On Thu, 30 Mar 2006 16:06:10 +0200
Takashi Iwai <[EMAIL PROTECTED]> wrote:
> At Mon, 27 Mar 2006 19:59:22 +0200,
> Clemens Ladisch wrote:
> >
> > Sergei Steshenko wrote:
> > > Clemens Ladisch <[EMAIL PROTECTED]> wrote:
> > > > ALSA's rate plugin currently is limited to linear interpolation,
> > > > too, so you would have to add a better resampler to it.
> > >
> > > Can't you guys (ALSA developers, I mean) define an API for resampling ?
> >
> > I think it would be easier (faster) to just add another resampler to
> > pcm_rate.c than to design and implement such an API.
> >
> > However, fell free to submit a patch that does it in whatever way you
> > like. (Then you wouldn't have to wait until I have enough time to get to
> > the resampler entry of my ToDo list. :-)
>
> Well, I've had such one on my local tree for testing purpose, but not
> released publicly. A rate-converter SDK and an external plugin using
> libsamplerate.
>
> Looks like we have enough demands on it. OK, I'll clean it up and
> commit to CVS soon later (currently sourceforge is again refusing
> connections...)
>
>
> Takashi
>
Thanks - this seems like a very logical solution/decision to me.
...
I was thinking about somewhat wider issue - "connection" of audion
applications to ALSA sound system.
I haven't yet used dmix, but I think the proper architecture should be:
|------|
audio_application_1----->---rate_converter_if_needed---->| dmix |
audio_application_2----->---rate_converter_if_needed---->| |
... | |--->card_driver
audio_application_N----->---rate_converter_if_needed---->| |
|------|
.
I.e. ALSA configuration should allow multiple instances of the rate converter
- an instance per audio source.
Regards,
Sergei.
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