
		
		

		
			





 Still, it's a blanket statement to claim that when you can't hear it, there is no difference.


----------



## nick_charles

Quote:


  Originally Posted by *CyberTheo* /img/forum/go_quote.gif 
_Lets conclude that the average listener can't hear any difference_

 

The evidence so far would seem to support the truth of the above assertion , but also there is pretty decent evidence that other non-average listeners have similar dificulty, including listeners who are "audiophile", audio professionals, and importantly young listeners and listeners with good hearing.


  Quote:


 , though an average listener probably doesn't have a high resolution system either, nor golden ears that so many self-proclaimed audiophiles have. 
 

Again the Meyer and Moran study used listeners with good hearing and used good if not perhaps stratosperic kit, but for the sake of argument lets say they could have got better listeners and better kit. 

 Then the argument is that a difference between 16 and 24 bit or between 44khz and 96khz ( or 1Mbit DSD) which is massive in absolute terms, 24 bits has 256 times more levels of discrimination, 96K more than doubles the bandwidth still can only be detected with mega-dollar kit and 17 year old ears. Given that most audiophiles who have high end kit are in their 30s + this means their hearing will no longer be good enough to pick up the subtle differences that their high end kit can render.

  Quote:


 Being a self-proclaimed audiophile like myself, I would like to assume I have good ears. 
	

	
	
		
		

		
		
	


	



 

Without wishing to be rude how old are you ? - if you are in your 20s then your hearing should be pretty good, but merely being keen on high fidelity audio will not make anybody's hearing better, you may be able to train your hearing but that will only make the most of what you have.

  Quote:


 Still, it's a blanket statement to claim that when you can't hear it, there is no difference. 
 

Perfectly true, that is why I like studies with big numbers in them.


----------



## bigshot

Quote:


  Originally Posted by *CyberTheo* /img/forum/go_quote.gif 
_Lets conclude that the average listener can't hear any difference, though an average listener probably doesn't have a high resolution system either, nor golden ears that so many self-proclaimed audiophiles have. Being a self-proclaimed audiophile like myself, I would like to assume I have good ears. 
	

	
	
		
		

		
		
	


	




 Still, it's a blanket statement to claim that when you can't hear it, there is no difference._

 

The difference between me and someone who just assumes they have ears capable of hearing the difference is that I actually did a test and found out whether I could or not.

 I'm not saying that there is no difference between 24 and 16. 24 bit is very useful for recording, because it gives you more flexibility in the mix. My point is, during playback of mixed and mastered music there is no audible difference. Why go to all the trouble for something you can't hear?

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Without wishing to be rude how old are you ? - if you are in your 20s then your hearing should be pretty good, but merely being keen on high fidelity audio will not make anybody's hearing better, you may be able to train your hearing but that will only make the most of what you have._

 

He's not going to have good hearing very long if he tries to hear the difference in dynamic range between 16 and 24 bit.

 See ya
 Steve


----------



## wnmnkh

Even with all of these setups I have, I just cannot distinguish the difference.

 And I think there are some research results that humans just cannot distinguish the difference between CD and SACD/DVD-A.


----------



## Chri5peed

Quote:


  Originally Posted by *CyberTheo* /img/forum/go_quote.gif 
_nor golden ears that so many self-proclaimed audiophiles have. Being a self-proclaimed audiophile like myself, I would like to assume I have good ears. 
	

	
	
		
		

		
		
	


	




 Still, it's a blanket statement to claim that when you can't hear it, there is no difference._

 

I believe good hearing can be learnt somewhat, i.e. if you spend weeks listening to music through an expensive/good system, going to a crappy system would be highly noticable...more than someone who didn't do the switch.

 This doesn't mean you can develop 'Golden ears', their owners are freaks of nature.
	

	
	
		
		

		
		
	


	




 edit - Lol, Gold-plated ears!


----------



## nick_charles

Quote:


  Originally Posted by *wnmnkh* /img/forum/go_quote.gif 
_And I think there are some research results that humans just cannot distinguish the difference between CD and SACD/DVD-A._

 

hmmmm let me think ....

  Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_Meyer and Moran (2007) in a peer reviewed journal paper published by the AES concluded that nobody (in over 500 trials) could reliably discern a difference between high res and 16/44.1 - except that you could detect the higher noise floor when some music was played really loud. _

 

 Quote:


  Originally Posted by *hciman77* /img/forum/go_quote.gif 
_so far the best experimental evidence 

Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback E. Brad Meyer and David R. Moran, 2007

 is that when you properly downsample High res audio to 16/44.1 that the difference is inaudible except at really high volume levels where the noise floor comes into play. 
_

 

 Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_No difference.
AES E-Library: Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback by Meyer, E. Brad; Moran, David R.
 Very extensive blind testing study with hundreds of trials including golden ears and many others showed that 24/96 and 24/196 is useless; the only reason many recordings in the hi-rez formats seem to sound better is because they were mastered better, since they're intended for the audiophile crowd._

 

 Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Now if it is data you are interested in. In 2007 Meyer and Moran of the BAS ran a large number of blind listening tests using a variety of systems and material and listeners. Their paper is avalable from the AES at $5 (for members, student membership is $30) it is very interesting reading. 

 Key findings, not one subject (n = 60) was capable of detecting the difference between high res and 16/44.1 at a sufficient level of confidence (95%), the total # of trials was 554 and correct answers 276 (49.819%). Subjects did up to 10 trials. One subject scored 8/10, two scored 7/10, no others scored even 7/10, even these few high performers are results what you would expect by chance with a large enough sample. Engineers and "Audiophiles" did slightly better at 52.7%. Females did worse at 38%, those with better high frequency hearing did worse(45.3%) as did younger listeners. There was no effect due to different systems.

 Now here is the kicker, Meyer and Moran did not even use noise-shaping on the A/D/A stage - this is handicapping the A/D/A stage big time, giving the High-res the best possible hand (metaphorically speaking)._

 

 Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_It really isnt *just* a matter of theory. There has been a rather well-publicised study examining how well listeners can differentiate between high res and 16/44.1 music. With 60 Audiophile, Engineer and Sound Engineering students as subjects the level of differentiation was 50% i.e complete chance._


----------



## ADD

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I've done that comparison on a ProTools workstation. No difference in normal playback._

 

G'day Steve,

 What ancilliary components did you use? (converter, amps, speakers / headphones?). And what type of music was this with and what listening levels? Have you tried these same tests on classical music at, say 5 - 8 dB below live concert levels?

 Anyway, I tried an experiment the other day, where I rendered two output files from Plogue bidule - one was 32 bit floating point (which was the resolution with which I did all the software mastering). Both these output files came from the exact same input file, whose original resolution pre-mastering was 24-48. The two output files were absolutely identical in every way (including output level) bar the resolution. The music I used was classical.

 I then ran them both in Foobar using the blind ABX plugin through my Musical Fidelity XCan3 and my PXC350 headphones. Listening level was around 82 dBc RMS, so soft bits around 40 - 50 dB with very rare and short peaks to about 101 dB. In 16 attempts I identified the 32 bit floating version 100% of the time versus the 16 bit (but it was not easy - I had to concentrate pretty hard).

 I don't have golden ears and I'm getting on a bit too, but technically my hearing is still medically classified as "normal". I do tend to agree with you that apart from the mixing and mastering process in 24 bit that allows for headroom, it shouldn't be likely the differences on the final output are audible.

 So with all that I am just trying to find a possible technical reason for my findings. I'm noticing the same thing when I use my digital recorder to record directly from vinyl - a 24 bit recording sounds different to a 16 bit one in a Foobar ABX comparison. Perhaps the equipment I am using is not good enough - in other words the 16 bit conversion is not good enough to be inaudible versus 24 bit (which might be the equivalent of "perfect" 16 bit for all I know), but if I used a very expensive converter, the 16 bit might be good enough for me not to hear the difference.

 Do you have any classical music test files where you can't hear the difference between 16 and 24 bit? I wouldn't mind sticking them into the Foobar ABX to see if I can hear anything.


----------



## nick_charles

Quote:


  Originally Posted by *ADD* /img/forum/go_quote.gif 
_Anyway, I tried an experiment the other day, where I rendered two output files from Plogue bidule - one was 32 bit floating point (which was the resolution with which I did all the software mastering). Both these output files came from the exact same input file, whose original resolution pre-mastering was 24-48. The two output files were absolutely identical in every way (including output level) bar the resolution. The music I used was classical._

 

Could you by any chance post these, I would be really interested in trying them out, do you still have the originals ?

 So you converted 24-48 files to 16/48 and 32/48 ?


  Quote:


 I then ran them both in Foobar using the blind ABX plugin through my Musical Fidelity XCan3 and my PXC350 headphones. Listening level was around 82 dBc RMS, so soft bits around 40 - 50 dB with very rare and short peaks to about 101 dB. In 16 attempts I identified the 32 bit floating version 100% of the time versus the 16 bit (but it was not easy - I had to concentrate pretty hard). 
 

Interesting and at last what appears to be a decent piece of evidence to contradict Meyer and Moran. 

 Where did you detect the differences ?

  Quote:


 So with all that I am just trying to find a possible technical reason for my findings. I'm noticing the same thing when I use my digital recorder to record directly from vinyl - a 24 bit recording sounds different to a 16 bit one in a Foobar ABX comparison. Perhaps the equipment I am using is not good enough - in other words the 16 bit conversion is not good enough to be inaudible versus 24 bit (which might be the equivalent of "perfect" 16 bit for all I know), but if I used a very expensive converter, the 16 bit might be good enough for me not to hear the difference. 
 

When you compare 16 and 24 bit digitizations do you do 2 different digitizations or one at 24 bit and render down to 16 bits. I ask as I have noticed that I can do 2 digitizations of the same input at the same settings and get slightly different results - when I say slightly I mean 100ths to 10ths of a db different at most frequencies, but variations at high frequencies can be quite large, showing some variability in the workings of my ADC. This is massively annoying as you would expect more consistent results and it may just represent the inherent error levels of my setup at 16 bits.


----------



## bigshot

I used classical music, but I was testing transfers from LP at the time, so the records I used were some Living Stereo and Sheffield Lab disks. I also used some finished mixes from the program I was working on at the time.

 However, you've set up a scenario there that would show the difference, if it could be detected... high volume, huge dynamics, headphones, etc. The only problem, which I'm sure you noticed, is that that kind of volume is not a comfortable listening level. It's not a natural way to listen to orchestral music either, unless you're trying to reproduce what the orchestra sounded like if you were sitting in its lap. The volume you used for comparison is very near the edge of causing hearing damage with extended listening.

 Even with that extreme situation, you still have to struggle to be able to discern a difference. It probably isn't going to be discernible at a normal listening level.

 There's one more thing you can check if you're interested. You should try a variety of dithers when you downsample the 24 bit track to 16. Some dithers work better with some types of music than others. The minute difference you're detecting might be due to the dither.

 See ya
 Steve


----------



## ADD

Hi Steve,

 I would agree with those points. Late lat night I tried the experiment again, rendering two output files natively at 24 bit and 16 bit (rather than the 32 float and 16 bit previously).

 This time around the blind test was interesting. I would get it right for the first 5 attempts, and then listening fatigue and brain fade would set in (the amount of concentration needed to detect anything is monumentally huge). By the time I got to 16 (my preset "target"), I would be lucky to have scored 40% - in all cases the early scores were good and my results declined thereafter. This thus represents a failure on my part to hear the difference.

 The actual SPLs through the headphones were still around 8 dB lower than what I would get in the concert hall, but I also noticed that if I were to have any "chance" at all, the passages needed to be of a moderate volume (I don't mean turning the volume up - I mean all the violins had to be playing mezzo forte or louder to begin with).

 The interesting thing was that I thought I might hear something in the very soft passages, but I couldn't. Even things like the tiny, ultra-quiet clattering of of the soles of Curzon's shoes on the piano pedal were indistinguishable for me between 16 and 24 bit. And of course I could not continually listen to forte passages because of the continued SPL exposure. 

 So I gave up because (a) it was too hard for me to tell and (b) continuing to do it would not have done my ears any good. 

 As for dithering (or more precisely, using programs to change resolution rather than natively recording in it), that was much more easy to notice - I scored better on that depending on the program and settings I used. None of them sounded bad per se, but I found that recording natively in 16 bit produced a better result than recording in 24 and dithering back to 16. By that I mean more faithful to the original.

 But of course the point of all of this discussion really ought to be about taking the exact same piece of hardware and making two recordings at 24 bit and 16 bit native resolution and being able to reliably ABX them.

 As for me, it is beyond me. However I can't dismiss the fact that there would be people around who hear better than me, who have incredbly discriminating ears and have huge powers of concentration. That being the case, in theoretical sense, I guess the jury is still out for me, in that such a feat _could_ still be possible. But if such people existed, I don't think they would be anything like normal human beings.

 But I will say at the very least that for normal humans (99.9% of us), 24 bit is only audible to our recording equipment chains 
	

	
	
		
		

		
			





. And I feel it gives us sufficient headroom such that by the time the final mix comes out of the (sometimes considerable) wash, it is still using at least 16 bits of precision.

 I'd have to say to anyone who remains adamant about these sorts of sonic differences, it's a totally different ball game when you have no crutches to hold onto and you are there blind ABXing with absolutely no hints, no placebos and total objectiveness. Suddenly all those differences you thought you were hearing - well, you no longer hear them.


----------



## ADD

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Where did you detect the differences ?_

 


 Hi Nick,

 I don't have the originals as I was badly running out of disk space, however I don't think we should read much into that first experiment. In that particular instance, I had run a 24-48 input file through Dolby headphone in Plogue Bidule, with Dolby Headphone apparently accepting 32 floating input (or converts the input to floating point) and spits out 16 bit and the end (well, that is how I understand it works).

 So I have think that I am hearing some abnormality in the equipment, in that I was asking my hardware to read 24 bit, process 32 bit in software and then render to 32 bit floating from a 16 bit output.

 Yes, I did hear differences, and it was mainly in the authenticity of massed violin sound - but as I say, I am not convinced this was a genuine way to conduct this sort of experiment.

 If you read my reply below Steve's, you'll see I tried again, but this time I used completely seperate equipment. I ran a native 24-96 digital file through the analogue outputs of my X-Fi soundcard and into my digital recorder, which can natively record at either 16 bit or 24 bit. I made three sets of paired recordings - each pair had a 16-44 and 24-44 recording.

 I then uploaded them to the PC and trimmed them so that they were of exactly the same length and start point, then blind ABXed the three pairs in Foobar. In this case, my scores were less than 50% on all of them, so effectively I could not hear differences amongst them.


----------



## ADD

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_So you converted 24-48 files to 16/48 and 32/48 ?_

 

The problem is that if you do this, you are hearing the "sound" of either the software alogrythms or hardware used to change the resolution. And in my experience, these are sometimes audible. One program of mine (Resample) is quite audible when doing this, another (r8brain) has two filters - a linear and minimum phase one. The minimum phase filter is quite audible in blind listening, the linear isn't (to my ears anyway).

 I think the most "pure" way to test this out is to use hardware that can record at both resolutions natively and then compare the results. Otherwise you are just hearing the "sound" of a piece of software or hardware that is capable of changing resolution of an original file, rather than one which records in the desired resolutions to begin with.


----------



## gregorio

Hi ADD,

 Reading through your posts, I still can't quite work out exactly how you're getting from 24/48 to 16/48. It's important to know this information as it could account for perceived differences.

 There are only 3 ways to get from 24 to 16bit audio; dithering, rounding and truncating. Truncating, while not adding much noise, does cause other more serious problems; odd harmonics, harshness and various other artefacts. Truncating is definitely the worst option of the three. Rounding, chops of the 8 LSBs (like truncating) but mathematically rounds off the result, second best option. Dithering is the best option, adding noise to the mathemetical process, converting all errors into white (uncorrelated) noise.

 End observation; a dithered 16bit converted file may sound slightly noisier at very high volumes but will be indistiguishable (to your ear) from the 24bit original. The amount of additional noise (compared to the original) you will hear will depend on the dithering program and the noise shaping settings but a good noise shaped dither should be below the noise floor of virtually any consumer system.

 Just out of curiosity, can you hear any difference between 32 and 24bit? If so, no matter how slight, it would indicate a problem with your software. In theory there should be no difference whatsoever (audible or measurable in any other way at the output of your DAC) between a 24bit file and a 32bit file with the last 8 LSBs all set to zero. BTW, this type of problem with software isn't unheard of!


----------



## ADD

Hi,

 This may sound very stupid, but I do not actually know how I am getting to 16 bit. Plogue Bidule offers a "recording" function, buty it's more traditionally the rendering type of function that would be used in DAW mastering to produce an output file. And that function simply lets me choose from 32 bit floating output, 32 bit output, 24 bit output and 16 bit output. So I don't know what the software is doing to produce an output file of lower resolution than the input file. It does have a dither checkbox too (as an overall Plogue Bidule setting), but I can't hear a difference between dithered output and non-dithered output.

 That said, if I use a program such as "ReSample" and dither 24 bit output to 16 bit output (obviously without changing the sampling rate) using it's best dithering algorythm, I can hear a difference in ABX listening.

 But I have tried the Plogue Bidule test several more times this week (listening at lower volumes than the first test, since the original volumes were slightly exaggerated compared to normal listening volumes) and I have consistently failed to successfully ABX 16 bit from any other resolution.

 So sorry, I know that does not really answer your question, apart from saying I can hear a difference when I use the "ReSample" program.

 I'd like to say though that after even more investigations into what I have been doing, I don't think I am really giving myself a fair crack at the whip with this, and particularly not as regards attempting to provide a meaningful answer as to whether humans can tell the difference between 16 and 24 bit.

 Those reasons - in my opinion - are as follows:

 1. I don't actually have any source recordings where the original noise floor or softest passages are low enough to explore the potential of 24 bit - thus far I have just been transcribing LPs.

 2. One of the devices I use (a Zoom H2 recorder) has been reported by various parties to only offer something like around 17 to 18 bits of resolution in 24 bit mode, owing to the noise floor. OK, it is better than 16 bit, but this then becomes an issue of whether I can hear the difference between 16 and 18 bit at best!

 3. I don't have great ears. I would honestly have to say that my hearing is likely average at the best, despite the gushing appraisal of an audiologist last year. You would not need to search very far at all to find someone who can hear better than I can imo.

 4. The equipment I have been using was just my X-Fi soundcard output and my PXC350 headphones.


----------



## gregorio

Quote:


  Originally Posted by *ADD* /img/forum/go_quote.gif 
_1. I don't actually have any source recordings where the original noise floor or softest passages are low enough to explore the potential of 24 bit ..._

 

Don't worry, no one does! The very finest quality pro-audio gear usually has a noise floor around -108db to -120dB, still along way away from the -144dB of 24bit and we haven't even considered the noise floor of the recording environment which is going to be higher still, even in a world class facility!

 Baring the above in mind, 17 to 18bits is entirely within the range expected of pro-audio gear. The very finest quality pro-gear sometimes gets close to 20bits (120dB dynamic range). As Bigshot, Nick and others mentioned a few pages back, you are not likely to find any commercial recordings (on any format) which utilise a dynamic range larger than about 60dB.

 I don't know what the Bidule program does either but my guess would be that unless you are checking the dither box then it's truncating. I'm a little surprised that you can't hear any difference between the truncated and dithered versions, I would expect you to hear something different if you turned the gain up quite high.

 Your experience with "ReSample" should not be happening! The (probable) 50 or 60dB range from your original recording should put the dithering noise way below the noise floor of the recording. If you are hearing a difference "ReSample" would definitely appear to be doing something dodgy. If you can get it, find a program that includes the Pow-R dithering algorithms. Most professional audio software includes Pow-R, quite a few others too.


----------



## Nocturnal310

I dont know why but since i have changed my laptop sound driver & changed the format to 24bit, 48000Hz.

 i ve been hearing very Crystallized sound, very sharp & i can also here static + vinyl effect from my HD205.

 I tried switching back to 16 bit ..4-5 times to confirm.. and yes there was a difference in quality of sound itself.

 used Low Frequency protection to remove the static..now even 128 kbps tracks are sounding like 192 kbps.

 I am an audio n00b...but to notice the difference this is what i did.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_you are not likely to find any commercial recordings (on any format) which utilise a dynamic range larger than about 60dB._

 

For the overall envelope in the time domain, sure. But if you do a series of windowed FFTs (what the ear does) you'll see individual troughs going below that in many recordings. You're thinking of overall volume level, whereas I'm talking about the increased resolution of detail--how close the gap between digital words with values x and x+1 is.


----------



## ADD

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I don't know what the Bidule program does either but my guess would be that unless you are checking the dither box then it's truncating. I'm a little surprised that you can't hear any difference between the truncated and dithered versions, I would expect you to hear something different if you turned the gain up quite high.._

 

Well I guess one reason is that I'm not turning the gain up high because then comparisons - particularly ABX ones - are pretty fatiguing and tough on the ear. The way I see it, if I can't hear the differences at normal listening levels, then it isn't something for me to worry about.

 I think there could be another reason though. I use Plogue Bidule to process vinyl transcriptions using Dolby Headphone and I believe the DH plugin author once said that nothing more than 16 bits comes out of Dolby Headphone anyway - regardless of what goes into it. So that being the case, it's arguable there would be something quite wrong if I could consistently hear differences in output bit depths. So in this instance, although checking or unchecking the dither box meant I could not hear any difference, it is possible that no dithering ever took place within Plogue Bidule per se - because 16 bits is all that the Dolby Headphone plugin spits out (so if any dithering was done, I guess it is part of Dolby Headphone's internal algorythms, but I am just speculating here).

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Your experience with "ReSample" should not be happening! The (probable) 50 or 60dB range from your original recording should put the dithering noise way below the noise floor of the recording. If you are hearing a difference "ReSample" would definitely appear to be doing something dodgy. If you can get it, find a program that includes the Pow-R dithering algorithms. Most professional audio software includes Pow-R, quite a few others too._

 

I agree that the results I get with ReSample are rather strange. That said, my main reason for wanting to resample is to convert my 24-96 vinyl transcription masters to 24-48 in order to input into Plogue Bidule (that isn't a restriction of Plogue Bidule - it is a restriction of Dolby Headphone). It is hard for me to imagine a more transparent resampling than r8brain Pro using it's linear filter and ultra-steep mode, but I am always up for comparisons. It is certainly better than either ReSample and also seems to be better than the hardware resampling of my X-Fi card (which is actually surprisingly good to me).


----------



## gregorio

Nocturnal - Similar problem to the conversation I'm having with ADD. I'm not quite sure how you're switching between bit depths but something is going on with your 16bit which shouldn't. The practical difference between 24bit and 16bit should not be noticeable in an MP3 file, even an MP3 at 320kbps. In fact, only in extreme (bordering on dangerous) situations should you be able to hear a difference with completely uncompressed audio.

 Crowbar - The increased detail you're talking about only exists in graphic representations of digital audio data stored on your computer. When output through a DAC there is no difference in resolution, as resolution is perfectly linear (at any bit depth). Remember, all the bits are doing is storing a quantised value of the wave amplitude. All I can say is that in my experience I've never heard of or seen a commercially released recording with more than a 60dB range and I am talking about peak values, not averages.

 ADD - It just goes to show how much processing is going on and all the variables we are actually listening to when we listen to a recording. I don't know the software you have mentioned (r8brain Pro) I mentioned Pow-R because it has been developed by a committee who license it's use to a number of different software manufacturers. It may not be the best dithering software on the market but it is very good and will produce 16bit files which are indistinguishable from 24bit masters. It's kind of the standard against which other dithering programs are measured.

