[time-nuts] And, for my next trick, 50Hz

Steve Rooke sar10538 at gmail.com
Fri Oct 3 17:18:46 UTC 2008


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 <sar10538 at gmail.com>:
> Tom,
> 2008/10/3 Tom Van Baak <tvb at leapsecond.com>:
>> 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

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