Is ffmpeg incorrectly interpreting .aif files?

Posted by marue on Server Fault See other posts from Server Fault or by marue
Published on 2012-10-27T18:55:13Z Indexed on 2012/10/27 23:04 UTC
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Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost.

For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as

Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s

my Ubuntu server says it is:

Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s

which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message

[pcm_s24be @ 0xcd4b580]invalid PCM packet
Error while decoding stream #0.0

which certainly is not true. The file is perfectly valid.

Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue?


ffprobe output:

FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini
  libavutil     49.15. 0 / 49.15. 0
  libavcodec    52.20. 0 / 52.20. 1
  libavformat   52.31. 0 / 52.31. 0
  built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease)
Input #0, aiff, from 'testfile.aif':
  Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s
    Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s

update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything.

Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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