[MPlayer-users] hwac3 not working - WORKING now

Jean-Pierre Gygax jpgygax at swissonline.ch
Sat Apr 26 18:29:04 CEST 2003

> Can't set audio device /dev/dsp to AC3 output, trying S16...
> That's the line that keys into your problem.  ON SBLive cards for
> example with alsa you have to provide the hardware for spdif output (the
> analog output of emu10k1(sblive) is hw:0,0 ):
> -ao alsa9:hw:0,1:surround51 -ac hwac3
> It might be a similar situation for the onboard SS51G

Thanks Jeremy,

and sorry it took so long to reply, I have been busy reconfiguring my box.
After getting the same symptom under OSS that I got with ALSA, I thought it
might be a good idea to go back to ALSA so I could try out your suggestion,
especially since I remembered that the optical output had at least been
turned on (visible red light coming out of the cable). Getting an
ALSA-enabled kernel 2.5.67 to work (again) turned out to be real hard,
because I had messed up my Linux installation quite a bit (kernel panic upon
boot, among other things), but I finally managed.

In short, everything is now working perfectly. The -ao option has nothing to
do with it, though - the problem was in the mixer settings. I wish I had a
better understanding of what is going on there, but the right setting to
have the pass-through working is - no setting at all! That's right, boot up
your machine, making sure that no script touches your mixer settings in any
way, and that's how you're going to get perfect AC3 pass-through.

In hindsight, that makes sense, only there had been so much misconfiguration
at first (not only with my Linux box but also with the settings of my
amplifier) that I was unable to isolate the problem. The one decisive
setting in alsamixer apparently is the "analog to IEC958" item, which must
be muted.

It's odd. I cannot seem to reproduce the symptom (horrible noise) any more.
I have found out through experimentation though that in order to play music
from another source than the pass-through (tried with a RealAudio file), I
need to configure alsamixer like this:

- set a capture on the "capture" item (whatever that actually does, I
- set a capture on the "mixer" item
- unmute PCM
- volume is then controlled through the "capture" and "PCM" levels

Can't say it's very satisfactory to inch forward like this. Is there a good
document somewhere that explains what all the stuff in alsamixer means?

Thanks for the help!

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