192 kHz in real life ... quite surprising ...

[okydoky]

Originally Posted by Plush

Hiya, Oky--

You're sending us press releases from a wide array of manufacturers but you are not showing that you understand the actual design limitations that contribute to a/d and d/a performance. You're the guy who sticks almost completely to theoretical and textbook specifications of 24 bit audio.

hi,

i understand much better than you would like to think. i am not a manufacturer. you guys are like the biblical children taunting each other in the market place.

if someone voices an opinion, you complain that it is not referenced. if someone posts references, you complain that it is reference material.

i do not stick only to theoretical textbook specifications by any stretch of the imagination. that's a different guy.

Originally Posted by plush

Sorry, but this theoretical huge dynamic range and dc to light frequency response must live in the real studio world. It is further constrained by real world (actual) musical and extra-musical frequency responses and dynamic ranges.

In plain language, there is very little information useable to the ear at very high frequency and most musical sources don't have anywhere near 110 dB dynamic range.

what dc to light frequency response are you talking about, plush? the argument is that the impulse response results are relevant at all points in the bandwidth [whatever bandwidth you are comfortable with].

extended dynamic range in converters is desirable, and necessary, because signals are often not presented in a manner that exercises all of the bits, and signals vary in amplitude during the course of a musical performance, so you need lots more dynamic range [bit depth]. low amplitude signals [and components of signals] are resolved with fewer bits, so the higher up in the bit depth you can keep the signal, the better off you are.

Originally Posted by plush

Your arguments are familiar because these same arguments have been rehashed over and over and over again for more than 20 years. I have heard them for at least that long and participated in debates about the subjects that you now cite as new areas of discussion. They are not new and you are not adding anything new here at all.

careful, now you are showing your age!

i am the one that said earlier that all the "nyquist theorum" stuff is old news. but there do seem to be a lot of people who do not understand it, or who think that it means that digital audio is perfect, or any number of other misconceptions.

Originally Posted by plush

I have the impression that you do not understand what you are talking about. Not right.

well, i am sorry you got the wrong impression.

Originally Posted by plush

Please don't send in any more pr materials from manufacturers or fake promotional data sheets from audio for video codec vendors.

well i am obviously not sending anything to you directly, and it seems to me that you may be being presumptuous by calling things "fake" with no basis for the statement. but i will leave all that to the manufacturers and authors that you are attacking or disputing

the best thing for someone of your stated knowledge and experience might be for you to simply not read my posts at all, since they are not directed to you anyway, and God knows you probably have heard everything that could ever be said about these things over many years as an accomplished industry veteran professional. why should you spend time reading stuff that is way beneath you.

i remember playing through a "plush" guitar amp years and years ago. i think it sounded good.

take care, and thanks for all you contributions [here, there, and everywhere]!

your wisdom is appreciated.


right.

[Ozpeter]

low amplitude signals [and components of signals] are resolved with fewer bits, so the higher up in the bit depth you can keep the signal, the better off you are.

Eh? If you compare a 16 bit recording vs a 24 bit recording, the additional 8 bits are added at the bottom, not somehow distributed across the full dynamic range. So the close-to-inaudible part of the dynamic range has a further 8 bits added to describe it. That's why when recording in 24 bits it isn't nearly so important to peak close to full scale - which is the chief point in using 24 bits to record.

I really think this thread should be locked.

[Plush]

not right.

[okydoky]

Originally Posted by Ozpeter

Eh? If you compare a 16 bit recording vs a 24 bit recording, the additional 8 bits are added at the bottom, not somehow distributed across the full dynamic range. So the close-to-inaudible part of the dynamic range has a further 8 bits added to describe it. That's why when recording in 24 bits it isn't nearly so important to peak close to full scale - which is the chief point in using 24 bits to record.

