I'm building a real-time audio resampler (think pitch bend) that needs to have a several different performance vs quality configuration options. I understand that I'll need to apply a low-pass filter before resampling in order to avoid aliasing.
The requirements for the first filter are this:
- very efficient
- better than nothing
For this filter, I don't mind if I have a very slow roll-off, or if there's significant phase distortion. I just want it to be as fast as possible, and to attenuate frequencies above nyquist to some degree.
The requirements for the second filter are:
- reasonably efficient
- significant reduction in frequencies above nyquist
I'm guessing that I'll want to go with one of the classic algorithms I hear about for the second filter (chebyshev, butterworth), and potentially something like a recursive moving average filter for the first.
I realize my question may so vague as to be useless ("very efficient" and "reasonably efficient" are obviously relative terms!) I realize that filter selection and design is an art that one can spend a lifetime on. I'm quite inexperienced with DSP and am just looking for a couple algorithms I can get my hands dirty with. I'm hoping someone with audio experience can pip up and say "here are my general purpose, go-to algorithms for efficient lowpass filtering".