[PD] Real-time frequency filtering and analysis

padawan12 padawan12 at obiwannabe.co.uk
Thu Mar 29 08:00:16 CEST 2007

I think this just reinforces what everyone is saying
about the errors. Either you make it into a crazy
competition to vote by banging your note the loudest
and the mostest, or you accept a margin of error.
As Jamie says the intensity and frequencies are all
going to vary and interact.

attached patch intented to investigate that, it makes
little clusters of 10 votes (idealised sinewave "xylophone")
separated in time by a value between 0 and 100ms at
one of 3 frequencies chosen at random.
What you wont get in reality is the notes being exactly
the same frequency, there should be a fair degree
of variance. When I used a focused noise source to
approximate the xylophones inharmonic spectrum
I couldn't see any correlation above chance. YMMV,
especially if you tweak the windows/blocks .

I got better initial results with the filter bank approach
I think.


On Wed, 28 Mar 2007 09:16:23 -0500
"Charles Henry" <czhenry at gmail.com> wrote:

> filtering in general may not be the best approach because some of your
> partials from one xylophone note will overlap with other note's
> partials.  They are inharmonic complex tones, which are not so easy to
> predict.... you'll probably have to measure the frequencies of each
> note of your xylophone to know exactly what the spectrum is like.
> In terms of averaging like Jamie suggested... suppose you want to
> compute the expectation of the power spectral density.  You would take
> the fft of the auto-covariance of your recieved signal, divided by the
> number of blocks in your time frame.  (dividing by a number of blocks
> will not in general be necessary, when all you need to do is find a
> peak, with pique~ as before)
> This will give you a very clear/accurate peak, without much
> jitter/noise to clean up.
> Chuck
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