Shawn Riley wrote:
>
> >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
I don't think that 3 would be good, since it would affect a rather
large range of frequencies ( the lowest band is for 38 Hz, as someone
recently mentioned, I think at least freq to 50 Hz would be affected )
david balazic
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