On Sun, Mar 8, 2020 at 11:41 PM Ethan Duni <ethan.d...@gmail.com> wrote: > FFT filterbanks are time variant due to framing effects and the circular > convolution property. They exhibit “perfect reconstruction” if you design the > windows correctly, but this only applies if the FFT coefficients are not > altered between analysis and synthesis. If you alter the FFT coefficients > (i.e., “filtering”), it causes time domain aliasing.
If the system is suitably designed (e.g. correct window and overlap), you can filter using an FFT and get identical results to a time domain FIR filter (up-to rounding/precision limits, of course). The appropriate window and overlap process will cause all circular convolution artefacts to cancel. In fact, one particularly computationally efficient method to apply a very long FIR filter (e.g. for reverbs) with low delay is to factor it into a low delay portion and one or more longer delay chunks and use naive convolution for the low delay portion and large overlapped FFTs for the high delay portions. _______________________________________________ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
