[time-nuts] Using cheap sound cards for measurements
Magnus Danielson
magnus at rubidium.dyndns.org
Mon Aug 24 19:36:11 UTC 2009
Hal Murray wrote:
> james.p.lux at jpl.nasa.gov said:
> [External clock at strange frequency.]
>
>> That's an interesting idea. I would imagine that the clock going into
>> the chip is probably some multiple of the sample rate (e.g. 48kHz*16*2
>> = 1.536 MHz), so you could pick the closest 1/N from 10 MHz and pump
>> that in.
>
>> However, what about the USB interface? These are inexpensive devices,
>> and I'll bet all the rates are carefully chosen so that everything
>> shares one clock.
>
> I guess somebody will have to take the lid off and look inside.
>
> Most USB gizmos that I've looked at have something like a 24 MHz crystal. I
> assume that is a sweet spot for cost. At the root hub, that turns into the
> clock/bit rate. At the device end, I think it's PLLed to the upstream clock.
24,576 MHz is common, as it is 512 x 48 kHz.
> My guess is that any claimed-to-be-good audio gear would have it's own audio
> clock just to avoid the wander as the PLL follows its view of the upstream
> clock.
When you lock the clock you want it to follow the source, but you as the
user needs to ensure the source is good. For professional systems, the
Audio Engineering Society (AES) have standardised this in the form of
AES-11 while the professional audio standard AES-3 do standardise the
jitter transfer between timing reference and output.
> I don't understand the audio numbers. Is there a crystal frequency that
> works well with all normal sampling frequencies? I don't see one if you want
> both 44.1 and 48 KHz. (There could easily be some sneaky scheme I don't know
> about.)
44,1 kHz is a consumer number, as a result of Sony/Philips working on
how to shoe-horn into the limits of the CD. They wanted 72 min music for
a suitable size (120 mm) and optical technology.
The professional audio prefers 48 kHz (a simple x 6 from traditional 8
kHz audio) and power of 2 multiples (24 kHz, 96 kHz, 192 kHz or 384
kHz). An older standard is 32 kHz, which has a simple relationship to
the modern series (2:3).
Even more hair-pulling is tossing in the 1000/1001 factor and its
inverse 1001/1000 for all places. Makes alot to cause troubles for
frequency syntesis. If we could do away with that, then I would be much
happier.
The 48 kHz sampling rate has known and defined relationship to frame
rate to TV standards, as defined in AES 5 and AES 11.
Professional rates when not infected by 1000/1001 factors makes sense,
is easy to correlate to frames, GPS and whatever is relevant for production.
Wordclock is very similar in behaviour, but has no real definition.
AES-11 has an informative annex covering it.
The AES-2id and AES-12id would be recommended reading for someone
wanting to peak into the issue of jitter (and wander) from the audio
perspective.
Unfortunatly the AES papers isn't free on the web. Other resources is
available. Julian Dunn have written several very good papers. He has
also written two of the Audio Precision (http://www.ap.com) application
notes, one on jitter and the other on AES/EBU digital audio interfaces.
None of these uncover the mysterious sample rate numbers thought.
Cheers,
Magnus
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