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• Playing With File

  8 septembre 2011, par Multimedia Mike — General

  I played with the ‘file’ utility a long time ago because I wanted to make it recognize a large number of multimedia formats. I had trouble getting my changes to take. But I’m prepared to try again after many years.

  Aiming at the Corpus
  In my local mirror of the MPlayerHQ samples archive, I find 9853 unique files. So I run all of them through the ‘file’ command :

    ’find /path/to/samples -type f -print0 | xargs -0 file —no-pad’
  

  My Ubuntu installation has file v5.04. I also tested against 5.07 and the latest, 5.08. Here is the number of files each version was unable to identify (generically marking as ‘data’) :

  5.04  1521
  5.07  1405
  5.08  1501
  

  That seems like a regression for v5.08 until I dug into the details and saw quite a few items like this, indicating that the MPEG detection could use some work :

  
   mov/mov-demux-infinite-loop.mpg : DOS-executable (
  +mov/mg-‹demux-infinite-loop.mpg : data
  
   image-samples/UNeedQT4.pntg : DOS-executable (
  +imY- samples/UNeedQT4.pntg : data
  

  Workflow
  These are just notes to myself and perhaps anyone else who wants to add new file formats to be identified by the ‘file’ command.

  First, download either the latest release from the FTP or clone from Github. Do the usual unpack, ‘./configure’, ‘make’ routine. To use this newly-built version and its associated magic file :

    ./src/file —magic-file magic/magic.mgc <file>
  

  To add a new format for ID, first, run the foregoing command to ensure that it’s not already identified. Then, check over the files in magic/Magdir and see which one might pertain to what you’re doing (it’s unlikely that your format will merit a new file in this directory). For example, for this round, I modified animation, audio, iff, and riff. Add or modify existing specs based on the copious examples in the directory and by consulting the appropriate man page (‘man 5 magic’).

  Finally, run ‘make’ again which will regenerate the magic file. Invoke the above command again to use the modified magic file.

  Before and After
  On a selection of formats taken from the samples archive (renamed and cut down to a kilobyte because detection typically only relies on the first few bytes), here is the “before” :

  amv :            RIFF (little-endian) data
  armovie :        data
  bbc-dirac :      data
  interplay-mve :  data
  mtv :            data
  nintendo-thp :   data
  nullsoft-video : data
  redcode :        data
  sega-film :      data
  smacker :        data
  trueaudio :      data
  vqa :            IFF data
  wavpack :        data
  wc3-mve :        IFF data
  wtv :            data
  

  And the “after” :

  amv :            RIFF (little-endian) data, AMV
  armovie :        ARMovie
  bbc-dirac :      BBC Dirac Video
  interplay-mve :  Interplay MVE Movie
  mtv :            MTV Multimedia File
  nintendo-thp :   Nintendo THP Multimedia
  nullsoft-video : Nullsoft Video
  redcode :        REDCode Video
  sega-film :      Sega FILM/CPK Multimedia, 320 x 224
  smacker :        RAD Game Tools Smacker Multimedia version 2, 320 x 200, 100 frames
  trueaudio :      True Audio Lossless Audio
  vqa :            IFF data, Westwood Studios VQA Multimedia, 418 video frames, 320 x 200
  wavpack :        WavPack Lossless Audio
  wc3-mve :        IFF data, Wing Commander III Video, PC version
  wtv :            Windows Television DVR Media
  

  After rerunning ‘file’ on the mphq corpus using the modified magic file, only 1329 files remain unidentified (down from 1501).

  Going Forward
  As mentioned, MPEG detection could probably be strengthened. However, a major weakness is QuickTime/MP4. Many files are not detected, probably owing to the many ways that QuickTime files can begin.

• ffmpeg for a android (using tutorial : "ffmpeg and Android.mk")

  7 avril 2016, par Matthias

  I am trying to compile ffmpeg for a android. I have found several posts on this theme but non of these seems to work. If tried to build ffmpeg like it is posted on [1]. Did anybody successfully compile ffmpeg using theses tutorial ?
  I am not sure how to realize step 4 to 5.

  STEP4 : Configuring ...

  STEP5 : cd to your NDK root dir, type make TARGET_ARCH=arm APP=ffmpeg-org

  It seems to me that building an application like it is explained in the tutorial in step 5 need some previous steps. Unfortunately I have no app in the folder to make. I am using the current android ndk release 3 and checked out the actual ffmpeg releases from [3] and [4]. I am thankful for every advice.

  [1] http://slworkthings.wordpress.com/
  [2] http://gitorious.org/ olvaffe/ffmpeg/ffmpeg-android
  [3] http://ffmpeg.org/download.html

• How to configure ffmpeg on ubuntu to convert *.3gp to pcm *.wav ? [migrated]

  31 juillet 2012, par Monica Sol

  I'm using linux Ubuntu ver 10.04.
  I need to convert file *.3gp to PCM *.wav. I'm using for that ffmpeg program.

  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 read some stuff on the Internet and I've made what there was written.
  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 http://ffmpeg.org/download.html (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

  than : time make && make install.

  Till this time everything was ok. After conversion (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 ? I'm linux newbie, but I really need to get this thing works.