The Optimal Sample Rate for Quality Audio

[Jud]

Find out why “more” is not always “better!”

 

Lavry tech

I would be interested to know what connections (if any) you have with this commercial product.

That statement sounds to me, like something that could have come from their Advertising Department.

 

"Lavry tech" works for Lavry. He's posted in the forums before on behalf of Lavry and as his nickname indicates, is completely forthcoming about his identity and affiliation. So nothing at all untoward about this, he's simply announcing to the forum the availability of Lavry's most recent paper.

 

The paper contains a nice explanation of how, as long as one satisfies the Shannon-Nyquist conditions (sample at over 2x the highest frequency of interest), one may thereby specify two musical signals with an arbitrary time differential between them. That is, the time between samples doesn't have to be equal to or shorter than the time differential between the musical signals. The signals are completely specified as long as Shannon-Nyquist requirements are satisfied, and faster sampling rates won't specify the time differential more exactly.

 

There are two areas related to issues raised by the paper on which I'd like to comment.

 

- An aspect of timing not discussed by the paper is transient response where the initial inharmonic attack of an instrument has a rise time faster than the wavefront of a 22.05kHz sine wave. If the steepness of that initial inharmonic attack transient is something that our ears and brains react to (as distinct from a continuous harmonic tone at 22.05kHz, which I think we can agree is inaudible), then it raises the possibility that systems based on reproduction of the audible harmonic range may not be adequate. Academic papers on the subject by Kunchur conclude people can discriminate such transients at about half the rise time of a 22.05kHz harmonic wave, indicating double the CD sample rate may be necessary to reproduce them *in the mathematically ideal case with perfect filtering*. Since filtering in the real world is not perfect, this would tend to support the use of sampling rates substantially above 2x CD rate.

 

- The Lavry paper first says there is no reason to retain any information above the audible range, then notes such information can have a substantial effect within the audible range. These two statements, it seems to me, are contradictory. If real world instruments produce ultrasonic information (and they do) which, as the Lavry paper admits (in fact emphasizes), affects the audible sound, then removing the ultrasonics alters the original audible sound, i.e., introduces distortion. So in order to faithfully reproduce the audible range, including the audible effects of the ultrasonics produced by the instruments, the goal should be to correctly reproduce the ultrasonics as well. Particularly with the limitations of real-world filtering referenced in the previous paragraph, this would seem again to point toward sampling rates above 2x the CD rate.

[Jud]

A fair amount of academic researchers have expressed concerns about Kunchur's work. There has been a number of attempts at replicating the results, and they have concluded that *if you remove the intermodulation distortion that causes the ultrasound to produce intermodulation components in the audible spectrum*, there are no audible difference if the ultrasonic component is removed.

 

I'd appreciate references if you have them handy. Anything that helps me better understand the issues is always welcome.

 

It might be the other way around - the ultrasonic information might cause intermodulation distortion that is audible in the sonic frequency range.

 

Yes, absolutely. My point was that the electrical impulse representing the ultrasonic and audible range waves cannot "know" whether their intermodulation creates distortion or a correct representation of an original event in which both ultrasonics and audible range waves were involved. Once one grants that ultrasonics affect audible frequencies, one cannot then simply declare that such interaction is always distortion and by declaring make it so.

 

Granting the premise that ultrasonics affect audible frequencies, one has two choices if there are ultrasonics in the original event: Keep the ultrasonics or attempt to remove them. If one keeps the ultrasonics, this is an argument for greater than 2x CD samplng rates in order to get to the Shannon-Nyquist frequency and then leave sufficient headroom for non "brick wall" filtering. If one attempts to remove the ultrasonics (which to my mind is in itself distortion, since one is removing something that was present in the original), either one uses "brick wall" filtering to try to remove them completely, which creates distortion; or one uses a more gently sloped filter, which leaves altered ultrasonics that will intermodulate with audible frequencies differently than the original, and which also may necessitate higher than 2x CD sampling rates. Thus in order to reproduce the original event properly, it seems best to try to reproduce the ultrasonics correctly.

[Jud]

Griesinger, Perception of mid-frequency and high-frequency intermodulation distortion in loudspeakers, and its relationship to high definition audio (http://www.davidgriesinger.com/intermod.ppt)

 

Julf, I don't understand what you intend by this.

