On Tue, 23 Nov 1999, Reckhard, Tobias wrote:
> > *Specifically* I'd be looking for someone intimately familiar with direct
> > form
> > II IIR filters, the Levinson-Durbin and Schur algorithms for producing
> > coefficient sets and the underlying linear and discrete math.
> >
> Sorry, I don't know these filter types. Would you have any literature on
> them? What do they do and what does IIR stand for?
IIR = Infinite Impulse Response, It's a class of filters as opposed to FIR
(finite impluse). It's covered in a lot of books.
Somewhere out there there is a version of the celp alg. source code
floating around that is very well commented. It uses LPC/LSP
(w/Levinson-Durbin I believe, I seem to recall autocoorlation) , pitch
tracking, codebook quantazition, and residule coding. Very informative.
> LPC = Linear Predicting Codec? (LP very probably means linear predict*, but
> what's the C?)
Linear Predictive coding. AFIR.
> Now an idea I have is to use a different set of FFT coefficients, or rather,
> to space them in a logarithmic way. If I understand correctly, you are
> transforming from the time domain to the frequency domain (which introduces
> an information loss, BTW). Normally, this results in a set of coefficients
> with arguments spaced evenly apart, e.g. you'd have coefficients for the
> energy of a signal at 1, 2, 3, 4, 5, ... Hz. What you want is a set of
> coefficients at logarithmic (base 2) intervals, e.g. at 1, 2, 4, 8, 16, ...
> Hz. You could achieve this by modifying the FFT code you use. Note that, to
> achieve losslessness, you need to keep track of the errors in all of your
> transformations and apply them when recreating the original.
If my understanding is correct, he's looking to create a LPC type alg that
better fits the mechanics of music while still operating in the time
domain.
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