J. L. Trantham wrote: > I have been enjoying this discussion. > > Since the original question was the desire to 'compare' the frequency of an > LPRO to a Z3801, it seems that you could consider that from two (at least) > perspectives. > > Before I begin, I confess that I am a novice in this arena and please > correct me in any area that needs it. > > The first perspective is the issue of frequency. That seems to me to be the > issue of the average frequency of the LPRO versus the average frequency of > the Z3801. Assuming that there is no gross difference of the 10 MHz > signals, a lissajous figure (X-Y display) on a scope with the appropriate > bandwidth amplifiers would be a reasonable initial approach. > > Assuming that they are both near 10 MHz and you do not know which is the > most accurate (although the Z3801 would seem to be the default standard), if > it takes 10 minutes for a single cycle of the lissajous figure to complete, > then it is 1 cycle per 600 seconds difference between the two and therefore > the two are within 1/600 Hz or 1.67 mHz of each other. If we assume that > they are both close to 10 MHz, then that is 1.67 parts in 10E-10 difference > between the two. Is my logic faulty? > > The other perspective is the issue of 'purity'. That is to say, what is the > 'frequency modulation' of the source? This, I think, is the issue of phase > noise. Correct? > > That is something that I have not yet had a chance to contemplate as far as > how to measure. It would appear to require a particularly stable (pure) > source as a reference though. Various multiplying or dividing protocols > would seem to introduce a host of other variables that would seem to be > difficult to account for though they might accentuate an impurity in the > signal in question. I have read Bruce's comments and I still do not > understand the basics of time stamping or how a sound card might provide > this. > > I would appreciate any direction for further reading regarding this and I > would appreciate any direction/correction/etc. in the thoughts above. > > Joe > > Joe
There is sufficient information available from the sound card samples to calculate the input signal at any time between 2 samples and in particular derive the time at which the signal crosses zero. This is the time stamp for that zero crossing. The frequency and ADEV of a signal can then be calculated from such a sequence of time stamps. However it is necessary to either calibrate the sound card sampling frequency or lock it to a known frequency. The method used to interpolate between samples is called WSK (Whittaker Shannon Kotelnikov) interpolation. 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.
