[FFmpeg-user] ac3 to mp3 converting bug

[Ingo Brückl]

I use to convert ac3 (dolby) audio files using ffmpeg linked with a static
version of lame to mp3 which was working fine so far. Since recently this
doesn't work any longer. The mp3 output file seems ok, but does contain
nothing but 0x55 blocks and thus is mute. A conversion from an other format
(like wav) to mp3 works fine, so it can't be lame.

Has anything changed recently that I've missed or is there a bug?

Ingo
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$ ffmpeg -i Broadway-5.1-48khz-448kbit.ac3 -acodec libmp3lame -ab 128k -ac 2 test.mp3 2> log1
ffmpeg version git-2012-07-01-f7cba73 Copyright (c) 2000-2012 the FFmpeg developers
  built on Jul  1 2012 17:27:19 with gcc 4.3.4
  configuration: --prefix=/usr --enable-shared --disable-static --enable-pthreads --enable-gpl --enable-postproc --enable-libmp3lame --extra-cflags=-I/var/tmp/usr/include --extra-libs=-L/var/tmp/usr/lib --disable-protocol=ipv6 --disable-ffserver
  libavutil      51. 63.100 / 51. 63.100
  libavcodec     54. 31.100 / 54. 31.100
  libavformat    54. 14.100 / 54. 14.100
  libavdevice    54.  0.100 / 54.  0.100
  libavfilter     3.  0.100 /  3.  0.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[ac3 @ 0x93ee3c0] max_analyze_duration 5000000 reached at 5024000
[ac3 @ 0x93ee3c0] Estimating duration from bitrate, this may be inaccurate
Input #0, ac3, from 'Broadway-5.1-48khz-448kbit.ac3':
  Duration: 00:00:33.98, start: 0.000000, bitrate: 448 kb/s
    Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s
[graph 0 input from stream 0:0 @ 0x9449920] tb:1/90000 samplefmt:s16 samplerate:48000 chlayout:0x60f
[auto-inserted resampler 0 @ 0x9462420] chl:5.1(side) fmt:s16 r:48000Hz -> chl:stereo fmt:s16 r:48000Hz
Output #0, mp3, to 'test.mp3':
  Metadata:
    TSSE            : Lavf54.14.100
    Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (ac3 -> libmp3lame)
Press [q] to stop, [?] for help
size=     121kB time=00:00:07.68 bitrate= 128.8kbits/s    
size=     243kB time=00:00:15.50 bitrate= 128.4kbits/s    
size=     342kB time=00:00:21.81 bitrate= 128.3kbits/s    
size=     464kB time=00:00:29.66 bitrate= 128.2kbits/s    
size=     532kB time=00:00:33.98 bitrate= 128.2kbits/s    

video:0kB audio:531kB subtitle:0 global headers:0kB muxing overhead 0.077004%
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$ ffprobe test.mp3 2> log2
ffprobe version git-2012-07-01-f7cba73 Copyright (c) 2007-2012 the FFmpeg developers
  built on Jul  1 2012 17:27:19 with gcc 4.3.4
  configuration: --prefix=/usr --enable-shared --disable-static --enable-pthreads --enable-gpl --enable-postproc --enable-libmp3lame --extra-cflags=-I/var/tmp/usr/include --extra-libs=-L/var/tmp/usr/lib --disable-protocol=ipv6 --disable-ffserver
  libavutil      51. 63.100 / 51. 63.100
  libavcodec     54. 31.100 / 54. 31.100
  libavformat    54. 14.100 / 54. 14.100
  libavdevice    54.  0.100 / 54.  0.100
  libavfilter     3.  0.100 /  3.  0.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[mp3 @ 0x91ad4c0] max_analyze_duration 5000000 reached at 5016000
Input #0, mp3, from 'test.mp3':
  Metadata:
    encoder         : Lavf54.14.100
  Duration: 00:00:34.00, start: 0.000000, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s