 Just because we may listen to a 24bit recording and hear better quality compared to a 16bit recording does not mean that 24bit is any better than 16bit. What we can say (and prove), is that given appropriate processing a 16bit audio file is indistinguishable from a 24bit file at any normal listening volume. However, it would seem that there is often quite a bit of inappropriate processing going on, due to software, hardware and quite often ignorance. Digital Audio is quite complex when we start looking at it in any detail. Unfortunately, manufacturers use and manipulate the fact that consumers (and quite a few pros) don't have an accurate understanding of the principles or issues. Professionally, certain manufacturers gain reputations for good quality digital audio processing and many of us pros take an interest in what processing is being done but in the consumer world I wouldn't have a clue about what goes on inside various boxes (or software) and many manufacturers are unwilling to discuss the details or how detrimental their processing is to the sound quality.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_When output through a DAC there is no difference in resolution, as resolution is perfectly linear (at any bit depth)._

 

Whether it's linear is besides the point. The sound intensity difference between a word x and the consecutive one, x+1, is the same regardless of what x is, but that difference is a function of various things, such as the analog volume control, gain, etc. You can make a 16 bit DAC S and a 24 bit DAC T such that word (hex) 0000 from S and 000000 from T produce equal power output, and word FFFF from S and FFFFFF from T also produce equal power output. Now both cover the same dynamic range, but T provides a much finer quantization.

 So you are confusing quantization and dynamic range. With increased bits per sample, you can either increase the dynamic range you cover, the quantization, or both. This far you have only made an argument that the total dynamic range of music is not usually above 60 dB. But that doesn't address the quantization within it. I could produce a 1 bit system where 0 is dead silent and 1 is 200 dB. That covers an enormous dynamic range, and trivially shows that dynamic range without considering quantization is nonsensical.

 As another example, consider the image pairs that were posted above, one 8-bit and another one 24-bit. They both cover exactly the same dynamic range! A black pixel in one is as dark as a black pixel in the other, and a white pixel in one is as bright as a white pixel in the other. All the difference comes from a quantization.

 It's unfortunate that terminology is used imprecisely. Consider HDR (high dynamic range displays)--something I'm closely familiar with since I helped my professor with one of the earliest prototypes in the university lab years ago. The dynamic range by definition is the contrast ratio. Yet if you just increase that and retain 8-bit per color quantization, you start getting banding artifacts as the difference between two consecutive brightness levels increases. So when people refer to HDR displays, it is never meant literally; rather, it does have increased contrast ratio, but also increased quantization.


----------



## Crowbar

Quote:


  Originally Posted by *ADD* /img/forum/go_quote.gif 
_I'd have to say to anyone who remains adamant about these sorts of sonic differences, it's a totally different ball game when you have no crutches to hold onto and you are there blind ABXing with absolutely no hints, no placebos and total objectiveness. Suddenly all those differences you thought you were hearing - well, you no longer hear them._

 

That implies that those saying there are differences are against blind testing, which I find very insulting. I have been banned from a forum for arguing against blind testing deniers.
 Moreover, ABX is not recommended as the blind test to use by the official standars body (ITU); tristimulus test with hidden reference (ABC+HR) is. I have pointed this out in this thread alone at least twice. Next time try reading a thread before exercising your itchy Submit Reply-clicking finger.


----------



## gregorio

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_So you are confusing quantization and dynamic range. With increased bits per sample, you can either increase the dynamic range you cover, the quantization, or both._

 

No Crowbar. In digital audio theory you can only increase the dynamic range by increasing the number of bits. The quantisation steps are not variable, they are always the same no matter how many bits the system is capable of. Adding another bit doubles the quantisation steps, halves the quantisation errors and therefore halves the amount of noise in the system. So the only advantage of more bits is a larger dynamic range, full stop! Think of it like a ladder, as we increase the number of steps we just make the ladder longer, the steps do not get closer together. Each time we add an additional bit of data we add the ability to encode another 6dB of dynamic range. This isn't negotiable or variable, it is true of every PCM digital system.


----------



## grawk

It's the same fairyland problem it's been all along. How do you have a discussion when on one hand, you have reality, and on the other, you have what crowbar thinks reality should be?


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_In digital audio theory you can only increase the dynamic range by increasing the number of bits. The quantisation steps are not variable, they are always the same no matter how many bits the system is capable of._

 

False. The quantization steps size is an analog quantity that depends on the current output of the dynamic elements within the DAC and the transimpedance and subsequent voltage gain of the I/V and further analog stages.

  Quote:


 Adding another bit doubles the quantisation steps 
 

Yes, and that agrees with what I said.
  Quote:


 halves the quantisation errors 
 

Yes, and that also agrees with what I said.
  Quote:


 and therefore halves the amount of noise in the system. 
 

It halves the amount of relative noise, not necessarily the absolute noise. And this only concerns quantization noise. The issue of a noise floor that is present even when the DAC is being fed samples of even the lowest value, 0, is separate. You can have a DAC that has such an analog noise level above say a couple of LSBs, yet a narrowband singal with 1 LSB magnitude will still be resolvable.

 Since the noise halved is relative, there's nothing here that addresses the absolute signal levels produced by the smallest and largest digital value, which defines the dynamic range.

  Quote:


 as we increase the number of steps we just make the ladder longer, the steps do not get closer together. 
 

As I pointed out above, you've failed to show that. Moreover, it's easy to disprove by a trivial contradiction. Here's an image example again. The mathematics of signal processing is identical with the audio case, other than an extension to two spatial dimensions instead of one time dimension. *The two images below have THE SAME dynamic range*, yet one is 24 bit and the other 4 bit. The difference is in quantization, and the first one has the ladder steps closer together, to use your terminology. If you disagree with my claim about dynamic range vs quantization, you must show that my analogy in image processing is invalid (which you can't really do, because it's the exact same math).


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_It's the same fairyland problem it's been all along. How do you have a discussion when on one hand, you have reality, and on the other, you have what crowbar thinks reality should be?_

 

Your inability to bring forth any supporting argument highlights your ignorance and makes your position irrelevant to any thinking reader. You're clearly the sort of pathological example of an intellectual failure which, unable to rationally justify its beliefs, tries to hide its inadequacy with arrogance and the pretense of authoritarian decrees that are impotent to sway the minds of all but the most cretinous sycophants.


----------



## gregorio

Crowbar - There has been plenty of proof provided throughout this thread. It's can be a little difficult to grasp but is really not that complicated. 

 I'm not very au fait with image processing but your analogy doesn't appear correct. Dynamic range in a digital audio system is based almost exclusively on bit depth. Dynamic range is all that bit depth represents. So the dynamic range of a 4bit audio system is vastly different to the dynamic range of a 24bit system. Once we get to 16bit then we are already beyond a human being's ability to hear any quantisation noise, at all but uncomfortable or dangerous levels. An analogy would be your two images, one at 512bit and one at 1024bit, could you tell a difference? Is there even equipment which is capable of representing 1024bit depth in an image, if so, are human eyes capable of telling the difference. If the answer to all these questions is no, then it's a good analogy between 16bit and 24bit audio.

 If this doesn't answer your question then I don't really know what you are on about. Of course the dynamic range is relative. The dynamic range of a 16bit system is 6dB more than the dynamic range of a 15bit system. Exactly where you define the peak or minimum value on your amp is irrelevant to digital audio theory.


----------



## grawk

Ok, I agree, it is possible to write your own soundfile type, where you define a range to be 120db, where a 4bit, 16bit, 24bit and 32bit file all define the same space, with more resolution. It's possible. You do this, and you've done it. No one else supports it, no one else uses it, nothing records it, nothing plays it back. As there are currently vast numbers of devices for recording, playing back, and manipulating the standards as they actually exist, continuing to talk about the new method you've defined is equivalent to talking about fairyland, because no one is going to implement it in the real world.


----------



## Nocturnal310

Can someone give the final verdict?

 I really wanna know this. Because outside .all these companies are doing this hype of 24bit ranting.

 Pls tell me in 100 words how 24 bit is better.


----------



## grawk

24 bits is better for capturing audio because it lets you leave more headroom in the recording process.

 24 bits is better in the playback process because it lets you have enough dynamic range to use soundwaves to set small animals on fire.


----------



## Nocturnal310

And for average listener with a Pair of headphones without amp or soundcard ..it doesnt make a difference rite?


----------



## grawk

It matters because some high quality recordings are available only in 24bit, but otherwise, no it doesn't make a difference if you listen at levels that wouldn't make your ears bleed.


----------



## gregorio

Nocturnal - Providing you are listening to a finished master (rather than recording audio) and provided decent quality dithering has been used, then there is no perceivable difference between 16bit and 24bit. There is a technical difference (where the theoretical noise floor is) but using even the highest quality playback equipment and environment, you won't be able to hear this difference in the real world. This difference is only related to the noise floor, in every other respect, the resolution and quality of 16bit is identical to 24bit.

 Even running your amp to near breaking point (and making your ears bleed) won't let you hear the difference because the noise floor of the amp (even a world class amp) is going to be higher than the noise floor of properly dithered 16bit audio.


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Nocturnal - Providing you are listening to a finished master (rather than recording audio) and provided decent quality dithering has been used, then there is no perceivable difference between 16bit and 24bit. There is a technical difference (where the theoretical noise floor is) but using even the highest quality playback equipment and environment, you won't be able to hear this difference in the real world. This difference is only related to the noise floor, in every other respect, the resolution and quality of 16bit is identical to 24bit.

 Even running your amp to near breaking point (and making your ears bleed) won't let you hear the difference because the noise floor of the amp (even a world class amp) is going to be higher than the noise floor of properly dithered 16bit audio._

 

This is mostly but not quite entirely true. In the Meyer and Moran paper they did find that it was possible to discern a difference *sometimes* with a select few recordings that had *unusually low *background noise levels when played at 112db or so, which is admittedly quite loud. But from what I read it was only the different noise level that gives the game away not some intrinsic perceptible quality difference.

 More interesting however was Blech and Yang's paper (2004) that found that 24 bit PCM and DSD were perceptibly indistinguishable in 97.3% of their tests.


----------



## Nocturnal310

Okay there mite be a difference...but can u elaborate on WHAT DIFFERENCE?

 i mean is it the details, the background noise, the softness? what basically improves when u r at 24 bit?


----------



## grawk

what improves is the thought that goes into the recording and hte hardware design, generally, that's it. It's shorthand for "this will probably sound better". In reality, no one needs the 24 bits, it just raises the bar and so the marketting changes.


----------



## bigshot

For playback of music in a home setting, high bitrates are unnecessary.

 See ya
 Steve


----------



## gregorio

Nocturnal - The only difference is the level of hiss (white noise) in the background. 24bit has a lower noise floor than 16bit, apart from this there is no difference. Under any normal listening conditions though, there is no way you're going to be able to hear it. You'd have to have your amp turned up so high that in the loud passages you would be close to the pain threashold of human hearing.

 Grawk - True, except that 24bit is not a gimmick, it's extremely useful for the recording engineer. Pretty much all professional recordings are now done in 24bit and have been for a few years. I personally switched to 24bit recording in 1997. However, when we're talking about the consumer, or even professionals just playing back completed (mastered) recordings, then you are entirely correct, it's just a marketing ploy. This ploy looks like being quite successful as it plays on the common misconception that more bits = greater accuracy and better quality sound.

 Nick - My caveat is the dithering that has been used. It's generally accepted that top class noise-shaped dithering will allow about 120dB of dynamic range. I say generally accepted because it's based on the perception of added noise. With a range of 120dB or even 112dB (as you mentioned) this is still beyond what most individual bits of pro-equipment are capable of. Sum the noise from all the equipment, together with the noise floor of the recording environment and I can't see how it would be possible to hear the noise floor of 16bit (properly dithered). My guess is the recordings where people could tell a difference used non-noise shaped dither, in which case it may be possible to hear the noise floor (-96dB) but you'd still have to have your amp cranked-up to uncomfortable levels.

 To put this into context, the Prism ADA-8XR is one of the top pro ADCs/DACs on the market. Many world class studios and audio post facilities use them (Recording and Production User List) and they cost about $12k per unit. The Prism (and the Lavry's) are the best converters I've ever heard. The noise floor of the Prism is given as 108dB. This is way above the noise floor of 24bit (144dB) and even above the noise floor of well dithered 16bit (120dB approx). So, if you can't hear the 16bit digital noise floor on a truely world class piece of pro-gear, what's the consumer going to notice?


----------



## grawk

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Grawk - True, except that 24bit is not a gimmick, it's extremely useful for the recording engineer. Pretty much all professional recordings are now done in 24bit and have been for a few years. I personally switched to 24bit recording in 1997. However, when we're talking about the consumer, or even professionals just playing back completed (mastered) recordings, then you are entirely correct, it's just a marketing ploy. This ploy looks like being quite successful as it plays on the common misconception that more bits = greater accuracy and better quality sound._

 

I have said that over and over in this thread too 
	

	
	
		
		

		
			





. I'm not a professional, but I've done a fair bit of location sound recording...24 bit was a godsend for recording. Playback just normalize and dither and 16 is way more than enough.


----------



## regal

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Nocturnal - To put this into context, the Prism ADA-8XR is one of the top pro ADCs/DACs on the market. Many world class studios and audio post facilities use them (Recording and Production User List) and they cost about $12k per unit. The Prism (and the Lavry's) are the best converters I've ever heard. ?_

 

You know the old Pacific Micro Model One smokes the new Lavry and Prism ADC's. Have you heard one?, its all discrete without all of the opamp garbage.


----------



## gregorio

Grawk - I wouldn't normalise. Keep your peak level to about -3dB otherwise you could end up clipping on some DACs.

 Regal - I haven't heard the Pacific Micro. In what way does it sound better?


----------



## Nocturnal310

But when i playback audio on 16 bit @ high Gain...there is crackling Noise in the background.

 This noise vanishes under 24-bit 48khz playback.

 Why is that?


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It's can be a little difficult to grasp but is really not that complicated._

 

I think, having studied digital signal processing in graduate courses in university, I don't have any problem grasping anything.

  Quote:


 your analogy doesn't appear correct. Dynamic range in a digital audio system is based almost exclusively on bit depth. Dynamic range is all that bit depth represents. 
 

This doesn't explain why my analogy is supposedly incorrect. It simply restates your claim that it is incorrect. The rest of your comment relies on that assumption and thus falls with it.

  Quote:


 An analogy would be your two images, one at 512bit and one at 1024bit, could you tell a difference? 
 

For a display with standard contrast ratio, these are much further from threshold of detectability than 16 and 24 bit in audio. The 120 dB figure is a scientific fact--it's what you'll find in any reference and encyclopedia.

  Quote:


 Is there even equipment which is capable of representing 1024bit depth in an image, if so, are human eyes capable of telling the difference. If the answer to all these questions is no, then it's a good analogy between 16bit and 24bit audio. 
 

It's not because 512 bit in a display with dynamic range extended to cover total darkness to blinding light, this still provides much finer difference between levels than is needed to create banding artifacts; i.e. increasing the digital word by 1 won't make a visible difference. However, in a digital audio system where levels are set so that the lowest is at the threshold of hearing and the highest at the threshold of pain, increasing the digital word by 1 if the system is 16 bit will make an audible difference.

 As for what equipment is able to display, look at HDR displays:
DR37-P - Extreme Dynamic Range LED Display - Dolby
 At 8 bit per color per pixel (24 bit per pixel RGB images), the quantization is fine enough that on a regular display where the dynamic range = contrast ratio is low enough that you can't see differences between consecutive digital values. On an HDR display such as the one above where the black is actually fully black (backlight array has LEDs in that location off) and the brightest possible white is as bright as staring directly into a lightsource (LEDs in that area behind the LCD fully on), the contrast ratio is far larger and 24 bit images would have banding artifacts, showing obvious bands of different brightness, like the 4 bit image example from above. 16 bit per color per pixel (48 bit RGB images) is sufficient to make the quantization fine enough for this increased dynamic range.

 In the case of audio, if the analog levels are set so that the dynamic range covers the full 120 dB audible, then 16 bit doesn't provide for fine enough quantization (which means you can't hear the difference between two levels different than one)--at least in some sections of the overall range, as hearing is not linear but logarithmic, whereas PCM is linear. This last bit actually would allow 16 bit coding to give fine enough quantization _if_ it didn't have to be linear, as threshold differences between levels are smallest at the bottom of the scale with quieter sounds. In other words, you can hear a difference between words (hex) 0012 and 0013 but not between 7005 and 7006 (same thing with images, differences between brighter levels are less detectable by the eye). But PCM is linear for implementation reasons and also for editing purposes where the absolute levels are often shifted, rather than efficient coding.


----------



## Crowbar

Quote:


  Originally Posted by *Nocturnal310* /img/forum/go_quote.gif 
_But when i playback audio on 16 bit @ high Gain...there is crackling Noise in the background.

 This noise vanishes under 24-bit 48khz playback.

 Why is that?_

 

Analog noise level in the 24-bit system is designed to be at a much lower absolute level.


----------



## regal

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Grawk - I wouldn't normalise. Keep your peak level to about -3dB otherwise you could end up clipping on some DACs.

 Regal - I haven't heard the Pacific Micro. In what way does it sound better?_

 

Much more accurate, the filter is not brickwalled, they used Dynamic-decimation filtering which mitigates the problem with brickwalls. So the treble is a lot more realistic. I think the anti-dithered they used is also beneficial.

 Some of the few studios left who care about SQ still use the Pacific Micro ADC, it is generally regarded as the best ADC ever made, sort of the golden age of digital when it was more popular.


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_Ok, I agree, it is possible to write your own soundfile type, where you define a range to be 120db, where a 4bit, 16bit, 24bit and 32bit file all define the same space, with more resolution._

 

Actually, it has nothing to do with defining a sound file or anything on the digital side. It has to do with the specifics of the analog electronics. As soon as two DACs, both 16 bit, have different ratios between their loudest output and the base level of their analog noise floor, they have different dynamic range. The bit depth just shows how finely that is quantized.

  Quote:


 As there are currently vast numbers of devices for recording, playing back, and manipulating the standards as they actually exist 
 

There are no standards about the specific analog levels, other than that the levels between consecutive words differ by the same voltage.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Even running your amp to near breaking point (and making your ears bleed) won't let you hear the difference because the noise floor of the amp (even a world class amp) is going to be higher than the noise floor of properly dithered 16bit audio._

 

There are power amplifiers with THD+N below 120 dB from full scale output, so their noise floor is inaudible even if you set the highest level to 120 dB (threshold of pain); see Home for one actually built, and a bunch more designs have been published at diyaudio forums.

 Your post is also misleading about dithering, because the best dithering by noise shaping just moves the noise from one part of the band to another--it's not magic. This has already been pointed out. With 96k audio you could move the nose above the audible frequency band, but with 44.1 you can't since there's not enough space between 20 and 22 kHz.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It's generally accepted that top class noise-shaped dithering will allow about 120dB of dynamic range._

 

[size=large]In one part of the frequency range! The noise can be moved around, but you can't decrease the total amount of quantization noise! The decreased quantization noise in one band is necessarily at the expense of increased noise in another part![/size]


----------



## Crowbar

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_its all discrete without all of the opamp garbage._

 

The number of commercial products with discrete circuits that have lower distortion than a high end opamp (say AD797) I can count on the fingers of one hand. Opamps are not garbage; only crappy opamps are.


----------



## grawk

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Grawk - I wouldn't normalise. Keep your peak level to about -3dB otherwise you could end up clipping on some DACs.

 Regal - I haven't heard the Pacific Micro. In what way does it sound better?_

 

What I would do is record with typical peaks at about -6, then bring the highest peak to 0, which generally put normal peaks at -3. But it's been several years now.


----------



## Crowbar

Anyone try this ADC? Grimm Audio
 The specs look good at THD+N of -115 dB. Discrete sigma delta. It was made by Putzeys, who previously did the UcD amps.
 Too bad it's DSD format though, I'm against that after I read prof. Hawksford's paper trashing it


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Analog noise level in the 24-bit system is designed to be at a much lower absolute level._

 

I would hesitate to make too much of this. To hear the noise difference between 16 bits and 24 bits requires absurd gain levels and very very quiet source material, in Meyer and Moran's study it was _unpleasantly loud_ - 14db above normal listening levels of 99db through speakers before anyone could detect the difference in noise 


 cracking to me makes me think there may be some fundamental problem with the driver/card but the poster has not given us anything really to go on.

 Also we are talking about a unidentified sound card, application and driver, I would like to have more details before making too many judgments.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Too bad it's DSD format though, I'm against that after I read prof. Hawksford's paper trashing it_

 

Have a read of Blech and Yang's 2004 paper. They did blind listening tests between DSD and High-bitrate PCM. 97.24% (from 145 trials) of (110) subjects could not discern a difference between the two formats with the same music tracks.

 Hawksford's gripe against DSD (like Bob Stuart of Meridian, another ARA member) is based on a particular world view that is highly theoretical and not related to empirical listening tests. i.e he can prove theoretically why DSD has to be at 3.07mhz minimum sampling rate in the same way that Julian Dunn proved that jitter is audible at 20ps. In real world testing both these contentions are harder to support.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Have a read of Blech and Yang's 2004 paper. They did blind listening tests between DSD and High-bitrate PCM. 97.24% (from 145 trials) of (110) subjects could not discern a difference between the two formats with the same music tracks._

 

We can't say whether the liming factor was other equipment. My biggest criticism of current audio hardware is the speakers/headphones, since they have enormous distortion compared to most electronics. It's what most needs improvement (besides room acoustics etc.) With spare time in between other project, I've been working on glow discharge drivers for that reason over the past two years (derivative of Dr Hill's Plasmatronic speakers), as distortion can be made arbitrarily lower by decreasing efficiency. I have built a working prototype but this is a long term project, at least a year left for polished design. I'm very interested in seeing whether some distortions from electronics become more audible when masking effects of the huge driver distortion is ameliorated, and I'm interested in comments on blind testing methodology design for maximizing test sensitivity. It's unfortunate that most manufacturers of high end audio are not interested because they'll lose out on stuff that obviously will fail any blind test (silver vs copper conductors, shakhti stones, cable stands, and most other things discussed on AudioAsylum). The irony of course is that it also impedes testing that has potential to provide firm evidence for other items that are controversial in terms of audibility, such as thermal memory distortion (one of the types of distortion that THD testing doesn't reveal), or consider the very detailed capacitor and cable measurements and simulations at Sign in to BT Yahoo! Online (note that the author, Cyril Bates, used to be one of the big skeptics) that provide convincing engineering evidence that these deserve a closer look (or listen, rather).


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_We can't say whether the liming factor was other equipment. My biggest criticism of current audio hardware is the speakers/headphones, since they have enormous distortion compared to most electronics._

 

That's absolutely true. And what's even more important than distortion is frequency response imbalances. That's what makes sound "harsh" or "muffled", "boomy" or "thin".

 See ya
 Steve


----------



## Crowbar

About DSD vs PCM, I remembered about this:
diyAudio Forums - Do all audio amplifiers really sound the same???

 The papers referenced are provided in another post:
diyAudio Forums - Do all audio amplifiers really sound the same???

 Actually, that whole thread is very interesting with the two sides duking it out; worth reading.


----------



## Crowbar

Er, head-fi mangled the link to Cyril Bateman's cap&cable measurements in my post #526; I just fixed it.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_About DSD vs PCM, I remembered about this:
diyAudio Forums - Do all audio amplifiers really sound the same???_

 

Masters and Clark did an interesting set of tests in the 1980s on Amplifiers - called "Do All Amplifiers sound the same" - with the amps they tested ranging from $12,000 monoblock boutique amps to $200 bog-standard receivers. 

