I'm writing a software analogue synth. I have been using the Overlap-add method to cheaply implement a low-pass filter using FFT.
My low-pass filter has had a constant impulse response; a constant cutoff frequency. Now I'd like to make it possible to vary the cut-off frequency while a note is playing (this is a common feature of analogue synths).
I believe this is impossible while still using the circular convolution theorem trick (that let's us use FFT to implement FIR-filters).
- Am I right in thinking so?
- If so, is there some other trick I can use to efficiently implement a band-pass filter whose cut-off frequency varies ("sweeps")?