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. > > > > Given the need for autocalibration -- > >> 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. > > > >> 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) > > 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 > > > Without any averaging the 10 bit ADC resolution limits the phase resolution to about 32ps with 10MHz mixer inputs. Bruce _______________________________________________ time-nuts mailing list -- [email protected] To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
