[music-dsp] measuring partial frequencies
peter at sonicreef.com
Tue Jan 13 23:23:01 EST 2004
I don't understand what measuring frequencies has to do with wavetable
I don't want to generate sounds, I just want to be able to tell the
frequency of the fundamental and it's partials.
Initially I thought, it would be trivial, but it seems more complex than
From: music-dsp-admin at shoko.calarts.edu
[mailto:music-dsp-admin at shoko.calarts.edu] On Behalf Of robert
Sent: Tuesday, January 13, 2004 2:52 PM
To: music-dsp at shoko.calarts.edu
Subject: Re: [music-dsp] measuring partial frequencies
> From: "Peter Thom" <peter at sonicreef.com>
> To: <music-dsp at shoko.calarts.edu>
> Subject: RE: [music-dsp] measuring partial frequencies
> Date: Tue, 13 Jan 2004 10:26:55 -0800
> Reply-To: music-dsp at ceait.calarts.edu
On 01/13/2004 10:26:55 -0800, Peter Thom at <peter at sonicreef.com> wrote:
> Yeah, I'm looking for frequencies other than the fundamental. FFT
> isn't accurate enough, I want to measure frequencies within 0.01Hz.
then, in my opinion, you want to seriously look at that Wavetable
Synthesis paper (and remember "true" or classic wavetable synthesis is
not the same thing as sampling "synthesis" which really isn't synthesis
at all). a frequency deviation of 0.01 Hz is essentially a slow
changing of phase.
> Would a bandpass filter and zero crossing counting do, or is that not
> accurate enough? How about autocorrelation?
you might use something like that to precisely measure the period of a
tone (which is useful for "pitch detection" or measuring f0), but it
does not separate the partials which you will need to do somehow.
> BTW, the tone is a piano tone, so it should be a relatively periodic
really, Peter, i don't know how deeply you want to dive into the math
(the paper was done that way, with continuous-time, because it was the
easiest way to be rigorous), but conceptually it is *not* that bad.
in terms of implementation, i presume these piano tones are WAV or AIFF
files, right? and you have the means to open and parse them, right?
what language are you planning to do this in? C? Matlab?
> Any help is greatly appreciated.
besides the means to put your sound file into an array of PCM data that
you can do math on, you will need:
1. a good pitch detection and pitch *riding* algorithm
might work, but i think ASDF will work a little better). this
algorithm also needs to measure how periodic the tone is at some
time within it so that you know when you crawled over the
transient or "attack" portion into the quasi-periodic portion.
2. a good intersample interpolation utility.
3. a DFT or FFT utility.
4. lotsa memory, which shouldn't be a problem nowadays.
what comes out will be a collection of phase-aligned, single period,
wavetables and after FFTing them, you will have the instantaneous
amplitude and instantaneous phase of each harmonic for that wavetable
snapshot. if the phase changes in time, that means the harmonic is
actually detuned away from its exact integer multiple of the
fundamental. so if you know the fundamental frequency (from the pitch
detector), the index number of the harmonic (from the bin number of the
FFT), and the rate of change of phase (from comparing the phase of some
particular harmonic of two neighboring wavetables), you have all of the
information for a precise measurement of the instantaneous frequency of
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