Which anti aliasing filter algorithm for efficient audio resampling? 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".
For a first cut, I would consider a windowed Sinc interpolation kernel for both resamplers.  Very narrow window with a small polyphase table for the fast one.  Much wider window with more phases in a larger, possibly interpolated lookup table for the slower higher quality filtered interpolator/resampler.  Lots of parameters (including choice of window) that can be adjusted for both quality and performance to meet widely varying requirements.