 

You said several academic researchers had expressed concern about Kunchur's work and were unable to reproduce that work. I was very interested to hear this, since I'd looked for such papers and been unable to find any, so I asked you to provide them. In return, you cite a single presentation by a non-academic (though a highly qualified person in certain areas of audio, mainly regarding reverberant vs. direct sound) that expresses no concerns whatever about Kunchur's work, one very good reason for which is because the presentation predates Kunchur's first paper on this subject by four years.

 

Furthermore, you stated the following with regard to the reason why "other researchers" were supposedly unable to duplicate Kunchur's work:

 

A fair amount of academic researchers have expressed concerns about Kunchur's work. There has been a number of attempts at replicating the results, and they have concluded that *if you remove the intermodulation distortion that causes the ultrasound to produce intermodulation components in the audible spectrum*, there are no audible difference if the ultrasonic component is removed.

 

But Kunchur's work on detecting time differences does not have to do with audibility of ultrasonic tones or their intermodulation products, so this cannot be a correct statement. The part of your statement above emphasized by asterisks is not a correct description of Kunchur's work but rather regards an experiment Griesinger ran for himself and reported in his 2003 presentation. In fact Griesinger is careful to state at slide 29 and again at slide 35 that audibility of time differences is an entirely distinct thing from the ultrasonic information he is concerned with, pointing out very small timing differences can be detected well within the audible band. This agrees completely with Kunchur 2007, where his experiments on timing differences were carried out using tones within the audible range.

 

So I am wondering if you have an explanation as to why you gave me this in response to my request for you to provide the references you claimed to have regarding concerns about Kunchur's work and repeatability of his experiments?

[Jud]

Originally Posted by Julf

A fair amount of academic researchers have expressed concerns about Kunchur's work.

 

How about Oohashi & co brain imaging results when sound with ultrasonics produces different brain responses than sound without ultrasonics?

 

Just like Kunchur, people were expressing concerns about Oohashi's work four years before it was done! ;-p

[Jud]

James Johnston has expressed doubts about Kunchur's results. I don't know if anyone has exactly replicated his experiment. But very similar experiments came up with fairly consistent results that don't agree with Kunchur. I am not sure it matters all that much either way. Kunchur basically said timing differences were audible to about half the accepted amount. Even 44.1 khz will have the timing accuracy for that. I believe, though might be remembering it wrong, that Kunchur is one saying 44.1 khz couldn't time accurately enough, and that is more where other people in the field disagreed with him.

 

Hi, Dennis. First of all, thanks as usual for referring me to something relevant I hadn't seen before. Having fun and learning (actually the latter pretty much assures the former) are my reasons for hanging out here.

 

Now, re Johnston and Kunchur: If you're referring to discussions in the Stereophile forums, I think Johnston's disagreement may be more with how Kunchur's work has been characterized than with Kunchur's specific conclusions. Johnston says 44.1kHz is good enough to specify the time difference in two signals to an arbitrarily short length. That's elementary sampling theory. As I explained above, I don't disagree; neither, I think, does Kunchur. In 2007, Kunchur was (at least as I minimally understand from reading the paper) talking strictly about audible range material. More recently, he has begun to talk about human ability to sense sharper amplitude rise times than occur in sound waves at the highest audible range. He has referred to this ability with the descriptor "ultrasonic," but always in quotes, i.e., not to be taken literally. I haven't seen anywhere that Johnston has questioned this, but he may have; frankly the signal-to-noise ratio at Stereophile was too much for me to make it to the end of the forum discussions. Anyway, you may be interested in slides from a 2010 presentation by Kunchur and others: http://www.physics.sc.edu/~kunchur/papers/Auditory-mechanisms-that-can-resolve-ultrasonic-time-scales.pdf

 

Have a look and let me know if Johnston has weighed in on this aspect of Kunchur's work specifically.

 

Now Oohashi's work is much more interesting. Especially as the ultrasonics don't go through the normal hearing channel of the ear. Instead they apparently stimulate the inner ear or some related area of the brain yet the ultrasonics enter through the eye sockets. Would like to know more about how high a frequency is needed, what ranges it covers etc. Maybe you should start a thread just on that Jud.

