This is probably a reasonable compromise for going in to the
kernel since it is efficient.  It leads to rather poor audio
quality though, so we might consider more complicated filters
in the future.

	-- Lennart

TAMURA Kent wrote:
> 
> In message "Re: Proposal: Audio: Sampling rate conversion"
>     on 02/03/04, Lennart Augustsson <lennart@augustsson.net> writes:
> > Can you describe the algoritm you use to perform the sample rate
> > conversion?
> 
> Simple resampling like "dst[i] = src[i*srcrate/dstrate]" is used
> for down-sampling, and linear interpolation is used for
> up-sampling.
> 
> --
> TAMURA Kent <kent2002@hauN.org> <kent@netbsd.org>

-- 

	-- Lennart