Re: Sample Rate Conversion Quality in ALSA

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At Mon, 21 Aug 2006 13:45:46 +0100,
James Courtier-Dutton wrote:
> 
> Marc Brooker wrote:
> > Hello,
> >
> > I am new to the ALSA development, and would be interested in
> > contributing in future.
> >
> > After installing the latest Ubuntu on my home machine, I noticed that
> > dmix is enabled by default. This is a very good move in my opinion, as
> > it makes Linux sound much more transparent to the end user. However, I
> > was concerned with the quality of the mixing algorithm and sample rate
> > conversion used by dmix.
> >
> >
> > Cheers
> >
> > Marc Brooker
> >   
> The problem is actually far more complicated than you described. A fair 
> amount of ALSA core will have to be modified to really fix the problem.
> The main problem is the buffer and period sized as they pass through dmix.
> For example, if the sound card hardware is running with 1024 samples per 
> period at 48000, and an application wished to use a sample rate of 
> 44100, the application should really get 940.8 samples per period. That 
> is nor possible, so the application gets 940. dmix then tries to convert 
> 940 samples to 1024 samples ready for the hardware. So, even if the dmix 
> algorithm used the super high quality sample rate conversion function, 
> the actual rate change applied would still be slightly off.

Well, the problem raised in the original post is rather the usability
issue.  As mentioned there, the quality issue can be solved by using
rate plugin in alsa-plugins package.  The question is how easily one
can set it up.

IMO, it's just a problem of lack of (so-called user-friendly)
configuration tool.


Takashi

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