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The Optimal Sample Rate for Quality Audio


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Hi Lavry Tech - While this paper may be interesting and very valuable from an engineering standpoint, your surrounding statements really hurt your credibility. If you post here on CA in an effort to educate there is no need to tout Dan as "One of the world’s top converter designers..." or to begin your post with "Interested in the facts?"

 

I can find several engineers and AES Fellows who contradict much of what Dan says. My point is there's not one set of facts.

 

I would appreciate something to support your assertion that my “surrounding statements really hurt” my credibility.

 

1.) “Interested in the facts?” is not a statement; it is a query with the specific goal of generating interest in a rather “dry subject” that has important implications for anyone serious about digital audio. It is relevant to the subject of the paper because the vast majority of “rebuttals” to Dan Lavry’s assertion that there is an optimal sample rate for high quality audio are based on opinion or subjective “test” results. We are not afraid of facts; and would be interested in hearing from the “AES Fellows who contradict much of what Dan says” in their own words. This is not a “new subject,” and during the years that have passed since the original Sampling Theory paper was published, no one has yet come forward with credible scientific evidence to the contrary.

 

2.) Regarding- “If you post here on CA in an effort to educate there is no need to tout Dan as "One of the world’s top converter designers..." or to begin your post with "Interested in the facts?"

One cannot really “educate” anyone else; one can only show them the way and hope they can educate themselves. I find it quite surprising that anyone associated with an online Forum would take the perspective that people who are not familiar with a very narrow field of electronics design would also not be interested in this subject. For example; I typed “Optimal sample rate” into Google, and Computer Audiophile was third on the list of results; which is something anyone, anywhere in the world can do.

 

Personally; I believe despite that fact that Dan Lavry is well known and respected in the professional audio industry, there are millions of people world-wide that are interested in the subject and are not aware of who Dan Lavry is; or why his fact-based argument might be more credible than either the opinions of people who lack anything even close to the depth of his understanding, or who have commercial interests in promoting lower quality audio as “better.”

 

In a world where nothing less than “extreme” even registers with so many who are overwhelmed by the amount of information available to them (useful and otherwise), I felt that a mildly provocative subtitle would help in the effort to bring attention to the subject.

 

Here is what Dan Lavry had to say:

“The industry is exposed to a well-financed campaign by large manufacturers trying to sell the false notion that faster sampling is better. There is a lot of advertising of higher sample rate conversion gear, aimed at benefiting the makers of such gear. A smaller converter manufacture has a choice. One can join the high sample rate crowd (making high sample rate converters) while riding the advertising hype that is well financed by larger companies. The alternative is to stay true to quality audio.

 

Lavry Engineering stands for quality audio. So we do what we can to steer the industry in the right direction in a manner that is transparent and does not benefit only our interests.

 

A few years back, I resisted the 192kHz sampling hype. That is when I wrote the paper “Sampling Theory” and refused to make higher sample rate gear. The hype died down and 44.1- 96kHz became mainstream again in professional recording and Mastering studios. A few years passed by and here we are again, this time with the pushing of 384kHz and even 768kHz. Again there is no credible engineering reason for it, and no supporting objective listening tests results.

 

We are trying to do our best to steer audio in the right direction. I am sorry to see that you seem to be focused on the paper introduction instead of the paper itself. I agree that the introduction was aimed towards getting people interested in reading the paper. I think that the Lavrytech introduction was a drop in the ocean compared to the well subsidized advertising hype for higher and higher sample rates for audio. I hope that people would concentrate more on the issue (the paper content) and less on the packaging (the announcement).”

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While it is possible to get "good" results with converters optimized to operate at sample rates higher than 96kHz; this does not "prove" that higher sample rates do not come without a cost. The unfortunate fact that, by design, a multi-bit sigma delta converter optimized to operate at 192kHz (or higher) cannot also be optimized to operate at 96kHz makes it nearly impossible to make a meaningful comparison of the effects of changing ONLY the sample rate via listening tests. Other factors such as differences in analog circuitry, jitter in conversion clocking, or even PC board layout can have a significant effect on the perceived "sound" of the entire converter unit; which makes comparing different models of converters useless in this regard.

 

To help illustrate the difference between scientific facts and subjective test results I submit the following argument:

 

How about that joker Columbus? Can you believe that B.S. about the world being "round?" Anyone with eyes can just look around and see that the world is FLAT. Anyone.

 

As Dan Lavry points out in the following response; the Nyquist theorem is not intuitive. Trying to apply "common sense" analogies such as comparing samples to pixels only confuses the matter.

