[FFmpeg-trac] #3180(undetermined:new): flv demuxer does not decode the aac audio properly

[FFmpeg]

#3180: flv demuxer does not decode the aac audio properly
-------------------------------------+-------------------------------------
             Reporter:  merbanan     |                     Type:  defect
               Status:  new          |                 Priority:  normal
            Component:               |                  Version:
  undetermined                       |  unspecified
             Keywords:               |               Blocked By:
             Blocking:               |  Reproduced by developer:  0
Analyzed by developer:  0            |
-------------------------------------+-------------------------------------
 ./ffmpeg -i rtmpt_stream2.flv
 ffmpeg version N-58535-g58010e5 Copyright (c) 2000-2013 the FFmpeg
 developers
   built on Nov 28 2013 21:28:49 with gcc 4.8 (Ubuntu/Linaro
 4.8.1-10ubuntu9)
   configuration:
   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
 Input #0, flv, from 'rtmpt_stream2.flv':
   Metadata:
     Encoder         : Omnia A/XE
     StreamTitle     :
     StreamUrl       :
   Duration: 00:00:11.87, start: 0.000000, bitrate: 35 kb/s
     Stream #0:0: Audio: aac, 44100 Hz, mono, fltp, 32 kb/s
 At least one output file must be specified

 FFmpeg identifies the channels by parsing the aac extra data which
 indicates that the file is a mono stream.


 ./ffmpeg -i rtmpt_stream2.flv out.wav
 ffmpeg version N-58535-g58010e5 Copyright (c) 2000-2013 the FFmpeg
 developers
   built on Nov 28 2013 21:28:49 with gcc 4.8 (Ubuntu/Linaro
 4.8.1-10ubuntu9)
   configuration:
   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
 Input #0, flv, from 'rtmpt_stream2.flv':
   Metadata:
     Encoder         : Omnia A/XE
     StreamTitle     :
     StreamUrl       :
   Duration: 00:00:11.87, start: 0.000000, bitrate: 35 kb/s
     Stream #0:0: Audio: aac, 44100 Hz, mono, fltp, 32 kb/s
 File 'out.wav' already exists. Overwrite ? [y/N] y
 Output #0, wav, to 'out.wav':
   Metadata:
     StreamUrl       :
     StreamTitle     :
     ISFT            : Lavf55.21.102
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono,
 s16, 705 kb/s
 Stream mapping:
   Stream #0:0 -> #0:0 (aac -> pcm_s16le)
 Press [q] to stop, [?] for help
 [aac @ 0x1f8b440] Parametric Stereo signaled to be not-present but was
 found in the bitstream.
     Last message repeated 248 times
 size=    1036kB time=00:00:11.91 bitrate= 712.3kbits/s
 video:0kB audio:1036kB subtitle:0 global headers:0kB muxing overhead
 0.007541%

 Decoding it shows that the aac stream contains parametric stereo.

 The onMetaData indicates that this stream actually contains a stereo
 stream.

 File onMetaData block
 DEBUG: Property: <Name:           duration, NUMBER:    0.00>
 DEBUG: Property: <Name:      audiodatarate, NUMBER:    32.00>
 DEBUG: Property: <Name:    audiosamplerate, NUMBER:    44100.00>
 DEBUG: Property: <Name:    audiosamplesize, NUMBER:    16.00>
 DEBUG: Property: <Name:             stereo, BOOLEAN:    TRUE>
 DEBUG: Property: <Name:       audiocodecid, NUMBER:    10.00>
 DEBUG: Property: <Name:            Encoder, STRING:    Omnia A/XE>
 DEBUG: Property: <Name:        StreamTitle, STRING:    >
 DEBUG: Property: <Name:          StreamUrl, STRING:    >


 So the suggestion is to change the demuxer to replace the aac extra data
 with a valid version. The replace condition should be based on codec, data
 rate and stereo flag.

--
Ticket URL: <https://trac.ffmpeg.org/ticket/3180>
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