[openbeos] Re: Resampling algorithms for audio mixer
- From: "Stefano D'Angelo" <zanga.mail@xxxxxxxxx>
- To: openbeos@xxxxxxxxxxxxx
- Date: Wed, 28 Nov 2007 19:22:56 +0100
2007/11/27, cyanh256@xxxxxxxxxxxx <cyanh256@xxxxxxxxxxxx>:
> The other thing I noticed while skimming through was that
> there doesn't seem to be any kind of encoder state
> preservation between calls to the resampling function.
> When working with small buffers (typical of soundcard
> buffers in BeOS -- I'm using a buffer size of 64 frames),
> there's the danger of inaccuracies appearing at the initial
> (or last) samples, which could become audible as a tone
> at these buffer sizes. Providing some way for the function
> to know what samples occured at the end of the previous
> buffer (such as passing a state struct into the resampling
> function) generally clears it up.
What about "interpolating remaining one sample behind the input
buffer" (so that you cannot have to interpolate values beyond the
innput buffer) and passing the last 3 or 4 samples of the previous
input buffer through a struct, as you suggested, so that the values
you now have to inteprolate before the input buffer are well defined?
Stefano
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