> Does anyone have a GOOD quality Sample Rate conversion algorithm ?
See any elementary discrete time audio processing book and look under 'integer
ratio rate conversion'. It's mostly an exercise in upsampling, lowpassing and
downsampling. The computational complexity is proportional to the product of
the ratio integers.
For example, going from 22050 to 44100, just make every other sample a zero,
then apply a lowpass (however you like) with a cutoff just below 11.025 kHz.
Now downsampling is needed in this case (well, the downsample factor is 1).
This is a potentially expensive algorithm (eg, 44.1kHz to 8kHz requires much
more suckage), but classically the most 'correct'. It will exhibit no blocking
effects and minimal aliasing (you'll be limited by the shape of your lowpass,
but that will be true of any approach). It can be implemented in either the
time or frequency domain (time domain is faster for low order filters, cheating
with an FFT is faster for high orders).
There are other slightly messier ways to do things with FFTs so long as one
pays attention to edge effects.
Monty
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