I really think this thread should be locked.

hi,

well, i am not here to argue or teach, but 24 bit dynamic range is not a free pass to record at low, or unnecessarily reduced, levels, like some people may think. there is a lot of confusion about all that, apparently. most of it has to do with people trying to avoid clipping.

i'm not sure why you are referencing 16-bit recording. nobody was discussing that, so i am not sure what you are trying to get at there.

anyhow, 24 bits is not "the best thing ever", its just the best thing that current component will reasonably allow. you're not even getting 24 bits, as a practical matter.

another thing that is often overlooked is the fact that complex musical signals are made up lots of low level components. so when you see your piano peaking nicely at -3dBfs, it does not mean that all aspects of the sound are at that level. some harmonics are significantly lower in amplitude, and they are still very important to the timbre of the instrument.

here a quote that you might find helpful. [wikipedia, probably] [emphasis added] [but don't be mad at me for just stating known facts].


"Analysis of the quantization error of low-amplitude signals reveals that the spectrum is a function of the input signal. The error is not noiselike (as with high-amplitude signals); it is correlated. At the system output, when the quantized sample values reconstruct the analog waveform, the in-band components of the error are contained in the output signal. Because quantization error is a function of the original signal, it cannot be described as noise; rather, it must be classified as distortion.

As noted, when quantization error is random from sample to sample, the rms quantization error E (sub)rms = Q(12) sup.1/2. This equation demonstrates that the magnitude of the error is independent of the amplitude of the input signal, but depends on the size of the quantization interval; the greater the number of intervals, the lower the distortion. However, the relevant number of intervals is not only the number of intervals in the quantizer, but also the number intervals used to quantize a particular level. A maximum peak-to-peak signal (as used in the preceding analysis) presents the best case scenario because all the quantization intervals are exercised. However, as signal level decreases, fewer and fewer levels are exercised as shown in Fig. 2.8. For example, given a 15-bit quantizer, a half-amplitude signal would be mapped into half of the intervals. Instead of 65,536 levels, it would se 32,768 intervals. In other words, it would be quantized with 15-bit resolution.

The problem increases as the signal level decreases. A very low-level signal, for example, might receive only single-bit quantization or might not be quantized at all. In other words, as the signal level decreases, the percentage of distortion increases. Although the distortion percentage might be extremely small with a high level, ) 0 dBFS, its percentage increases significantly at low-amplitude levels.

The error floor of a digital audio system differs from the noise floor of an analog system, because in a digital system the error is a function of the signal. the nature of quantization error varies with the amplitude and nature of the audio signal. For broadband, high amplitude input signals the quantization error is perceived similarly to white noise."


right.

[David Spearritt]

Originally Posted by Ozpeter

If you compare a 16 bit recording vs a 24 bit recording, the additional 8 bits are added at the bottom, not somehow distributed across the full dynamic range.

Not sure this is correct Peter. With a 24 bit recording, the amplitude quantization is 256 times more abundant uniformly. As an example:

16 bit recording: Plus or minus 32767 amplitude steps possible.
24 bit recording: Plus or minus 8388607 amplitude steps possible.

If we send a sinewave waveform of 1 V P-P to each system set so that 1V peak is 0dB, the maximum amplitude words recorded will be 32767 and 8388607 respectively.

Now if we reduce the signal to 0.5 V P-P, the amplitude words will be 16384 and 4194304 respectively, so the top 0.5V is sliced up into 4194304/16384 or 256 more steps.

[okydoky]

Originally Posted by David Spearritt

Not sure this is correct Peter. With a 24 bit recording, the amplitude quantization is 256 times more abundant uniformly.

hi,

correct. and the issue is that the ratio of the signal's amplitude to the amount of "steps" does worsen as you go lower in the bit depth. and of course quantization noise [to the extent that is of any moment]. there is effectively less accuracy down in the "basement". unavoidable with digital audio, it seems to me, dither notwithstanding.


right.

[David Rick]

The quote below (apparently from Wikipedia) is a pretty good summary of the trouble with undithered digital audio. But properly implemented dither converts a system with granular amplitude resolution into one without. (It does so at the cost of noise, but analog systems have noise as well.) End of argument.

Really, I think this has nothing to do with whether 192k converters are worthwhile or not. If it actually turns out that there's an audible advantage to them (and I'm keeping an open mind), it will not be because Nyquist, Shannon, et. al. were wrong. It will not be because signal detection theory is wrong. It will be because we failed to understand something important about human perception, and equipment designers were led to make poor tradeoffs due to that gap in understanding. (And at the low end of the market, it will be because they really didn't care!)

