[pulseaudio-discuss] Emulate Alsa A52 plugin in pulseaudio ?

[Jim Duda]

Tanu,

Okay, what you are describing makes sense.  We route the 6 channels back through alsa and the a52 encoding, then out the 
actual device driver.

The front-spdif was leftover stuff in my asoundrc file.

I also need to send the raw AC3, DTS stream from some applications to the external digital decoder (mplayer and xine). 
Currently, I do this by using alsa:device=spdif in the applications which require this mode, instead of the stereo 
upmix.

So, to accomplish this, do I define two alsa-sinks?

module-load module-alsa-sink sink_name=ac3_out device=a52encode channels=6 rate=48000
module-load module-alsa-size sink_name=ac3_raw device=surround51:0

Then my .asoundrc has this:

pcm.!default {
   type pulse
   device ac3_out
}

pcm.passthrough {
  type pulse
  device ac3_raw
}

pcm.a52encode {
   type a52
}

The default would do stereo upmix from 2 to 6 channels through a52 encoder.

The passthrough would send the raw ac3/dts stream out the hardware.

Does this make sense?  (I would test now, but I have to go off and build 0.9.8 first, since FC7 uses 0.9.6 currently).

Thanks,

Jim


"Tanu Kaskinen" <tanuk at iki.fi> wrote in message news:20080228205506.GA11317 at a9a.mannikko1.tontut.fi...
> On Thu, Feb 28, 2008 at 03:09:45PM -0500, Jim Duda wrote:
>> I would like to use pulseaudio on a machine which I have the sound card attached to an digital decoder.  I'm using 
>> the
>> alsa A52 plugin to perform a stereoupmix from 2 channels to six channels such that I get the same stereo out of the
>> front and rear speakers.
>
>> Can I use the remap module to copy 2 channels to 4?  The front speaker and sub woofer would be nice too.
>
> Yes you can, but there shouldn't be need for that. Since
> 0.9.8 PulseAudio has supported automatic up- and downmixing,
> which probably does what you want. If you have 0.9.8 and it
> still doesn't work, check that you haven't disabled the
> feature in daemon.conf by saying disable-remixing=yes.
>
> If I've understood your setup correctly, you would need to
> encode the output of PulseAudio to AC-3. I don't have any
> experience in that field, so the following is just my best
> guess how it would work:
>
> Your new ~/.asoundrc:
>
> pcm.!default {
>    type pulse
> }
>
> pcm.a52encode {
>    type a52
> }
>
> # What's this for?
> pcm.front-spdif {
>    type plug
>    slave.pcm "iec958"
> }
>
>
> Comment out module-hal-detect and module-detect in
> /etc/pulse/default.pa. Add this line instead:
> module-load module-alsa-sink sink_name=ac3_out device=a52encode channels=6 rate=48000
>
> -- 
> Tanu Kaskinen