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# Thread: Dan Lavry, design guru new white paper on the Optimal Sampling Rate for Audio!

1. ## Dan Lavry, design guru new white paper on the Optimal Sampling Rate for Audio!

http://www.lavryengineering.com/pdfs...lity_audio.pdf

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

2. Common sense at last. Funny how red book keeps on getting better as you add room treatments and better loudspeakers...16/44 is by far the smallest problem, if problem it is, in even the very best installations.

Probably the biggest barrier to realism is the way music is recorded. Five mics for a drum kit...how can it sound like a real drum kit?
Darren

Sent from my HTC Sensation Z710e using Tapatalk 2

3. Yet My Golden Ears require at least 24/196 to listen to “Muskrat Love” The Captain and Tennille (1976)

4. ## My understanding of Nyquist

If I understand correctly; for the Nyquist sampling to result in a perfectly reconstructible waveform (at the receiving end) the sampled waveform must be repetitive and the waveform must be sampled for a period that is long in comparison to the waveform. This can be seen on a 'scope where a sampling rate not much greater than 2f results in a very good looking waveform on the trace of the scope for a repetitive waveform. However, audio is not quite like that as the waveforms are not repetitive but rather they are transitory. I dont therefore see how they may be reconstructed perfectly unless they are sampled at somewhat more than the 2f of Nyquist. How much more would, I guess, be dependant upon the required allowable error in reconstruction??

Perhaps you would elucidate??

AndyN

5. Originally Posted by AndyN
If I understand correctly; for the Nyquist sampling to result in a perfectly reconstructible waveform (at the receiving end) the sampled waveform must be repetitive and the waveform must be sampled for a period that is long in comparison to the waveform. This can be seen on a 'scope where a sampling rate not much greater than 2f results in a very good looking waveform on the trace of the scope for a repetitive waveform. However, audio is not quite like that as the waveforms are not repetitive but rather they are transitory. I dont therefore see how they may be reconstructed perfectly unless they are sampled at somewhat more than the 2f of Nyquist. How much more would, I guess, be dependant upon the required allowable error in reconstruction??

Perhaps you would elucidate??

AndyN
Are you sure that the waveform must be repetitive? AFAIK At any given instant the time varying signal will comprise particular frequency components, but the frequency components do not have to remain constant from one sampling point to another. I was under the impression that the sampling theorem applies to any (band limited) time variant signal. Of course the signal cannot change in a which requires a frequency higher than nyquist to make the transition.
That said I would have guessed that in a music signal the waveforms are not as transitory as all that relative even to a 44.1kHz sampling rate- the drummer does not hit the drums 44 thousand times a second.

6. Originally Posted by darrenyeats
Common sense at last. Funny how red book keeps on getting better as you add room treatments and better loudspeakers...16/44 is by far the smallest problem, if problem it is, in even the very best installations.

Probably the biggest barrier to realism is the way music is recorded. Five mics for a drum kit...how can it sound like a real drum kit?
I would much rather listen to a well recorded, well mixed and mastered mp3 than to a shoddily recorded/mixed/mastered 24/192 FLAC.

7. Originally Posted by magiccarpetride
I would much rather listen to a well recorded, well mixed and mastered mp3 than to a shoddily recorded/mixed/mastered 24/192 FLAC.
+1

Try the recently remastered "Blue In Green" from Kind Of Blue - Miles Davis

(Of course, I'm not listening to an MP3 version)

8. Originally Posted by SlimChances
Yet My Golden Ears require at least 24/196 to listen to “Muskrat Love” The Captain and Tennille (1976)
Love your pic. Now I know how some here hear so well. I wasn't away of a "Beats" version.

bfl

9. Originally Posted by magiccarpetride
I would much rather listen to a well recorded, well mixed and mastered mp3 than to a shoddily recorded/mixed/mastered 24/192 FLAC.
life's too short to listen to mp3, who records/mixes/masters in mp3 ? The amount of reviews of digital music reduces the chance of having to listen to shoddily recorded/mixed/mastered 24/192 FLAC.

I enjoy 16/44.1 up to 24/96 (24/192 is a bit of a luxury due to space etc) Could I pick one over the other, I doubt it. The higher sample rates seem to be more atmospheric - again could be better production.

10. Originally Posted by banned for life
+1

Try the recently remastered "Blue In Green" from Kind Of Blue - Miles Davis

(Of course, I'm not listening to an MP3 version)
Hi Banned for life!

Can you give more details of this recently remastered version of Kind of Blue? Where can I get a copy and in what format?
Cheers

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