[time-nuts] And, for my next trick, 50Hz
bruce.griffiths at xtra.co.nz
Fri Oct 3 19:58:26 UTC 2008
>> 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?
Try WKS interpolation (or even linear interpolation if you have enough
samples of the signal transitions) to achieve sub sampling period
All the information required to do this is contained within the sample
This is how jitter is measured with a digital oscilloscope with ps or
even sub ps resolution even though no digital oscilloscope yet has a
sample rate anywhere near 1 THz.
>> 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.
All sound cards of any use have separate ADCs (usually on the same
chip), one ADC for the L channel and one for the R channel.
Since they tend to use sigma delta ADCs and don't use incremental form,
switching an ADC dynamically between channels is impractical due to the
long group delay of the associated digital filters.
The point of such a measurement is to measure the internal measurement
noise of the sound card system.
> So do I get two Thunderbolts now.
> 73, Steve
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