aac distrorted after transcoding (regression)

[mpan]

When transcoding AAC to another format, output is distorted. I have noticed this situation only for one file (sample provided below), no other AACs I have tested seem to be affected. Also it seems that bug doesn't depend on output format.

Everything was fine in 0.7.6. The problems started after update (from ArchLinux repo) to version N-34586-g33feba3. I have also tested this with the latest version from git and the problem persists.

Files are available here: ​http://mpan.pl/pub/ffmpeg-report/
Original sample:

• test.aac

Sample after conversion to pcm:

• test-good.wav - version produced by 0.7.6
• test-bad.wav - version produced by N-34586-g33feba3
• test-latest.wav - version produced by N-34918-g4f7ad4c (latest)

Corresponding logs:

• test-good.log, test-bad.log, test-latest.log

Smaller (1MB) versions of sample (just in case someone doesn't want to download the 8M ones):

• short-*

Platform:
Linux 3.1.1-1-ARCH x86_64 Pentium(R) Dual-Core CPU E5300 @ 2.60GHz GenuineIntel

[mpan]

Forget the command:
ffmpeg -i test.aac test-blahblah.wav

[Carl Eugen Hoyos]

Regression since 981503905399a25ba179f587efda4a18ac9f4675.

[Carl Eugen Hoyos]

$ ffmpeg -loglevel 99 -i test.aac -t 50 out.wav
ffmpeg version N-34938-g7cdfce4, Copyright (c) 2000-2011 the FFmpeg developers
  built on Nov 17 2011 14:13:00 with gcc 4.5.3
  configuration: --cc=/usr/local/gcc-4.5.3/bin/gcc
  libavutil    51. 26. 0 / 51. 26. 0
  libavcodec   53. 34. 0 / 53. 34. 0
  libavformat  53. 20. 0 / 53. 20. 0
  libavdevice  53.  4. 0 / 53.  4. 0
  libavfilter   2. 48. 1 /  2. 48. 1
  libswscale    2.  1. 0 /  2.  1. 0
[aac @ 0x138f760] Format aac probed with size=131072 and score=50
[aac @ 0x1395b00] err{or,}_recognition separate: 1; 1
[aac @ 0x1395b00] err{or,}_recognition combined: 1; 1
[aac @ 0x1395b00] Unsupported bit depth: 0
[aac @ 0x1395b00] Input buffer exhausted before END element found
[aac @ 0x138f760] max_analyze_duration 5000000 reached at 5015510
[aac @ 0x138f760] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'test.aac':
  Duration: 00:00:51.47, bitrate: 43 kb/s
    Stream #0:0, 110, 1/28224000: Audio: aac, 44100 Hz, 2 channels (FC), s16, 43 kb/s
[pcm_s16le @ 0x13a5a00] err{or,}_recognition separate: 1; 1
[pcm_s16le @ 0x13a5a00] err{or,}_recognition combined: 1; 1
[aac @ 0x1395b00] err{or,}_recognition separate: 1; 1
[aac @ 0x1395b00] err{or,}_recognition combined: 1; 1
[aac @ 0x1395b00] Unsupported bit depth: 0
Output #0, wav, to 'out.wav':
  Metadata:
    encoder         : Lavf53.20.0
    Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x1395b00] Input buffer exhausted before END element found
Error while decoding stream #0:0
size=    8608kB time=00:00:49.96 bitrate=1411.2kbits/s
video:0kB audio:8608kB global headers:0kB muxing overhead 0.000522%