[MPlayer-users] audio just stops when playing streamed source w/AC3-liba52

[michael higgins]

Hello --

I've been musing over this issue for a while now. I've just now managed
to reproduce it and compare the result. The audio just stops at a
certain point when using the liba52 codec? I've tried my best to give 
complete info... I apologise in advance for any lack of effort on my part.

Anyway, here's [with some output snipped] what happens:

MPlayer dev-CVS-040401-10:59-3.2.2 (C) 2000-2004 MPlayer Team

CPU: Intel Pentium II Klamath/Pentium II OverDrive 297.8 MHz (Family: 6,
Stepping: 4)
Detected cache-line size is 32 bytes
CPUflags:  MMX: 1 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0
Compiled for x86 CPU with extensions: MMX

[snip the usual personal .conf stuff]

Falling back on default (hardcoded) input config

Playing pnm://rm.bbc.co.uk/worldservice/live/livenews.ra.
Resolving rm.bbc.co.uk for AF_INET...
Connecting to server rm.bbc.co.uk[212.58.240.61]:7070 ...
PNM:// fd=6
Cache size set to 128 KBytes
Connected to server: rm.bbc.co.uk
Cache fill: 18.75% (24576 bytes)    REAL file format detected.
======= WAVE Format =======
Format Tag: 28260 (0x6E64)
Channels: 1
Samplerate: 11025
avg byte/sec: 16000
Block align: 278
bits/sample: 16
cbSize: 0
===========================
demux_real: invalid chunksize! (0)
Clip info:
  name: BBC World Service
  author:
  copyright: (C) British Broadcasting Corporation 2004
==========================================================================
Opening audio decoder: [liba52] AC3 decoding with liba52
No accelerated IMDCT transform found
AC3: 1.0 (mono)  11025 Hz  16.0 kbit/s
Using MMX optimized resampler
AUDIO: 11025 Hz, 2 ch, 16 bit (0x10), ratio: 2000->44100 (16.0 kbit)
Selected audio codec: [a52] afm:liba52 (AC3-liba52)
==========================================================================
Checking audio filter chain for 11025Hz/2ch/16bit -> 11025Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 11025 hz, little endian signed int
AF_pre: 11025Hz 2ch Signed 16-bit (Little-Endian)
alsa-init: requested format: 11025 Hz, 2 channels, Signed 16-bit
(Little-Endian)
alsa-init: 1 soundcard found, using: hw:0,0
alsa1x: 11025 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit
Little Endian
AO: [alsa1x] 11025Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 11025Hz/2ch/16bit -> 11025Hz/2ch/16bit...
Video: no video
Starting playback...
A:   2:32:13.9  0.7% 23%


Okay. Here's the thing. The audio just stops here. The stream apparently
continues to download, but the audio is silent. No error is reported.

I have to kill it, (not quit):

MPlayer interrupted by signal 2 in module: decode_audio
alsa-uninit: pcm closed

And do it again:

[ same open and connect stuff snipped ]

Playing pnm://rm.bbc.co.uk/worldservice/live/livenews.ra.

[ same stream info stuff snipped ]

Starting playback...
A:   2:32:14.0  0.7% 23%

And again, suddenly no audio, but stream apparently continues, and I 
have to kill it.

So, why this magic number of 2:23:14?

I'll try to test a different stream in the morning to see if this number
is the same.... okay, it's been morning. Tested on a different stream, 
no issue. Played for 3 1/2 hours, no glitches (just some initial 
underruns).

Here's the stuff about this stream that works fine:

Playing rtsp://128.208.34.90:554/broadcast/live.rm?cloakport=80,554,7070.
Connecting to server 128.208.34.90[128.208.34.90]:554 ...
Cache size set to 64 KBytes
Connected to server: 128.208.34.90
Cache fill: 12.50% (8192 bytes)    REAL file format detected.
======= WAVE Format =======
Format Tag: 29793 (0x7461)
Channels: 2
Samplerate: 44100
avg byte/sec: 176400
Block align: 1024
bits/sample: 16
cbSize: 20
Unknown extra header dump: [0] [2] [1e] [0] [5] [0] [0] [4] [a] [0] [0] 
[0] [0] [4] [8] [0] [8] [8e] [0] [2]
===========================
demux_real: invalid chunksize! (0)
Clip info:
  name: KUOW Public Radio Live
  author: KUOW Public Radio
  copyright: 2003 KUOW Public Radio
==========================================================================
Opening audio decoder: [realaud] RealAudio decoder
opening shared obj '/usr/local/lib/codecs/atrc.so.6.0'
Audio codec: [5] 176 Kbps Stereo Music - RA 8
Audio bitrate: 176.400 kbit/s (22050 bps)
AUDIO: 44100 Hz, 2 ch, 16 bit (0x10), ratio: 22050->176400 (176.4 kbit)
Selected audio codec: [raatrc] afm:realaud (RealAudio ATRAC3)
==========================================================================
Checking audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 44100 hz, little endian signed int
AF_pre: 44100Hz 2ch Signed 16-bit (Little-Endian)
alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit 
(Little-Endian)
alsa-init: 1 soundcard found, using: hw:0,0
alsa1x: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit 
Little Endian
AO: [alsa1x] 44100Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
Video: no video
Starting playback...
alsa-play: xrun of at least 35.136 msecs. resetting stream
alsa-play: xrun of at least 25.034 msecs. resetting stream
alsa-play: xrun of at least 0.063 msecs. resetting stream
alsa-play: xrun of at least 96.105 msecs. resetting stream
alsa-play: xrun of at least 249.537 msecs. resetting stream
alsa-play: xrun of at least 61.421 msecs. resetting stream
alsa-play: xrun of at least 75.380 msecs. resetting stream
alsa-uninit: pcm closed%

[the time read ~ 3 1/2 hours. I guess the elapsed time line goes away 
when you quit, rather than kill the proggy].

...

So, if there's any advice to be granted by the list as to how best to 
help find the problem or arrive at a fix, I'd love to hear it. All I can 
understand is that it's using a different audio codec,

[a52] afm:liba52 (AC3-liba52)

v.s.

[raatrc] afm:realaud (RealAudio ATRAC3)

-- mike higgins