[FFmpeg-user] libaacplus seg fault
Mike F
paziu at yahoo.com
Fri Mar 22 21:13:05 CET 2013
disabling FFT code on libaacplus-2 is the "answer".
libaacplus-2.0.2 # ./configure --with-fftw3=no
and
# ffmpeg -v debug -y -i audio.raw -acodec libaacplus -ar 48k -ab 64k -ac 2 audio.libaacplus.aac
works just fine....
I should find this info without posting here.... sorry.
Mike
----- Original Message -----
> From: Mike F <paziu at yahoo.com>
> To: FFmpeg user questions <ffmpeg-user at ffmpeg.org>
> Cc:
> Sent: Friday, March 22, 2013 3:36 PM
> Subject: Re: [FFmpeg-user] libaacplus seg fault
>
>t ied 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
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
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