[FFmpeg-trac] #3205(undetermined:new): ffmpeg detects input as aac when processing corrupted pcm_s24le and pcm_s32le input
FFmpeg
trac at avcodec.org
Sat Dec 7 13:52:19 CET 2013
#3205: ffmpeg detects input as aac when processing corrupted pcm_s24le and
pcm_s32le input
-------------------------------------+-------------------------------------
Reporter: Fusl | Type: defect
Status: new | Priority: minor
Component: | Version:
undetermined | unspecified
Keywords: | Blocked By:
Blocking: | Reproduced by developer: 0
Analyzed by developer: 0 |
-------------------------------------+-------------------------------------
While trying to test some raw audio formats for under-the-hood processing
in my scripts, i came across this "issue":
How to reproduce with pcm_s24le acodec:
{{{
% ffmpeg -i inputfile -t 30 -acodec pcm_s24le -f wav - > test.wav
ffmpeg version N-58613-g26b526e Copyright (c) 2000-2013 the FFmpeg
developers
built on Nov 30 2013 01:24:40 with gcc 4.8 (SUSE Linux)
configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libaacplus --enable-x11grab
libavutil 52. 55.100 / 52. 55.100
libavcodec 55. 44.100 / 55. 44.100
libavformat 55. 21.102 / 55. 21.102
libavdevice 55. 5.102 / 55. 5.102
libavfilter 3. 91.100 / 3. 91.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
Input #0, mp3, from 'inputfile':
Duration: 01:07:01.94, start: 0.000000, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
Output #0, wav, to 'pipe:':
Metadata:
ISFT : Lavf55.21.102
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 44100 Hz,
stereo, s32, 2116 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 -> pcm_s24le)
Press [q] to stop, [?] for help
size= 7749kB time=00:00:30.01 bitrate=2115.1kbits/s
video:0kB audio:7749kB subtitle:0 global headers:0kB muxing overhead
0.001285%
tail -c +8144 test.wav | ffmpeg -i pipe:0 -y test.mp3
ffmpeg version N-58613-g26b526e Copyright (c) 2000-2013 the FFmpeg
developers
built on Nov 30 2013 01:24:40 with gcc 4.8 (SUSE Linux)
configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libaacplus --enable-x11grab
libavutil 52. 55.100 / 52. 55.100
libavcodec 55. 44.100 / 55. 44.100
libavformat 55. 21.102 / 55. 21.102
libavdevice 55. 5.102 / 55. 5.102
libavfilter 3. 91.100 / 3. 91.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
[aac @ 0x185c900] Format aac detected only with low score of 1,
misdetection possible!
[aac @ 0x185d2c0] get_buffer() failed
[aac @ 0x185d2c0] Reserved bit set.
[aac @ 0x185d2c0] invalid band type
[aac @ 0x185d2c0] More than one AAC RDB per ADTS frame is not implemented.
Update your FFmpeg version to the newest one from Git. If the problem
still occurs, it means that your file has a feature which has not been
implemented.
[aac @ 0x185d2c0] Input buffer exhausted before END element found
[aac @ 0x185d2c0] Number of bands (3) exceeds limit (2).
[aac @ 0x185d2c0] Number of bands (46) exceeds limit (37).
[aac @ 0x185d2c0] Reserved bit set.
[aac @ 0x185d2c0] channel element 2.10 is not allocated
[aac @ 0x185d2c0] Reserved bit set.
Last message repeated 1 times
etc. (continued on https://ezcrypt.it/P97n#cOHJwrl0Qve2Sx7KCVFJy0wX )
}}}
How to reproduce with pcm_s32le acodec:
{{{
ffmpeg -i inputfile -t 30 -acodec pcm_s32le -f wav - > test.wav
ffmpeg version N-58613-g26b526e Copyright (c) 2000-2013 the FFmpeg
developers
built on Nov 30 2013 01:24:40 with gcc 4.8 (SUSE Linux)
configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libaacplus --enable-x11grab
libavutil 52. 55.100 / 52. 55.100
libavcodec 55. 44.100 / 55. 44.100
libavformat 55. 21.102 / 55. 21.102
libavdevice 55. 5.102 / 55. 5.102
libavfilter 3. 91.100 / 3. 91.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
Input #0, mp3, from 'inputfile':
Duration: 01:07:01.94, start: 0.000000, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
Output #0, wav, to 'pipe:':
Metadata:
ISFT : Lavf55.21.102
Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 44100 Hz,
stereo, s32, 2822 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 -> pcm_s32le)
Press [q] to stop, [?] for help
size= 10332kB time=00:00:30.01 bitrate=2820.1kbits/s
video:0kB audio:10332kB subtitle:0 global headers:0kB muxing overhead
0.000964%
tail -c +8144 test.wav | ffmpeg -i pipe:0 -y test.mp3
ffmpeg version N-58613-g26b526e Copyright (c) 2000-2013 the FFmpeg
developers
built on Nov 30 2013 01:24:40 with gcc 4.8 (SUSE Linux)
configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libaacplus --enable-x11grab
libavutil 52. 55.100 / 52. 55.100
libavcodec 55. 44.100 / 55. 44.100
libavformat 55. 21.102 / 55. 21.102
libavdevice 55. 5.102 / 55. 5.102
libavfilter 3. 91.100 / 3. 91.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
[aac @ 0x31f7900] Format aac detected only with low score of 1,
misdetection possible!
[aac @ 0x31f82c0] get_buffer() failed
Last message repeated 2344 times
[aac @ 0x31f7900] Stream #0: not enough frames to estimate rate; consider
increasing probesize
[aac @ 0x31f7900] decoding for stream 0 failed
[aac @ 0x31f7900] Could not find codec parameters for stream 0 (Audio:
aac, 0 channels, fltp, 342 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
pipe:0: could not find codec parameters
}}}
--
Ticket URL: <https://trac.ffmpeg.org/ticket/3205>
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