[time-nuts] Sub Pico Second Phase logger

[Joseph M Gwinn]

Bruce,


time-nuts-bounces at febo.com wrote on 12/16/2008 09:05:55 PM:

> Joseph M Gwinn wrote:
> > Bruce,
> >
> >
> > time-nuts-bounces at febo.com wrote on 12/15/2008 05:31:27 PM:
> >
> > 
> >> Joseph M Gwinn wrote:
> >> 
> >>> Bruce,
> >>>
> >>>
> >>> time-nuts-bounces at febo.com wrote on 12/15/2008 04:34:34 PM:
> >>>
> >>> 
> > [snip]
> >>
> >> I'll look into doing this [MDEV and ADEV].
> >> Real time filtering and decimation may be impractical, in the short 
term
> >> at least, as most signal processing libraries only process 16 bit 
> >> samples.
> > 
> >> Most real time spectrum analysis programs are similarly afflicted in
> >> that they only process 16 bit samples.
> >> 
> >
> > I don't see why we would need realtime filtering.  Data reduction can 
be 
> > offline, so we ought to be able to use 32-bit or 64-bit arithmetic.
> >
> > Given that we will inspect Allan Deviation data in a log-log plot, one 
can 
> > save much processing time by spacing the tau values to be computed 
> > uniformly in log tau.  I've played with this in Mathematica, and it 
does 
> > work and yields a large speedup factor.  It should also help with 
Plotter 
> > and Win2K limits.  One trick is to ensure that one computes each tau 
value 
> > at most once.  This check is needed because with close spacing, the 
round 
> > function will yield the same tau values multiple times for small 
values of 
> > tau.
> >
> > Joe
> >
> > 
> Joe
> 
> Real time processing certainly isn't required to characterise the 
performance.
> However some may be tempted to do this, it's probably possible with a 
sufficently fast machine.

If we are looking for thermal effects, with a characteristic timescale of 
tens of minutes to hours, the concept of realtime can be very generous.


> I was just highlighting a problem with some available signal processing
> libraries which may have been developed before sound cards with
> resolutions of more than 16 bits became available.
> Some (perhaps most) real time spectrum display software also has this
> problem (eg baudline, Virtins etc).

I would assume that there are newer libraries now, and libraries available 
as source code can be updated and recompiled.

20 Log[ 2^16 ]= 96 dB.  This isn't awful, and we will get the entire 
16-bit range if the ADC is 24 bits (with ENOB of 19-20 bits) and we scale 
and round the samples properly.

As I think about it, the 16-bit limit must be for embedded signal 
processing code, and math libraries intended for use on ordinary computers 
will be at least 32 bit or 64-bit float, so it should not be difficult to 
come by the necessary code.


> It isn't necessary to use a pair of mixers and an offset source to
> characterise the sound card, driving both sound card inputs from the
> same audio source should suffice.

Yes.  One input at a time, with the other input shorted, so we can also 
see the crosstalk.


> The audio source need not have low ultra low distortion (the IF output
> signals in a dual mixer system won't have ultra low distortion) or very
> high frequency stability (the IF output signals in a dual mixer system
> 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.

 
> A standard RC audio oscillator with distortion lower than 1% or so
> should suffice.
> At least the resultant frequency fluctuations should thoroughly exercise
> 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 it
> can potentially introduce extraneous noise and other artifacts such as
> 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 words 
that have zeroes in the accumulator bits that are truncated (that is, do 
not make it into the sin/cos lookup table).


Step one of planning an experiment is to decide on the objectives.  The 
large scale objective is to determine which sound cards are suitable for a 
number of time-related tasks, so we should enumerate and describe these 
tasks. 

Task 1.  The immediate task is to receive and digitize the sinewave output 
from a mixer, the sinewave being the offset frequency coming out of a DMTD 
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 not 
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 be 
done in two channels in parallel.  The signals are at the same frequency 
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 function 
to the measured data.

And so on.  We need to list the tasks, and to use this task list to inform 
the experiment design.