Hearing tests showed the ability to discern jitter above a few hundred nanoseconds rms.
http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf

Others claim the ability to detect jitter in the picoseconds range...

It would be a conservative assumption that jitter in the range of tens-hundreds of picoseconds will be practically not discernible.

Usually integrated oscillators are composed of the classical inverting gate oscillator, with external CQC, and selfbiasing R, which has practically no rejection (~6dB) of the power supply noise. As it's usually on the same die with noisy digital circuitry, the gate threshold will jump around, producing timing errors, also the slew-rate is quite low, which just worsens the situation.

As most digital circuitry is less affected by jitter, the best solution is to place a clean oscillator near the D/A conversion, where the most critical timing point is, and through buffers clock the rest of the digital circuits - eventually galvanic isolation might be implemented, to pollute less the analog part with digital noise. To minimize jitter, digital clock inputs should be driven by fast slew-rate circuitry.


On 5/10/2012 12:25 AM, Hal Murray wrote:
was Subject: Re: [time-nuts] Faster than light of a different type
(Probably my fault.)

[email protected] said:
What I found funny was that the Audiophlie and light thread drew such
attacks when it hit home to me as exactly what the Time-Nuts mission is
about.  The Audio thread touched on some real world time and freq research
...

I too enjoyed the technical discussions.  Thanks for your contributions.

It's the audiophool bashing that people are complaining about.  Sure, it's
fun, but only at the right time and it gets old quickly.  The problem is that
with large groups, there are different opinions of when and how much is
appropriate.  The long tail on opinions of reasonable can annoy a lot of
people.

-----------

Back to technical stuff...

As a practical matter, is clock jitter or phase noise from a typical low cost
crystal and decent board layout a significant problem in audio gear?  How
hard is it to measure?

Is clock accuracy a practical problem?  How good are people with perfect
pitch?  It wouldn't surprise me if there are a few that are much much better
than others, but how good is that relative to 50 PPM which I can get in a low
cost crystal?

Video geeks solved their clock distribution problems by using frame buffers.
Is there a similar trick for audio?  Is there a need for it?

--------

I know clocking is a serious problem in fancy DSP systems.  For example,
modern radar has gone digital.  In that context, clock jitter can be
important.  Standard procedure is don't run your clock through a FPGA because
it will add jitter.

Part of the problem is that they are doing magic down conversion in the ADC.
(I can't think of the term.)  Suppose you have a 100 MHz signal with a 1 MHz
bandwidth.  You don't have to sample at 200 MHz.  You can sample at 2 MHz and
your signal will alias down.  It's turning what is normally a bug into a
feature.  The catch is that the errors/noise due to clock jitter happens at
the high frequency, in this case multiplying the noise by 100.  (Your
sample/hold at the front end has to work at the high frequency and your
anti-aliasing filter gets more interesting.)

There has been an interesting change in the specs for ADCs and DACs over the
past 20(?) years.  They used to be specified using terms like DNL and INL and
No-missing-codes.  Modern high-speed ADCs are specified with terms like ENOB
and SFDR.  Data sheets often include several plots of a batch of samples run
through a DFT so you can see the noise floor and such.

Here is a reasonable glossary:
   http://pdfserv.maxim-ic.com/en/an/AN641.pdf

I don't remember comments/specs about clock jitter in the data sheets but I
haven't looked at one in a few years.  I'll have to keep an eye out the next
time I'm browsing.




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