[FFmpeg-user] libaacplus seg fault

Fri Mar 22 17:54:00 CET 2013

Hi All,

just got the latest ffmpeg git and libaacplus-2.0.2

ffmpeg version N-51213-g076c1c9 Copyright (c) 2000-2013 the FFmpeg developers
  built on Mar 22 2013 12:30:28 with gcc 4.5.2 (Gentoo 4.5.2 p1.1, pie-0.4.5)

 configuration: --enable-libx264 --enable-libxvid --enable-libfaac --enable-libmp3lame --enable-libvorbis --enable-gpl --enable-nonfree --enable-libtheora --enable-x11grab --enable-vdpau --enable-libfdk-aac --enable-libass --enable-libvpx --enable-libaacplus
  libavutil      52. 22.101 / 52. 22.101
  libavcodec     55.  1.100 / 55.  1.100
  libavformat    55.  0.100 / 55.  0.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 48.100 /  3. 48.100
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100

once executed the following:

# ffmpeg -v debug -y -i audio.raw -acodec libaacplus -ar 44100 -ab 32k -ac 2 audio.libaacplus.aac

i get:

Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set libav* logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-i' ... matched as input file with argument 'audio.raw'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libaacplus'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
Reading option '-ab' ... matched as AVOption 'ab' with argument '32k'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option 'audio.libaacplus.aac' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set libav* logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file audio.raw.
Successfully parsed a group of options.
Opening an input file: audio.raw.
[dts @ 0x34e6520] Format dts probed with size=8192 and score=51
[dts @ 0x34e6520] File position before avformat_find_stream_info() is 0
[dca @ 0x34e6e40] Stream with high frequencies VQ coding
[dts @ 0x34e6520] max_analyze_duration 5000000 reached at 5002667 microseconds
[dts @ 0x34e6520] Estimating duration from bitrate, this may be inaccurate
[dts @ 0x34e6520] File position after avformat_find_stream_info() is 950272
Input #0, dts, from 'audio.raw':
  Duration: 01:49:48.78, start: 0.000000, bitrate: 1535 kb/s
    Stream #0:0, 471, 1/90000: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file audio.libaacplus.aac.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument libaacplus.
Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an output file: audio.libaacplus.aac.
Successfully opened the file.
[abuffer @ 0x34ea3e0] Setting entry with key 'time_base' to value '1/48000'
[abuffer @ 0x34ea3e0] Setting entry with key 'sample_rate' to value '48000'
[abuffer @ 0x34ea3e0] Setting entry with key 'sample_fmt' to value 'fltp'
[abuffer @ 0x34ea3e0] Setting entry with key 'channel_layout' to value '0x60f'
[graph 0 input from stream 0:0 @ 0x34e7c00] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[aformat @ 0x34eac20] Setting entry with key 'sample_fmts' to value 's16'
[aformat @ 0x34eac20] Setting entry with key 'sample_rates' to value '44100'
[aformat @ 0x34eac20] Setting entry with key 'channel_layouts' to value '0x3'
[audio format for output stream 0:0 @ 0x34ea940] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
[auto-inserted resampler 0 @ 0x34ec2c0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz
Output #0, adts, to 'audio.libaacplus.aac':
  Metadata:
    encoder         : Lavf55.0.100
    Stream #0:0, 0, 1/90000: Audio: aac, 44100 Hz, stereo, s16, 32 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (dca -> libaacplus)
Press [q] to stop, [?] for help
[dca @ 0x34e6e40] Stream with high frequencies VQ coding
Segmentation fault
gentoo3 My.Brother.The.Devil.2012.LiMiTED.720p.BluRay.X264-7SinS # ffmpeg -v debug -y -i audio.raw -acodec libaacplus -ar 48k -ab 32k -ac 2 audio.libaacplus.aac
ffmpeg version N-51213-g076c1c9 Copyright (c) 2000-2013 the FFmpeg developers
  built on Mar 22 2013 12:30:28 with gcc 4.5.2 (Gentoo 4.5.2 p1.1, pie-0.4.5)
  configuration: --enable-libx264 --enable-libxvid --enable-libfaac --enable-libmp3lame --enable-libvorbis --enable-gpl --enable-nonfree --enable-libtheora --enable-x11grab --enable-vdpau --enable-libfdk-aac --enable-libass --enable-libvpx --enable-libaacplus
  libavutil      52. 22.101 / 52. 22.101
  libavcodec     55.  1.100 / 55.  1.100
  libavformat    55.  0.100 / 55.  0.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 48.100 /  3. 48.100
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set libav* logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-i' ... matched as input file with argument 'audio.raw'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libaacplus'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '48k'.
Reading option '-ab' ... matched as AVOption 'ab' with argument '32k'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option 'audio.libaacplus.aac' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set libav* logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file audio.raw.
Successfully parsed a group of options.
Opening an input file: audio.raw.
[dts @ 0x17da520] Format dts probed with size=8192 and score=51
[dts @ 0x17da520] File position before avformat_find_stream_info() is 0
[dca @ 0x17dae40] Stream with high frequencies VQ coding
[dts @ 0x17da520] max_analyze_duration 5000000 reached at 5002667 microseconds
[dts @ 0x17da520] Estimating duration from bitrate, this may be inaccurate
[dts @ 0x17da520] File position after avformat_find_stream_info() is 950272
Input #0, dts, from 'audio.raw':
  Duration: 01:49:48.78, start: 0.000000, bitrate: 1535 kb/s
    Stream #0:0, 471, 1/90000: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file audio.libaacplus.aac.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument libaacplus.
Applying option ar (set audio sampling rate (in Hz)) with argument 48k.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an output file: audio.libaacplus.aac.
Successfully opened the file.
[abuffer @ 0x17de3e0] Setting entry with key 'time_base' to value '1/48000'
[abuffer @ 0x17de3e0] Setting entry with key 'sample_rate' to value '48000'
[abuffer @ 0x17de3e0] Setting entry with key 'sample_fmt' to value 'fltp'
[abuffer @ 0x17de3e0] Setting entry with key 'channel_layout' to value '0x60f'
[graph 0 input from stream 0:0 @ 0x17dbc00] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[aformat @ 0x17dec20] Setting entry with key 'sample_fmts' to value 's16'
[aformat @ 0x17dec20] Setting entry with key 'sample_rates' to value '48000'
[aformat @ 0x17dec20] Setting entry with key 'channel_layouts' to value '0x3'
[audio format for output stream 0:0 @ 0x17de940] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
[auto-inserted resampler 0 @ 0x17e02c0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
Output #0, adts, to 'audio.libaacplus.aac':
  Metadata:
    encoder         : Lavf55.0.100
    Stream #0:0, 0, 1/90000: Audio: aac, 48000 Hz, stereo, s16, 32 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (dca -> libaacplus)
Press [q] to stop, [?] for help
[dca @ 0x17dae40] Stream with high frequencies VQ coding
Segmentation fault

Am I doing anything wrong here?

Thanks,

Mike