[MPlayer-users] pulseaudio failing at passthrough

[Alexander Roalter]

If I play back AC3 audio with ao pulse and S/PDIF passthrough, I get

Playing download/www_lynnemusic_com_surround_test.ac3.
libavformat version 55.22.103 (internal)
libavformat file format detected.
[ac3 @ 0x7fb8560f44a0]Estimating duration from bitrate, this may be 
inaccurate
[lavf] stream 0: audio (ac3), -aid 0
Load subtitles in download/
==========================================================================
Trying to force audio codec driver family hwac3...
Opening audio decoder: [hwac3] AC3/DTS pass-through S/PDIF
hwac3: switched to AC3, 448000 bps, 48000 Hz

AUDIO: 48000 Hz, 2 ch, ac3be, 448.0 kbit/29.17% (ratio: 56000->192000)
Selected audio codec: [hwac3] afm: hwac3 (AC3 through S/PDIF)
==========================================================================
AO: [pulse] 48000Hz 6ch s16le (2 bytes per sample)
[format] Sample format big-endian AC3 not yet supported
[libaf] Reinitialization did not work, audio filter 'format' returned 
error code -2
[libaf] Unable to setup filter system can not meet sound-card demands, 
please send a bug report.
Couldn't find matching filter/ao format!
Audio: no sound


Exiting... (End of file)

What gives?


Even stranger, with dts audio:

Playing dts.vob.
libavformat version 54.28.100 (internal)
MPEG-PS file format detected.
VIDEO:  MPEG2  720x576  (aspect 3)  25.000 fps  9800.0 kbps (1225.0 kbyte/s)
Load subtitles in ./
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
libavcodec version 54.58.100 (internal)
Selected video codec: [ffmpeg2] vfm: ffmpeg (FFmpeg MPEG-2)
==========================================================================
==========================================================================
Trying to force audio codec driver family hwac3...
Opening audio decoder: [hwac3] AC3/DTS pass-through S/PDIF
hwac3: switched to DTS, 1536000 bps, 48000 Hz

AUDIO: 48000 Hz, 2 ch, ac3le, 1536.0 kbit/100.00% (ratio: 192000->192000)
Selected audio codec: [hwdts] afm: hwac3 (DTS through S/PDIF)
==========================================================================
AO: [pulse] 48000Hz 6ch s16le (2 bytes per sample)
[format] Sample format little-endian AC3 not yet supported
[libaf] Reinitialization did not work, audio filter 'format' returned 
error code -2
[libaf] Unable to setup filter system can not meet sound-card demands, 
please send a bug report.
Couldn't find matching filter/ao format!
Audio: no sound
Starting playback...
[VD_FFMPEG] Trying pixfmt=0.
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
The selected video_out device is incompatible with this codec.
Try appending the scale filter to your filter list,
e.g. -vf spp,scale instead of -vf spp.
Movie-Aspect is 1.78:1 - prescaling to correct movie aspect.
VO: [xv] 720x576 => 1024x576 Planar YV12
V:   7.0 169/169 10%  0%  0.0% 0 0 


Exiting... (Quit)


Why does it even think that the format is litle endian AC3 for a dts stream?

Also, I fail to compile the most recent svn version because of

stream/stream_dvdnav.c:622:35: error: 'MSGTR_DVDnumChapters' undeclared 
(first use in this function)




-- 
Cheers,
Alex