[time-nuts] Digital Mixing with a BeagleBone Black and D Flip Flop
kb8tq at n1k.org
Sun Oct 12 19:16:15 UTC 2014
If you are mixing down to 10 Hz, and are looking for 1x10^-7 on the 10 Hz, that equates to a stability / accuracy spec of 0.1 ppm on the ADC clock.
A 20 to 100 ppm offset on the clock is not all that unusual. Calibrating out initial offset to < 1 ppm is pretty simple. If you can poke a counter onto the clock output, you can get a lot closer than that.
The clock probably drifts in the 0.1 to 1.0 ppm / C range. If your room cycles 2 C every hour, correcting that to 0.1 ppm is quite do-able via a “spare” channel. You can record on an input or monitor a signal generated on an output. If your room is stable enough / your stability is good enough you may get away without doing real time correction.
Unless your clock has really bad specs, it’s ADEV should be below 1x10^-9 at 0.1 to 10 seconds. That’s 100X better than what you are after.
If the clock is a lot worse than these numbers, the audio properties of the ADC will begin to degrade. That more than anything else is what makes this all work.
On Oct 12, 2014, at 2:37 PM, Hal Murray <hmurray at megapathdsl.net> wrote:
> anders.e.e.wallin at gmail.com said:
>> Does it matter that the ADC in the sound-card is probably clocked by a
>> crystal clock that is 50ppm off and has bad ADEV?
> You can calibrate the clock on the ADC.
> One way is to feed a known reference frequency in on the other channel.
> (That's assuming you have a stereo setup and don't need the second channel
> for something else.)
> Another way is to compare the sample rate with the PC clock. That will
> correct for any long term drift but may not track shorter transients.
> These are my opinions. I hate spam.
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