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I'm using Ubuntu 10.04. I need to convert a .3gp file to PCM WAV. I'm using ffmpeg for that.

When it's installed from repository by using aptitude install ffmpeg it's installing some basic version of it and I cannot convert what I need.

I've installed the latest yasm ver.1.1.0 and the newest x264 - 0.125.2208. After that I got ffmpeg using git from the official homepage with git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg.

I`ve tried to configure ffmpeg by myself using:

./configure --enable-gpl --enable-version3 --enable-postproc 
--enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame 
--enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

Then: time make && make install.

Till this time everything was ok. After conversion with

ffmpeg -i audiotest.3gp -f s16le -ar 8000 -acodec pcm_s16le audio.wav

I wanted to check information about this PCM *.wav file (ffmpeg -i audio.wav) and I`ve got this error:

~# ffmpeg -i audio.wav

ffmpeg version N-42619-g6b7849e Copyright (c) 2000-2012 the FFmpeg developers
  built on Jul 21 2012 00:50:52 with gcc 4.4.3
  configuration: --enable-gpl --enable-version3 --enable-postproc --enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

  libavutil      51. 65.100 / 51. 65.100
  libavcodec     54. 41.100 / 54. 41.100
  libavformat    54. 17.100 / 54. 17.100
  libavdevice    54.  1.100 / 54.  1.100
  libavfilter     3.  2.100 /  3.  2.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[aac @ 0x943d4e0] Format aac detected only with low score of 1, misdetection possible!
[aac @ 0x9443740] channel element 0.0 is not allocated
    Last message repeated 2 times
[aac @ 0x9443740] 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 @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
[aac @ 0x9443740] Number of bands (7) exceeds limit (2).
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] SSR 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 @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Number of bands (31) exceeds limit (1).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
[aac @ 0x9443740] channel element 0.7 is not allocated
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Number of scalefactor bands in group (62) exceeds limit (41).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.15 is not allocated
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Number of scalefactor bands in group (48) exceeds limit (41).
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Number of bands (31) exceeds limit (4).
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] SSR 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 @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.3 is not allocated
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (35) exceeds limit (16).
[aac @ 0x9443740] Number of scalefactor bands in group (63) exceeds limit (41).
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Number of bands (38) exceeds limit (10).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] channel element 0.7 is not allocated
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 2 times
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] decode_band_types: Input buffer exhausted before END element found
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
    Last message repeated 1 times
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Number of bands (4) exceeds limit (1).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Number of bands (31) exceeds limit (8).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Number of bands (31) exceeds limit (2).
[aac @ 0x9443740] Number of bands (28) exceeds limit (1).
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x943d4e0] decoding for stream 0 failed
[aac @ 0x943d4e0] Could not find codec parameters for stream 0 (Audio: aac, 4.0, s16, 383 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[aac @ 0x943d4e0] Estimating duration from bitrate, this may be inaccurate
audio.wav: could not find codec parameters

Can anyone help me with this? What I'm doing wrong?

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migrated from stackoverflow.com Jul 31 '12 at 3:57

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Please show us the full, uncut output of the actual encoding process, not from your output file. –  slhck Aug 1 '12 at 10:13

1 Answer 1

As suggested by the first error message:

Format aac detected only with low score of 1, misdetection possible!

it's detecting the input file type incorrectly. Specify the input file format using the -f option like so:

ffmpeg -f s16le -i input.wav

and it should work better.

However, if you just want to get information about the file, you should use FFprobe instead. It's generally packaged with FFmpeg, takes similar options and provides information in a much easier to parse format. The -show_format and -show_streams options should give you most all the information you need about a file.

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