 When driven below clipping with non problematic speaker loads at the same output levels it was extremely difficult to pick out amps despite wildly varying prices. 

http://www.bruce.coppola.name/audio/Amp_Sound.pdf

 Bear in mind Masters and Clark worked to a > 50% standard, these days nobody would use less than > 75% for ABX testing.

  Quote:


 The papers referenced are provided in another post:
diyAudio Forums - Do all audio amplifiers really sound the same??? 
 

One of which is The Blech and Yang paper I mentioned. The interesting one is in German so I cannot read it.


----------



## gregorio

Crowbar - I'm not going to keep repeating what's already been written in this thread. The whole point of noise shaping is that it takes the energy away from the area where our ears are most sensitive to areas where our ears are very insensitive. Therefore giving a perceived dynamic range of about 120dB with 16bit. Yes, the same amount of noise is still there but you can't hear it! For the last time, 120dB is way beyond what is required in practice. You have quoted an exceptional amp with a 120dB range, great but that's just one bit of kit in a whole chain. What about the noise from the analogue section of the DAC, what about the cabling between DAC, amp and speakers, what about the speakers themselves and what about the listening environment. 120dB range is absolutely pointless, no one will ever release a recording with a 120dB range and no one would ever be able to listen to it. Read the thread Crowbar and try to understand it.

 Nocturnal - Something weird is going on with your system. Crackling is definitely not a consequence of 24bit vs 16bit, although it could happen with certain material if truncation instead of dithering was used to get down to 16bit from a 24bit original. There are a couple of other possibilities: Crackling in digital audio is usually the most obvious symptom of serious jitter. It could also be that as the audio is taken down to 16bit it is normalised, which could cause clipping, which can manifest as crackling.

 Grawk - What would probably be better, is to add a little compression or limiting and have no peaks in your finished mix (16bit) above -3dB. Any audio (even the odd transient) above -3dB could cause very obvious digital distortion on some DACs. Normalising is generally a process to be avoided in digital audio. You should also aim to record your maximum peaks (transients) probably no higher than -12dB, you don't have to worry about maximising your signals, that's the whole point of 24bit recording! Having peaks at -3dB (or worst still, normalised) means that probably the 8 or 10 LSBs are just noise and you are going to get more noise because your pre-amps will have to be working that much harder.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Read the thread Crowbar and try to understand it._

 

You're the one that doesn't understand by letting your feelings and hand-waiving arguments override rigorous scientific thinking.


----------



## kipman725

If available I go for the 24bit/96KHz recordings as they are the maximum quality that my soundcard will playback and disk space is cheap. Also the mere release of >16bit recordings shows attention to sound quality which is more important than the actual 24bit recordings.


----------



## bigshot

One of the best recordings I've ever heard (Pentatone's Stravinsky Chamber works by Jaarvi) was a DSD SACD hybrid. I put my SACD player in the closet because it sounds as good off the redbook layer as it does on the SACD layer.

 See ya
 Steve


----------



## gregorio

Quote:


  Originally Posted by *kipman725* /img/forum/go_quote.gif 
_Also the mere release of >16bit recordings shows attention to sound quality which is more important than the actual 24bit recordings._

 

Not really, just about all recordings for the last ten years have been made at 24bit, some of them are great, some are garbage. It's nothing to do with the bit depth.

 Also, careful with your assertion that 24/96 is the best quality available on your soundcard: We've proved (hopefully) in this thread that 24bit is no better, as far as SQ quality is concerned, than 16bit. Your sound card may be performing better at 96k than at 44.1 but that will depend on your individual DAC, sampling frequencies are independent of bit depth.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_We've proved (hopefully) in this thread that 24bit is no better, as far as SQ quality is concerned, than 16bit._

 

Nothing of the sort.


----------



## grawk

ok, it's proven excluding fairyland.


----------



## Crowbar

Rather, it's only proven in your fairyland. None of the arguments presented from your side have held up. Yet you persist in your belief. Well, that's how absolute faith is--it's the same with religious people. You're no different.


----------



## grawk

How is it faith that music doesn't require more than 96db to reproduce faithfully, so the extra bits in playback aren't helpful?


----------



## bigshot

One of my favorite Mark Twain quotes is, "It's better to keep one's mouth shut and risk being thought a fool than to open it and remove all doubt."

 See ya
 Steve


----------



## gregorio

Crowbar - If you go through this thread there are links to scientific studies, published uncontested papers and respected magazine articles. Not to mention the Nyquist/Shannon theorum upon which modern digital audio is based. All of which prove that a digital audio recording can be reconstructed back into an analogue waveform with perfect linearity at any bit depth. It's just the noise floor which varies at different bit depths and the noise floor of 16bit (using modern dithering) gives us a dynamic range which exceeds the dynamic range which can be reproduced in a listening environment and greatly exceeds the dynamic range which the human ear can comfortably listen to. What part of this don't you understand or which of the world renown scientists do you disagree with?


----------



## 12Bass

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_All of which prove that a digital audio recording can be reconstructed back into an analogue waveform with perfect linearity at any bit depth._

 

In theory there is no difference between theory and practice. In practice there is. - Yogi Berra


----------



## nick_charles

Quote:


  Originally Posted by *12Bass* /img/forum/go_quote.gif 
_In theory there is no difference between theory and practice. In practice there is. - Yogi Berra 
	

	
	
		
		

		
		
	


	


_

 

_There is nothing so practical as a good theory_ - Kurt Lewin

_He who loves practice without theory is like the sailor who boards ship without a rudder and compass and never knows where he may cast._ - Leonardo Da Vinci

 Having said that Theory makes jitter audible at 20ps and DSD unlistenable due to high frequency spuriae


----------



## Nocturnal310

all words.... Audio cant be read .

 i am completely untechnical person in this field..

 can u pass me a link to 2 similar samples... one at 16 & other at 24 bit.??


----------



## nick_charles

Quote:


  Originally Posted by *Nocturnal310* /img/forum/go_quote.gif 
_all words.... Audio cant be read .

 i am completely untechnical person in this field..

 can u pass me a link to 2 similar samples... one at 16 & other at 24 bit.??_

 

No, but take a 24 bit sample and downsample it to 16 bits with dither and you should have a suitable point of comparison. There are many links to high res and redbook files on the web. All that I have found so far have had small but significant differences between the redbook and high res samples that make comparisons unreliable...also I have not found any decent length samples that are different only in bit-depth.

 Audacity which is freeware allows you to do this. Just load up a 24 bit sample, in the preferences (under the edit menu) make sure the output for wav is set to 16 bit pcm and dither is set (I use triangle), in the Track Pop down menu *next to the filename *choose set sample format, select 16 bit then export as wav.

Audacity: Documentation

 Bingo !


----------



## bigshot

Why do you think there are differences between the samples you find on the web? Is someone trying to stack the deck?

 See ya
 Steve


----------



## nick_charles

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Why do you think there are differences between the samples you find on the web? Is someone trying to stack the deck?

 See ya
 Steve_

 

That is a leading question if ever I saw one. 

 No, I do not think that at all. I think that the recordings are just not quite identical, generally the differences are small but in one one case (The Dragon Boats example) there is a distinct spike on the start of one sample but not the other. In some cases the track lengths are not identical and/or the alignment is not identical.

 If you are going to argue that x is superior to y because of independent variable z then you have to make sure that all other variables are the same.

 When I have downsampled the 24 bit variants to 16 bits I have found no audible difference but that is just me and is not to say that others might not find differences...as it were. For instance in the Blech and Yang study 4 people did detect an audible difference between PCM and DSD, okay it was 4/110 but the result still stands.


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_If you are going to argue that x is superior to y because of independent variable z then you have to make sure that all other variables are the same._

 

Absolutely, that is the problem with much of the misconception about digital audio. People listen to a 24bit file and it may sound better than 16bit so they conclude that 24bit is better. There are so many complex processes which take place in converting to and from digital that a true comparison between the two bit depths is not always as easy or obvious as it may at first appear. Ignorance of this fact is why we often see two files at different bit depths which may sound so different. The use of a good noise-shaped dither, as opposed to truncation, is a typical case in point.

 It's a bit like driver road testing a Ford and then a Ferrari and noticing that the Ford handles better than the Ferrari. The driver then concludes that Fords handle better than Ferraris. 

 However, the Ford was driven in the dry and the Ferrari was in the wet and an inspection of the Ferrari reveals tyres which are completely shot. We all know enough about driving to realise the falsehood of the conclusion that Fords handle better than Ferraris and to understand the difference that tyres and weather conditions make. The more accurate conclusion would be that in general a Ferrari will handle better than an ford but depending on certain conditions, a Ford may handle better than a Ferrari. Same is true of 24bit vs 16bit. All else being equal there is no perceivable difference between the two but there are conditions under which 24bit can sound better than 16bit and indeed vice versa.

 Bare in mind that the quality of the recording studio's ADCs (and other equipment) and the ability of the Recording engineer, Producer and Mastering Engineer are each going to have way more influence on the audio quality than the difference between 16 and 24bit. So in theory and in practice 16bit and 24bit are indistinguishable, provided all other variable are equal! The reason I've used the term "in Theory" is because in practice it can be very difficult to make all the other variables equal.

 The whole argument really is a non issue. It's a bit like with our comparison of the performance of the Ford vs the Ferrari and then getting seriously hung up over the colour of the gear shift lever. In fact this isn't a great analogy because we can perceive the difference between the two differently coloured gear shifts but we can't perceive the difference between 24 and 16bit!!


----------



## regal

Gregorio, I agree fully with this last post.

 Understand that there are many that actually think a digital cable from their transport to DAC is carrying little 1's and 0's ! 

 There is a huge education gap to be overcome in explaining digital audio to people. 

 All digital audio debates come down to one overriding factor that is more important than any of our playback equipment: *mixing and mastering of the material. * 

 A good recording sounds better from a iPod source than a bad one on a $20K TEAC.


----------



## Solan

Nyquistâ€“Shannon sampling theorem - Wikipedia, the free encyclopedia

_The conclusion, that perfect reconstruction is possible, is mathematically correct for the model, but only an approximation for actual signals and actual sampling techniques._


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Not to mention the Nyquist/Shannon theorum upon which modern digital audio is based._

 

Having formally studied the mathematics of sampling theory, I know exactly what this is. And it says NOTHING about bit depth. The Shannon sampling theorem has to do with sampling frequency, and the mathematics assumed that individual samples are infinite precision real numbers. Quantization of samples introduces a further limit on reproduction.

  Quote:


 It's just the noise floor which varies at different bit depths and the noise floor of 16bit (using modern dithering) gives us a dynamic range which exceeds the dynamic range which can be reproduced in a listening environment 
 

Again you're failing to distinguish between quantization noise and the analog noise of the electronics. Completely different.

  Quote:


 and greatly exceeds the dynamic range which the human ear can comfortably listen to. What part of this don't you understand or which of the world renown scientists do you disagree with? 
 

I don't disagree with renowned scientists. You're misunderstanding, misrepresenting, and misapplying the theory. You keep referring to various sources and fail to address the clear argument I've presented that quantization and dynamic range are not the same thing. Dynamic range is simply the highest signal divided by the lowest signal. The highest signal is the voltage output for the largest digital word. The lowest signal in digital is zero, however this is not achievable due to the analog noise floor. Thus the dynamic range is simply the SNR of the analog outputs. The bit depth on the other hand determines how finely you subdivide this range.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_There is nothing so practical as a good theory - Kurt Lewin

He who loves practice without theory is like the sailor who boards ship without a rudder and compass and never knows where he may cast. - Leonardo Da Vinci_

 

And the one to end them all:
_A witty saying proves nothing. --Voltaire_


----------



## Crowbar

By the way, I don't put much stock in the listening tests mentioned here. I agree that the recording and mastering differences account for most of the audible difference. My argument is that current technology can be used to record and reproduce audio at bit depth greater than 16 bit that, when requantized to 16 bit, will sound different to a trained ear.


----------



## Solan

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Having formally studied the mathematics of sampling theory, I know exactly what this is. And it says NOTHING about bit depth. The Shannon sampling theorem has to do with sampling frequency, and the mathematics assumed that individual samples are infinite precision real numbers. Quantization of samples introduces a further limit on reproduction._

 

I am not a student of sampling theory myself. Only a humble associate prof of mathematics at a minor college. But your point addresses the essential issue as far as the theory of vinyl vs CD vs SACD goes, and proves (yes, proves) that we cannot dismiss the need for comparing by listening tests on purely theoretical grounds. 

 There is a point at which digital resolution is better than even the best resolution of the best human ear, but listening evidence says CDs are not near that point. I don't know about SACD vs vinyl, though, since I have never listened to SACD. But SACD is a dying format, as is the CD. What CD has over vinyl is not sound quality, but merely convenience. Same goes for SACD, though the sound quality may be comparable to vinyl. But more convenient than scratchable, little plastic disks are files. More and more stores are selling lossless downloads, which are equivalent to CDs, and some (like Linn Records) also offer 24bit/96kHz (I've also seen 192kHz on some eclectic sites) eating into SACD territory. So drop the CD/SACD purchase plans and go for a high-end DAC instead, folks!

 But there is a special magic to vinyl, even if some digital format should prove to have better sound quality. It is the spinning! The look of a spinning disk is soothing by itself --especially my old Tangerine Dream Live at Poland picture LP-- and induces a state of relaxation that opens up the blood vessels and ear canals, thereby improving audio quality. Bass sounds cleaner, deeper and more controlled on vinyl, too, as long as you look at the spinning LP. A small plastic disk whizzing in a closed cabinet just doesn't do the same.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_ My argument is that current technology can be used to record and reproduce audio at bit depth greater than 16 bit that, when requantized to 16 bit, will sound different to a trained ear._

 

Then show us some real evidence for this, do your own controlled listening tests or find some strictly controlled tests that support your assertion. So far the evidence using current (as actually found commercially) technology and controlled listening tests suggests that apart from noise level detected under extreme conditions there is scant evidence for perception of a quality difference between 16 and 24 bits.


----------



## nick_charles

...


----------



## Solan

Nick, I see you regard me as a troll for bringing in vinyl. If it offends anyone that I brought up vinyl, my apologies, and to my defense I must admit that I confused this thread with another one on CD vs SACD vs Vinyl that I was reading almost at the same time:

SACD vs CD vs VINYL..

 Aside from that, I'll dislike Nick for a whole week or so for calling me a troll. We Norwegians are very ethnically sensitive to such charges (especially from Swedes and Brits), since trolls are kind of a national anti-symbol.


----------



## nick_charles

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_Nick, I see you regard me as a troll for bringing in vinyl. If it offends anyone that I brought up vinyl, my apologies, and to my defense I must admit that I confused this thread with another one on CD vs SACD vs Vinyl that I was reading almost at the same time:
 ._

 

I was not offended, but this thread has already taken several sideturns and the digital vs vinyl one tends to get **very** heated, not that this thread has not already been quite animated...

  Quote:


 Aside from that, I'll dislike Nick for a whole week or so for calling me a troll. We Norwegians are very ethnically sensitive to such charges (especially from Swedes and Brits), since trolls are kind of a national anti-symbol. 
 

My mistake I am really sorry, I did not know that troll was used an ethnic slur with regard to Norwegians, I have never heard anyone use it that way, I sincerely apologise for that.

 Maybe I should have used this....


----------



## bigshot

The answer about which is better, vinyl or digital is very simple to solve. Just get a good capture card and make a CD of an LP. I've done this. The CD capture of the LP is totally indistinguishable from the original LP.

 Both LP and CD are capable of great sound. The differences are always a function of pressing and mastering quality.

 See ya
 Steve


----------



## Solan

No prob, Nick. So let's get back to Nyquist/Shannon. The theorem covers the case where your sample amplitudes are _exact_ and not mere n-bit approximations. With approximation, you will not be able to cover all frequencies as exactly.

 Just curious: Doesn't the theorem also state that within the time interval, the sound wave must be an unaltered combination of sine waves? No variation of amplitudes or combinations? Mathematically, we can of course approximate any sound wave within a time interval by an infinite series, but note that this series is infinite, and that even a normal sound wave requires sine waves past 20kHz to have its audible portion accurately represented that way.

 So ... back or further on to audibility. There have been no real blind tests, have there? Would it be possible to make one here at Head-fi? Say someone could make two music files that would be indistinguishable by size and easy markers, but with the same sound recorded, and in true and false 24-bit depth and perhaps with true and false 96kHz. Then spread those to head-fi'ers willing to test if their ears and gear are up to the challenge?


----------



## CyberTheo

Simple experiment? Use a good capture card to make a CD of a CD and a CD of an LP. Compare the two reproduced CDs. It may not be as simple as that. I bet you'll still hear difference because the original CD and vinyl are not be mastered the same. Indeed, "The differences are always a function of pressing and mastering quality."; I totally agree.


----------



## nick_charles

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_No prob, Nick. So let's get back to Nyquist/Shannon. The theorem covers the case where your sample amplitudes are exact and not mere n-bit approximations. With approximation, you will not be able to cover all frequencies as exactly._

 

The bit-ness gives you uncertainty over amplitude not frequency, but 16 bits is a range of 65536 values , 24 bits gives you more _resolution_ but is this difference humanly detectable ?

  Quote:


 Just curious: Doesn't the theorem also state that within the time interval, the sound wave must be an unaltered combination of sine waves? No variation of amplitudes or combinations? Mathematically, we can of course approximate any sound wave within a time interval by an infinite series, but note that this series is infinite, and that even a normal sound wave requires sine waves past 20kHz to have its audible portion accurately represented that way. 
 

No sampling system has infinite bandwidth so it is always an approximation, but in empirical testing dating back to the 1970s low-pass filters at 20, 18 and even 16K have been used with no audible consequences.

  Quote:


 So ... back or further on to audibility. There have been no real blind tests, have there? Would it be possible to make one here at Head-fi? Say someone could make two music files that would be indistinguishable by size and easy markers, but with the same sound recorded, and in true and false 24-bit depth and perhaps with true and false 96kHz. Then spread those to head-fi'ers willing to test if their ears and gear are up to the challenge? 
 

A few people here have done 16 bit vs 24 bit blind tests, I have done a few using FooBar's ABX plug-in which does not let you use cheats like file size.

 I have downconverted 24 bit files to 16 bits successfully and tested them that way.

 However to get two different format files with the same size you then have to upconvert one of them which adds another variable, also if you downconvert a 96K file to 44.1K it is quite obvious from the spectrograms which is which as the 96K file has a boatload of energy above 22K.

 Having said that it doesnt really matter how big the files are as long as you have a decent ABX comparator that shields you from that knowledge.

 As for blind tests done "Professionally/Academically" there have been a few but they have not separated bit-depth from sampling rate.

 Meyer and Moran SACD/DVD-A vs 16/44.1
 Blech and Yang PCM vs DSD


----------



## Solan

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_The bit-ness gives you uncertainty over amplitude not frequency, but 16 bits is a range of 65536 values , 24 bits gives you more resolution but is this difference humanly detectable ?_

 

Well, this is a theoretical point, and may not be useful, but the number of frequencies below 20kHz is infinite. It requires infinite information to pinpoint a frequency exactly. With less than infinite information, you get an approximation, and how good that approximation is, is limited on the amount of information.

 This is for one single frequency. And if that one single frequency was all there was to music, then it would be easily covered with fewer bits than 16 if all you wanted to was to fool the human ear. But music is a changing composition of many frequencies, making the amount of information ... vast.

 But as I said: this was a mere theoretical clarification. 

 It might be that 24bit/96kHz and such is needed by sound engineers only, to avoid aliasing when mixing. I would love to be in a position to test all this for myself. I would in particular be interested in non-audible sensations of sound, like feeling of space or its opposite, listening fatigue ... longer-term effects that you may not pin-point just by listening for differences over a short time interval. Why? Because that is my primary experience further down the quality spectrum: poorer quality isn't necessarily always heard as such right away, but is rather felt in terms of lacking space or increasing fatigue with listening and so on.


----------



## bigshot

Listening fatigue has to do with frequency response imbalances, not resolution- especially not resolution that falls beyond the threshold of perception.

 See ya
 Steve


----------



## ADD

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Listening fatigue has to do with frequency response imbalances, not resolution- especially not resolution that falls beyond the threshold of perception.

 See ya
 Steve_

 

Agreed, but I would add accurate phase coherence to that as well.


----------



## 12Bass

While I think that brickwall filtration artifacts may be a greater contributor to errors in digital sound reproduction, perhaps it is worth noting that most of the early generations of CD players were not able to resolve full 16-bit resolution - many were 14-bit or worse, with significant problems in low-level linearity.


----------



## gregorio

12Bass - The linearity of 14bit should have been identical to that of 16bit, although with just a little more noise. In practice ADCs and DACs have improved significantly since the early '80s and the sound today is far better. Brickwall filters are now mainly implemented in the digital domain and result in a lot fewer artefacts.

 A similar situation still exists today though. Have a look at the specs of your 24bit DAC, what is the dynamic range (or where is the noise floor)? I can pretty much guarantee that it's 120dB or less. 24bit is 144dB, so your DAC is not in fact a 24bit converter but at best a 20bit converter and probably only an 18bit converter. As far as I'm aware, there are no true 24bit DACs on the market!!

  Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_With less than infinite information, you get an approximation, and how good that approximation is, is limited on the amount of information._

 

That's not how digital audio works. With less information you get more quantisation errors. Properly dithered, all quantisation errors are converted to uncorrelated (white) noise. So you get exactly the same accuracy (100%, perfectly linear) at any bit depth but the fewer bits you use the more noise is going to be added to your perfectly linear reproduction. In other words, 24bit audio has the same perfect resolution as 1bit audio ... don't believe me? Listen to an SACD and compare it to a 24bit audio file. The 24bit audio file has roughly 16,000,000 times the (so called) "resolution" than the 1bit audio used on all SACDs, does your 24bit file sound 16,000,000 times better than an SACD?!


----------



## bigshot

Quote:


  Originally Posted by *ADD* /img/forum/go_quote.gif 
_Agreed, but I would add accurate phase coherence to that as well._

 

That's not nearly as common of a problem as frequency spikes. But I suppose with headphones, that might be annoying over time.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *ADD* /img/forum/go_quote.gif 
_Agreed, but I would add accurate phase coherence to that as well._

 

That's not nearly as common of a problem as frequency spikes. But I suppose with headphones, that might be annoying over time.

 See ya
 Steve

 Edit: I just realized that this is an element of overly compressed lossy audio that can be annoying. Maybe it isn't that rare.


----------



## coolkwc

24bit VS 16bit differences? The number is difference..lol


----------



## Solan

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_That's not how digital audio works. With less information you get more quantisation errors. Properly dithered, all quantisation errors are converted to uncorrelated (white) noise. So you get exactly the same accuracy (100%, perfectly linear) at any bit depth but the fewer bits you use the more noise is going to be added to your perfectly linear reproduction._

 

If you use 16bit, there is _no difference_ in the digital representation of a sine wave of 1000Hz and one of 1000.000001Hz over a whole second! If you use 24bit, their representation will be different, and you will have to go down to a smaller difference for the sine waves to be equal over a whole second. For shorter time intervals, you can of course get the same digital representation with a smaller frequency difference.

 I doubt any DACs would sample the whole CD before sending out analog signal, btw. So the practical frequency accuracy is no better than how long sequences the DAC uses when it converts digital to analog. So identical signals over a given time interval will give identical output, meaning that your recorded 1000.000001Hz signal will come out as 1000Hz ... and you'll probably have frequences a lot further from 1kHz ending up as 1kHz in the final stage. And that is just simple sine waves.

