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• How to change metadata with ffmpeg/avconv ?

  14 juillet 2012, par tampis

  I am writing a python script for producing audio and video podcasts. There are a bunch of recorded media files (audio and video) and text files containing the meta information.

  Now I want to program a function which shall add the information from the meta data text files to all media files (the original and the converted ones). Because I have to handle many different file formats (wav, flac, mp3, mp4, ogg, ogv...) it would be great to have a tool which add meta data to arbitrary formats.

  My Question :

  How can I change the metadata of a file with ffmpeg/avconv without changing the audio or video of it and without creating a new file ? Is there another commandline/python tool which would do the job for me ?

  What I tried so far :

  I thought ffmpeg/avconv could be such a tool, because it can handle nearly all media formats. I hoped, that if I set -i input_file and the output_file to the same file, ffmpeg/avconv will be smart enough to leave the file unchanged. Then I could set -metadata key=value and just the metadata will be changed.

  But I noticed, that if I type avconv -i test.mp3 -metadata title='Test title' test.mp3 the audio test.mp3 will be reconverted in another bitrate.

  So I thought to use -c copy to copy all video and audio information. Unfortunately also this does not work :

  :~$ du -h test.wav # test.wav is 303 MB big
  
  303M    test.wav
  
  
  
  :~$ avconv -i test.wav -c copy -metadata title='Test title' test.wav
  
  avconv version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the
  
  Libav    developers
  
  built on Jun 12 2012 16:37:58 with gcc 4.6.3
  
  [wav @ 0x846b260] max_analyze_duration reached
  
  Input #0, wav, from 'test.wav':
  
  Duration: 00:29:58.74, bitrate: 1411 kb/s
  
      Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  File 'test.wav' already exists. Overwrite ? [y/N] y
  
  Output #0, wav, to 'test.wav':
  
  Metadata:
  
      title           : Test title
  
      encoder         : Lavf53.21.0
  
      Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, 1411 kb/s
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (copy)
  
  Press ctrl-c to stop encoding
  
  size=     896kB time=5.20 bitrate=1411.3kbits/s    
  
  video:0kB audio:896kB global headers:0kB muxing overhead 0.005014%
  
  
  
  :~$ du -h test.wav # file size of test.wav changed dramatically
  
  900K    test.wav

  You see, that I cannot use -c copy if input_file and output_file are the same. Of course I could produce a temporarily file :

  :-$ avconv -i test.wav -c copy -metadata title='Test title' test_temp.mp3
  
  :-$ mv test_tmp.mp3 test.mp3

  But this solution would create (temporarily) a new file on the filesystem and is therefore not preferable.

• Revision 67750 : define(’_CACHE_CONTEXTES_AJAX’,true) ; est trop expérimental pour ...

  15 novembre 2012, par kent1@… — Log

  define(’_CACHE_CONTEXTES_AJAX’,true) ; est trop expérimental pour l’utiliser sur tous les sites installant le plugin, autre solution à trouver

• Lossless codec for bayer data

  21 août 2016, par vhdirk

  I’m working with lots of camera’s which capture in BG bayer pattern natively.

  Now, every time I record some data, I save it to the disk in the raw bayer pattern, in an avi container. The problem is, that this really adds up after a while. After one year of research, I have close to 4TB of data...

  So I’m looking for a lossless codec to compress this data. I know I could use libx264 (with —qp 0), or huffYUV, dirac or jpeg2000, but they all assume you have RGB or YUV data. It’s easy enough to convert the bayered data to RGB, and then compress it, but it kind of defeats the purpose of compression if you first triple the data. This would also mean that the demoasicing artefacts introduced by debayering would also be in my source data, which is also not too great. It would be nice to have a codec that can work on the bayered data directly.

  Even more nice would be that the solution would involve a codec that is already supported by gstreamer (or ffmpeg), since that’s what I am already using.