> 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.)
> 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.)
> 3- Remove the lowest frequency band(s) from the FFT data
The resampler uses a low pass filter, so I guess it'd just be a case of
modifying the code to reshape the filter.
I don't know how many points you need to do a 16Hz filter though
Scott Manley (aka Szyzyg) /------ _@/ Mail -----\
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