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Old 24th October 2012, 04:43 PM   #491
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I lost a rather lengthy post on this due to some network screwup, so this may be a bit brief.

A little known but pervasive limitation of parallel quantized PCM is the instantaneous waveform reconstruction accuracy which at 0.8 Nyquist is about 5%. This waveform reconstruction error phenomenon also does not reveal itself to continuous waveform distortion measurements of any competent digital audio system design. This is a major concern with A/D conversions in systems that perform waveform envelope detection in the digital domain where errors over a tenth of a percent or so are not acceptable. To a slightly lesser extent, this factor drives the oversampling ratio design of quality digital oscilloscopes.

Also, this error only decreases as a first order function of the Nyquist ratio, so to obtain 'traditional' 'digital distortion' figures of under 0.01%, the Nyquist ratio would have to be under 0.2%, or for a 20khz bandwidth system, about 40 hz. This error is independent of sampling word length and is purely a result of the actual sampling conversion rate.

I believe this is a contributing factor that has caused major transient, imaging stability, HF, and even midrange problems in redbook audio and, from the above description, even 192Ks/S parallel schemes would not really come close to eliminating it. Perhaps this is a reason that massively oversampled (at the actual A/D conversion) and 'one bit' digital audio has usually appeared to have an advantage in some areas such as 'smoothness'.

Last edited by thoriated; 24th October 2012 at 04:57 PM.
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Old 24th October 2012, 05:35 PM   #492
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Actually, the instanteous amplitude reconstruction *error* at 0.8 Nyquist is 5%.
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Old 24th October 2012, 05:37 PM   #493
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VERY interesting info Thoriated. Im surprised that the conversion is at sub nyquist levels. Id imagine with my limited knowledge (comms from uni), that that would cause gross errors. 5% doesnt sound brilliant to me, even though i understand BERs of 0.1% are acceptable for telephony, i wouldnt want my hifi chain at that level.
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Old 24th October 2012, 05:58 PM   #494
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If the sampling rate is 44.1kb/s (CD red book) What is the Nyquist frequency? 22.05kHz? or 44.1kHz?
If so then 0.8 of Nyquist is 17.64kHz, or 35.28kHz?
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Old 24th October 2012, 06:20 PM   #495
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Nyquist frequency is half the sampling frequency.

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Old 24th October 2012, 06:37 PM   #496
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Quote:
Originally Posted by tinitus View Post
is copper top testing our ears now
Yes, sort of. Back in the vinyl days, I was acutely aware of the inferiority of tracks on the inside tracks of LPs as they slow down; the words "harsh" and "grainy" spring to mind.

Yet High End people claim to be able to hear night and day differences between sample rates, or one DAC and another, yet never mention the fact that there is huge sonic degradation towards harshness and graininess as they listen to a side of an LP - just before they have to get up to turn it over.

If they can't hear it, how can they hear the difference between 192 and 44.1? If they can hear it, but it still sounds OK to them, what does that say for their judgement when recommending equipment to the rest of us?
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Old 24th October 2012, 06:55 PM   #497
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Originally Posted by 5th element View Post
192 for digital delay, for time aligning/phase aligning drivers is somewhat necessary. The minimum you can delay by is one sample period and 96 and 48 don't quite give you the precision required.
Fair enough but do you really need to adjust to better than 0.010 milliseconds which is about 1 sample period at 96k?
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Old 24th October 2012, 06:56 PM   #498
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Originally Posted by CopperTop View Post
If they can hear it, but it still sounds OK to them, what does that say for their judgement when recommending equipment to the rest of us?
It is passive approach, it is endless listening to judgments. Active approach is to try for yourself, and end the quest on your own decision, what sounds better, and what is different in cost of ownership only.
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Old 24th October 2012, 07:13 PM   #499
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Yet High End people ....
who ?
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Old 24th October 2012, 07:21 PM   #500
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Originally Posted by 5th element View Post
192 for digital delay, for time aligning/phase aligning drivers is somewhat necessary. The minimum you can delay by is one sample period and 96 and 48 don't quite give you the precision required.
Possibly true for 'bit perfect', but you can have any arbitrarily tiny delay you like if you re-sample the incoming audio.
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