 To me, that lends credence to the claim that harmonics are not accurately reproduced by digital - not accurately _enough_, to be more precise, if the bits are too low.

 But as I said before, it must be tested by ear, and for all effects, even "inaudible" ones.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_In other words, 24bit audio has the same perfect resolution as 1bit audio ... don't believe me? Listen to an SACD and compare it to a 24bit audio file. The 24bit audio file has roughly 16,000,000 times the (so called) "resolution" than the 1bit audio used on all SACDs, does your 24bit file sound 16,000,000 times better than an SACD?!_

 

That's apples and oranges, since the conversion process is different.

Delta-sigma modulation - Wikipedia, the free encyclopedia


----------



## regal

Gregario,

 Consider this plausible scenario:

 Music was recorded 24 bit and compressed so that only the top 6 MSB's are used. Still a 24 bit file.

 The engineer takes this file and dithers to 16 bit. Now you have what roughly 3 bits of resolution?

 I think this is what is happening in the studios now days and why the native bit depth is preferred.


----------



## grawk

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_Gregario,

 Consider this plausible scenario:

 Music was recorded 24 bit and compressed so that only the top 6 MSB's are used. Still a 24 bit file.

 The engineer takes this file and dithers to 16 bit. Now you have what roughly 3 bits of resolution?

 I think this is what is happening in the studios now days and why the native bit depth is preferred._

 

Take a 24 bit file with 6 MSB. Dither to 16, you still have 6 MSB.

 As for Solan's example, you're talking about sample rate, not bit depth. The bits represent loudness, the sample rate is how many times a second the waveform is recorded.


----------



## Solan

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_As for Solan's example, you're talking about sample rate, not bit depth. The bits represent loudness, the sample rate is how many times a second the waveform is recorded._

 

Sorry I forgot to state that my example presupposed 48kHz sampling rate.


----------



## grawk

If you change the bit depth and not the sample rate, you'll have exactly as many samples of the waveform in both 16 and 24 bits, so your example isn't following what's really happening.


----------



## Solan

A 24-bit sample @ 48kHz can tell the difference between the two freqiencies, whereas a 16-bit sample @ 48 kHz can't. It's a bit like graph paper, where the horizontal axis is time and the vertical axis is sound pressure. In digital sampling, the curve is made jagged: it follows the boxes of the graph paper. The sampling rate determines the width of the boxes and the bits determine their height. Finer dimensions in either dimension will help tell different signals apart.


----------



## nick_charles

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_If you use 16bit, there is no difference in the digital representation of a sine wave of 1000Hz and one of 1000.000001Hz over a whole second! If you use 24bit, their representation will be different, and you will have to go down to a smaller difference for the sine waves to be equal over a whole second. For shorter time intervals, you can of course get the same digital representation with a smaller frequency difference._

 

This seems counterintuitive, but interesting, do you have a reference for this ?

 Also even if there is a difference between 16 and 24 bits in terms of rendering the difference between 1000hz and 1000.000001hz it is so far below the human capacity (.5% at 1000 hz (Roederer, 1973) to tell the difference that it really is pretty moot. Even with rapid changes the sensitivity only goes up to 0.016% bst case, way below your example.


----------



## gregorio

Solan - you're falling into the old trap. The graphical representation of digital audio is completely irrelevant, it's how the DAC decodes that digital representation that matters. Digital audio is always represented as a stepped output, the more steps the closer to an original waveform it will visually appear. However, what comes out of your DAC is not stepped, it's linear, at any bit depth. Your DAC has decoded the digital data. If you get a zip file of a book, do you judge the book by examining the contents of this zip file? ... No, what's in the zip file is irrelevant and bares no resemblance to what the book will actually look like once it's extracted back out (un-zipped) to the original. Digital audio is just an encoding system, similar in some ways to zip.

 To be honest I've never tried comparing a 1k sine wave with a 1.000001k sine wave. This fine difference is again beyond the human ear to differentiate (as Nick states). Bottom line is that if 24bit can represent the difference, so can 16bit, reproduction from your DAC should be 100% linear, no exceptions! BTW, there is no way an analogue system can differentiate this degree of pitch change, due to the constant maintenance and re-alignment which is required. If fact sometimes a pitch accuracy of a semi-tone was quite good going between different playback devices!

 A good way of thinking about digital audio is to consider that bit depth encodes the amplitude (volume) and sample rate encodes the frequency (pitch). This is pretty much what Grawk was saying.

 Grawk was also correct when replying to Regal. The difference between 16bit and 24bit is all in the eight LSBs. More bits = lower noise floor.


----------



## regal

The issue with the steps being referred to is aliasing. What comes out of the DAC at 20Khz (44.1hz sampled rate) is these larger "steps" result in high frequency distortion (supersonic) which must be dealt with. So this is a valid argument IMO.


----------



## gregorio

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_The issue with the steps being referred to is aliasing. What comes out of the DAC at 20Khz (44.1hz sampled rate) is these larger "steps" result in high frequency distortion (supersonic) which must be dealt with. So this is a valid argument IMO._

 

Don't know where you got this from Regal?! It's not a valid argument because it doesn't really make sense. Bit depth has nothing directly to do with aliasing. With digital audio you do get alias images but this is related to the sampling rate not the bit depth. Even at 44.1kFs/s these alias images are supersonic (>22kHz) and beyond human hearing anyway, not to mention that these alias images are completely removed (filtered out) during the initial A to D conversion process. What comes out of your DAC at 20kHz should not be distorted, if it is you need to get your DAC fixed! This thread is about bit depth rather than sample frequency though.

 Solan - "That's apples and oranges, since the conversion process is different." Actually it's not that different. Modern studio PCM ADCs sample pretty much the same way as DSD ADCs, the difference is this result is then decimated back to a standard PCM sample frequency. Your link to delta-sigma modulation is interesting but irrelevant, as AFAIK, delta-sigma modulation is employed in the same way in PCM converters as in DSD converters.


----------



## regal

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Don't know where you got this from Regal?! It's not a valid argument because it doesn't really make sense. Bit depth has nothing directly to do with aliasing. With digital audio you do get alias images but this is related to the sampling rate not the bit depth. Even at 44.1kFs/s these alias images are supersonic (>22kHz) and beyond human hearing anyway, not to mention that these alias images are completely removed (filtered out) during the initial A to D conversion process. What comes out of your DAC at 20kHz should not be distorted, if it is you need to get your DAC fixed! This thread is about bit depth rather than sample frequency though.

 ._

 


 I agree it has nothing to do with bit depth but I didn't start the discussion about staircasing.

 You are wrong that all DAC's filter aliasing images. Most of the DAC's that are en vogue now do not do any filtering past DAC chip. Even though we can't hear the supersonic distortion it still wrecks havock on any gain or buffer stages in the audio chain and forces distortion down into the audio band.


----------



## bigshot

Can you give me a link to a specs page or test results for a DAC that shows that no filtering is being done and distortion is present in the audio band? I'd be interested in seeing that.

 See ya
 Steve


----------



## Solan

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Solan - you're falling into the old trap._

 

No. I just happen to understand the math, and am trying to explain it, and to explain why for instance 16bit and 48kHz can't distinguish between two frequencies I've given.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_The graphical representation of digital audio is completely irrelevant, it's how the DAC decodes that digital representation that matters. Digital audio is always represented as a stepped output, the more steps the closer to an original waveform it will visually appear. However, what comes out of your DAC is not stepped, it's linear, at any bit depth. Your DAC has decoded the digital data._

 

That sounds like a poor DAC to me. Linear (1. order) interpolation will still leave a jagged graph. You should at least get a 2. order interpolation to smoothe out the edges.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_To be honest I've never tried comparing a 1k sine wave with a 1.000001k sine wave. This fine difference is again beyond the human ear to differentiate (as Nick states)._

 

And I have not claimed otherwise. I am merely straightening out an erroneous statement which (as I've stated) possibly has nothing other than theoretical ramifications.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Bottom line is that if 24bit can represent the difference, so can 16bit, reproduction from your DAC should be 100% linear, no exceptions!_

 

"Linear" meaning exactly what, in this last context?

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_BTW, there is no way an analogue system can differentiate this degree of pitch change, due to the constant maintenance and re-alignment which is required._

 

You are probably right.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_If fact sometimes a pitch accuracy of a semi-tone was quite good going between different playback devices!_

 

Musicians often accept two essentially different tones as the same tone. Think 12-tone vs natural tone scale.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_A good way of thinking about digital audio is to consider that bit depth encodes the amplitude (volume) and sample rate encodes the frequency (pitch). This is pretty much what Grawk was saying._

 

A good first approximation, I agree, but none the less wrong. Insufficient resolution in the time scale will render the system incapable of telling two frequencies apart, but so will insufficient resolution in the bit depth dimension. It's an AND, not an exclusive OR.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Grawk was also correct when replying to Regal. The difference between 16bit and 24bit is all in the eight LSBs. More bits = lower noise floor._

 

So if we try to synthesize our viewpoints here, we could say that the difference between a 1000Hz and a 1000,0001Hz sine wave pvr a second may well be considered noise captured by the noise floor. I can live with that.


----------



## Solan

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_This seems counterintuitive, but interesting, do you have a reference for this ?_

 

I think your best bet is simply to consult a local mathematician. Show him my post and ask him to draw up the graph and explain. Any mathematician should be able to do that with ease.

  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Also even if there is a difference between 16 and 24 bits in terms of rendering the difference between 1000hz and 1000.000001hz it is so far below the human capacity (.5% at 1000 hz (Roederer, 1973) to tell the difference that it really is pretty moot. Even with rapid changes the sensitivity only goes up to 0.016% bst case, way below your example._

 

Unless you believe Ninja lore, that difference is far beyond human perception. If we play the waves side-by-side, they _will_ interfere and create a new wave with an amplitude swinging at a frequency of .000001 Hz, but given that that takes hours to complete a single cycle, we'd have to rely on memory, not perception, to tell it's there.


----------



## grawk

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_So if we try to synthesize our viewpoints here, we could say that the difference between a 1000Hz and a 1000,0001Hz sine wave pvr a second may well be considered noise captured by the noise floor. I can live with that._

 


 You're still missing the key point:

 bit depth has nothing to do with the frequencies recorded, just the volume. The differences between the two sine waves doesn't have anything to do with the noise floor. It's not going to be captured because the resolution isn't high enough, and you wouldn't hear it because your ears aren't that accurate. But if the volume was high enough, then the difference would be above the noise floor.


----------



## bigshot

This is why normal people never are able to glean the basic concepts from threads like this. It starts out all right with basic information being presented. Then folks who can't abide being proven wrong muddy the waters. Eventually, everyone, even the ones who know the score, is off in tangents and the blatant and obvious gets buried under a pile of niddling and irrelevant details. I guess as long as everyone is enjoying themselves, it's OK.

 See ya
 Steve


----------



## Crowbar

Then you should leave this thread. I've proven you wrong a number of times already.


----------



## Solan

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_bit depth has nothing to do with the frequencies recorded, just the volume. The differences between the two sine waves doesn't have anything to do with the noise floor. It's not going to be captured because the resolution isn't high enough, and you wouldn't hear it because your ears aren't that accurate. But if the volume was high enough, then the difference would be above the noise floor._

 

If you plot the difference between the two sine waves, what would you see over the course of a second, grawk? You'd see a sound wave with a very, very low amplitude. Low enough, in fact, that it would not peek past the 16th bit. But it _would_ peek past the 24th. Hence the logical conclusion that 24bits can spot this difference within 1 second whereas 16bits can't. 

 As for volume, you can't get past 0dB here, which is max.

 As for noise floor, that which can't peek past the 16th bit is below the 16bit noise floor.


----------



## Crowbar

Solan, inter-sample peaks can exceed 0 dB. You can't make a digital filter that guarantees with 100% certainty it will never produce such pathological cases. But at least one can decrease the chance, so one should be aware of this issue.


----------



## Solan

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Solan, inter-sample peaks can exceed 0 dB. You can't make a digital filter that guarantees with 100% certainty it will never produce such pathological cases. But at least one can decrease the chance, so one should be aware of this issue._

 

Good point. Well, I guess you could always introduce an ugly clipping at 0dB, but I assume that would hardly be an improvement.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Then you should leave this thread. I've proven you wrong a number of times already._

 

Do you mind if I ask in general terms how old you are?

 Thanks
 Steve


----------



## Crowbar

28 in a week.


----------



## bigshot

Thanks
 Steve


----------



## gregorio

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_You are wrong that all DAC's filter aliasing images. Most of the DAC's that are en vogue now do not do any filtering past DAC chip. Even though we can't hear the supersonic distortion it still wrecks havock on any gain or buffer stages in the audio chain and forces distortion down into the audio band._

 

I would have been wrong, had I said that! What I said was; the alias images are filtered out during the initial A to D conversion process. So that's an ADC I'm talking about, not a DAC. There is no ADC which doesn't filter out the alias images, it's an integral part of the digitisation process in PCM digital audio.

 Solan - "That sounds like a poor DAC to me. Linear (1. order) interpolation will still leave a jagged graph. You should at least get a 2. order interpolation to smoothe out the edges." - Absolutely true, if standard interpolation was the process by which digital audio was reconstructed. Unfortunately for you it isn't and therefore your statement is completely inaccurate. You need to be looking up the process of a dithering quantiser, rather than the process of interpolation. If you're really hung up on interpolation, look at the special case of the Shannon interpolation formula but this still needs to be considered along with the dithering quantiser. The dithering process completely removes the "jagged" edges to give a perfectly linear output, there is no guess work or inaccuracies in the end product. Providing the sampling theorem rules are being adhered to, a sound wave can be *perfectly* recreated from the digital sample, this is a basic tenet of digital audio theory.

 BTW, Musicians don't accept two tones as one, unless you are talking about harmonics or frequency modulated synthesis. The 12 tone scale was used and developed by the Serialists (esp. Shoenberg) and still falls within the framework of the diatonic scale system which musicians have been accustomed to since the C18th.

 "If you plot the difference between the two sine waves, what would you see over the course of a second, grawk? You'd see a sound wave with a very, very low amplitude." - I don't really understand this, what do you mean by "plot the difference", you'd only see a very low amplitude if there was a small difference in the amplitudes of the input sine waves and besides, what relevance does this have to the real world of sound? As other parts of this thread have already explored, if the difference is in the 5 or 6 LSBs of a 24bit audio file there is no way in the real world that it can be reproduced or heard, so it's irrelevant. I agree with your statement about insufficient resolution in the time scale but this is defined by the sample rate, not the bit depth, the two are not related.

 Crowbar - The filters in ADCs or DACs are designed to remove frequencies above the Niquist point, they are not designed to prevent an input signal from clipping. However, some converters include a limiter to help prevent clipping, DigiDesign's 192 converter for example. I have to say Crowbar, I've found BigShot to be one of the most knowledgeable contributors to this thread and I can't think of an occasion when you've proven him wrong but I can think of plenty of occasions when you've been proven wrong. So I don't think your comment to Bigshot was appropriate.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Providing the sampling theorem rules are being adhered to, a sound wave can be *perfectly* recreated from the digital sample, this is a basic tenet of digital audio theory._

 

*One of the assumptions of the sampling theorem is that the samples are accurate! If your samples are quantized, then that assumption is violated.*

 From The Sampling Theorem
  Quote:


 Quite surprisingly, the Sampling Theorem allows us to quantize the *time axis without error* for some signals....In contrast, no one has found a way of performing the *amplitude quantization step without introducing an unrecoverable error*. 
 

The sampling theorem is defined with samples being *infinite precision real numbers*. The mathematics assume continuous, not discrete, amplitude. I have brought this up already and you've failed to address it. There is nothing "perfect" about reproduction with errors!
 Dithering only moves around the quantization error energy, it doesn't make it go away.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_*One of the assumptions of the sampling theorem is that the samples are accurate! If your samples are quantized, then that assumption is violated.*

 From The Sampling Theorem

 The sampling theorem is defined with samples being *infinite precision real numbers*. The mathematics assume continuous, not discrete, amplitude. I have brought this up already and you've failed to address it. There is nothing "perfect" about reproduction with errors!
 Dithering only moves around the quantization error energy, it doesn't make it go away._

 


 And yet despite these theoretical, technical and implementation limitations digital even at humble 16/44.1 provides a remarkably good performance in terms of noise and dynamic range, extremely difficult to distinguish from high resolution formats, perceptually transparent when compared with original source material and with robust error correction.


----------



## Solan

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_And yet despite these theoretical, technical and implementation limitations digital even at humble 16/44.1 provides a remarkably good performance in terms of noise and dynamic range, extremely difficult to distinguish from high resolution formats, perceptually transparent when compared with original source material and with robust error correction._

 

I think we have reached consensus there, so we have proceeded into ultra-audiophile territory where the most miniscule differences lie - differences that maybe only sophisticated equipment can pick up.


----------



## tfarney

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_I think we have reached consensus there, so we have proceeded into ultra-audiophile territory where the most miniscule differences lie - differences that maybe only sophisticated equipment can pick up._

 

My Cairn terrier can hear it. My terrier can hear a fly cut a path through the room and leap in front of it to snap the bugger's very life from mid-air. Good boy, Duke. Wanna try the Senns?

 Tim


----------



## Solan

Post a pic of Duke with headphones, Tim.


----------



## tfarney

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_Post a pic of Duke with headphones, Tim. 
	

	
	
		
		

		
		
	


	


_

 

You'd love Duke. Duke is so doggone lovable, I'm suspicious of anyone who doesn't love him. I should have had Duke when I was young and single. The dude is a chick magnet.

 Tim


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_perceptually transparent when compared with original source material_

 

What do you mean? If by original source material you're referring to the live performance, then that cannot be--no speaker arrangement can reproduce the spatial positioning that you get from the live performance. This would be the first time on this site I come across someone claiming they can't differentiate from being at a live perormance and listening through speakers/headphones. Even if your equipment has zero distortion, you still don't have the right geometrical setup to get correct positional sound.


----------



## Crowbar

Quote:


  Originally Posted by *Solan* /img/forum/go_quote.gif 
_I think we have reached consensus there, so we have proceeded into ultra-audiophile territory where the most miniscule differences lie - differences that maybe only sophisticated equipment can pick up._

 

With sensitivity range of 120 dB and the ability to do both time domain and frequency domain analysis at the same time, the ear is one of the most sophisticated.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_What do you mean? If by original source material you're referring to the live performance, then that cannot be--no speaker arrangement can reproduce the spatial positioning that you get from the live performance. This would be the first time on this site I come across someone claiming they can't differentiate from being at a live perormance and listening through speakers/headphones. Even if your equipment has zero distortion, you still don't have the right geometrical setup to get correct positional sound._

 


 No, I am referring to thing such as A/D/A loops after the output of LP playback , in a different thread a musician member has mentioned how the digital feed via speakers is indistinguishable from the live feed via speakers.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_With sensitivity range of 120 dB and the ability to do both time domain and frequency domain analysis at the same time, the ear is one of the most sophisticated._

 

In other respects the human auditory system is remarkably crude and PCM can easily render differences in frequency and volume that humans cannot detect. Read some of the psychophysics literature and you will see that we are very poor at some auditory tasks.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_the digital feed via speakers is indistinguishable from the live feed via speakers_

 

That just says something we already know--speaker distortion dominates; ergo, we need better speakers.


----------



## Solan

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_That just says something we already know--speaker distortion dominates; ergo, we need better speakers._

 

We need headphones.


----------



## Crowbar

Better ones. Just look at typical distortion plots for headphones at headphone.com


----------



## bigshot

I guess we all agree then... until better headphones and speakers technology arrives, redbook is perfectly capable of reproducing sound faithfully enough for current home audio purposes. If we want to improve the sound quality of our systems, instead of upping bitrates, we should be paying attention to the distortion levels of our speakers and headphones and trying to achieve flatter response through equalization and room treatments.

 See ya
 Steve


----------



## Crowbar

The technology exists. Plasmatronic speakers from near 30 yrs ago with some small modification can do < 0.1% distortion over their frequency range (700 and up), most of which is 2nd harmonic and thus largely inaudible. Nelson Pass has a pair, and a few other people. I'm working on a helium-less derivative with a 400-500 crossover point.
Hill Plasmatronics | Google Groups


----------



## gregorio

Crowbar - The human ear does not have a dynamic range of 120dB. The ear has a safety system to help prevent permanent damage when listening to loud sounds. It's called TTS (Temporary Transient Shift), which essentially reduces sensitivity so that high SPLs don't do too much damage. Depending on the individual, TTS usually starts kicking in around 85dBSPL or so and can reduce sensitivity by as much as 40dB. So although the human ear can hear 120dB it does not have a 120dB range. 120dB is around the point that physical pain will be felt and listening for anything but the briefest moment at this level will cause permanent damage. For this reason no-one will ever put out a commercial recording with anything near a 120dB range. So far in this thread, I don't think anyone has discovered a recording with more than about a 60dB range and I can't see this ever changing by much.

 So, it's not just a case of inventing a sound system which could take full advantage of the CD standard, we would also need to redesign the human ear itself!! I agree that there are many weak links in the playback chain and improving speakers, environment etc., is going to provide a better listening experience but moving up to 24bit for the consumer is and always will be a waste of time (and storage space)!


----------



## Crowbar

Your argument is about to get crushed once again. Starting to see a pattern of this in this thread!
  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It's called TTS (Temporary Transient Shift)_

 

There is no such thing as temporary transient shift. It's temporary *threshold* shift.
  Quote:


 TTS usually starts kicking in around 85dBSPL or so and can reduce sensitivity by as much as 40dB. So although the human ear can hear 120dB it does not have a 120dB range. 
 

The dynamic range compression you describe is not instant. You were confused because you thought it said transient, but lacking reading comprehension is no excuse for propagating nonsense. *The top of the two graphs below shows that a 15 second of exposure to continuous 120 dB induces no theshold shift!*






 BOO-YAH! *blows smoke away from gun barrel*


----------



## Solan

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I agree that there are many weak links in the playback chain and improving speakers, environment etc., is going to provide a better listening experience but moving up to 24bit for the consumer is and always will be a waste of time (and storage space)!_

 

Storage is dirt cheap. I'm not saying that you should store all your albums at 24bit/192kHz, but an album at that file quality is a mere 2GB at worst, which --now that 1TB costs $250-- is a cost of 50 cents. Music not worth 50 cents storing is not worth listening to.


----------



## charonme

Would a larger bit depth result in better reproduction when the sound is quiet and we turn up the volume? (I'm not saying doing this would be wise or practical) Of course that would only be the case if the original recording and the playback equipment had that resolution too. Also, I believe this could be equally achieved using some kind of reply gain information, although CDs don't support it.
 .
 Btw I'm amazed that such a simple question induced a 62 page discussion. Hasn't everything already been said here? It's already hard to search for some useful information in this thread and trolling is increasing. Isn't it time for a lock?


----------



## nick_charles

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_Would a larger bit depth result in better reproduction when the sound is quiet and we turn up the volume? _

 

So far as published controlled listening tests have shown to date, this is the only scenario where you can unequivocably say that there is empirical evidence for the superiority of high res audio in and of itself with some selected recordings. But, this superiority may relate more to noise levels than any underlying qualitative superiority. 

  Quote:


 Btw I'm amazed that such a simple question induced a 62 page discussion. Hasn't everything already been said here? It's already hard to search for some useful information in this thread and trolling is increasing. Isn't it time for a lock? 
	

