> I have 3 ideas, but I'm not sure if they'd even work, let alone how to code them if
>they did...
> 1- Try simulating the behaviour of ideal capacitors/resistors/op-amps/etc. Maybe a
>highpass filter could be "built" out of computer code & put just before the
>resampling function. (Or is *that* what the "convolution approach" is? Oops.)
Well, you can simulate any electronics in an algorithm; the other way around
can be quite difficult ;-) If you simulate some RC or RCL or something
network, you end up having a IIR filter. That's easier to program directly...
> 2- Resample the input file to something ridiculously low like 40Hz, & subtract this
>from the original WAV. (Although that would probably require the above method anyway.)
Resampling requires filtering --> you end up with the same problem.
> 3- Remove the lowest frequency band(s) from the FFT data
dct/mdct you mean, I guess. But this removes everything up to 38 Hz
(see previous post), and introduces some horrible aliasing as well.
Ciao,
Segher
>
> Shawn
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