>Does anyone know how a high pass filter is usually implemented? The
>convolution approach (like the low pass filter) seems like it would be
>expensive and require a lot of extra internal buffering: a 10Hz
>signal takes 4410 samples to represent one period. To get good
>frequency resolution (so you can tell the difference between the 10Hz
>signal and a 20Hz signal, for example) I would guess your window size
>would have to be at least twice that, or 8820 samples?
>
>Mark
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
Shawn
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