	
	
		
		

		
		
	


	



 

If you think this is long you should look at some of the old Vinyl vs CD threads


----------



## Febs

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Crowbar - The human ear does not have a dynamic range of 120dB. The ear has a safety system to help prevent permanent damage when listening to loud sounds. It's called TTS (Temporary Transient Shift), which essentially reduces sensitivity so that high SPLs don't do too much damage. Depending on the individual, TTS usually starts kicking in around 85dBSPL or so and can reduce sensitivity by as much as 40dB. So although the human ear can hear 120dB it does not have a 120dB range. 120dB is around the point that physical pain will be felt and listening for anything but the briefest moment at this level will cause permanent damage. For this reason no-one will ever put out a commercial recording with anything near a 120dB range. So far in this thread, I don't think anyone has discovered a recording with more than about a 60dB range and I can't see this ever changing by much._

 

 Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_The dynamic range compression you describe is not instant. You were confused because you thought it said transient, but lacking reading comprehension is no excuse for propagating nonsense. *The top of the two graphs below shows that a 15 second of exposure to continuous 120 dB induces no theshold shift!*_

 

Crowbar, what in those graphs contradicts anything that gregorio wrote? He said:

 -"TTS usually starts kicking in around 85dBSPL": The graphs actually show that with a significant exposure length, TTS can occur at even lower exposure levels than 85dB SPL, but it is clear from the graphs that TTS occurs at 85 dB SPL even with relatively short (less than 2 minute) exposure.

 --"and can reduce sensitivity by as much as 40dB": The graphs show that TTS at 85 DB exposure is 42.5 dB, and TTS is even higher for higher exposure levels.

 --"120dB is around the point that physical pain will be felt and listening for anything but the briefest moment at this level will cause permanent damage." Given the resolution of the graphs, it is difficult to tell whether in fact there is no threshold shift at 15 seconds of exposure, but it appears that the onset of TTS from a 120 dB exposure begins in less than 20 seconds. I suppose you could debate whether 20 seconds of exposure constitutes "the briefest moment," but your chest-thumping seems a bit uncalled for.

 Also, it's worth noting that the graphs that you posted are captioned, "Hypothetical growth and recovery of threshold shift after various single and continuous exposures to noise centred near 4 kHz."

 Edit: gregorio's statement that "120dB is around the point that physical pain will be felt" seems to be supported by another page from the same site that you used for you graphs, which indicates that the threshold of pain is between 115 dB and 140 dB: Threshold_of_Pain


----------



## bigshot

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_Would a larger bit depth result in better reproduction when the sound is quiet and we turn up the volume?_

 

Yes if you like to listen to your music at a volume level near or beyond the threshold of pain. But odds are the music wasn't mixed to have that sort of dynamic range anyway. In the real world, redbook is as good as you could hope for.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Crowbar, what in those graphs contradicts anything that gregorio wrote?_

 

Crowbar is just arguing for the sake of arguing. He's talking purely in theory. Gregorio knows how the technology is applied. That's what really counts.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Crowbar, what in those graphs contradicts anything that gregorio wrote?_

 

It directly contradicts the central statement of his post that is his conclusion of what the effect means for this thread:
  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_So although the human ear can hear 120dB it does not have a 120dB range._

 

The graph shows this is obviously false, since the effect takes a good deal of time to kick in. That means that not only for transients (which is what I've been arguing this far), but for even a 15 second extended 120 dB loudness the ear maintains a 120 dB range! I'm glad the issue of TTS came up for I wouldn't have found this graph that so succinctly confirms what I've been saying and obliterates the opposition.

  Quote:


 Given the resolution of the graphs, it is difficult to tell whether in fact there is no threshold shift at 15 seconds of exposure, but it appears that the onset of TTS from a 120 dB exposure begins in less than 20 seconds. 
 

Clearly you haven't read much of this thread if you think I anywhere suggested we listen to continous 120 dB sound. If you had done your homework, you'd see that I've only argued for capturing transients of that magnitude.

  Quote:


 Also, it's worth noting that the graphs that you posted are captioned, "Hypothetical growth and recovery of threshold shift after various single and continuous exposures to noise centred near 4 kHz." 
 

Note also that 4 kHz is the frequency to which the ear is most sensitive. If the frequencies involved are different, then TTS takes even longer to kick in and the graph's sigmoid curves would be shifted to the right.

 Also a note about the second graph: the recovery times shown are for after you've reached full TTS for the given dB level. Thus what should be immediately obvious is that what matters is not just loudness, but loudness and time--specifically whether you've exceeded a threshold shift of 50 dB or not--if you have reached that much shift on the first graph along one of those curves, then you'll have some permanent shift because no curve on the recovery graph drops more than 50 dB. Note, however, that in order to get this damaging 50 dB shift *you would have to listen to continuous 120 dB for 5 minutes*.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Crowbar is just arguing for the sake of arguing. He's talking purely in theory. Gregorio knows how the technology is applied. That's what really counts.

 See ya
 Steve_

 

Gregorio clearly knows jack about aural physiology, as I've nicely demonstrated. Indeed, I thank him for providing me with that opportunity with his blunder.


----------



## bigshot

This isn't a contest to see who "wins". It's an exchange of information. Continually claiming "victory" adds nothing to the discussion.

 see ya
 Steve


----------



## Crowbar

I do it to indicate a certain level of annoyance of someone whose position is based on ignorance and/or belief rather than science.


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Clearly you haven't read much of this thread if you think I anywhere suggested we listen to continous 120 dB sound. If you had done your homework, you'd see that I've only argued for capturing transients of that magnitude._

 

I've read the entire thread. It's a shame that you apparently cannot engage in the discussion without resorting to gratuitous insults.

 Under what circumstances could one listen to a recording that used a full 120 dB of dynamic range? No real-world listening environment has an ambient noise level of 0 dB. Even an anechoic chamber has an ambient noise level of 10 to 20 dB. If one were to listen to a recording with a full 120 dB dynamic range in that environment, at a volume sufficient to hear the quietest parts of the recording, the loudest transients would be in the range of 130 to 140 dB. That _*is*_ loud enough to risk instantaneous hearing damage. In a more typical listening environment, the transients would be even louder.


----------



## charonme

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_ Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
Would a larger bit depth result in better reproduction when the sound is quiet and we turn up the volume?

 

Yes if you like to listen to your music at a volume level near or beyond the threshold of pain._

 

I'm sorry, I don't see how listening at painful levels relates to my question. Second, even if there were few (or none) such highly dynamic recordings, one can surely be digitally generated


----------



## grawk

Because the only time you'd ever need more than 16 bits is if you're listening to music where there's more than 96db between the quietest and loudest moments.


----------



## gregorio

Thanks for adding that Febs. Crowbar is off on one again, he's already been told about the dynamic range above the noise floor but no, he wants to revolutionise the music industry and deafen (or kill) us all in the process!

 Crowbar - You are not going to find any definitive figures on when TTS cuts in, how long it takes to cut in or at what point permanent damage is going to be caused. You'll find many reports which purport to be definitive but the bottom line is it's all essentially guess work and very rough averages because it varies from person to person. However, one thing is clear, no one should be listening to anything even approaching 120dB, if they don't want to damage their hearing. To suggest anything else is *foolish and dangerous*!!!

 Charonme - what grawk is trying to explain is that if you have a very dynamic recording and have to turn up the volume to hear the quiet passages, when the loud passages come along you're going to damage your hearing. It's entirely possible to make a recording with a large dynamic range but no one in their right mind would create such a recording for two reasons:

 1. Most listeners would find it very annoying, having to keep changing the volume during a song.

 2. If someone forgot to turn down the volume during a loud passage they could easily cause hearing damage and be tempted to sue the record company for making a dangerous product. It's immoral, illegal and just plain bad business to knowingly put a product on the market which is dangerous! The record company wouldn't have a leg to stand on as it's no more difficult to make an equally high quality product which isn't dangerous.

 Grawk - Bare in mind that the theoretical dynamic range of 16bit is 96dB as you stated but in practice with good noise-shaped dither it's actually closer to 120dB. Hence why these last few pages have concentrated on the 120dB figure.


----------



## charonme

I know and understand all that, that's why I explicitly stated "*when the sound is quiet*" in my question.
 .
 Btw I heard classical recordings at a hi-end show where the quiet parts were so quiet that (if I could) I would have to turn up the volume to make something of them but the loud parts were loud enough even at that level.


----------



## nick_charles

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_I know and understand all that, that's why I explicitly stated "*when the sound is quiet*" in my question.
 ._

 

Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback 
 JAES Volume 55 Issue 9 pp. 775-779; September 2007 
 Meyer, E. Brad; Moran, David R.

 This paper describes blind listening tests between High res sources and 16/44.1 bottlenecks. You will have to buy it to read the whole text, $5 for AES members $20 for non members. Various summaries and expansions exist on the internet and it is the one paper that has caused the most heated debate between High Res proponents and High res skeptics. On the other hand it is one of the best papers out there to have tackled this issue. The authors address the issue in a reasonably neutral tone. The nice thing about this paper is the large sample size, they conducted over 500 trials.


----------



## gregorio

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_I know and understand all that, that's why I explicitly stated "*when the sound is quiet*" in my question.
 .
 Btw I heard classical recordings at a hi-end show where the quiet parts were so quiet that (if I could) I would have to turn up the volume to make something of them but the loud parts were loud enough even at that level._

 

I think we understood what you meant. The problem is that if we were to make a product as you suggest, we would be relying on the consumer to turn the volume up in the quiet passages and down again in the loud passages. Not every consumer is going to do this in the right place every time (or potentially damage their sound system, hearing or both) and to be honest most consumers wouldn't want to.

 The recording which you heard at a show is a classic example of making a recording with too large a dynamic range. It certainly sounds like the mastering engineer may have made a bit of a mistake. That sort of dynamic range would still be well within 16bit though.

 Ever seen a feature film which had such a big dynamic range it almost made you soil yourself?! Feature film soundtracks are always 16bit and are encoded on to the film using very lossy data compression that's roughly equivalent to about 80kbps (yes eighty!) per channel. I don't hear too many people complaining about the quality or resolution of film soundtracks, when in reality it's roughly the same quality as listening to a 160kbps MP3 ripped from CD!


----------



## grawk

I realize you can get higher than 96, but even 96 is way more than practical.


----------



## bigshot

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_I'm sorry, I don't see how listening at painful levels relates to my question. Second, even if there were few (or none) such highly dynamic recordings, one can surely be digitally generated_

 

It's been said before, but I'll say it again... In order to create a dynamic range wide enough to require a higher bitrate than redbook, playback would require a volume level high enough to create permanent hearing damage in the listener. Even the most dynamic musical recordings fall into the 55dB range. That is well within the scope of redbook. Yes, you could turn the volume up high enough to hear the difference, but you sure as heck wouldn't want to listen to music that way. Perhaps if you listen to recordings of jet aircraft taking off, you might need higher bitrates.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I do it to indicate a certain level of annoyance of someone whose position is based on ignorance and/or belief rather than science._

 

Please don't feel obligated to ensure that everyone is as annoyed as you are.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback 
 JAES Volume 55 Issue 9 pp. 775-779; September 2007 
 Meyer, E. Brad; Moran, David R._

 

This paper was discussed and criticized at the diyhifi forums. I'm not going to bother repeating here what's been said already. You know how to use a forum search.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_In order to create a dynamic range wide enough to require a higher bitrate than redbook, playback would require a volume level high enough to create permanent hearing damage in the listener._

 

That is false, as the graph I posted shows--unless you listen sufficiently long along an applicable loudness curve--that's continuous--to get 50 dB of shift, there will be no permanent damage (the graph even explicitly states that the region of possible damage is above the 120 dB curve):


----------



## Crowbar

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_Even an anechoic chamber has an ambient noise level of 10 to 20 dB._

 

So? It's been discussed already that the ear can pick up signals several dB below the noise floor. Ambient noise is spread over a wide bandwidth, whereas signal energy is concentrated in specific bands. Tell me, why are you making me repeat something that's already come up? It's no excuse that you didn't read the thread!


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Crowbar - You are not going to find any definitive figures on when TTS cuts in, how long it takes to cut in or at what point permanent damage is going to be caused._

 

They don't need to be definite, only approximate. As long as the graph is within an order of magnitude, it's clear that transients of 120 dB will not cause hearing damage if the average intensity of the sound is much lower. What the graph shows very clearly that it takes continued listening to such power in order to result in permanent phase shift (damage).

  Quote:


 You'll find many reports which purport to be definitive but t
 he bottom line is it's all essentially guess work and very rough averages because it varies from person to person. 
 

So at first you rely on TTS literature to make your case, then when I show what it actually says, you attack it! You should be a politician!

  Quote:


 no one should be listening to anything even approaching 120dB 
 

Don't misrepresent what I've been arguing. No one should listen to continuous 120 dB. But there's nothing wrong with representing 120 dB transients, and transient level can often exceed average intensity by a large amount.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_This paper was discussed and criticized at the diyhifi forums. I'm not going to bother repeating here what's been said already. You know how to use a forum search._

 

Could you humour me and summarise the arguments against the paper and I will reflect on them.

 I trust you have read this paper in full ?

 The paper was bound to solicit criticism as it challenges a fundamental assumption of high res audio proponents, but consider this , they did over 500 trials and in not one case was any human being able to reliably detect a difference between High res and red book at normal to loud listening levels. Now you may critique the paper on some minor methodological grounds but not one person detected a difference.

 Surely if a fundamental audible quality difference exists between High res and red book due to its superior technology one person at least would be expected to detect a difference under controlled conditions ?


----------



## Crowbar

I think a large part of the criticism was equipment used. There were also criticisms about testing methodology. I suggest you find the discussion at either diyhifi.org or diyaudio.com
 From what I remember I was actually the one that started the thread in question, before I got banned at those sites (for unrelated reasons).


----------



## grawk

Let me guess, you got banned for being an obnoxious f-tard who continued arguing based on strawman arguments rather than actually dealing with the real world?


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I think a large part of the criticism was equipment used. _

 

That horse won't run. M&M listed their kit in a later exposition they used three different high res transports Sony, Yamaha and Philips, decently high quality amps and speakers. At least two of the sources had exemplary measurements and one was given the both a subjective huge thumbs up by Stereophile, though that of course means nothing at all but also the Stereophile measurements seal of approval by John Atkinson. All 3 had dynamic ranges of over 100db varying from 103db to 108db. Bear in mind the best dynamic range for any high res source I have ever seen is 112db.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_That is false, as the graph I posted shows--unless you listen sufficiently long along an applicable loudness curve--that's continuous--to get 50 dB of shift, there will be no permanent damage (the graph even explicitly states that the region of possible damage is above the 120 dB curve)_

 

I invite you to take a typical high quality classical digital recording and boost the peaks up to 120dB. Listen to it through your experimental speakers with the super low distortion levels. Even if it doesn't blow the voice coils or make you deaf right away, it will certainly be an unpleasant experience and one that your neighbors will remember for many years to come.

 See ya
 Steve


----------



## Crowbar

Nothing to do with any technical argument; it's for constant spelling/grammar corrections and a personal attack (which was worth it, for the person deserved it).


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Nothing to do with any technical argument; it's for constant spelling/grammar corrections and a personal attack_

 

Congratulations!

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_or make you deaf right away_

 

Why are you insisting on trying to scare people with false information? The very topic of threshold shift that gregorio brought up, when examined shows that transients at 120 dB do not cause any damage, only prolonged exposure to continuous 120 dB does.

  Quote:


 it will certainly be an unpleasant experience and one that your neighbors will remember for many years to come. 
 

An unpleasant experience is a handgun firing. 120 dB is over 100 times less sound power! Typical ear plugs are 20 dB attenuation, thus people in the firing range with hearing protection are exposed continuously to 120 dB transients--one with every shot. Note that's WITH hearing protection!


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_So? It's been discussed already that the ear can pick up signals several dB below the noise floor._

 

Yes, I saw you make that assertion earlier in the thread, but it does not diminish my point. You don't quantify what you mean by "several dB." Suppose the ambient noise level is 40 dB. And suppose further that the ear can pick up signals 5 dB below that. To hear both the loudest and quietest parts of a recording with a 120 dB dynamic range, one would need to turn up the volume to the point that the loudest peaks were 155 dB.

  Quote:


 Tell me, why are you making me repeat something that's already come up? It's no excuse that you didn't read the thread! 
 

Interestingly enough, I re-read much of the thread, and one thing that became immediately clear is that you resort to name-calling whenever someone disagrees with you. 

 In any event, I raise the point because you are making the false assumption that making full use of 120 dB of dynamic range means that the listener would be exposed to a maximum volume level of 120 dB SPL.


----------



## Crowbar

In-ear-canal headphones provide 35 dB isolation. That means in a 40 dB environment, with the headphones you would have about 5 dB noise, and you could detect signals around 2 dB.


----------



## Febs

That assumes that there is no noise at all inside the ear.


----------



## grawk

and with 96db snr, your peaks would be at 98db,a nd youd be deaf


----------



## charonme

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I think we understood what you meant. The problem is that if we were to make a product as you suggest..._

 

I completely agree with everything you said except this: I didn't suggest making such a product, I'm rather against it. I just said it's possible, I even tried it myself out of curiosity (I generated a sine wave spanning across 32bits).

  Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_...In order to create a dynamic range wide enough to require a higher bitrate than redbook, playback would require a volume level high enough to create permanent hearing damage..._

 

Yes, I believe that too. I also believe that 16bits is more than enough for final music products. But these true facts are still unrelated to and outside the context of my original question. In the setup of my question a recording can very well be presented in a 24bit format without any risk of pain, because only quiet sounds are recorded.

 So if I may, I would like to specify in more detail the conditions for your statement: 1.) the 16+ bit recording must contain both quiet and loud parts 2.) the listener must listen to the whole recording, 3.) must listen to everything at a fixed amplification level 4.) must be able to hear the quiet parts in a sufficient detail. 

 Otherwise a multibit recording could be done and played without pain: if 1.) is not met the listener would turn up or down (depending on whether quiet parts or loud parts are missing) the volume sufficiently to hear everything comfortably or if 2.) is not met, he would listen to the quiet parts only, if 3.) is not met he would turn up the volume for quiet parts and down for loud parts or if 4.) is not met, he would turn down the volume, but not hear the quiet parts.


----------



## gregorio

Crowbar - It's becoming apparent that you are only capable of dealing with absolutes. Unfortunately, there are few precise absolutes when it comes to hearing and you seem to be deeply disturbed by this notion, to the point that you start insulting others. Take the example of you patting yourself on the back for supposedly proving me wrong on how long it takes for TTS to kick in and ask yourself; why will TTS be kicking in and what is happening to my hearing before it kicks in? ... Is any of this starting to sink in?

 You are just going to have to accept that there are no precise figures, or rather there are lots of precise figures and reports but they usually do not agree with each other. Therefore: Depending on the person, 120dB could cause severe pain and hearing damage, to others it may be no more than just a slightly uncomfortable loud noise, in a very few cases it could even cause death!


----------



## regal

The whole argument against 24 bit breaks down when you ABX an HDCD encoded track. Just ABX the original flac (non decoded) vs the decoded 20 bit track and play thru a non HDCD 24/96 DAC. HDCD can be decoded to 20 bit with HDCD.EXE (someone cracked it last year.) I got 14 out of 14 with the foobar ABX.


----------



## nick_charles

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_The whole argument against 24 bit breaks down when you ABX an HDCD encoded track. Just ABX the original flac (non decoded) vs the decoded 20 bit track and play thru a non HDCD 24/96 DAC. HDCD can be decoded to 20 bit with HDCD.EXE (someone cracked it last year.) I got 14 out of 14 with the foobar ABX._

 

Interesting.

 Can you explain this a bit more ? Can you break down the exact steps you took so it can be replicated ? I have a few HDCDs and I will have a go at it.

 Why FLAC ? why not WAV format so there is one level less of processing ?

 Also is HDCD really a comparable example as it is a bit of a bodge, only 1 of the "extra" 4 bits is used for DR expansion iirc it isnt true 20 bits is it ?

 Did you verify that the average, min and max energy levels were comparable between the two files ? If one is even 1db (perhaps less) louder in parts it would be easy to tell apart. 

 Anyway if you could tell me how to do this I will have a go at it myself.

 Thanks.


----------



## regal

1. Rip the CD with EAC
 2. Use HDCD.exe to make a 24 bit decoded version 
 3. Level match with Goldwave.
 4. Run ABX with the Foobar.


 Yes HDCD decoded is a genuine 20 bit file per verified with CoolEditPro


----------



## nick_charles

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_1. Rip the CD with EAC
 2. Use HDCD.exe to make a 24 bit decoded version 
 3. Level match with Goldwave.
 4. Run ABX with the Foobar.


 Yes HDCD decoded is a genuine 20 bit file per verified with CoolEditPro_

 

Thanks for ther procedure.

 I am seeing contradictory information on this. I had a look into this and as I see it some say HDCD is 16 bits but encoded in the LSB is a DR compression signal which looks like harmless low level noise to a non HDCD player but is picked up by the HDCD decoder which then adjusts the dynamic range on the 4 least significant bits effectively expanding it, even Pacific talk about "effectively" 20 bits. 
	

	
	
		
		

		
			





 Anyway I will have a go at this myself.


----------



## bigshot

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_Yes, I believe that too. I also believe that 16bits is more than enough for final music products. But these true facts are still unrelated to and outside the context of my original question. In the setup of my question a recording can very well be presented in a 24bit format without any risk of pain, because only quiet sounds are recorded._

 

Dynamic range is the difference between the loudest part and the quietest part. A dynamic range of 96dB means that the quietest sound recorded is 96dB below the loudest. If the loudest peak on your recording is a quiet sound below 20dB, you don't have a 96dB range any more. You have a 20dB dynamic range. Simply normalizing the quiet peak up near the top of redbook would give you the exact same dynamic range with plenty of room to spare.

 Dynamic range doesn't extend up, it extends down. A peak is a peak, and it sounds exactly the same no matter if it's 16 bit or 24 bit. The difference between the two is how far down in volume *below the peak* the sound resolution goes. Is that clearer?

 See ya
 Steve


----------



## charonme

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_1. Rip the CD with EAC
 2. Use HDCD.exe to make a 24 bit decoded version 
 3. Level match with Goldwave.
 4. Run ABX with the Foobar._

 

between 2 and 3 I'm missing a step where you make a 16bit version from the 24bit version with some good dithering and noise shaping

  Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Is that clearer?_

 

Yes, thank you. As I'm new to this, could you please explain something else if it is not completely OT? Imagine you have an analog recording which is a result of a DAC from either 16bit digital data or 24bit data. Is there any way of telling whether it came from 16bits or 24bits (including theoretical ways or sci-fi technology)?


----------



## grawk

sure, if the loudest moment and the quietest moment are less than 96db apart, then it may as well have been 16 bits.


----------



## Solan

It's been interesting to see how the debate has been about theoretical bests and all that, but ...

*BUT*

 What is the difference between 16bit and 24bit in _actual, implemented difference?_


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_Dynamic range is the *difference* between the loudest part and the quietest part._

 

Why are you *lying*?
 Definition of dynamic range: "Dynamic range is a term used frequently in numerous fields to describe the *ratio* between the smallest and largest possible values of a changeable quantity"
 The smallest value occurs when the DAC realizes an output for the digital code word 0. It is the analog noise floor of the equipment. Since that can be made arbitrarily low regardless of bit depth, the fraction giving you dynamic range can be made arbitrarily large regardless of bit depth. As an example, a 1 bit system could be trivially constructed where the analog noise floor is 1 dB @ 1 m when played through a speaker (digital word is 0), and the highest level (digital word 1) is 120 dB. So you have a 1-bit DAC with 120 dB dynamic range. Therefore, bit depth defines the quantization and has nothing to do with the dynamic range.

