[time-nuts] Using cheap sound cards for measurements

[Bruce Griffiths]

Locating the zero crossing of the signal isnt the best approach.
If one uses some DSP the phase shifts between the 2 beat notes can be
measured without the sound card sampling frequency stability making a
significant contribution to the noise.

Lux, Jim (337C) wrote:
>>> Wouldn't the cards need to be synchronized, though?  Essentially,
>>> you're still  comparing the two signals with each other, just
>>> doing it in software, rather than in hardware, in the classical
>>> time interval counter scheme counting 1Hz (or 123Hz).  Syncing
>>> inexpensive cards is a real chore (and the only reason to be
>>> thinking about using this in the first place is to keep the cost
>>> to a minimum, otherwise, you might as well build a special
>>> purpose little box with counters & A/Ds, and an interface)
>>>       
>> I'm not sure it's that important (or helpful) for the ADCs to share a common
>> clock.  Presumably the ~100 Hz beatnotes being digitized are on the order of
>> 1/100000 of the frequencies being measured.  That means that a microsecond
>> of synchronization error between the ADCs would have an effect similar to a
>> picosecond-scale error on the DUT/reference sides of the mixers.
>>
>> Getting microsecond precision out of an audio ADC is going to require
>> processing multiple successive samples, and IMHO it will also require  some
>> kind of auto-calibration scheme since sound-card clocks probably drift more
>> than 1 ppm per minute or so anyway.
>>     
>
> They're not that bad. Fairly high aging, fairly substantial variation with temperature, but that actually stays pretty constant.
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>   Given the need for autocalibration  --
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>> probably through a high-frequency sidetone sent to both channels in phase --
>> the difference in complexity between supporting two ADC clock domains and
>> one is probably not a deal-killer.
>>     
>
> Yes.. but if you start feeding multiple signals into the ADC (e.g. a calibration pilot tone), then you start running into intermod effects from the inevitable ADC nonlinearities. I don't have a good intuitive feel for just how good the digitizing needs to be for this approach; I guess if I want to go further, I need to sit down and do the math.
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>> Most installations would probably need to use a beatnote frequency
>> closer to  1 Hz, so that would take a lot of pressure off the ADC clocks.  It
>> *might*  be enough to get you out of the autocalibration business, but my guess
>> is  that matching the phase tempco of the (AC-coupled) sound card inputs
>> might still be necessary for good long-term results.
>>     
>
> But at 1Hz, you're down in the LF rolloff of the ADC.  They probably roll off around 10-20 Hz, and none too predictably (e.g. they just slap a suitable cheap ceramic capacitor in series with the audio as a DC block)
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>   
A typical high end sound card rolls of at around 1Hz or so.
> But that DOES bring to mind an even cheaper approach.. the DATAQ $25 data acquisition unit. 4 10 bit ADCs at 1kHz or so
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Without any averaging the 10 bit ADC resolution limits the phase
resolution to about 32ps with 10MHz mixer inputs.

Bruce