 

I'm considering it, but have some trepidation that more heat than light may result.

[Jud]

I believe this and Oohashi's work match quite well, and also with my own experience on passive sonar and such. It's not about hearing ultrasonic sine waves alone, it's about sensing waveform shapes and changes steeper than "audible" (as continuous sines) frequencies. I believe hearing is closer to wavelet analysis filter banks than Fourier frequency breakdown. (I've spent quite some time on WVD analysis and etc, perfectly matching these)

 

Kunchur is not a neurologist, but he does cite work by others regarding different parts of the brain being used to process continuous tones versus changes in amplitude. My recollection of reading some of the work he cites, and of reviewing the cited papers in Google Scholar, is that this (different parts of the brain being used) is not controversial.

[Jud]

Kunchurs experiments only shows that the ear can detect quite small timing differences. The problem with Kunchur is that he makes statements that, especially when taken out of context, imply that a 44.1 kHz sample rate system can't reproduce those timing differences (despite the fact that his experiments don't address that issue at all).

 

Not to rehash old discussions, but: Kunchur's experiments reported in 2007 did indeed focus as you say on timing differences between two continuous signals. For audible signals, Shannon-Nyquist proves 44.1kHz is adequate to handle such timing differences to any arbitrarily brief length of time.

 

But certainly by 2010, Kunchur was focusing more on the audibility of inharmonic transients with such a steep rise in so short a time that 44.1kHz sampling rates are not adequate to reproduce them. Kunchur sometimes refers to such signals as "ultrasonic" (always using quotes) to point out the paradoxical fact that humans appear capable of responding to them as audible stimuli even though the amplitude rises are steeper than those of sine waves at the highest audible range for continuous tones. It appears from neurological research, as Kunchur has pointed out, that different parts of the brain handle these transient stimuli than are used for continuous tones.

[Jud]

Meaning it [measurement] will usually eliminate the wrong factors until someone comes up with a whole new way of looking at things. That's science!

 

While we must be conscious that measurements are not the whole story, let's not disregard the fact that they're an inescapable part of figuring out why and how things happen as they do, and what might work to make them happen more along the lines you'd like them to. There are people like Keith Johnson in audio, or Richard Lenski in biology, who are extraordinarily painstaking and careful in designing and setting up tests or experiments and taking the data from them, and that data in turn assists immeasurably (heh) in helping these fields progress.

 

Intuitive leaps in science and engineering may have a grip on the public imagination, but to me, anyone who announces that there is a "whole new way of looking at things" has a virtually 100% chance of being a crackpot.

[Jud]

The only fairly recent work I can find is the AES panel contribution you already linked to. I find it interesting that the only actual experimental research work he refers to is the original work about temporal differences - all the rest is pretty much just theoretical speculation. It might be right, but it is still pretty much unconfirmed by any actual observations.

 

I do remember being frustrated that I couldn't find more clear-cut information in Kunchur's published work about rapid-rise inharmonic transients specifically. I will say that the evident care with which he designed, set up, and conducted his experimental work published in 2007 (building his own equipment in some critical cases to ensure it met the desired specs) went some way in convincing me he wasn't just making stuff up, but was discussing something about which he had some actual data.

 

There's also other published research work on such inharmonic transients, and on the differences in how they are processed in the brain versus continuous tones, as well as remarks by audio engineers in various sources (for one very informal example, see Miska in this thread) so it is not as if Kunchur is all alone in this. If he were, no matter how careful he appears to be in his experiments, I'd be considerably more skeptical.

[Jud]

It occurs to me to look at this question a slightly different way (thanks Barry for planting the germ of the idea).

 

Almost all DACs do oversampling to reach some much-higher-than-RedBook rate (e.g., 352.8 or 384kHz) before filtering. Doesn't this constitute at least a tacit, if not explicit, admission by most audio engineers who've worked on the problem that filtering at these much higher rates produces better results than filtering at RedBook sample rates?

 

This being so, what would be the preference of most on this board for attaining these higher rates - start at RedBook rates and interpolate most values through sample rate conversion, or start at rates that are as high as possible and interpolate few or even no values?

[Jud]

So? The ESS SABRE³² Reference ES9018 chip can upsample 24-bit 192 kHz material to no less than 1536 kHz, even (...and it uses a 32-bit internal data path to go with that).