  Quote:


 A dynamic range of 96dB means that the quietest sound recorded is 96dB below the loudest. 
 

Dynamic range is a ratio.

  Quote:


 The difference between the two is how far down in volume *below the peak* the sound resolution goes. 
 

Following from the above, the bit depth resolution (viz. quantization) determines the step size between two consecutive levels, not how low you can go (the latter being determined by the fraction of the output for the highest digital word divided by the analog noise floor).


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Why are you *lying*?
 Definition of dynamic range: "Dynamic range is a term used frequently in numerous fields to describe the *ratio* between the smallest and largest possible values of a changeable quantity"_

 

And "decibel" is defined as a "logarithmic unit of measurement that expresses the magnitude of a physical quantity (usually power or intensity) relative to a specified or implied reference level." In other words, a decibel expresses a *ratio*. So Bigshot's statement should hardly be accused of lying for making the statement, "Dynamic range is the difference between the loudest part and the quietest part. A dynamic range of 96dB means that the quietest sound recorded is 96dB below the loudest."


----------



## Crowbar

Quote:


  Originally Posted by *Febs* /img/forum/go_quote.gif 
_"Dynamic range is the difference between the loudest part and the quietest part."_

 

If one insists using the word "difference", then it must be stated as follows: "The logarithm of dynamic range is the difference between the logarithms of the loudest part and the quietest part." Simply assuming that the quantities in question are already converted into log space is unacceptable, because in the general case when talking about mathematics, math in linear space is assumed; they're completely different in log space.

 Now let's look at the other statement:
  Quote:


 A dynamic range of 96dB means that the quietest sound recorded is 96dB below the loudest. 
 

And if the quietest sound is silence? Suppose that the ADC's analog front end is a preamp and we set the gain/attenuation so that the lowest ambient noise maps to digital word 0. Thus, after quantization, the dynamic range is infinite because either you divide by 0 in linear space, or subtract logarithms for dB-based calculation, but log(0) is negative infinity.


----------



## grawk

The range of sound covered by 16 and 24 bits isn't the same. The -96db point is exactly the same on a 24 bit recording as it is on a 16 bit recording. The number of steps is exactly the same. 24 bits allows more information to be captured below what would be the lowest point of a 16 bit recording. If you set your levels in recording such that the ambient noise floor is too high (set the levels too hot) then you won't ever capture the additional information, because it will all be in the area of headroom. 

 Just because crowbar doesn't understand the basic concepts and is adversarial in all his responses doesn't mean that bigshot (or anyone else) is lying.


----------



## Crowbar

grawk is making a claim without presenting any justification. I have presented justification for my claims. Things are pretty clear cut here. I'm going by the scientific method; you, by blind faith. Next you'll be telling us gawd created the Earth 6000 years ago despite evidence I present to the contrary.


 The -96 dB point is only the same if the 0 dB points of the two DACs are matched to correspond to the same physical analog output level. There is every reason they should not be the same. I've already covered this.


----------



## charonme

Please correct me, I'm trying to understand this here:
 By increasing the bit depth we increase the possible dynamic range, because we will have those extra red possible sample values available, which are unfortunately too quiet for us, so we have to turn up the volume, but then the peaks will be too loud.





 What if the recording doesn't use those extra red sample-values? Are then all the additional bits wasted? Aren't the new *yellow* possible sample values still available for more detail? In other words, can we still benefit from a higher bit-depth without the need of increasing the dynamic range and the need to increase the volume?


----------



## bigshot

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_Imagine you have an analog recording which is a result of a DAC from either 16bit digital data or 24bit data. Is there any way of telling whether it came from 16bits or 24bits (including theoretical ways or sci-fi technology)?_

 

I might not be understanding what you mean... but if you used an analogue tape recorder to capture both 24 bit and 16 bit audio on tape, you would be hard pressed to know which was the higher bitrate, because the analogue tape recorder likely has a narrower dynamic range and higher noise floor than either bitrate.

 Does that answer it?

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Why are you *lying*?_

 

Semantic quibbles and exaggerated accusations are a great way to make yourself look like a horse's ass. I'd suggest that you pay a little more attention to trying to understand what a person is saying, and less time trying to manufacture arguing points out of thin air.

 My theory is that the anonymity of the internet encourages certain types of people to exhibit personality traits that they would never hope to get away with in real life. I'm struggling to hold the concept in my head that you're probably a really nice person face to face. Help me with that, would ya?

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_What if the recording doesn't use those extra red sample-values? Are then all the additional bits wasted? Aren't the new *yellow* possible sample values still available for more detail? In other words, can we still benefit from a higher bit-depth without the need of increasing the dynamic range and the need to increase the volume?_

 

As I understand it, the increased resolution is all down in the quietest parts... The resolution of the loud parts is identical. Human hearing is less critical with resolution of really quiet stuff than it is with moderately loud stuff, so the bottom line is that even if the increased resolution crosses over into sound beyond the threshold of perception, the difference is still imperceptible. By the time it matters, there's no difference in resolution any more.

 See ya
 Steve


----------



## gyrodec

bigshot - I have argued exactly what you just did in a thread a few months ago, but I don't think it can be true on further reflection. The 17 to 24 bits do indeed define uselessly quite stuff if they are the only bits turned on. BUT, what if the 8 bit were also on, equal to, say, -48db (from as 0db as all bits on), then by setting any of the 17-24 level bits would allow you to more accurate define levels between that outout and whatever the 8bit and just the 16bit on would be (-48.001db, say) - i.e we could define -48.00001. (The fractions are illustartive, I know they aren't the right numbers, its just to explain the idea).

 So, doesn't that mean we DO get more resolution in the critical areas as well. Please show me how I am wrong, if I am, because I reaaly want to nail this part of the debate down, for my own education.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_the anonymity of the internet encourages certain types of people to exhibit personality traits that they would never hope to get away with in real life_

 

I certainly do not rely on anonymity, and have no problem being called out.

 Borislav Trifonov
 3895 W 24 Ave
 Vancouver, BC
 Canada V6S 1L8
 (778) 688-6139


----------



## Crowbar

I think gyrodec is saying something similar to what I am.
 The point is that whether the extra bits add levels only below, depends on the absolute analog level matching between the two DACs.
 I'm going to argue this on a professional forum and see if someone can actually give an actual counterargument instead of just making claims without backing them up.


----------



## gyrodec

My point is slightly different from your's crowbar. I don't think the level matching has any real baring on my argument, as no matter what signal level is being handled, the extra 17-24 bits will always allow intermediate levels to be defined. My example of using the 8bit at -48db was tie it to bigshots point about the "moderately loud" part of the range being the important one.

 I will be very interested in what you find from the pro site however crowbar. Keep us posted.


----------



## Filburt

24 bit audio has higher granularity than 16 bit audio from -96dB to full scale as it resides in the upper 16 bits, not the lower 16. That is, there are exponentially more values between bits 8 and 24 than there are between bits 1 and 16 (e.g. 2^24 - 2^8 (16776960), versus 2^16 (65536) to represent -96dB to -0dB). -0dB is always in the MSB, so in a 24 bit dac it just doesn't use much of the available values in its higher bits when decoding 16 bits, though I guess you could run some sort of interpolation algo to make use of the additional values.


----------



## Crowbar

gyrodec, looks like Filburt just clarified what you meant. That is exactly right, and formally settles the issue. *It's mathematical proof that the extra bits in a 24 bit system provide finer quantization even within the range covered by a 16 bit system*.


----------



## Crowbar

For some people that don't understand the math, let me provide a simple 2 vs 3 bit example.
 In binary, highest words on the 2-bit and 3-bit are values 11 and 111, respectively, and this is our 0 dB.
 Levels are then:
 11 | 111
 ... | 110
 10 | 101
 ... | 100
 01 | 011
 ... | 010
 00 | 001
 ... | 000
 It's pretty obvious that while the extension of the higher bit depth system is not only downwards, but between samples as well. I think the confusion of Bigshot et al. is due to the fact that they are looking at dB going down as if it were a linear scale. But dB is a log scale, so linearly you are going down into fractions of a signal. But that going down applies between the other levels of a large quantization step system as well, not just between the its lowest non-zero level and zero.

 To show how unintuitive doing this sort of thing in dB is, let's convert the above scale to dB with respect to the highest level. Using the dB formula for voltage (scale by a factor of 0.5 if you prefer dB of power), 20*log10(ratio), we get (to two significant decimal digits):
 -0.0 dB | -0.0 dB
 .......... | -1.3 dB
 -3.5 dB | -2.9 dB
 .......... | -4.9 dB
 -9.5 dB | -7.4 dB
 .......... | -10 dB
 floor.... | -17 dB
 .......... | floor
 The graphs obviously don't align and the left is shifted down a bit, but I'm limited by text formatting. What's clear again though is that the dB extension not only goes down, but between quantization levels of the lower-bit system as well. In a 24 bit system, there are 2^8=256 steps between 16 bit quantization levels, including the 256 steps below the floor of the 16 bit system--and Bigshot et al. only were thinking of the latter, which has been now proven to be false.

 So Bigshot, gregorio, and Febs, will you be men enough to admit you were wrong? I'm not going to bother asking grawk.


----------



## grawk

The higher granularity is at the -96 - -140db range, not the 0 to -96db range. 16bit and 24bit use the exact same bits to represent 0 to -96.


----------



## Crowbar

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_16bit and 24bit use the exact same bits to represent 0 to -96._

 

Obviously they don't, as Filburt and gyrodec have pointed out. I shall present a formal proof by contradiction now:
 Pick a particular 16 bit word. Let us represent it in hexadecimal, so a 4 bit block is 0 through F.
 Example 16-bit word: 8D25
 Corresponding 24-bit word: 8D2500
 Now pick the next 16-bit word lower than this one.
 Second 16-bit word: 8D24
 Corresponding 24-bit word: 8D2400
 Your claim implies there is no increase in levels between 8D25 and 8D24 when you go to 24 bits. Now we take word 8D2419. This is between the two 24-bit words corresponding to the consecutive 16-bit words. Theferore, your claim is false.
 QED

 The downwards extension you're talking about is a subset of this: lowest non-zero 16-bit word being 0001, whereas for 24-bit you get 0000xy where for any x and y you have a bunch of levels. The finer quantization at the bottom is the exact same as between all other levels, as has been shown multiple times with basic mathematics.


----------



## grawk

Your proof ignores the way things are actually implemented. 0db is represented as 0, -96db is represented as -2^16. In a 24 bit dac, -96db is still represented as -2^16.


----------



## Crowbar

Since I build DACs I call BS. A 16 bit signal sent to a 24 bit DAC is zero-padded to the right. What you're saying can only be true if the 16 bit signal was stuffed into the lower bits--this is NEVER done. You would get extremely quiet output.


----------



## gyrodec

crowbar, your last point about zero padding is the key that the others, and myself for quite a while, are missing. I am now 100% sold on greater resolution of voltage levels in 24 bit audio vs 16 bit, unless you specifically rig the 24 bit DAC to work as a 16 bit and not to zero pad as is/should always be done.

 Please guys, think about it. Don't just argue because it' crowbars point - I know its tempting. The dynamic range numbers are not the whole story. Reread crowbars post of 2:23 today, it realy is compelling if you stop to think.


----------



## gyrodec

For the record, I'm not saying you can hear the difference - I haven't done the test so I can't say. I'm just saying that there is a resolution difference at normal signal levels.


----------



## Crowbar

I really want everyone to understand this. It's not complicated at all, just exercise your brain a little bit and you'll get it. I'm hoping this diagram will help explain the confusion that a dB scale is doing when you start talking about dB when discussing quantization. This shows the quantization levels for a 16 bit and 24 bit system if the max is set to 0 dB. Obviously there should be far more lines, 2^16 in the first and 2^24 in the second, but I can't really draw that. Excuse my MS Paint 
	

	
	
		
		

		
			













 One interesting thing to note is that on the logarithmic dB scale, the spacing between lines on the bottom is the same on both sides, it just looks like one has been shifted down. And that's how it gets you. You must remember that a difference of say 10 dB depends on where in the scale you are. The difference between -120 and -130 dB is far, far tinier than the difference between -10 and -20 dB!

 In the linear graph, you can see that the extension downards is really quite small--you're just getting closer to zero because you have more bits to play with--*the 8 lowest bits of a 24-bit word means you can deal with values a fraction of the power of the smallest bit of a 16 bit word*. That gives you the finer quantization between the other levels as well, not just the bottom. Note that 0 V cannot be represented on the dB scale because it's at negative infinity in dB. Of course, the real analog noise floor of a DAC will be higher than that; for the best 25-bit DAC, around -135 dB (ESS Sabre)

 (Edit: added linear graph)


----------



## Filburt

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_The higher granularity is at the -96 - -140db range, not the 0 to -96db range. 16bit and 24bit use the exact same bits to represent 0 to -96._

 

Granularity is greatest towards the highest bit, not the lowest, and -96 to -144 resides in the lower bits. It _has to_ be this way; think binary. Smaller values are represented by lower bits being turned on or off in combination, and higher values include a greater number and magnitude of bits in order to represent the value. In a dac, those higher magnitude bits produce a greater output amplitude (e.g. upper-bit resistors in a ladder dac produce greater current at the output). It is physically impossible* for the dac to represent lower amplitudes using the upper bits and higher amplitudes using the lower bits, which is what would be required in order to have granularity higher in the lower amplitudes (e.g. there are far more values between bits 16 and 24 (16711680) than there are between bits 1 and 8 (256)). 

 When doing 16 bit playback, 16 and 24 bit converters _do_ use the exact same bits: MSB to the 16th bit under it (which is LSB on a 16 bit converter). 16 bit playback in a 24 bit converter will still only use 65536 possible values, however, the 24 bit converter will still be _capable_ of providing 16776960 values to those upper 16 bits. This is easy to test, since the MSB provides the converter's maximum amplitude. If 16 bit playback resided in the lower 16 bits, which it would have to in order to limit granularity to 65536 values, the maximum amplitude on 16 bit recordings would be that provided by the 16th bit instead of the 24th (MSB). This would make your 16 bit playback peak at considerably lower amplitude than 24 bit recordings, have terrible SNR, and a boatload of distortion. I don't know about your DAC, but on mine what I see is the same amplitude at peak irrespective of whether I play back 24 bit or 16 bit audio.

 edit* - Technically, I guess, you could represent lower amplitudes using upper bits by using some sort of stage thereafter which will invert the value relationship, but that isn't something that dacs, as they are presently designed, do. Furthermore, such a design choice would be asinine since, in audio, it is _by far_ more important to have granularity higher as amplitude rises, since audibility of quantisation error rises with amplitude (obviously).


----------



## jenghanHsieh

Crowbar, 

 I've followed this intensive thread for some time and I've read through every page.

 Your drawings are fine and express your idea clearly, however, I found that you keep neglecting (IMHO) a key point that some people said about the digital audio theory. 

 That is ( in my own words ) : 
 Quantization Error is NOT an issue for digital audio reconstruction, because
 statistically, a long (infinite) sequence of Quantization Errors can be dithered
 and transformed into *uncorrelated noise * ( ie WHITE NOISE --- which they called the 'noise floor' ).

 I know your demonstrations and drawings pretty well, if not perfectly. 
 And I know what quantization and quantization error are. But if the others are correct about the digital audio THEORY ( juxtaposition of Sine waves of band-limited frequencies and amplitude... ), sampling theroem, anti-aliasing, dithering ... etc, then 

Quantization Error is NOT an issue for digital audio reconstruction
Quantization Error is NOT an issue for digital audio reconstruction
Quantization Error is NOT an issue for digital audio reconstruction

 I don't think they are ignorant about Quantization Error ( or granularity ), instead they try to tell us that the Quantization Errors can be eliminated 'statistically' during the ADC/DAC process, thus such error will not be an issue and only the noise floor matters.


  Quote:


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

 One interesting thing to note is that on the logarithmic dB scale, the spacing between lines on the bottom is the same on both sides, it just looks like one has been shifted down. And that's how it gets you. You must remember that a difference of say 10 dB depends on where in the scale you are. The difference between -120 and -130 dB is far, far tinier than the difference between -10 and -20 dB!
 ...

 (Edit: added linear graph)_


----------



## Crowbar

Theoretically optimal dither cannot be achieved in practice. Moreover, it trades quantization errors for noise, and thus you lose dynamic range. Flat dither on 16 bit results in about 12 dB increase in noise. If it's noise shaped, you can cut it in some areas of the bandwidth, at the expense of raising it in others. But the noise energy remains the same. This is not so bad if your sampling rate is high, say 96 ks/s, since then you can move more of it above the audio band, but not for 44.1 ks/s.
 Quantization errors also cause intermodulation with the signal due to limitations of the analog electronics. If dithering is used, most of this distortion is transformed to noise--that's better but it's still additional noise, that also gets added to the figure above.
 So even with an optimally dithered system the quantization error IS an issue, albeit indirectly--it results in significantly increased noise.


----------



## charonme

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_As I understand it, the increased resolution is all down in the quietest parts..._

 

Did I get what you said correctly? Did you mean that by increasing the bit-depth we only gain those red available sample values but *not* the *yellow* ones? Is my drawing incorrect? Or are you saying that we can still use the yellow sample values, but it will not be audible anyway? - I believe that's true with 24bits, but isn't it audible enough when going from eg. 4bits to 16bits?

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It's pretty obvious that while the extension of the higher bit depth system is not only downwards, but between samples as well._

 

Can "_downwards_" be equated with the red possible sample values and "_between samples_" with the *yellow* ones in my drawing?

  Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_The higher granularity is at the -96 - -140db range, not the 0 to -96db range. 16bit and 24bit use the exact same bits to represent 0 to -96._

 

By "0 to -96" are you referring to the green scale sans the *yellow* values? Why wouldn't you use the yellow values?


----------



## jenghanHsieh

Quote:


  Originally Posted by *charonme* /img/forum/go_quote.gif 
_Can "downwards" be equated with the red possible sample values and "between samples" with the *yellow* ones in my drawing?_

 

The visual resolution & perception is determined by the canvas size (screen) and the viewer's distance. The audio is determined by the maximum amplitude (0db), the gain (amplification, voltage swing... whatever audio equipment that translate the 0db to horribly noisy sound wave), and the threshold of pain.

 While it make sense to use a huge LCD screen up close for programming, and perhaps managing large spreadsheet because our eyes can easily focus on a small part of the surface area and ignore the rest ( without being harmed ). 

 But if we use the 'super-zoom' in audio reproduction. A sudden loud noise (HUGE Canvas) may damage our ears ... 

 IMHO this is one major difference between digital audio & image .


----------



## charonme

Quote:


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

 

True, true, I agree. But sometimes you don't have to zoom to see that higher resolution is better, for example if you're standing 10m away from a 2x2m canvas, you can clearly see the picture is better with 16x16pixels than 4x4pixels. The same applies to 4bit vs. 8bit audio. You don't have to amplify to hear the improvement.


----------



## Febs

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_So Bigshot, gregorio, and Febs, will you be men enough to admit you were wrong? I'm not going to bother asking grawk._

 

I'm certainly "man enough" to acknowledge the validity of the point that you and Filburt are making regarding granularity. My recent posts in this thread, however, have dealt more with the practical implications of using 120 dB of dynamic range, so I'm not sure that your explanation of granularity directly impacts that discussion.


----------



## gregorio

Crowbar (and Filburt) you are completely missing the whole point. Yes, there are more quantisation points in 24bit than 16bit, this is obvious. The point you are missing is what difference it makes to the output of a DAC. Both 24bit and 16bit (or for that matter 4bit) are going to produce a completely linear output when decoded to analogue. Same resolution, same granularity, everything. This is a basic tenet of digital audio theory. What changes with higher bit depths is fewer quantisation errors (due to the increased number of quantisation points). However, quantisation errors manifest as white noise. So the difference between the high bit rates and low bit rates is just the amount of noise, that's all, there are no other consequences or differences in quality between higher and lower bit depths, just noise. Each doubling of the number of quantisation points (adding an additional MSB) will result in half the number of quantisation errors, half the noise and therefore an additional 6dB of dynamic range. As already discussed, using 16bit the level of this noise is going to be below the threshold of hearing. Using good quality dither to create 16bit output will result in the noise being perceivable around the -120dB level, for 24bit it's around -150dB or lower. We've gone over this same info a number of times now in this thread and provided ample links to accepted proofs. Crowbar - Either you are incredibly dense or you are just deliberately trying to spoil this thread!

 Charonme - The additional resolution of the new yellow parts (in your diagram) are available and are used, it's this extra resolution which causes fewer quantisation errors, resulting in a lower noise floor and therefore the additional dynamic range. Your diagram isn't quite correct though, because the added dynamic range is caused by the higher number of quantisation points in the yellow section compared to the green, there are no red bits as such.

 jenghanHsieh - Spot on. Quantisation errors are converted to white noise in any process which requires conversion, quantisation or re-quantisation. This means the ADC, the mastering process and the DAC.


----------



## gyrodec

gregorio, I have only read the start and end of this thread, so would you mind giving me a link to the reference for your last post. I've learn't that doubting you usually ends in being proved wrong, but I want to get more detail on what you are saying. It looks like I might have to 180 in less that 24 hours. Man, I hate having to do that, but live and learn - or you're just pretending to live.


----------



## nick_charles

Let me see if I have this right ?
 ------------------------------

 As I understand it a 16 bit system representing a nominal 2V signal will cater for values between -2V and + 2V, the number of steps available is 65536 (0 - 65536) so the voltage is quantised according to 

 2 - (-2)
 ======
 65536

 = 4/65536 

 So each "step" is nominally 0.00006103515625V lets say 0.00006 for simplicity, so 

 -2.00000 to - 1.99994 = 0000000000000000
 -1.99994 to - 1.99988 = 0000000000000001
 -1.99988 to - 1.99982 = 0000000000000010
 -1.99982 to - 1.99976 = 0000000000000011

 and so on...

 1.99976 to 1.99982 = 1111111111111100
 1.99982 to 1.99988 = 1111111111111101
 1.99988 to 1.99994 = 1111111111111110
 1.99994 to 2.00000 = 1111111111111111


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_So Bigshot, gregorio, and Febs, will you be men enough to admit you were wrong?_

 

I'm afraid when you get into long paragraphs punctuated by rows and rows rows of numbers with decimal points, I glaze over. I'm more interested in the application of technology than the details of the theory behind it. I'll let those with an interest in numbers and graphs address your points.

 I can only tell you what I found from working with ProTools workstations in the studio and carefully testing their performance... The only difference in sound quality between bitrates is the ability to pull up very low level audio in 24 bit without loss in resolution. In the normal range of dynamics in music, the waveform on the screen is exactly the same and it sounds exactly the same, regardless of whether it's 24 or 16. A 24 bit final mix of a music track, properly dithered down to 16 bits is *identical* for all intents and purposes. 24 bit is perfectly suited for mixing. 16 bit is perfectly suited for playing back music. At a certain point, more numbers and bigger file sizes don't guarantee better sound. High fidelity is high fidelity.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *jenghanHsieh* /img/forum/go_quote.gif 
_While it make sense to use a huge LCD screen up close for programming, and perhaps managing large spreadsheet because our eyes can easily focus on a small part of the surface area and ignore the rest ( without being harmed ). 

