At Thu, 29 Jun 2006 23:11:05 +0100,
Alan Horstmann wrote:
>
> On Tuesday 20 June 2006 10:56, you wrote:
> > At Mon, 19 Jun 2006 22:12:57 +0100,
> >
> > Alan Horstmann wrote:
> > > Another alternative would be for ALSA to create at least two devices, one
> > > for all playback channels, the other for all capture, eg /dsp0 and /adsp0
> > > or something else. It would then be more like the raws OSS in its
> > > behaviour. That would also solve the problems here. Obviously a full
> > > implementation as per attached description would also be fine!
> >
> > It's easy over aoss. Having the following
> >
> > pcm.dsp0 {
> > type asym
> > playback.pcm "plughw"
> > }
> >
> > pcm.dsp1 {
> > type asym
> > capture.pcm "plughw"
> > }
> >
> > would give you /dev/dsp0 for playback and /dev/dsp1 for capture.
> > You can replace "plughw" whatever you like, of course.
>
> I'm not very familiar with these conf files, and have tried for hours to get
> this to work. However, at best
> aoss audacity
> results in
> ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) playback slave is not defined
> ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) capture slave is not defined
>
> Have I missed other bits?
>
> Also, won't using aoss introduce a performance hit?
Err, I should have tested before posting. How about below?
pcm.dsp0 {
type asym
playback.pcm "plughw"
capture.pcm "null"
}
pcm.dsp1 {
type asym
playback.pcm "null"
capture.pcm "plughw"
}
Takashi
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