[FFmpeg-user] libaacplus seg fault
Mike F
paziu at yahoo.com
Fri Mar 22 20:36:04 CET 2013
tied a few earlier versions of ffmpeg, same problem.
tried theree different releases of libaacplus ( 2.0.0 - 2.0.2 ) - aacplusenc seg faults every time on a wav file
========================
Complete name : audio.wav
Format : Wave
File size : 1.10 GiB
Duration : 1h 51mn
Overall bit rate mode : Constant
Overall bit rate : 1 411 Kbps
Writing application : Lavf55.0.100
Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Codec ID : 1
Duration : 1h 51mn
Bit rate mode : Constant
Bit rate : 1 411.2 Kbps
Channel(s) : 2 channels
Sampling rate : 44.1 KHz
Bit depth : 16 bits
Stream size : 1.10 GiB (100%)
=========================
went to libaacplus-1.1.0, aacplusenc does not seg fault, but ffmpeg requires version => 2.0.0
i do not know why all libaacplus => v2.0.0 are puking on my box....
this does not seem to be a problem with ffmpeg... ( my guess )
Thanks,
Mike
----- Original Message -----
> From: Mike F <paziu at yahoo.com>
> To: "ffmpeg-user at ffmpeg.org" <ffmpeg-user at ffmpeg.org>
> Cc:
> Sent: Friday, March 22, 2013 12:54 PM
> Subject: [FFmpeg-user] libaacplus seg fault
>
>
>
> 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
>
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