 But if we use the 'super-zoom' in audio reproduction. A sudden loud noise (HUGE Canvas) may damage our ears ..._

 

Bzzz! Audio loudness in the image domain corresponds to brightness. Resolution in images corresponds to sampling rate in audio. You're getting mixed up.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Same resolution, same granularity, everything._

 

Only if you have theoretically perfect dither. You can't do that in practice.

  Quote:


 However, quantisation errors manifest as white noise. 
 

What I said above. Also, if you're doing noise shaping, the noise isn't white (obviously).

  Quote:


 Using good quality dither to create 16bit output will result in the noise being perceivable around the -120dB level 
 

Absolutely false. Flat dither gives -84 dB (undithered is -96 dB but with signal-correlated quantization errors). If you use noise shaping instead of flat dither, you can make it lower in some areas, but then you have to move that noise energy into other areas of the bandwidth. Your claim of -120 dB can only be achieved in a narrow part of the band unless you want to raise noise by a large amount in another part. The noise energy throughout fs/2 is still going to be that of a flat dither's -84 dB.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I'm afraid when you get into long paragraphs punctuated by rows and rows rows of numbers with decimal points, I glaze over._

 

Technology is based on mathematics, and to hear something like this in modern times is startling. We're talking grade school basic math here, that should have taken you less time to understand than writing your long post.
 Febs and gregorio already conceded the finer quantization point, and have moved on to something else.


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Let me see if I have this right ?_

 

Looks right. Output will vary within DACs, but you're dividing it into equal steps in the linear space, which gives a logarithmic subdivision in log space. In a dithered system the actual value of each sample will be affected by its context of other samples, so it really becomes a timelike distribution. That decorrelates the error from the signal but raises the noise level--total error energy cannot be decreased (if you could do it it would mean you're getting more information than the theoretical maximum of the digital stream).


----------



## bigshot

Dithering works. It doesn't take a million fancy equations to prove that... just a pair of ears. Go listen in a controlled way to what your theories actually sound like. A properly dithered 16 bit bump down from a 24 bit mix sounds exactly the same. You can obsessively use math to cross every T and dot every I, but it isn't going to result in better sound.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_just a pair of ears._

 

Everyone's ears, on any equipment? Or perhaps you think your ears and equipment are the very best... that's what your post reads like.
 Dither does work--at raising the noise floor.

  Quote:


 A properly dithered 16 bit bump down from a 24 bit mix sounds exactly the same. 
 

With your ears, on your equipment. Audiology says 120 dB, thus obviously there are ears that do not fit with your little picture. Audiology is a science, by the way, and they measure stuff. With representative samples of subjects and correct statistics. Your anecdotal experience is completely irrelevant.

  Quote:


 You can obsessively use math to cross every T and dot every I, but it isn't going to result in better sound. 
 

Dither only exists because people obsessed over math. So does digital audio, and indeed, electronics of any sort.


----------



## bigshot

Feel free to use any human ears. Dithering works. The noise floor of digital is so low, even raised by dithering, it would still make no difference. Redbook has dynamics and signal to noise to spare. It's perfectly capable of reproducing sound to a level where human ears can't discern any improvement from bigger files and more numbers. If you want better sound, chasing down extra zeros in digital theory isn't going to do it... Much better to study acoustics and learn miking technique. The aspects of sound reproduction that need work are the beginning stages of recording and mixing and the end ones of playing back on speakers in rooms. The middle steps are just about as good as it's going to get.

 See ya
 Steve

 P.S. You keep citing a 120dB figure based on audiology. That isn't based on recorded music. Redbook provides plenty of dynamic range. Anything more than that would be uncomfortable to listen to in a human habitat. If you are recording a person dancing soft shoe next to a blast furnace, 24 bit would be a must. For music, redbook is more than you need.

 To make it clear...

 A 120 dB dynamic range in music played on a home stereo would be completely unlistenable. You would either be flinching in pain at the volume of the peaks, or leaping up and down to adjust the volume whenever the dynamic changes. 50dB is plenty.


----------



## Crowbar

You are contradicting the scientific fact that the ear covers a 120 dB dynamic range. It's as simple as that.


----------



## gyrodec

No he's not, he's saying that yes the ear can hear 120dB, but no recording will ever be made for domestic human listening with more than redbooks 96dB for obvious and often stated reasons. Indeed, virtually no commercial recording has ever been produced with more than 50dB dynamic range. It has nothing to do with the theroetical, or indeed practical, limits of human hearing, but about what will ever be produced for sane practical reasons.

 (Sorry for answering for you bigshot.)


----------



## regal

so each step is 60 uV, do we think we could hear a 60 uV difference in sound at normal listening volumes? Maybe when amplified with the Rolling Stones conert gear?


----------



## b0dhi

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_so each step is 60 uV, do we think we could hear a 60 uV difference in sound at normal listening volumes? Maybe when amplified with the Rolling Stones conert gear?_

 

That's not an easy question to answer. This paper shows that the air displacement measurement precision of the cilia can be as low as fractions of a nanometer, but that precision isn't equal across the range of motion, and the ear has a very complex hearing mechanism. You can do subjective studies but I have doubts as to their validity in cases such as this. What's measurable by the ear and what's consciously detectable are are not necessarily the same.


----------



## regal

Its generally accepted that our hearing can't differentiate less than 1 to 3 dB. Should be an easy caclulation.

 60uV multiplied gain of 3 in the headphone amp = .18 mV.

 A typical headphone has a sensitivity of 95 dB/mW and a current of 20 mA. So .18mV*20mA = 3.6uW. then 3.6uW*95*1000 = .324 dB.


 So 16 bits gives .36 dB steps, hearing can't tell this from 1 dB.


 16 bits provides plenty resolution


----------



## b0dhi

The decibel value you quoted is generally related to the overall volume of the sound, that is, the integral of the sound energy over time, not the precision of air displacement measurement per sample. Also, it's a value derived by subjective assessment, which may not show what the ear is actually capable of.


----------



## regal

Well .36dB isn't an order of magnitude less than 1 dB, so with different headphones or amplifiers its possible there may be a slight chance we could hear 60 uV but I doubt it.

 You also have to consider the noise floor in the calc, I'll let someone else try that calculation.


----------



## bigshot

Do you know what 1dB difference actually sounds like? It isn't easy to detect. With direct A/B comparison, you *might* just barely be able to detect it. I doubt that one human being in a ten thousand can detect a difference of less than .5 dB. Even if they could hear, it, it isn't going to make a lick of difference to their enjoyment of music. As for the cilia stuff, you aren't really serious about that, are you?

 Let's not waste time talking about volumes less than 1dB or stuff up around 120dB- there's no point. Talk about the stuff in the middle that really matters.

 See ya
 Steve


----------



## Crowbar

The "cilia stuff" is actually even more sensitive than b0dhi says.

 From: Sensitivity of Human Ear
  Quote:


 The threshold of hearing corresponds to air vibrations on the order of a tenth of an atomic diameter. 
 

This is several orders of magnitude less than a nanometer.

 Anyways, a question such as "do we think we could hear a 60 uV difference" is meaningless since it depends on amplifier gain, driver sensitivity, ambient noise, isolation (in case of headphones), distance from the ears, etc. It's like asking is a car hitting a wall while its engine is doing 3500 rpm going to kill the driver. Well, if it's in 1st gear not likely; if in 5th, another matter. It only confuses things to think in absolute levels; the only thing that matters is relative levels, and an implied assumption that they will be matched to a specific dB at the ear.


----------



## Crowbar

Quote:


  Originally Posted by *gyrodec* /img/forum/go_quote.gif 
_No he's not, he's saying that yes the ear can hear 120dB, but no recording will ever be made for domestic human listening with more than redbooks 96dB_

 

If the recording is flat-dithered, the noise floor will be -84 dB. If it's noise shaped, then it could be lower in some areas, but necessarily higher in others.
 Ambient noise over the mid and upper frequencies can be brought down to the threshold of hearing through the use of in-ear-canal headphones like the Etymotic. You don't need an anechoic chamber. 120 dB, as has been demonstrated by the graph I posted, causes no hearing damage unless the sound is continuously at that level for some time--something I'm obviously not proposing. In many classical concerts I go to I hear a dynamic range over 100 dB--dozens of times greater than the flat-dithered 84 dB a 16-bit system can manage (since dB is a logarithmic scale, a difference of 16 dB means a linear factor of 40 in units of power)


----------



## regal

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_. In many classical concerts I go to I hear a dynamic range over 100 dB--dozens of times greater than the flat-dithered 84 dB a 16-bit system can manage (since dB is a logarithmic scale, a difference of 16 dB means a linear factor of 40 in units of power)_

 

So you go to concerts with a dB meter in hand or were you implanted one in your brain at birth?

 Seriously your argument on the necessity of 24 bit only holds some water in the mastering phase, no one masters a recording to live sound dynamic range. No one even comes close. If they did you would need horn speakers to hear all of it.


----------



## Crowbar

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_So you go to concerts with a dB meter in hand or were you implanted one in your brain at birth?_

 

No, I cite others' data:
Designing, Building, and Testing ... - Google Book Search
 100 dB is given for "typical seat, classical music concert"
 Most other places I've seen cite higher numbers (because usually those measure at the conductor location or near-stage microphones).

  Quote:


 your argument on the necessity of 24 bit 
 

My argument is for the insufficiency of 16 bit. 24 bit simply happens to be the only standard alternative (and one adopted because it adds a standard byte to the word length).

  Quote:


 If they did you would need horn speakers to hear all of it. 
 

Why? The only way you would say something so incredibly stupid is if you're confusing speaker efficiency with dynamic range! There's absolutely no relation--I'm startled anyone at an audio forum would post such ludicrous nonsense. As Babbage put it, I am not able rightly to apprehend the kind of confusion of ideas that could provoke such a question.

 Dynamic range is measured in dB. Speaker efficiency is dB SPL per watt and tells you *nothing* of the dynamic range the speaker can represent. I'm still shocked you made such a huge, embarrassing blunder. But if memory serves me correctly, this happened in another thread as well. Your ignorance, like your arrogance, knows no bounds.


----------



## regal

You're the one saying a 60uV signal feeding an amp can be differentiated, you would need very effecient speakers to even have a chance of hearing this. Its ignorant to think a dynamic driver has the resolution to pull this off. I am think a large amp and horn speakers maybe. You're not looking at the big picture (could be that spl meter implanted in your brain.)


----------



## b0dhi

Until those involved address the issue of precision of measurement, _not_ threshold of measurement, this conversation will keep going around in circles like it has since I stopped reading it 30 pages ago. Bit depth isn't only relevant at the two extremes - it's relevant throughout the entire representable range. The extra 8 bits in 24bit sound *don't* just encode quiet sounds. They encode sound with greater _precision_ at all volumes. The question is whether that extra precision is at all detectable by the ear (objectively, not by subjective trial). That is *not* the same thing as asking whether a sound of that _quietness_ is detectable by the ear. 

 If the ear can detect air displacement with a _precision_ greater than the resulting air displacement precision representable with a particular bit depth (and associated converstion and amplification equipment running at, say, 100dB SPL), within any frequency band, then higher bit depth is necessary. Otherwise, it is not.

 I don't think higher dynamic range is a good justification for higher bit depth, as it would necessitate equipment that dynamically adjusts the dynamic range during playback so that it could be listened to at quieter volumes when the listener wants to. Otherwise, much of the quiet parts of the recording would be inaudible during regular/low volume listening.


----------



## Crowbar

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_You're the one saying a 60uV signal feeding an amp can be differentiated, you would need very effecient speakers to even have a chance of hearing this._

 

Not at all. I would simply use higher gain in the amplifier. 20 dB gain is not uncommon in power amplifiers. Also, many people use preamps. In that case you can have much higher gain. Speaker efficiency is irrelevant--one simply uses a more powerful amp.


----------



## 22906

There is a distinction between the quietest audible sound and the audible dynamic range. I don't understand how the discussion about atomic-scale vibrations is relevant.

 Also, there is a distinction between the dynamic range and absolute sound pressure level. I'm curious as to where in the absolute volume scale the 120 dB was measured.

 It makes sense that people are not going to reproduce concerts in their homes at the same level as they were originally. Also, it seems like in most real environments except for something like a concert or right next to a plane crash the audible dynamic range will be limited by the maximum absolute sound pressure level and never actually reach the theoretical limit.

 Isn't there a better way to quantify audio 'resolution' other than dynamic range?


----------



## Crowbar

In terms of SPL, 0 dB is 10^-12 W/(m^2), just under the threshold of hearing when that intensity is at the ear.

 One could argue that distortion is another way the perception of resolution is compromised. But that is not straightforward. Some people perceive increased jitter as more micro-detail and thus added resolution. It's best we don't go into such things as it doesn't (directly) depend on bit depth or sampling rate.

 I also disagree withour argument pro dynamic range compression. The goal of any music system in my view is to be transparent--that is, indistinguishable from the real thing. A good audio system is an aural virtual reality machine. Now, obviously there are problems with various parts of the signal chain, but that was already covered in the early pages of this thread.


----------



## regal

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Not at all. I would simply use higher gain in the amplifier. 20 dB gain is not uncommon in power amplifiers. Also, many people use preamps. In that case you can have much higher gain. Speaker efficiency is irrelevant--one simply uses a more powerful amp._

 



 Say 20X gain. 20x 60 uV = 1.2mV for the speaker to hear the quietest passage. Do you think even a horn speaker will have this audible after just hearing a 96db passage? Let alone a 20 bit range. You have missed the most important fallacy of your argument.


----------



## 22906

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_In terms of SPL, 0 dB is 10^-12 W/(m^2), just under the threshold of hearing when that intensity is at the ear.

 One could argue that distortion is another way the perception of resolution is compromised. But that is not straightforward. Some people perceive increased jitter as more micro-detail and thus added resolution. It's best we don't go into such things as it doesn't (directly) depend on bit depth or sampling rate.

 I also disagree withour argument pro dynamic range compression. The goal of any music system in my view is to be transparent--that is, indistinguishable from the real thing. A good audio system is an aural virtual reality machine. Now, obviously there are problems with various parts of the signal chain, but that was already covered in the early pages of this thread._

 

If I understand correctly you're saying the only reason 16 bit is insufficient is that it doesn't provide enough dynamic range? I'd be more concerned about resolution, which you say isn't even related to bit depth.

 My argument isn't about artificially compressing the sound. Its about playing back recordings in real environments, which would necessarily reduce the absolute sound pressure level below 120 dB, just because people would damage their ears otherwise, and therefore compromise the dynamic range. Dynamic range is just a ratio of the quietest sound to the loudest sound. It isn't really a good criteria to use when trying to justify a higher bit depth.


----------



## regal

m3, the speakers are the limiting factor not our ears. You are saying dynamic range is not important but resolution is. No speaker can differentiate the difference between a few milliVolts.

 The argument gets rediculous when you run the numbers.


----------



## 22906

You're right but its useful to assume that the rest of the chain is ideal when you are focusing on bit depth of the recording.

 Usually when you are isolating a specific quality of the reproduced sound it doesn't have a significant impact on sound quality by itself. The best we can do is to try and optimize each part from a theoretical standpoint, and fine tune the end result afterwards.

 Also, my guess is that speakers are limited in resolution by some mechanical threshold of the drivers rather than their efficiency. You can't quantify the sound in terms of volts, because you could just make an amplifier or a speaker of different gain/sensitivity to compensate and make small voltages into audible sounds.


----------



## regal

Quote:


  Originally Posted by *m3_arun* /img/forum/go_quote.gif 
_You're right but its useful to assume that the rest of the chain is ideal when you are focusing on bit depth of the recording.

 Usually when you are isolating a specific quality of the reproduced sound it doesn't have a significant impact on sound quality by itself. The best we can do is to try and optimize each part from a theoretical standpoint, and fine tune the end result afterwards._

 


 No, you always take the system as a whole, its called engineering. Your argument is true for the mixing/mastering phase. For the end user you look at the entire chain and 16vs 24 doesn't matter whatsoever for playback.


----------



## b0dhi

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_No, you always take the system as a whole, its called engineering. Your argument is true for the mixing/mastering phase. For the end user you look at the entire chain and 16vs 24 doesn't matter whatsoever for playback._

 

Apparently your understanding of engineering is to make an arbitrary assessment based on an irrelevant and misinformed calculation (the one you did in the last page), then ignore the fact that there are non-dynamic speakers out there, and decide that somehow 16v24bit argument is settled.

 I hope I never come across something you've engineered.

 I'm simply amazed at the level of ignorance consistently displayed in this thread. I was hoping that after 30 pages of not reading it there would've been some progress. Sadly, there has been none. This is my last post in it.


----------



## regal

I stand by the calculation, in fact it is better than what you posted which was nothing but opinion. Funny I have never had a problem with things I've engineered. But personal attacks are easy. Glad to see you leave the thread.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_If the recording is flat-dithered, the noise floor will be -84 dB._

 

That is double the range of most musical recordings, and at normal listening volumes, well under the room tone of even the quietest room. Inaudible.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_The "cilia stuff" is actually even more sensitive than b0dhi says._

 

Rhetorical question: How many more cilia does 24 bit sensitize as opposed to 16 bit? Irrelevant.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_In many classical concerts I go to I hear a dynamic range over 100 dB_

 

I guess you are a member of the orchestra that only plays rehearsals, because you won't get that kind of sound from a seat among the audience.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I also disagree withour argument pro dynamic range compression. The goal of any music system in my view is to be transparent--that is, indistinguishable from the real thing._

 

That is the statement of an armchair theorist who doesn't understand what people who actually produce recorded music do for a living. The goal of a recording is to create a well balanced performance _in people's homes._ It's totally impossible to reproduce exactly the directionality and dispersion of music in a real setting. The best you can do is just take two points and create a binaural recording. But 99% of recorded music isn't interested in making a high splat on the wall of "realism" like that. The goal is to create an acoustic performance that is crafted specifically for the home environment and optimized for that venue. This is your fundamental misunderstanding, and your stubborn refusal to see how music reproduction relates to music presentation is taking you down a blind alley of theory that doesn't work well in practice. You shouldn't be surprised when other people, particularly people in the music business, don't follow you down that path. 

 You could go ahead and create a technology that has 150dB of dynamic range and frequency response flat up to the range only bats can hear, but it wouldn't be utilized because it's unnecessary. The only people who would buy into it would be what Ken Rockwell in the photo field calls "measurebators"- people who care more about specifications on a piece of paper than the way the equipment actually performs in the real world.

 See ya
 Steve


----------



## bigshot

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_I hope I never come across something you've engineered._

 

Sorry, but you don't know what you are talking about and your REALLY don't know how good of an engineer he might be. You don't even know the questions to ask to find out. For someone complaining about ignorance in this thread, you're doing a pretty good job of being a poster boy for it yourself.

 See ya
 Steve


----------



## charonme

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_...fewer quantisation errors, resulting in a lower noise floor and therefore the additional dynamic range..._

 

Ah, I've never understood it like that, thank you for clarification. I truly learned a lot today, really. [size=xx-small]I'm not being sarcastic, even though it sounds like that in the context of this arrogance-filled thread.[/size]


----------



## gregorio

Charonme - Glad it helped. There is so much discussion about the technicalities of what exactly happens when converting to digital, bit depths, resolution etc. Some people get so caught up in this detail they seem to forget that it's all about what comes out of the DAC that really matters. It's not really important how many quantisation steps a given bit depth provides, what is important is how this resolution manifests itself when we actually listen to a finished product. the bottom line ends up being extremely simple, it's just the level of noise.

 Mentioning noise brings me back unfortunately to Crowbar. What are we to do with you? First the concert hall; yes Crowbar you do sometimes get transient peaks of up to 120dB in a live symphony performance. So how would we recreate this performance in the home? Easy, have a standard orchestral CD with about a 60dB dynamic range, whack the volume up so that the peaks hit 120dB and turn your TV on in the background to emulate the noise floor of a concert hall which is usually somewhere around 55-60dB and there you have it, all from just a so called low-res CD!!! You seem to have a great brain for facts, figures and insulting people but completely lack basic common sense when it comes to the rudimentary application of professional audio. For the last time, you cannot have a recording with a dynamic range of 120dB!!! Why is this so difficult for you to understand?

 2. Dithering, yes Crowbar you get exactly the same energy with noise-shaped dither as with no noise shaping. However, this energy is moved into areas of the frequency spectrum where we cannot hear it. At roughly 10-12kHz our hearing rapidly deteriorates in sensitivity, same with the low frequencies. A sound at 60Hz requires roughly 50dB more gain to appear the same volume as a sound at say 3kHz. In other words, if we redistribute the dither noise to <60Hz it is going to sound about 50dB quieter than if its at 3kHz. This is why noise-shaped dither works, why the perceived noise floor of 16bit can be as low as -120dB and why your figure of -84dB is so mis-leading.


----------



## Crowbar

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_No speaker can differentiate the difference between a few milliVolts._

 

That's false.


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_How many more cilia does 24 bit sensitize as opposed to 16 bit?_

 

Obviously it's a matter of degree. Cilia are binned by frequency, not amplitude. Yet again you show your ignorance.

  Quote:


 because you won't get that kind of sound from a seat among the audience. 
 

The reference I cited specifically says from a seat in the audience. A book is far more reliable than you are (especially when that number can be found at multiple places).


----------



## Crowbar

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_The only people who would buy into it would be what Ken Rockwell in the photo field calls "measurebators"- people who care more about specifications on a piece of paper than the way the equipment actually performs in the real world._

 

It's very clear the standards I'm interested in matching is those set by the physiology of the human auditory system. And the numbers are clear. The 120 dB number is quoted everywhere. Thus, to guarantee transparency, it is what's needed. Moreover, it is most definitely not your place to say that any quality less than perfect transparency with respect to the ear is good enough. That's like the average person saying that iPod quality is good enough for everybody (when in fact it's not good enough for everybody--even if it's good for most people, it's irrelevant unless you're making a mass market product).


----------



## bigshot

Can you name your favorite recording that has a 120dB dynamic range?

 See ya
 Steve


----------



## Crowbar

Considering the sparseness of non-16/44.1 recordings, how can I do that? Besides consumer adoption, another reason 24/96 isn't more widely adopted is because of the Ludditism of people like you.


----------



## 22906

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It's very clear the standards I'm interested in matching is those set by the physiology of the human auditory system. And the numbers are clear. The 120 dB number is quoted everywhere. Thus, to guarantee transparency, it is what's needed. Moreover, it is most definitely not your place to say that any quality less than perfect transparency with respect to the ear is good enough. That's like the average person saying that iPod quality is good enough for everybody (when in fact it's not good enough for everybody--even if it's good for most people, it's irrelevant unless you're making a mass market product)._

 

What does transparency have to do with the ear? Theoretically, you don't have to know anything about the ear to perfectly recreate the original sonic environment.

 I think the only valid justification of 24-bit recordings is a band-aid for DACs that can't fully achieve 16-bit resolution.


----------



## LFF

I get this question all the time regarding my LP transfers. Here is my response:

 The current fashion in digital audio right now is "bit depth". As has been the standard practice for decades, audio is being sold by the numbers, whether meaningful or meaningless. And of course, everybody involved in digital audio transfer is now shouting about their bit depth. 16-bit? 24-bit? 32-bit? Hey, why not 64-bit while they're at it? Logic would suggest that the greater the bit depth, the better the sound. Accordingly, LP to CD transfer companies always point out their 24-bit process to make you think their transfers will sound better. But is there a genuine advantage to higher bit-depth transfers?