 

Yes. The point is that since nearly all DACs already have sample rates of at least 352.8/384 going to the filtering process, why not feed them with native high sample rates, rather than RedBook, or 96kHz as Dan Lavry urges, then have the computer or DAC upsample by 4x or 8x just to reach the rate at which you're going to filter?

[Jud]

This is a very interesting way to look at it Jud.

 

My pure speculation as to why this might not be the best way is just that speculation. Is it more difficult to use 8x sample rates with traditional interfaces like AES/EBU? More jitter? Is sending an AES stream from something like a Mykerinos card more difficult at 8x? I also wonder if async USB is a completely different story when it comes to 8x.

 

Again, I have no idea just some thoughts.

 

I don't know either, though goldenpiggy and Julf have provided helpful responses.

 

Looking toward the near future, it seems to me we might point toward a "have our cake and eat it too" world, where we would no longer have to choose which we preferred, to remain "bit perfect" or feed 192kHz (or 352.8/384, or DSD) to the DAC. Unless there is some very good technical reason why 4x/8x/DSD bit rates are and will remain audibly inferior to RedBook, it seems to me the higher rates are what we ought to be aiming for as much as possible in recording, and then staying with those rates all the way into the DAC. Seeing the measurable differences between sample rate converters at SRC Comparisons, eliminating SRC in the chain to the extent possible may be a worthwhile goal.

[Jud]

Hi Jud,

 

As to your last point, about eliminating SRC in the chain being worthwhile, I think this is *very* dependent on the individual algorithm. As I mentioned above, I've heard algorithms that sound quite transparent to my ears and perhaps coincidentally, perhaps not, there are no measurable harmonics showing in the 1 kHz test. (See "iZotope 64-bit SRC, steep, no alias".) To my ears, the results, unlike most of the competition, are *not* brightened and hardened. They sound much like the unconverted original.

 

Hey, Barry. There are two reasons I was thinking eliminating SRC to the extent possible would be a good thing. The first was what you have said in the paragraph above should not be too much of a concern, at least with the better SRCs available. (Nice to hear that, and kind of confirms my listening preference to upsample to 192kHz with iZotope 64-bit.)

 

Why use SRC? * * * Further, if done at higher rates and the results later converted to 44.1 with a high quality algorithm (such as iZotope's), *some* of the benefits of the higher rates are preserved. In other words, I've found it creates a better sounding 44.1 version than if the SRC was eliminated and all mastering done at 44.1.

 

That's the other reason for staying at higher rates - not losing sample points and then having to interpolate them back. As well as some SRCs can do that second step (interpolating), why take the first (throwing away actual sample data) if you don't have to?

 

There will nearly always be some necessity for converting rates - as you mentioned, for the sake of EQ and other processing. But to the extent it can reasonably be avoided, and original data in the recording preserved, why not?

 

Julf - I'm not such a Polyanna as to imagine the music distribution companies will suddenly all decide RedBook (let alone mp3) should go the way of the dodo. I'm just suggesting that when we're thinking of what our computers should feed our DACs, and what our DACs should be able to process, to the extent practical we ought to aim for equipment and software that can preserve hi-res material in its original format all the way through the DAC; and furthermore, that we ought to aim for material that needs little or no oversampling once it hits the DAC, since I'm guessing SRC in the DAC may not work as well as something like, e.g., iZotope.

[Jud]

If that implies upsampling everything to the highest rate the DAC is capable of, I don't see much point in just adding empty air into the audio files.

 

That wasn't actually what I was trying to say. Whether to upsample or stay at the native rate is a subject of numerous lively discussions here and elsewhere with people whose opinions I respect on both sides. I personally like upsampling in the computer, but that is with my current DAC. Whether it would be the case with other DACs (at least those that do the typical oversampling) I don't know. (N.B., I think the interpolation facilities of the best SRC software are good enough that while the additional samples are not actual recorded samples, they are not quite "empty air" either.)

 

What I was referring to is that it would be quite nice to have material recorded in 176.4/192 or even 352.8/384, or DSD, that would require only 2x (in the case of 176.4/192) or no oversampling at all in either the computer or the DAC. There is at least some non-negligible amount of material available in 176.4/192 and DSD, though one could wish for both more and cheaper.

