Tom, Sorry to post in response to my previous post but perhaps this system could work if the input signal did have a quite a slow rise time. I was thinking more of coupling the output of the proposed 1KHz divider through quite a small amount of low-pass filtration. If the 1KHz was passed through a low-pass filter at the fundamental we would get a sine-wave with a relatively slow rise time and the ability to get more samples around the zero crossing point. Taking it a stage further, we could pass our square wave though an integrator to produce a linear sawtooth which would provide the best way to measure any phase shift. Now I think this could work to gauge 1ppm in 10 seconds but you would really need a short-term stable NTP disciplined clock to do it. I wonder what the average allan variance would be for this setup over a 10 second period.
73, Steve 2008/10/4 Steve Rooke <[EMAIL PROTECTED]>: > Tom, > > 2008/10/3 Tom Van Baak <[EMAIL PROTECTED]>: >> Ah, if you are a time-nut of course you will bite the bullet and >> get a GPSDO. But you should also not give up on the PC idea. >> If it works it will be a great gift to many people (OK, maybe not >> the pico- and nanosecond crowd, but to regular millisecond >> kind of folks). > > Well, that's the way I plan to go now but as you say there is indeed > some merit in continuing with the idea of using NTP to check the > frequency of an oscillator as per my original idea. Certainly I > believe that figures in the milisecond area are probably achievable > without too much trouble. > >> And, I no longer think long sample times are required. >> >> Consider the following. Assume you can get NTP to give you >> ms or sub-millisecond accurate time-stamps. Also assume you >> divide down your UUT to something like 1 kHz and feed that >> into one channel of your sound card (and as Bruce pointed >> out, maybe the slower the rise time the better in this case). >> >> Now, collect 16-bits of waveform data at 44.1 kHz. First, note >> that it's not exactly 44.100000 kHz -- but over time, as your >> circular input buffers fill up, you can use NTP time-stamps to >> calculate what the sampling rate precisely was/is. > > This would produce a lot of data and be quite intensive in processor > time. Sampling the input waveform is easy as it's done in hardware on > the sound card but if we are time-stamping each sample with an > accurate wall time value that is going to be a lot of system calls and > I wonder if this may come out of sync at times. > >> Then, looking at your highly oversampled waveform data, you >> can calculate the phase of your UUT frequency relative to the >> now precisely known sound card sampling rate. Over time you >> will see the phase drift, which then directly gives you the UUT >> frequency error. > > It's a very interesting idea and I may have a go at this one. > >> So what you end up doing is using the sound card like a high >> resolution vernier between NTP timekeeping on the inside and >> your UUT on the outside. I bet you a Thunderbolt that you can >> measure to 1 ppm within ten seconds. > > Lets see, phase shift of 1ppm in 10 seconds at a sampling rate of > 44KHz. So our error is 10^-6 so over 10 seconds becomes a 10^-5 > change. Now 44KHz is a rate of 4.4x10^4 or 0.44x10^5. So that looks > like we would come up a bit short on data to verify the 1ppm > difference, IE. only 0.44 sample to indicate the error which would not > show up. at 10ppm we would have 4.4 samples to show the difference > which would be more workable. Is my logic wrong here or when do I get > my Thunderbolt? > >> p.s. For extra credit, tee your UUT into both channels, do twice >> the math, and see if you can measure both differential phase, >> and differential phase drift between them. > > This would really just check the phase difference between samples for > the two channels of the sound card and I would expect that to remain > fairly constant. It's an interesting point though, I wonder if both > channels are sampled simultaneously or in a serial fashion. If that > was the case, and assuming that the samples were equally spaced > between the two channels, you may get the equivalent of an 88KHz > sampling rate which would just push the ability of this system to > measure a 1ppm difference. I guess it depends on if the sound card > uses two A2D converter or just one and switches this between channels. > I think that switching it between channels may be a bit of a messy > affair due to the settling time needed before the sample is taken on > each channel. > > So do I get two Thunderbolts now. > > 73, Steve > -- > Steve Rooke - ZL3TUV & G8KVD > Omnium finis imminet > -- Steve Rooke - ZL3TUV & G8KVD Omnium finis imminet _______________________________________________ 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.
