[time-nuts] Checking accuracy of Rubidium standards

Bruce Griffiths bruce.griffiths at xtra.co.nz
Sun Nov 9 02:44:56 UTC 2008


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



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