Re: Quality resampling code for libasound

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On Tue, 2007-02-20 at 23:38 +1100, Jean-Marc Valin wrote:
> >>> That's one part making alsa-lib rate plugin so complicated.
> >>> Anyways, the resampling algorithm plugin doesn't take care much about
> >>> it, so far.
> >> Are you telling me people use the high-quality sinc in libsamplerate
> >> just to have the plugin drop samples once in a while? Can we just have a
> >> resampler that actually produces valid output once and for all? I guess
> >> it's another good reason to go directly with a non-plugin resampler.
> > 
> > No, I mean that there are many black magics in the alsa rate plugin to
> > calculate the sampling buffer size constant so that the resampler side
> > doesn't take care of it so much.  And this has nothing to do with the
> > fact whether it's an external plugin or not.
> 
> OK, so plugin or not, how do you actually make sure you process all
> samples and not just the ones that are the most convenient? I can't
> believe I'm the only one here who actually cares about having correct
> audio coming in/out the soundcard.

You are not alone (at least two of us? :-)

Linear interpolation works great as long as you don't have to listen to
the output :-) Maybe nobody does... If somebody needs linear interp
(slow processors?) they should be able to ask for it, but the default
should be something more usable (and a better match to the "Advanced"
part of the ALSA acronym). 

-- Fernando



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