[Jud]

Hi Jud,

 

I believe we agree on this.

I was not arguing *for* SRC, just mentioning situations where I believe it provides a benefit.

 

Yep, understood.

 

In terms of converting to a higher sample rate for playback, I prefer not to simply because I'd rather listen to files at their native rate. If I *was* going to use SRC to attain a higher than native sample rate, I'd have to dedicate quite a bit of time to converting the music in my library using an off-line process. (Even the best SRC I've heard is, in my view, not at its best when performing the conversion in real time, i.e. during playback.) Since there are too many other things I prefer to do with my time, I listen to music in my library at the rate at which it was delivered.

 

Don't know if the iZotope packaged with Audirvana+ can do offline SRC. While it may not be at its best doing on-the-fly SRC (though I wonder what the effect is of Audirvana putting the file in memory before playback - does this mean the SRC is effectively accomplished offline?), I like it better than without, so that's what I choose in my particular setup.

 

Julf -

 

With the processing power of the chips used increasing constantly, the non-realtime algorithms used on a PC today will be implemented in real time in a DAC in a couple of years. Do we want to increase the cost of our equipment significantly to overcome that temporary performance gap?

 

In the meantime I'll make do with iZotope bundled with Audirvana+. Since I'd have bought Audirvana+ anyway, iZotope's very good SRC capabilities are effectively free to me. For the future, at any given time it will simply depend on whether pre-DAC or in-DAC upsampling sounds better to me with the particular software and DAC I have.

 

And it will be a cost issue - and again I am questioning (but that is just my personal opinion) if the potential improvement in going beyond 96/24 is justified in terms of complexity and cost, compared to concentrating the effort to areas where the returns are much clearer (speakers and room). For some people it might be, for others not.

 

As a practical matter, the effective cost to me is whatever I want to spend on hi-res files. That expense comes in relatively small increments, versus speaker or room changes that would take considerably greater one-time outlays. (To say nothing of the fact that my wife and I are happy with the speakers and the room as they are.) I made sure I preferred the DAC I bought last year (first new one in about 20 years) to my old one with RedBook files, since that is what I have most of. Thus the additional hi-res capability was a bonus.

 

Others' calculations may be very different - whether to replace a not very old DAC with a new one considerably more expensive than mine, for example, versus spending the same on speakers. Even in that situation, I might go for the DAC. There is something in my nature that prefers giving the best possible signal to the speakers I have, versus changing speakers and giving them a signal that isn't quite as good. Of course if I hadn't been extremely well contented with my current speakers for a couple of decades, I might feel differently.

[Jud]

 

"An examination of converter IC data sheets will reveal that virtually all audio converter ICs deliver their peak performance near 96 kHz. The 4x (176.4 kHz and 192 kHz) mode delivers poorer performance in many respects. In most cases, noise, distortion, pass-band ripple, stop-band attenuation and other key performance measurements are significantly better in the 2X (88.2 kHz and 96 kHz) mode of operation.

 

Every A/D and D/A conversion IC that we have tested performs better at 96 kHz than at 192 kHz. In most cases THD+N, SNR, passband ripple, and stopband attenuation are all poorer at 192 kHz than at 96 kHz. Based upon these tests, I am not surprised that there is not yet any conclusive evidence that 192 kHz is better than 96 kHz. Given the current state of the art, 192 kHz should sound poorer than 96 kHz. 192 kHz provides additional bandwidth between 48 kHz and 96 kHz but there is no real evidence that this is useful given the limitations of our microphones, speakers, and hearing. 192 kHz adds useless bandwidth while decreasing performance."

 

From what Converter Manufacturers Don’t Want You to Know! http://www.benchmarkmedia.com/discuss/sites/default/files/Upsampling-to-110kHz.pdf

 

Wow, what a credible paper (not). First we're "informed" that ASRC (asynchronous sample rate conversion) is better at removing jitter than synchronizing with the digital input (i.e., adaptive interface). There's no mention whatever that an asynchronous USB interface, taking the clock from within the DAC itself, is better than either at reducing jitter.

 

Then we are told in breathless tones about Benchmark determining that using sample rates well above Shannon-Nyquist minimum allows non-"brickwall" filtering. Glad that Benchmark figured this out, only 25 years or so after the first consumer DACs employing oversampling.

