>> 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/ )
I am really looking for implimented and tested source code.
Any other advice ?
Thanks
Mark Gilbert.
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