Hi My assumption is that to do any of this, some sort of calibrated clock on the sound card would be needed. Either a mod to the card, or some luck with the built in clock source. It's going to be needed weather you use an FFT or some other DSP process. For a "fully locked" system, mod the sound card for an external clock and whip up a phase locked loop to drive it.
The mix down source would likely be purely digital. Inserted after you had the sample stream. My first choice would be to use a 2 KHz "stupid clock" (0 1 0 -1) with an 8 KHz sample rate. Pretty simple math. Bob On Mar 22, 2011, at 6:23 PM, Alberto di Bene wrote: > On 3/22/2011 5:24 PM, Bob Camp wrote: >> The other approach would be to simply take the samples, do a mix down, and >> get the phase from an ATAN calculation on the I/Q results. That would give >> you pure phase and thus frequency. > How do you intend to generate the numeric LO stream to use for the mix down ? > It must have a sampling rate > with a precision comparable to what you intend to measure... > > 73 Alberto I2PHD > > > _______________________________________________ > 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. _______________________________________________ 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.
