I'm working on a 10-second sound, sampled at 44.1 khz.
I want to do filtering, and have a desired EQ (equalization) curve that varies over time, as suggested here (here $f0=250\ Hz$)
How to implement a time-varying filter?
How to go from this bad idea of zeroing FFT bins
[0, 0, ..., 0, 1, 0, ..., 0, 1, 0, ..., 0, 1, 0, ... 0] | | | | | freq bins for f0, 2*f0, 3*f0 are kept | most freq bins are zeroed
to a good filtering?
Then, how to make this vary over time? Should I use overlap-add, etc.? Then if so, can't I do it inside a STFT framework?
Would you have a pseudo code or Python demo?
PS: I've read http://recherche.ircam.fr/anasyn/roebel/amt_audiosignale/VL3.pdf, but I can't go from here to a working algorithm.