 

Dan Lavry 's response:

I have been making the case against higher sample rates for audio for a long time. I have encountered no credible arguments to my paper “Sampling Theory”. The same is true for my recent paper “The Optimal Sample Rate for Quality Audio”. I encounter some that want to counter the message by “shooting the messenger”. Meanwhile the facts I preset are correct and UN-challenged. I realize that reading the papers demands time and concentration. So here is a shorter description of many of the points I presented in the papers. Let’s refrain from diverting the conversation away from the topics.

 

1. Sampling is not intuitive. SAMPLING IS NOT ANALOGUS TO PIXELS! A more detailed picture may require more pixels, but more audio detail does NOT require more samples. There is an “electronic tool” (filter) that enables recovering ALL of the audio from a limited number of samples. It is not intuitive and requires much study. In fact it is counter-intuitive and goes against “everyday common sense.” This is the reason why the marketing of “more samples is better” is successful in convincing so many of the false notion.

2. Nyquist theorem (theorem is a PROVEN theory) tells us that recovering ALL the audio intact does require the sampling rate (frequency of sampling) to be at least twice as fast as the highest signal (audio) frequency. Theory demands a perfect “reconstruction tool” filter. In practice, real world filters require sampling a little faster than twice the audio bandwidth. For 20 KHz audio bandwidth, the theory requires at least 40 KHz sample rate. The 44.1 KHz standard provides 4.1 KHz margin. The margin for the filter (from the theoretical filter) is 100*(44.1KHz-2*20KHz)/(2*20KHz) = 10.25%

3. Some people argue that we need more than 20 KHz for audio. The decision as to how wide the audio range is should be left to the ears. Say we agree to accept a 25 KHz as the audio bandwidth. When using 88.2 KHz sampling, (and 25 KHz for the audio bandwidth) the margin is i100*(88.2KHZ-2*25KHz) /(2*25KHZ) = 76.4%.

4. At 96 KHz sampling and 25 KHz audio, the margin is 92%. At 96 KHz sampling and 30 KHz audio the margin is 60%. At 192KHz sampling and 30KHz the margin is 220%!. For anyone crazy enough to claim they hear or feel 40 KHz, when sampling at 192 KHz the margin is still 140%. At 384 KHz sampling the margin is 380%!

5. Some argue that at 44.1 KHz the margin of 10.25% is tight, and that theoretical filters fail to provide a near perfect reconstruction. Others argue that 20 KHz audio is too small to accommodate some ears. Such arguments support some reasonable increase in sampling rate. Many argue that 44.1 KHz rate is good enough. Others disagree. But few will argue with the statement that 44.1 KHz is at least pretty close to acceptable. In order to accommodate those that want improvements, let’s increase the margin by a factor of say 2. You want more, OK, by a factor of 4. You want more audio bandwidth? OK let’s raise it to a factor of 5… And all that is more than covered by the use of 96 KHz sample rate!

6. A few manufacturers are starting to advocate 384 KHz and even 768 KHz sample rates. When audio sampled at 44.1KHz is considered as being somewhere between “not perfect” and “near perfect”, the notion of sampling 870% faster (for 384KHz) or even 1741% (for 768KHz) faster than a CD makes no sense. I expect even the least competent of designers to be able to design a filter that does not require such huge margins. I would also expect any converter designer to have enough background to know that more samples are not analogous to more pixels! I would expect converter designers to insist that their marketing department knows that, instead of closing their eyes to the crock of steering audio in the wrong direction. I also understand it is not easy when one’s job is on the line.

7. It is not wise to keep increasing the sample rate unnecessarily. The files keep growing, and faster sampling yields less accuracy. Yet the marketing of higher sample rates has no basis, other than some spreading of misinformation. The latest I saw claims that faster sampling yields better stereo location (time resolution). The argument is false. Faster sampling offers the ability to process wider bandwidth, but has no impact what so ever on stereo location!

8. Faster sampling for capturing bandwidth that we do not hear (ultrasonic) is not wise. If we did not hear it (or feel it) we don’t need it. If we did hear it (or feel it) it is not ultrasonic, it is audible bandwidth (by definition). Ultrasonic energy may cause problems by spilling over to the audible range (intermodulation distortions). At best case, ultrasonic energy adds nothing to audio while requiring faster sampling, thus larger files and slower file transfers. In reality there is another price to pay; the faster one samples, the less accurate the result.

 

Dan Lavry

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