Pat,

I am not sure I agree on jitter.  My simplistic view of a digital audio
stream is that if plotted on a graph, the 'y' coordinates are provided
by the digital amplitude of the sample, the 'x' coordinates are implied
by the (assumed) regular sample period.  So to plot a point on the graph
accurately, you need to have both x and y coordinates correct. Having
the sample at the wrong time is just as bad as having a sample that is
the wrong size.

I suppose the real question is whether the level of jitter experienced
in the clock derived from S/PDIF is in any way significant from an audio
perspective. If my maths was up to it, I could probably take a jitter
distribution and convert it into either an implied noise or a distortion
figure in the derived audio signal. I would then be able to see how many
picoseconds of jitter mattered or whether this is negligable. Any link
to analysis on this would be welcome.

If the NAIM ring buffer approach catches on then I guess this all
becomes irrelevant. In the scheme of tricky hardware, reading in a
jittered signal into a ring buffer and reclocking out at a stable rate
with a local accurate clock is pretty basic. It feels like something
that could be very easily and cheaply implemented in all external DACs.

To put this in perspective, and again at the risk of showing my age, I
have seen timebase correctors that performed a very similar function
de-jittering a video signal built using delay lines that dated from the
60s. For an even older example, I have recently visited the telegraph
museum at Porthcurno in Cornwall where they have on display a reclocking
system that takes a jittered high speed morse signal from a
transatlantic cable, derives a clock from the incoming data, resamples
and de-jitters - and this is pre second world war and built with gears,
motors and switches.

To be fair, neither approach could cope with much jitter, and neither
had the luxury of large ring buffer storage, but it goes to show that
there is nothing new under the sun.


-- 
Andy8421
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