I think one of the more interesting avenues for investigation is how we should best shape the impulse response of digital decimation and anti-image filters. Peter Craven wrote an excellent paper on this subject: AES Preprint 5822.

Rather than rehashing the same old pseudo-technical arguments (and driving Dan Lavry nuts in the process!), we should be talking about how to do better experiments to find out what Ivo, me, and many others think they are hearing. If those experiments show that we aren't really hearing anything, I'm perfectly ok with that. But we should repeat these experiments every decade or so, because critical listening is a skill, and people continue to get better at it.

David L. Rick

"Analysis of the quantization error of low-amplitude signals reveals that the spectrum is a function of the input signal. The error is not noiselike (as with high-amplitude signals); it is correlated. At the system output, when the quantized sample values reconstruct the analog waveform, the in-band components of the error are contained in the output signal. Because quantization error is a function of the original signal, it cannot be described as noise; rather, it must be classified as distortion.

As noted, when quantization error is random from sample to sample, the rms quantization error E (sub)rms = Q(12) sup.1/2. This equation demonstrates that the magnitude of the error is independent of the amplitude of the input signal, but depends on the size of the quantization interval; the greater the number of intervals, the lower the distortion. However, the relevant number of intervals is not only the number of intervals in the quantizer, but also the number intervals used to quantize a particular level. A maximum peak-to-peak signal (as used in the preceding analysis) presents the best case scenario because all the quantization intervals are exercised. However, as signal level decreases, fewer and fewer levels are exercised as shown in Fig. 2.8. For example, given a 15-bit quantizer, a half-amplitude signal would be mapped into half of the intervals. Instead of 65,536 levels, it would se 32,768 intervals. In other words, it would be quantized with 15-bit resolution.

The problem increases as the signal level decreases. A very low-level signal, for example, might receive only single-bit quantization or might not be quantized at all. In other words, as the signal level decreases, the percentage of distortion increases. Although the distortion percentage might be extremely small with a high level, ) 0 dBFS, its percentage increases significantly at low-amplitude levels.

[okydoky]

Originally Posted by David Rick

The quote below (apparently from Wikipedia) is a pretty good summary of the trouble with undithered digital audio. But properly implemented dither converts a system with granular amplitude resolution into one without. (It does so at the cost of noise, but analog systems have noise as well.) End of argument.

hi,

yes, but it does not improve accuracy. it merely removes quantization noise.

i am not sure what you are getting at. dither only goes so far, and it is not without its own downside.

Originally Posted by david rick

Really, I think this has nothing to do with whether 192k converters are worthwhile or not. If it actually turns out that there's an audible advantage to them (and I'm keeping an open mind), it will not be because Nyquist, Shannon, et. al. were wrong. It will not be because signal detection theory is wrong. It will be because we failed to understand something important about human perception, and equipment designers were led to make poor tradeoffs due to that gap in understanding. (And at the low end of the market, it will be because they really didn't care!)

respectfully, there is no doubt among people using them that 192kHz converters are "worthwhile". that is not even an issue in many places outside of these types of discussions.

sigma-delta converters with the amount of dsp that lavry and other throw at them to get anywhere near the touted performance have significant latency that removes them from serious consideration for tracking and overdubbing in rock and pop, unless you are direct monitoring. they might squeak by at 192kHz in some situations [live tracking of large groups, maybe], by that will be a cold day, as they say.

the need to minimize latency will only increase.

as to "audible advantage", people should realize that the issue is not necessarily only what can be perceived and identified in the moment, but also the long term effects of exposure to stimulus [such as digital audio] over time.

and it is clear that people can be dumbed down pretty quick, too.

so i am with you on the idea of keeping on top of things and reevaluating and trying to make progress.

Originally Posted by david rick

I think one of the more interesting avenues for investigation is how we should best shape the impulse response of digital decimation and anti-image filters. Peter Craven wrote an excellent paper on this subject: AES Preprint 5822.

i should read that before saying anything about it.

Originally Posted by david rick

Rather than rehashing the same old pseudo-technical arguments (and driving Dan Lavry nuts in the process!), we should be talking about how to do better experiments to find out what Ivo, me, and many others think they are hearing. If those experiments show that we aren't really hearing anything, I'm perfectly ok with that. But we should repeat these experiments every decade or so, because critical listening is a skill, and people continue to get better at it.

