[time-nuts] Sub Pico Second Phase logger
Joseph M Gwinn
gwinn at raytheon.com
Thu Dec 18 23:00:15 UTC 2008
time-nuts-bounces at febo.com wrote on 12/17/2008 06:26:00 PM:
> >> [BG] It isn't necessary to use a pair of mixers and an offset source
> >> characterise the sound card, driving both sound card inputs from the
> >> same audio source should suffice.
> > [JG] Yes. One input at a time, with the other input shorted, so we
> > see the crosstalk.
> >> The audio source need not have low ultra low distortion (the IF
> >> signals in a dual mixer system won't have ultra low distortion) or
> >> high frequency stability (the IF output signals in a dual mixer
> >> won't necessarily have particularly high frequency stability).
> > But ... but ... but ... I thought Time Nuts used only atomic frequency
> > refs, and crystals only if oven stabilized.
> If one mixes down a 10MHz source to 100Hz the fractional frequency
> instability (of the beat frequency) is magnified by a factor of 1E5 over
> that of the 10MHz source.
> This assumes that the offset source has significantly lower instability
> than the source under test.
> In the special case when the offset source and the test source are phase
> locked the offset frequency will have much greater stability.
Yes. One approach is to use the two 10 MHz signals as the clocks of a
pair of DDS chips programmed to generate ~ 1MHz and ~1 MHz + 10 Hz. When
mixed, these will yield a 10 Hz difference signal.
The same game can be performed in the software driving a soundcard, as
> >> A standard RC audio oscillator with distortion lower than 1% or so
> >> should suffice.
> >> At least the resultant frequency fluctuations should thoroughly
> >> the phase extraction algorithms.
> >> Another option would be to low pass filter the output of a divider.
> >> Using a sound card to generate the test signal is also possible but
> >> can potentially introduce extraneous noise and other artifacts such
> >> phase truncation spurs.
> > If one chooses the test frequencies correctly, one can eliminate the
> > spurs. The trick is to choose frequencies that lead to DDS tuning
> > that have zeroes in the accumulator bits that are truncated (that is,
> > not make it into the sin/cos lookup table).
> This just adds another layer of complexity for little immediate gain.
> Making the algorithms robust against small drifts in beat frequency is
> more useful in the general case (when 2 different test sources are being
> compared) than just assuming that the the beat frequency is very stable
> and fixed.
Yes, but I'm not sure we are solving the same problem.
I suppose the sound card could drive a simple PLL signal cleanup circuit.
> > Step one of planning an experiment is to decide on the objectives. The
> > large scale objective is to determine which sound cards are suitable
> > number of time-related tasks, so we should enumerate and describe
> > tasks.
> > Task 1. The immediate task is to receive and digitize the sinewave
> > from a mixer, the sinewave being the offset frequency coming out of a
> > apparatus. Offset frequencies will range from 10 Hz to 1 KHz, will be
> > known with great precision from the design of the apparatus, and need
> > be measured. This sinewave is high amplitude (at least one volt rms,
> > matched to the needs of the soundcard) and very high SNR. This will
> > done in two channels in parallel. The signals are at the same
> > but differ in phase. The intent is to extract the phases of these two
> > sinewaves, the difference in phase being the ultimate output.The phase
> > of a signal will be extracted by least-squares fitting of a sine
> > to the measured data.
> > And so on. We need to list the tasks, and to use this task list to
> > the experiment design.
> The immediate task is actually to evaluate sound cards for their
> suitability, preferably without the added cost and complexity of a DDS
> LO and mixer.
Suitability for what? That is the point of enumerating tasks.
I don't see where Task 1 above requires or even mentions a specific
implementation, such as a DDS LO and mixer.
> Once this evaluation is done, using a mixer and a DDS based LO to
> generate a beat frequency is the next step.
> Eliminating the mixer and DDS allows a greater number of participants at
> this stage than would otherwise be the case.
> 10Hz resolution whilst avoiding phase truncation spurs is impractical
> with a DDS chip by itself.
> Depending on the DDS and its clock frequency, the frequency spacing of
> phase truncation spur free outputs may be as large as several kHz.
Is this true of concatenated DDS chips? I recall a patent to the
> A few divide and mix stages will be required to achieve a spur free
> resolution of 10Hz.
That is a traditional approach. But are there alternate approaches that
have now become practical?
> A DDS chip with higher resolution phase outputs after truncation such as
> the AD99XX series are better in this respect than the earlier
> AD98XX series.
Actually, if we use a sound card to generate the test signals, the "DDS"
will be a bit of non-realtime math code in our computers. If we choose
the sample window size and test frequency correctly, we can arrange for
very low spurs and other errors. The spur reduction is largely due to the
fact that being offline one can use all of the phase bits to compute
sin/cos values, rather than truncating phase to say 14 bits.
The algorithm is something like this: Figure out how many samples there
will be per cycle of the test frequency. Adjust test frequency slightly
to eliminate any residue. Compute a full cycle of exact phase values.
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