 

Then comes the stuff above, about DAC chips and performance at 96kHz. Of course nothing specific is said about the chips used by good designers in DACs that work well at 4x rates or with DSD, leading one to wonder whether this is a third instance in the same article of Benchmark leaving out important information. (My guess, based on the rest of the article, would be yes.)

[Jud]

The Simple fact is many of us, oh I don't know. Maybe the 40% of people who do NOT live in urban areas. Most of us only get one shot at stuff. We do not have an array of things delivered to us, play with them and send them back. Like children in a sandbox trying out toys.Aloha Berry, McLearn sells out a fleet of cars before any are delivered. We do not have an array of things available for us to test, any more than test drive every car that was ever made before we become wedded to a brand. Pretty much, we read all this B.S. about what things we sound, or drive like. We are forced to buy it, and we be stuck with it for many years.

We might make adjustments if we can afford it, but I think you will find in the real world most of us do not get to sample the material things that we own before it is to late to reasonably turn them back in. We have no way to test the equipment or the outlandish claims of Madison avenue advertising. I have bought cars before they were released. There was nothing to compare them with.

 

It is the same with pre-orders on audio equipment. "Buy on spec, shall be the whole of the law." So let me make this clear. I picked my playback system without listening to it. Most of the things, material, objects that we buy are based only on the specifications, without audition of any of the components, and were not listened to at all until the system was assembled at home (or on stage).

 

I asked Roger McQuinn about this at the Troubadour in California one night when he got back from playing Banjo up at Bob Dylan's house up in the canyon. He told me he just walked into a McIntosh store and bought a complete system and had it delivered to his house. Player, pre-amp speakers the whole bit. My dad ordered tons of cars before they came out. Do you think a test drive in a car that you will have for many years has any real meaning. I laugh at people taking test drives. Do you need a test drive to own a Porsche or a Ferrari?

 

No you put you money down wait forever as time will stretch like you are sitting on a hot stove, and as long as it is up to specs, specs that you may have only heard about. You will be happy.

 

 

I don't believe city people even live in a place where they can crank the volume up like I listen to it in the country. What do you know about sound. Neil Young plays in a Barn. You take a 60x40 concert stage and try to reproduce it in a closet. Just like you drive your 200-mph Shelby's and live with short spurts in the realm of the densely packed.

 

So let me make this clear I picked my playback system without listening to it. I bought my Porsche without driving it. I have gotten past the point of no return with chicks based only on her Specs.

 

How could anyone hear a sound system before they set it up?

 

It is not even possible, there are two many variables. The material, the room, the harmonicas of everything including the comfortableness of the listener. Once things settle in they do not sound the same, anymore than the twelfth night feels like the first time. I have never heard of anything so ridiculous in all of my life. That people listen to there playback system before they bought it.

 

Hope music is your therapy, 'cause the above sounds bitter. Maybe you meant at least some of it tongue in cheek?

 

The majority of places I've lived for the past 30+ years had more livestock than people nearby. But when I was ready to plunk down serious change on an audio system, I spent most weekends driving hundreds of miles to three states to audition equipment over a period of about a year before I had settled on a complete system. Everything I've bought since then I listened to for a good long time before I purchased; or it cost less than $200; and/or it was made by someone I'd had years of good experience with before that.

 

(Same with cars, by the way - any manufacturer I hadn't had experience with before, I at least researched to the max, including frequency of repair statistics, and test drove, maybe even had it on extended rental. I wouldn't buy a Porsche without a test drive; maybe a Ferrari, but that I could likely resell for as much or more than I paid. I rented an Alfa Romeo convertible for a week once; looked great, sure sounded like a romantic thing to have, but I wound up liking the Acura Integra a lot better. I was on a healthy expense account, so price didn't really matter, just liked driving the Acura.)

 

The result has been that I've loved listening to music on my system for the 20+ years that I've had my system. I tend to keep components, and even cables, for decades. Same amp and pre-amp for 20+ years, recently replaced speaker cables and DAC after the originals lasted 20+ years, have the same turntable going on 30 years.... Yeah, it's more time and effort at the beginning, but you get to settle back and enjoy yourself for a *long* time afterward if you do it right to start with.