David L. Rick

please understand that "driving dan lavry nuts" is not even remotely a factor in any of this. that sounds silly. and nobody here does not understand dan lavry's argument or anything like that. it is found to be incomplete, not only by me, but by many other people. however, he is entitled to have his own view of things, just as others are.

dan has his very own web site, and it is expertly set up to discuss things in the manner that he wants to discuss them. its not like he moved to a different planet, and its not like you can't get him on the phone. i'll bet he would be pleased as punch to have a lot of you guys join his forum, too!

its usually weird when any manufacturer spends inordinate amounts of time arguing their positions away from their own web site.

you cannot reasonably expect the entire audio community [or at least the gearslutz community] to walk on eggshells to make sure nothing is said that contradicts one person's opinion.

people talk about all kinds of crazy stuff on these boards that is diametrically opposed to the truth, as far as i am concerned. i don't freak out about it. there are professional organizations for people that want to limit membership to those who share there own views.


right.

[Plush]

Sorry, but the whole reason that dither is used IS to improve accuracy. That is why we use dither. It toggles the least significant bits.

Oky--may I refer you to the book, The Art of Digital Audio by John Watkinson ?(Focal Press)

At least by reading this book you will be able to make a halfway credible discussion on some internet forums. You are not making a credible discussion impact at this time.

[Chris Lago]

Wow this thread is still going strong? What a waste. Get back in the studio guys (I'm waiting on my 8core Mac Pro delivery... Damnit).

[Ozpeter]

Not sure this is correct Peter. With a 24 bit recording, the amplitude quantization is 256 times more abundant uniformly.

David, you've shocked me! If what you say is true, then truncating a 24 bit recording to 16 bits wouldn't work, would it?

Bits and resolution - the way I've seen it explained elsewhere is like this.

Imagine you have a 16 bit ruler and a 24 bit ruler.

If the 16-bit ruler measured 100 cm (or 1000 mm) the 24 bit one should measure 150 cm (or 1500 mm).

The theoretical maximum dynamic range of a 16-bit recording is 96 dB and that the theoretical maximum dynamic range of a 24-bit recording is 144 dB, so the 24-bit ruler should be 1.5 times the length of the 16-bit ruler.

But in neither ruler would the gradations be evenly spaced. Sound (as expressed in dB) is logarithmic. As we went from the top of the rulers down to the bottom, the gradations would get closer and closer together. They would line up perfectly until we got to the bottom of the shorter ruler, after which they would continue to get closer and closer together until they got to the bottom of the second ruler. In this example, the 16-bit ruler is divided into 65,536 parts and the 24-bit ruler is divided into 16,777,216 parts. But remember, those parts are not evenly spaced. Using our 100 and 150 cm rulers, the "top" 100 cm of the 150 cm ruler is divided into 65,536 parts, and the "bottom" 50 are divided into 16,711,680. All that extra resolution is used to describe the "bottom" 48 dB dynamic range.

You get those extra 48 dB (theoretically) below the 96 dB you've already
got, not above. It's very simple to test this. Take a 16-bit signal that peaks at 9 dBFS (your "fully used" 16-bit signal) and run it into a device that has a 24-bit digital input. Does it peak at -48dB? No, it peaks at 0. 0 dBFS is always 0dBFS, whether it's on an 8-bit signal or a 24-bit signal. You can't go any higher, just lower.

[okydoky]

Originally Posted by Plush

Sorry, but the whole reason that dither is used IS to improve accuracy. That is why we use dither. It toggles the least significant bits.

hi,

no, old boy. you are confusing resolution with accuracy. yes, it does have the effect of toggling the lsb. more specificially, it toggles them randomly [otherwise it does not work]. that does not necessarily improve accuracy. it removes quantization noise.

"The results of the process still yield distortion, but the distortion is of a random nature so its result is effectively noise. Any bit-reduction process should add dither to the waveform before the reduction is performed." [dither - wikipedia (citing Pohlmann, Ken: Prinicples of Digita Audio, inter alia].

