Opened 11 years ago
Closed 11 years ago
#3205 closed defect (fixed)
ffmpeg detects pcm audio as aac with score 1
Reported by: | Katie Holly | Owned by: | |
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Priority: | minor | Component: | avformat |
Version: | git-master | Keywords: | aac |
Cc: | Blocked By: | ||
Blocking: | Reproduced by developer: | no | |
Analyzed by developer: | no |
Description
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
Attachments (2)
Change History (7)
comment:1 by , 11 years ago
comment:2 by , 11 years ago
Component: | undetermined → avformat |
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Keywords: | aac added |
Version: | unspecified → git-master |
I am not convinced that there is a bug but please provide a sample.
comment:3 by , 11 years ago
Sorry for the double attachment, i thought the sample file isn't going to be saved because of it's size of 7.6 MiB.
by , 11 years ago
by , 11 years ago
Attachment: | patchaacdec.diff added |
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comment:4 by , 11 years ago
Status: | new → open |
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Summary: | ffmpeg detects input as aac when processing corrupted pcm_s24le and pcm_s32le input → ffmpeg detects pcm audio as aac with score 1 |
I deleted the attachments that are imo not ideal to reproduce the issue (although it made sense that you explained how they were created) and attached both a sample and a possible patch.
$ ffmpeg -i test ffmpeg version N-58868-ge2f800f Copyright (c) 2000-2013 the FFmpeg developers built on Dec 7 2013 21:45:51 with gcc 4.7 (SUSE Linux) configuration: --enable-gpl libavutil 52. 56.100 / 52. 56.100 libavcodec 55. 45.100 / 55. 45.100 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [aac @ 0x2d18100] Format aac detected only with low score of 1, misdetection possible! ...
comment:5 by , 11 years ago
Resolution: | → fixed |
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Status: | open → closed |
Fixed by Martin Storsjö since 6dd007ad - thank you for the report!
Ah, and expected output of both commands:
pipe:0: Invalid data found when processing input