[music-dsp] Anti-Aliasing Filter With FFT/iFFT
dwerner at experimentalscene.com
Sun Sep 19 05:34:00 EDT 2004
vesa norilo wrote:
> You could do it like this but I guess you'd rather not...
> - Every time your sample playback frequency changes, you'll have to
> reconstruct your anti-aliasing filter. What about when a note starts
> mid-block? Part of the block requires one antialiasing filter, part of
> it another. What about when there's a vibrato? This could mean you
> have to reconstruct the filter and FFT it at all control rate frames.
> Plus the problem of filtering different parts of your block with
> different filters.
The plugin I'm writing is for DarkWave Studio which never changes the
samplerate or block size mid-block, so this isn't a problem. Do you mean
that I have to change the filter when the note changes? I thought you
would only have to change the filter when the oversampling rate, sample
rate or block size changes.
> - The filter orders where FFT starts to have a significant advantage
> over FIR are probably very much overkill for sampler interpolation.
> Commercial samplers have for some time used 4th to 8th order
> resampling and for filter orders so small, FFT is going to be slower
> than straightforward FIR.
I assume by order you mean the amount of oversampling, what about the
number of zero crossings in the filter?
> For this application, you might want to stick with Lagrange
> interpolation (which is equivalent to sinc with binomial window)
I'll consider Lagrange, I haven't read about it yet, but If you think
it's good I'm sure it's worth a look, thanks.
- Daniel Werner
More information about the music-dsp