"Note that dither can only increase the resolution of a sampler, it cannot improve the linearity, and thus accuracy does not necessarily improve." [analog to digital converters - wikipedia (citing Kester, Walt: The Data Conversion Handbook, inter alia)].


i am going to overlook all the snide remarks.



right.

[okydoky]

Originally Posted by Ozpeter

David, you've shocked me! If what you say is true, then truncating a 24 bit recording to 16 bits wouldn't work, would it?

hi,

guess what, your right, it [truncation] doesn't work.

you're way off on the rest of your post. the bits are "evenly spaced" [except in some more exotic purpose-built sampling devices not useful in audio, where only a small part of the signal's dynamic range is of interest]

you appear to be confusing dBfs with bit resolution, among other things.




right.

[Ozpeter]

you're way off on the rest of your post. the bits are "evenly spaced" [except in some more exotic purpose-built sampling devices not useful in audio, where only a small part of the signal's dynamic range is of interest]

In this forum, I shouldn't have to be dealing with very common misconceptions about how digital audio works, but here goes... I'll rehash some good explanations from elsewhere again and start from the ground up -

The first digit of a 16-bit word is called the Most Significant Bit (MSB). Thinking about a waveform as going up and down and up and down over a zero point, when the value is positive, the MSB is a "0", and when it's negative the MSB is a "1". So the "loudest" positive signal would be 0111 1111 1111 1111 (which is 32,767 expressed in base ten). A signal that's 6 dB down would be 0011 1111 1111 1111, or 16383. (32768, or 1000
0000 0000 0000, would actually be the lowest possible position below the zero crossing, which for all intents and purposes would be the same thing.)

Si we're not going from 1111 1111 1111 1111 down to 0000 0000 0000 0000, but from 0111 1111 1111 1111 (as close to 0 dB FS as we can get on the positive side of the zero crossing) down to 0000 0000 0000 0001 (one "step" above the zero crossing) to 0000 0000 0000 0000 (the zero crossing itself) to 1111 1111 1111 1111 (one step below the zero crossing) to 1000 0000 0000 0000 (the maximum amplitude on the negative side of the zero crossing). As we move to the right, each bit repesents a value closer and closer to the zero crossing, which is why as we increase our bit depth we can capture a wider dynamic range...we're capturing those signals closer to the zero crossing with more resolution. With 16 bits we can theoretically capture signals up to 96 dB below FS accurately. With 24 we can get closer and closer to the zero crossing and (theoretically) capture signals up to 144 dB down. Those extra bits are used when representing louder signals as well, but since they're representing such a quiet component of the signal they make no audible difference whatsoever.

Bear in mind that with digital audio we're always counting down from the top, not up. So putting it the most simple possible way, it's not unlike counting down from zero using two digits -

-01, -02, -03, -04, -05... -99

Now we could do the same with three digits -

-001, -002, -003, -004, -005... -099... -999

Now it's obvious that -01 = -001. And the two counting systems remain the same until the two digit system runs out of values at -99 (=-099). Then the three digit system carries on downwards from -100 down to -999. The fact that it does that doesn't alter the fact that the first 99 values are the same. It just goes a lot further down.

All of this is of course way off the original topic, but it's important not to let wildly incorrect statements about the way digital audio works lie unchallenged - otherwise people read and believe them - especially in a forum which considers itself to conduct its discussions at a professional standard.

[klaukholm]

Ozpeter, thanks for the excellent explanation.

Would it be possible for you to start a new thread on the topic of bit depth and audio, with the above post as the starting post.

[Ozpeter]

Would it be possible for you to start a new thread on the topic of bit depth and audio, with the above post as the starting post.

I would hope that it wouldn't be necessary!

Well, OK, but I would trust that it would be a very short discussion.

[klaukholm]

Unfortunately I think it is.
There are a lot of misconceptions out there, and wikipeida is not helping.

I was not meaning that it should be a discussion. I was hoping for a few concise and accurate posts followed by a lock and sticky by Admin.

[ISedlacek]

Quite few pages before, I already wrote that the only reasonable reply to the original post would be to say: I have tried and compared and my impression is .... (whatever). It would make a real sense. 200 post have passed and practically no one reported his own experience, just these lifeless theoretical battles ) Please, try it for yourself and let us know the results ... Anything else is just very little useful for the practical life.

