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From: "John Jardine"
Subject: Re: Audio Phase meter cct wanted
Date: Wed, 20 Nov 2002 02:33:39 -0000
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NNTP-Posting-Date: 20 Nov 2002 02:22:11 GMT
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Tom Bruhns wrote in message
> So...now that I have you hooked on the basic idea, I'll toss out just
> a bit more info.
> First of all, if you generate your sine wave in the same processor, or
> at least locked to the same clock, you'll be ahead. That way, you can
> get the sine frequency exactly on the DFT's frequency. Interestingly,
> you can use the same algorithm to generate a sine wave; just
> initialize the state variables to values representing the phase and
> amplitude you want. It's possible for roundoff errors to cause the
> amplitude to increase or decay, or the frequency to be slightly off,
> but at least it's one way to do the job. If you have to measure the
> phase between two signals you don't generate, but which you know to be
> from the same original source and therefore the same frequency, then
> it would be helpful to lock the digitization clock to them.
> Next, it can be helpful to sample at exactly 4 times or 8 times the
> input frequency. If you use 4 times, for example, then the phase
> shift between successive samples is just exactly 90 degrees, and in a
> "canonical" biquad, the state variables will be samples of a sine wave
> 90 degrees out of phase. Then it becomes relatively easy to find the
> phase angle of that wave, just the arctangent of the ratio of the two
> state variables. You really start to get into trouble with finite
> precision arithmetic if the number of samples per cycle is large and
> you are using a canonical biquad.
> However, if the freq is low and you want to take more samples to get
> more data to include in your processing, you can set up a little
> different filter topology. One that has worked very well for me is to
> have each state variable be the output of a "digital integrator" which
> implements y(k+1)=c*u(k)+y(k)...just add the scaled input to the
> current value to get the next value. A cascade of two of those stages
> looks very similar to a cascade of two analog integrators, and that's
> the basis of an analog state variable filter. Then you arrange the
> feedback to get the complex pole pair on the unit circle in the
> z-plane...it all ends up looking very much like what you'd do with the
> analog state variable filter to do the same thing. Then the two state
> variables (the digital integrator outputs) are very nearly 90 degrees
> out of phase, for that case of lots of samples per cycle.
> Results: the instruments we build are general purpose vector spectral
> analysis instruments. They work by digitizing the input and
> performing FFTs. They are only digitizing to 14 or 16 bits, but I
> have no trouble displaying stable phase relations down to about a
> millidegree or so, if the inputs are reasonably "clean". So with only
> audio range signals, you can see the difference in length between
> cables feeding the "same" signal to two channels. 10 millidegrees at
> 20kHz, for example, is only about 1.4 nanoseconds. You can generally
> do things like swapping channels and averaging to remove the
> systematic error in the measuring instrument, if your inputs are
> stable. I would expect to be able to do as good at a single frequency
> with an inexpensive stereo delta-sigma ADC and a processor that can
> handle the math fast enough and to enough bits of precision. Guess
> that could all be a PC with a sound card. :-) Doesn't even have to
> be fast if you can do the processing after acquisition of the signal.
> You might even be able to do the math in Excel or Matlab or Scilab.
I feel *The man* himself is divulging here some crown jewels. Tom, you have
aquired massive status, sufficient to cause me to print out your words for
The mere mention of a stable millidegree or so, brings a small but perfectly
formed, tear of emotive pleasure to my eye. If only that life could be so
Yes. All my kit frequencies are locked to each other with a seperate locked
'sampling sync'. The I.F. frequencies (via 2 off DDS's) I am looking at are
a (sub-sampled also) *very* low audio so am taking a whole rake of samples.
Signal processing at the moment is via PC using floating point. I've about 4
FIR's in the signal paths at the moment so the addition of a sharp selector
should fit in the programming loops nicely.
You guys sound to have some lovely work. I'm envious!.
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