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Old 9th November 2013, 08:21 AM   #11
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Originally Posted by Monte McGuire View Post
How could you verify this? Well, you could look at the amplitudes of the two samples, and determine the timing difference that would cause the observed voltage difference between the two sets of samples. What's the limit to your resolution here? The S/N ratio of the sampled waveform
The problem I have with this argument is that it only works for sine waves. Sine waves are single frequencies, therefore they extend infinitely in time. They can't ever begin or end, because these are discontinuities that would imply other frequency components.

So this kind of "time resolution" is really just phase resolution, only good for resolving the phase difference between two sine waves. Other studies have shown that allpass filters aren't audible, suggesting that the ear doesn't care too much about these phase differences.

I still believe the time resolution of CD quality digital audio is adequate for the purpose, I just don't think the Shannon-Nyquist type arguments constitute proof. I think the real issue is the tradeoff between aliasing, ringing in high-order anti-alias filters, and high frequency extension. You have to choose a sample rate high enough that none of these effects is audible.
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Last edited by scopeboy; 9th November 2013 at 08:25 AM.
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Old 9th November 2013, 08:57 AM   #12
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Kunchur and Yamaha are wrong in saying that timing resolution depends on the sample rate. Their fault lies in the fact that they think digital signals are not band limited and that the staircase wave you see in a sample editor is the actual sound wave that the da converter outputs..

A very simple demonstration clearly debunks this myth. Its done in this video at 20min 50sec.
D/A and A/D | Digital Show and Tell (Monty Montgomery @ xiph.org) - YouTube
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Old 9th November 2013, 09:44 AM   #13
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Originally Posted by mohammed li View Post
Kunchur and Yamaha are wrong in saying that timing resolution depends on the sample rate. Their fault lies in the fact that they think digital signals are not band limited
I don't think so.
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Old 9th November 2013, 09:45 AM   #14
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The maximum amount of information in any signal (including vinyl) is given by the Shannon-Hartley theorem:
Click the image to open in full size.
https://en.wikipedia.org/wiki/Shanno...artley_theorem
Timing resolution is a form of information, so this applies here.

If you run the numbers than the CD format can handle more information than the best vinyl reproduction systems.
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Old 9th November 2013, 09:45 AM   #15
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Originally Posted by 45 View Post
I don't think so.
Your wrong.
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Old 9th November 2013, 10:27 AM   #16
SY is offline SY  United States
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Quote:
Originally Posted by mohammed li View Post
Kunchur and Yamaha are wrong in saying that timing resolution depends on the sample rate.
Kunchur was incorrect, and many people with actual training and background in DSP have corrected him in the intervening years; that's the nature of science and its greatest strength.

Yamaha apparently has different people writing different pages- this most recent bout of quote-mining is more about latency than actual time or phase resolution. The page linked in the closed thread correctly states that the temporal resolution for a recorded and played back signal using a 48kHz sampling rate (standard in studios) is on the order of magnitude of a nanosecond, worst case, and explicitly says that this is inaudible (being a thousand times lower than any experimentally determined human threshold). And, of course, even the very best mechanical recording and playback systems are far worse than this, as anyone who has ever done a spectral analysis of tape or turntables can readily see.

Quote-mining is a poor substitute for understanding, but perhaps useful in propagandizing. Fortunately, the engineers who designed the CAT scanner that saved my life were more interested in getting things correct than rationalizing nostalgia.
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Old 9th November 2013, 10:38 AM   #17
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Originally Posted by Barleywater View Post
16bits is noise floor about -130dB of full scale.
...
? 20*LOG(2^16) = 96.33 dB
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Old 9th November 2013, 10:39 AM   #18
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Noise floor, not signal-to-noise. The integration is important!
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Old 9th November 2013, 10:44 AM   #19
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Originally Posted by SY View Post
Kunchur was incorrect, and many people with actual training and background in DSP have corrected him in the intervening years; that's the nature of science and its greatest strength.

Yamaha apparently has different people writing different pages- this most recent bout of quote-mining is more about latency than actual time or phase resolution. The page linked in the closed thread correctly states that the temporal resolution for a recorded and played back signal using a 48kHz sampling rate (standard in studios) is on the order of magnitude of a nanosecond, worst case, and explicitly says that this is inaudible (being a thousand times lower than any experimentally determined human threshold). And, of course, even the very best mechanical recording and playback systems are far worse than this, as anyone who has ever done a spectral analysis of tape or turntables can readily see.

Quote-mining is a poor substitute for understanding, but perhaps useful in propagandizing. Fortunately, the engineers who designed the CAT scanner that saved my life were more interested in getting things correct than rationalizing nostalgia.
Thanks for correcting me.
The Kunchur paper and some Yamaha papers have been used over and over again to attack the "poor timing of redbook". I just did not read the links.
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Old 9th November 2013, 10:46 AM   #20
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To be clear, although he made a fundamental mistake in interpretation, Kunchur's experimental results are solid and have not, to my knowledge, been refuted.
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