[klaukholm]

Originally Posted by oky****

here are some relevant quotes you requested, and a link to an article they are taken from:

right.

I did not ask for any quotes.

Originally Posted by oky****

i believe that the poster who said "analog is infinite resolution" [or words to that effect] is correct, particularly within the context that he was obviously speaking.

The reason why I responded to you was your misuse of the term infinity which has a very clear and specific meaning in the word of physics, and this is a misuse of that term.

[okydoky]

Originally Posted by Ozpeter

In this forum, I shouldn't have to be dealing with very common misconceptions about how digital audio works, but here goes... I'll rehash some good explanations from elsewhere again and start from the ground up -

The first digit of a 16-bit word is called the Most Significant Bit (MSB). Thinking about a waveform as going up and down and up and down over a zero point, when the value is positive, the MSB is a "0", and when it's negative the MSB is a "1". So the "loudest" positive signal would be 0111 1111 1111 1111 (which is 32,767 expressed in base ten). A signal that's 6 dB down would be 0011 1111 1111 1111, or 16383. (32768, or 1000
0000 0000 0000, would actually be the lowest possible position below the zero crossing, which for all intents and purposes would be the same thing.)

Si we're not going from 1111 1111 1111 1111 down to 0000 0000 0000 0000, but from 0111 1111 1111 1111 (as close to 0 dB FS as we can get on the positive side of the zero crossing) down to 0000 0000 0000 0001 (one "step" above the zero crossing) to 0000 0000 0000 0000 (the zero crossing itself) to 1111 1111 1111 1111 (one step below the zero crossing) to 1000 0000 0000 0000 (the maximum amplitude on the negative side of the zero crossing). As we move to the right, each bit repesents a value closer and closer to the zero crossing, which is why as we increase our bit depth we can capture a wider dynamic range...we're capturing those signals closer to the zero crossing with more resolution. With 16 bits we can theoretically capture signals up to 96 dB below FS accurately. With 24 we can get closer and closer to the zero crossing and (theoretically) capture signals up to 144 dB down. Those extra bits are used when representing louder signals as well, but since they're representing such a quiet component of the signal they make no audible difference whatsoever.

Bear in mind that with digital audio we're always counting down from the top, not up. So putting it the most simple possible way, it's not unlike counting down from zero using two digits -

-01, -02, -03, -04, -05... -99

Now we could do the same with three digits -

-001, -002, -003, -004, -005... -099... -999

Now it's obvious that -01 = -001. And the two counting systems remain the same until the two digit system runs out of values at -99 (=-099). Then the three digit system carries on downwards from -100 down to -999. The fact that it does that doesn't alter the fact that the first 99 values are the same. It just goes a lot further down.

All of this is of course way off the original topic, but it's important not to let wildly incorrect statements about the way digital audio works lie unchallenged - otherwise people read and believe them - especially in a forum which considers itself to conduct its discussions at a professional standard.

hi,

you apparently just don't want to understand. your analysis is wildly incorrect. in 24 bit audio, the louder components are also resolved with greater accuracy than they are in 16 bit audio.

you guys are certainly contentious, but uninformed. boring.


right.

[okydoky]

Originally Posted by klaukholm

I did not ask for any quotes.

hi,

you seemed to be asking to me.

Originally Posted by klaukholm

The reason why I responded to you was your misuse of the term infinity which has a very clear and specific meaning in the word of physics, and this is a misuse of that term.

i'll be blunt. i think the reason you are responding is because you are mad that i am posting anything at all.

you seem to think that this forum is your turf or something, and you are lobbying to exclude me.

i think that's childish, but i have been overlooking it for the most part.

i did not use the term "infinity", nor do i believe it was "misused". are you a physicist?


right.

[Ozpeter]

you apparently just don't want to understand. your analysis is wildly incorrect. in 24 bit audio, the louder components are also resolved with greater accuracy than they are in 16 bit audio.

As this is a side issue to the original purpose of this discussion, it's now been peeled off to Bit depth revisited - I won't pursue it further here.

[Remoteness]

Come on folks - in the future please consider working out your differences via PMs and/or emails.

It's just not fair to everyone else, especially when it gets ugly.

Let us learn from our mistakes and move on to a better place.