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" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> 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
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