 To answer that question, let's look at how bit depth relates to digital audio. The function of bit depth is to determine dynamic range. A greater bit-depth gives you more potential numbers between the zero-crossing point of the waveform and the peak, thus greater amplitude is possible. Or, to put it another way, a greater differential between the peak level and the noise floor. 24-bit might be, theoretically, quieter than 16 bit (which is already dead quiet). The popular way of thinking is that 24-bit has "higher resolution" than 16-bit, but this is fallacy. *Resolution is determined by sampling frequency, not bit depth.* To illustrate, picture 44,100 orange crates standing in a row. Those crates represent one second of CD audio. Bit depth measures the size of those orange crates. A 16-bit crate can hold 65,535 oranges, and a 24-bit crate can hold 16,777,216 oranges. Thing is, even if they held a TRILLION oranges, there's only ever going to be 44,100 of them. The resolution remains unchanged.

 The question then becomes, just how many bits do we need? The dynamic range of 16-bit digital recordings is around 90 dB. Dead quiet to full-blown symphony. But we must consider how much of that 90 dB dynamic range we're actually going to use when transferring albums to CD. The best turntables are going to be in the 70's. The very best vinyl is going to be in the 50's if you're lucky, with most commercial releases hovering in the 40s. Yep. 40's. So - we only need a about half of the 90 dB dynamic range that 16-bit makes available to us. Why don't LP-to-CD services mention this little detail while touting their 24-bit transfers?

 So now let's look at the resolution issue. The standard for CD audio is 44,100 kHz. This will theoretically give response to 22,050 Hz, according to the Nyquist limit. It is extended to 22,050 Hz to yield a flat composite response to 20,000 Hz when the multi-pole low-pass filter is factored in. Otherwise you would hear the quantization noise. But how do we listen to music? We cannot wire sound directly into our brains, we must use speakers. A speaker cone cannot stop at one point, then magically re-appear at another. It has no choice but to travel through all intermediary points as it goes from peak to peak. Thus, a speaker creates it's own infinite level of quantization while it reproduces the source. They don't mention that either, do they?

 But here's the most puzzling thing of all. The common blurb for transfer websites is "Our transfers have higher resolution because we record at 24 bits at such-and-such kHz and then downsample to make the CD." Great, but for the fact that whenever you resample, you ADD NOISE. So what they're saying is "Our transfers sound better because we add noise to the signal."

 When you figure THAT one out, please let me know.


----------



## LFF

I know the above deals with LP to CD only. Here is a paper for any of those interested in anything Hi-Rez v. CD quality.


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Considering the sparseness of non-16/44.1 recordings, how can I do that_

 

Just about everything is recorded at 24 bit today. If people wanted it, there certainly could be companies catering to the demand. But luddites aren't why there are no 120dB range recordings... regular people are. You're the only one who is so fascinated with "accurate" numbers. Most people want listenable music. Assuming one's stereo can reproduce it accurately, a 120dB dynamic range would be extremely unpleasant to listen to.

 Here's a fun little experiment for you. Take a recording of a Mahler symphony and bump it up to 24 bit. With gain riding, try to use as much dynamics as the fomat will allow. Then play the thing back on your own stereo and see if you can stand it.

 See ya
 Steve


----------



## Crowbar

Quote:


  Originally Posted by *m3_arun* /img/forum/go_quote.gif 
_Theoretically, you don't have to know anything about the ear to perfectly recreate the original sonic environment._

 

It's common-sensical that transparency is not meant in absolute terms, for that is a physical impossibility even in theory (quantum uncertainty guarantees you cannot produce an absolutely identical sound field, with "identical" even relaxed to refer to the possible set of gauge transformations). So what should it be relative to? Obviously to aural perception, and thus the physiological and neural mechanisms involved with it! Why did you make me explain something so trivially obvious? Or perhaps you just like trolling?


----------



## Crowbar

Quote:


  Originally Posted by *LFF* /img/forum/go_quote.gif 
_Resolution is determined by sampling frequency, not bit depth._

 

Wrong: "the time resolution of nyquist sampled audio is rather much better than 1/fs (infinite in principle, the uncertainty of a single transient's time being limited by noise as in any bandwidth and SNR limited channel)" and "An analogue channel limited to 22kHz and 93dB SNR will have the same timing uncertainty as a dithered 44kHz/16bit digital channel"
 Note that SNR depends on bit depth, and the above says exactly that time resolution depends on SNR (which depends on bit depth)--*not* sampling rate.
 Source: Bruno Putzeys at PSW Recording Forums: Bruno Putzeys => What the hell is "Watts Transient Aligned filter"
 For those not aware, Putzeys was Philips' class-D chief, invented UcD, and has published in the AES, so I suggest one give his comments proper consideration.
 Now, I made an argument in the referenced thread that one can trade sampling rate for bit depth (and the reverse) but, as was pointed out, the tradeoff in practice cannot be made perfect (that's why, for example, 1-bit sigma-delta is not used in high quality converters, and the modulators are 5- or 6- bit nowadays).


----------



## regal

I think no reasonable person on the 24 bit side could possible consider dynamic range the crux of thier argument. Obviously 96dB is enough range, so now its resolution they are holding on to. I would say that if you are on the 24 bit camp you would have to think that a 60uV signal is not low enough. You believe that that a resolution smaller than 60 uV is needed? Or you would have to believe that a standard 2Vrms out is too small a signal. Now show me an amplifier and speaker that can actually differentiate a 60uV step sound. I would say not in any home. You would literally need to buy the Rollings Stones sound system before you could say a 60uV resolution isn't enough.


----------



## Crowbar

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_Obviously 96dB is enough range_

 

96 dB is undithered digital, which has serious quantization errors. Dithered 16 bit is 84 dB. How many times do I have to repeat basic facts because they won't get through regal's thick skull?

  Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_show me an amplifier and speaker that can actually reproduce a 60uV sound_

 

There are amplifiers that can produce 120 dB:
Home
 Other amplifiers that do 100-110 dB have been around since the 80s, including Bob Cordell's MOSFET one with Hawksford error correction.

 As for speakers, any speaker can produce any dynamic range that is above thermal noise in the coil on the low end and speaker mechanical failure at the high end.
 "the dynamic range of a conventional speaker is in the region of 100 dB"
 Source: Loudspeaker - US Patent 7116790
 Note that the patent is talking about dynamic range without significant distortion, so the SNR is actually higher. But let's stay with the conservative estimate of 100 dB and compare it to 84 dB. The 16 dB give a sound intensity difference of 40 times! Not even close.


----------



## 22906

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_It's common-sensical that transparency is not meant in absolute terms, for that is a physical impossibility even in theory (quantum uncertainty guarantees you cannot produce an absolutely identical sound field, with "identical" even relaxed to refer to the possible set of gauge transformations). So what should it be relative to? Obviously to aural perception, and thus the physiological and neural mechanisms involved with it! Why did you make me explain something so trivially obvious? Or perhaps you just like trolling?_

 

Lol. Chill out, I don't see why this is such an emotional subject.

 Absolute transparency is an easier ideal to work towards than tailoring the sound to the human ear, because all people hear things differently. And, I'm pretty sure classical theory gives us a very good approximation as to how sound works.


----------



## Crowbar

The point is absolute transparency is an impossibility, so your post made no sense. Transparency is only needed to the point where a difference cannot be detected by a human ear under optimal circumstances--so obviously those limits are set by otophysiology.


----------



## nick_charles

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_96 dB is undithered digital, which has serious quantization errors. Dithered 16 bit is 84 dB. How many times do I have to repeat basic facts because they won't get through regal's thick skull?_

 

Bob Katz disagrees with you on the relevance of this figure, arguing that the subjective dynamic range is more like 115db for a signal that has been dithered when converted from 24 bits down to 16 bit, which as I understand it is how things are generally done these days. 

Digital Domain - Dither


----------



## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_96 dB is undithered digital, which has serious quantization errors. Dithered 16 bit is 84 dB._

 

84dB is still MUCH more than any music designed for playback in the home. You are insisting that you require 120dB of dynamic range because some speaker design book you read said that human ears can hear that range. You haven't taken into account that anything over 50dB is uncomfortable to listen to, because it requires such a high volume level on playback.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_There are amplifiers that can produce 120 dB_

 

...all the way down to 20Hz? Assuming you have speakers that give you flat response down to 20Hz, how much power would you need to push them? How much cost are we talking about to put together this theoretical system? Do you foresee the average public wanting to buy into 120dB recordings that play back with all 120dB? I sure don't! I don't even think you would listen to them more than once.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_How many times do I have to repeat basic facts because they won't get through regal's thick skull?_

 

If you're going to continue with the insults and ad hominem attacks, folks are going to start offering you the same lack of respect. We've been very patient with you so far in this thread. Patience has its limits. Act your age.

 See ya
 Steve


----------



## gregorio

Bigshot - Don't let Crowbar get to you, he's doing it deliberately. He's taking the thread round and round in circles because he doesn't have a clue how music works or how it's recorded. Because he's too immature to admit he's wrong, he's going to try his best to spread confusion and mis-information in the hope of convincing some readers that he's not wrong. It's a shame that Crowbar and one or two others have or are trying to sabotage a thread which could help dispel some myths and allow more informed choice of equipment rather than relying on some of the manipulative marketing hype about 24bit. Let him make a 24bit recording with a 120dB range and then we can sit back, watch his customers complaining and suing him and have a good laugh over a glass of beer! This would all be quite funny if what he was suggesting wasn't so dangerous. He's full of BS but if by some horrendous quirk of fate he does make this 120dB recording, it will be interesting to see how many people are actually killed by it. And no, I'm not joking, there will be some people who will die as a direct consequence of trying to listen to a recording with a 120dB dynamic range! What was his latest? Oh yes, bit depth and sample rate are interchangable, nice facts and figures to prove it, true genius, complete boll*cks! I can't wait for that new CD format of 2048bit resolution and 1 sample per hour. Perfect for recording that "on location" low frequency audio of galaxies colliding ... don't forget to take a large diameter condenser microphone and an umbrella!

 Regal - I've followed your sub-thread and can't believe it, have you actually read any of this thread? Stepped output, 0.36dB per quantisation step, what are you talking about? You're looking at a pretty picture of a digital waveform and thinking, "well how does the speaker respond to those little steps". I like your thinking and your argument, except for one thing, it's all complete and utter cr*p!!! There is no stepped output, and what's this extra resolution you're talking about, there isn't any of that either. The resolution ONLY defines the dynamic range, nothing else! There isn't any resolution left over to make music sound better! This isn't just my opinion, this is how digital audio works and you won't find a single serious scientist or article in any respected journal anywhere in the world which disputes it ... why are you?

 Nick - Who is Bob Katz? OK, so he's arguably the most respected and influential mastering engineer the world has ever known. Means nothing to me, I think we should all believe Crowbar instead, after all, he's posted on forums, seen hi-res photos of real recording studios and done lots of interesting things!!


----------



## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_the subjective dynamic range is more like 115db_

 

From that site:
 "But the dynamic range is far greater, as much as 115 dB, because we can hear music below the noise."
 We can indeed hear signals below the noise floor--but that signal must be encoded in the first place. With noise shaped dither you could achieve such a dynamic range, but only in a narrow part of the audio band--at the expense of more noise in other parts of the band. And with 44.1 kHz audio most of the noise will be in the audio band since you can't stuff all the noise in the 20 kHz to 22 kHz gap margin you have available.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_He's full of BS....how many people are actually killed by it._

 

How exactly are you not full of BS for suggesting that something is dangerous when the graph I referenced shows that 120 dB is perfectly safe unless sustained continuously for some time? Can you answer this?

  Quote:


 The resolution ONLY defines the dynamic range, nothing else! 
 

*Only if you have theoretically perfect dither--not achievable in practice*. The issue is touched upon by Putzeys in the thread I referenced above. I suggest you read his posts in that thread (both pages).


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## bigshot

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Bigshot - Don't let Crowbar get to you, he's doing it deliberately._

 

Yeah... I know that. My comment wasn't made in anger or frustration. It was the proverbial shot across the bow.

 See ya
 Steve


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## Crowbar

I'm not trying to "get to" anyone. My concern is with misinformation being spread in this thread, and it would be unethical for me to allow false beliefs to be promoted unchallenged.


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## monolith

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I'm not trying to "get to" anyone. My concern is with misinformation being spread in this thread, and it would be unethical for me to allow false beliefs to be promoted unchallenged._

 

After you say that, I feel forced to post this (regardless of who's right here, because I don't know that myself):


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## Crowbar

Dude, your avatar is scary!

 [edit: I took the pic out since some people around here have thin skin]


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## monolith

Another classic.

 Anyway, back to the regularly scheduled discussion. I'm finding it quite interesting.


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## gregorio

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_How exactly are you not full of BS for suggesting that something is dangerous when the graph I referenced shows that 120 dB is perfectly safe unless sustained continuously for some time? Can you answer this?

 Only if you have theoretically perfect dither--not achievable in practice. The issue is touched upon by Putzeys in the thread I referenced above. I suggest you read his posts in that thread (both pages)._

 

Oh, you've got a graph, that's different then! If you've got a 20 year old in perfect health, 120dB probably won't damage them long term unless they listen at this volume often. However, someone with say an undiagnosed heart condition could easily be killed. Sorry, have I damaged your theoretical little world, would it help if I made you a pretty graph? Thanks for proving again that you haven't a clue about real audio or how real people respond to it.

 It's got nothing to do with perfect dither. Any old dither, noise-shaped or not will provide a perfectly linear output. That's the whole point of digital audio! You've read the proofs of this and seen the articles we've put links to earlier in this thread. What part of it do you think all the real scientists and recording professionals in the world have got wrong?

 "My concern is with misinformation being spread in this thread, and it would be unethical for me to allow false beliefs to be promoted unchallenged."

 HAHAHAHA, truely priceless!!! The promotion of misinformation and false beliefs describes quite accurately pretty much every message you have ever posted to this thread!! Who on earth are you trying to convince and why? I'm trying to understand the motivation for your ridiculous posts, beyond the simplest conclusion which is that you're a compulsively lying misanthrope with a superiority complex.


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## gregorio

Just for the benefit of others, please ignore the rubbish that Crowbar is trying to peddle. For example, with dither the dynamic range of 16bit audio is 84dB. This is a lie, designed to mislead you. The dynamic range of CD is 96dB, not 84dB. Dither does not change this 96dB figure, indeed, this figure of 96dB can be extended (to about 115dB) because as Bob Katz states, we can hear below the noise floor with noise-shaped dither. This is because the noise is moved into areas where we can't hear it, leaving just the music. Again, Crowbar is lying when he states that only a narrow band of freqs are free from noise.

 BTW, while you're at it, ignore the last link Crowbar posted to a message on a recording site. The message is out of context and deals with self dithering during the 24bit ADC process and truncation effects during re-quantisation and could be misleading if applied to our argument of 16bit. Heaven forbid that Crowbar would deliberately try to mislead us though!


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## regal

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_
 Regal - I've followed your sub-thread and can't believe it, have you actually read any of this thread? Stepped output, 0.36dB per quantisation step, what are you talking about? )_

 



 A DAC output covers continuously its dynamic range per the digital theory, but it can only "stop" on discreet values depending on the range. Do you think a DAC output is infintisimal? A 16 bit DAC can only output 32767 levels, in between these is not a step as you thought I was saying, but the DAC cannot hold between the levels. You are badly misunderstanding the theory if you think that, but again I think you are just fighting a lanuage barrier, so quit calling people's post's crap and spend some time in England or the USA so you can learn to understand people.

 Now my point is that going above 32767 levels is pointless with modern drivers in a home setting. 100,000W system maybe.


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## gregorio

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_A DAC output covers continuously its dynamic range per the digital theory, but it can only "stop" on discreet values depending on the range. Do you think a DAC output is infintisimal?_

 

That's exactly what I think and know, along with every other informed audio professional on the planet! The output of a DAC is linear ... hello do you speaky English, do you even know what a linear output means? A 16bit DAC outputs a continuously varying voltage (analogue sine waves) with infinte resolution, ie. Perfectly, not just 32,000 or any other finite number of different levels. What goes into a DAC is a quantised digital datastream, what comes out has been converted into sine waves, that's why it's called a digital to analogue converter, duh!

 "If a function f(t) contains no frequencies higher than W cps, it is completely determined by giving its ordinates at a series of points spaced 1/(2W) seconds apart. 

 In essence this means that an analog signal that has been digitized can be *perfectly* reconstructed if the sampling rate was 1/(2W) seconds, where W is the highest frequency in the original signal." (Wikipedia: Nyquist-Shannon Sampling Theorum).

 This isn't rocket science, it's rule one lesson one of digital audio, surely you haven't managed to get all the way through this thread without realising this simple fact? Notice that the statement above did not mention bit depths, that's because the statement is true at any bit depth.

 Regal - you're right, I am fighting a language (or lanuage as you spell it) barrier, I'm speaking English and you're speaking garbage!! If you spent some time on planet Earth instead of in cookoo-land maybe you can learn not to talk utter crap!!


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## regal

A Dac outputs an alternating current or voltage with an RMS power level. There is no such thing as a sine wave without an RMS level outside of theory. A DAC chip cannot output an infinite level of power or division of power. If it could we would not even need 16 bit DAC's, just a lot of amplification. But I forgot all you know is theory, no practical knowledge of electronics.


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## gregorio

Quote:


  Originally Posted by *regal* /img/forum/go_quote.gif 
_A Dac outputs an alternating current or voltage with an RMS power level. There is no such thing as a sine wave without an RMS level outside of theory. A DAC chip cannot output an infinite level of power or division of power. If it could we would not even need 16 bit DAC's, just a lot of amplification. But I forgot all you know is theory, no practical knowledge of electronics._

 

Of course a DAC cannot output an infinte level of power, why do you think you need an amp? But it can only output an infinite division of the amplitude, anything else would not be analogue audio. 

 "If it could we would not even need 16 bit DAC's". Hurrah, are you starting to understand or are you demonstrating your ignorance again? You're absolutely right, we don't need 16bit DACs, a 1bit DAC can reproduce excellent quality audio, or do you think that CD is 32765 times higher quality than SACD? But I forgot, you've got no understanding of the theory and no understanding of the practical application either, in fact, you're going for a fully comprehensive level of ignorance!!


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## regal

So how are you going to signal a DAC chip to output 100,000 different RMS power levels with a 16 bit data stream ?


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## gregorio

So how are you going to do it with a 1 bit data stream from an SACD? Can't you read English, neither 1bit, 16bit or 24bit DACs can output 100,000 different values, they can only output sine waves, which have an infinte number of values because they are CONTINUOSLY varying!


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## regal

Sine waves have to have an amplitude in the physical world, my God its EE101.


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## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Of course a DAC cannot output an infinte level of power, why do you think you need an amp? But it can only output an infinite division of the amplitude, anything else would not be analogue audio. _

 

Sorry, but this is wrong. No system with a finite bandwidth and a finite dynamic range can be other than finite. Continuous and infinite are not the same thing. A 16 bit DAC has a determined level of precision and infinite it cannot be, it may (or may not) do a _perfect_ job of translating the supplied values into discrete voltage levels but it cannot for instance render a value represented as 1024.87654332 into a voltage as it cannot receive such a value. 

 DACs are also not always perfectly linear especially at the high frequency end it is common for a DAC to be out by 0.25 to .5db. Look at some actual measurements almost all DACs have a response that tails off slightly heading for 20K. 

 In the days of early CD players reviewers used to talk about whether a CD player could even manage 14 or 15 bits linearity, let alone the nominal 16 bits.


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## bigshot

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_DACs are also not always perfectly linear especially at the high frequency end it is common for a DAC to be out by 0.25 to .5db._

 

I think the technical term for that is "as close as dammit".

 See ya
 Steve


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## nick_charles

Quote:


  Originally Posted by *bigshot* /img/forum/go_quote.gif 
_I think the technical term for that is "as close as dammit".

 See ya
 Steve_

 

Sure in pragmatic terms you will get no argument from me here, but this is not the same as perfect, perfectly linear ,infinitely resolving , and so on. I would be just as A-R if someone said vinyl had infinite resolution. When you are using numbers to do something your precision is bounded by the precision of your number system. For most folks Pi is 3.14 but try sending a rocket to the moon using that........


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## bigshot

I listen to music with pragmatic ears!

 See ya
 Steve


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## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_However, someone with say an undiagnosed heart condition could easily be killed._

 

So now you're a cardiologist, eh? LOL, sorry, but that's now how it works.


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## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_This is because the noise is moved into areas where we can't hear it, leaving just the music._

 

Since you're dealing with 44.1 kHz audio, you can only move it in the 0-22.05 kHz band. You only have about 2 kHz lying outside the audio band, and there is no dither in existence that can stuff all the noise into that area. That means that if you use noise shaping, you will have increased noise within areas of the audio band. The average noise energy remains equivalent to that of the flat dither's -84 dB.

  Quote:


 The message is out of context 
 

Don't misrepresent what I did. I didn't post a link to a message but to a thread, and there are a number of posts relevant in that thread.

  Quote:


 deals with self dithering during the 24bit ADC process and truncation effects during re-quantisation and could be misleading if applied to our argument of 16bit. 
 

24 bit -> 16 bit is also requantization! The mathematics of dithering is the same; doesn't matter if you do it after the ADC or before DAC.


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## bigshot

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_You're an idiot_

 

pot... kettle... black.

 See ya
 Steve


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## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_What goes into a DAC is a quantised digital datastream, what comes out has been converted into sine waves, that's why it's called a digital to analogue converter, duh!_

 

The conversion from the stepped output of a DAC to a smoother signal is done by the analog filter. However, that just hides the lower resolution of a lower bit-depth word, sine if the steps are bigger, the filter in the time domain can be seen as doing interpolation. It still doesn't mean you have infinite detail--that is impossible since it implies infinite information density which is forbidden in physics. You're being purposely misleading.

  Quote:


 an analog signal that has been digitized can be *perfectly* reconstructed if the sampling rate was 1/(2W) seconds, where W is the highest frequency in the original signal." (Wikipedia: Nyquist-Shannon Sampling Theorum). 
 

The mathematics of Nyquist assume non-quantized samples (infinite precision). This has already been pointed out.


----------



## Crowbar

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_which have an infinte number of values because they are CONTINUOSLY varying!_

 

They have an infinite number of values but resolution is limited because they are produced by the analog reconstruction filter from a finite number of output voltages the DAC chip produces! These output waveforms are continuous but there are finite number of them that can be produced.


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## Crowbar

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Sorry, but this is wrong. No system with a finite bandwidth and a finite dynamic range can be other than finite. Continuous and infinite are not the same thing. A 16 bit DAC has a determined level of precision and infinite it cannot be, it may (or may not) do a perfect job of translating the supplied values into discrete voltage levels but it cannot for instance render a value represented as 1024.87654332 into a voltage as it cannot receive such a value._

 

It seems gregorio is confused because he's looking at the output of a DAC's analog reconstruction filter and assumes, incorrectly, that since the filter produces a continuous waveform that implies infinite resolution. That is, of course, false. It just converts the output signal from one basis to another. The basis functions of a DAC's output are step functions. The basis functions of the analog filter output are (approximately, since it's minimum phase not linear phase) sine waves. Low resolution in the DAC's staircase output is converted into low resolution expressed as increased smootheness of the output waveform. You're losing details either way.

 You cannot have infinite resolution signal of any sort--even an analog signal is not infinite in resolution.


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## Crowbar

[deleteme]


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## Zanth

I've locked the thread. 52 pages to read through to figure out exactly where I have to start deleting to clean up the mess. It will reopen at a later date.


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