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• Using FFmpeg in .net ?

  17 juin, par daniel

  So I know its a fairly big challenge but I want to write a basic movie player/converter in c# using the FFmpeg library. However, the first obstacle I need to overcome is wrapping the FFmpeg library in c#. I've downloaded ffmpeg but couldn't compile it on Windows, so I downloaded a precompiled version for me. Ok awesome. Then I started looking for C# wrappers.

  



  I have looked around and have found a few wrappers such as SharpFFmpeg (http://sourceforge.net/projects/sharpffmpeg/) and ffmpeg-sharp (http://code.google.com/p/ffmpeg-sharp/). First of all, I wanted to use ffmpeg-sharp as its LGPL and SharpFFmpeg is GPL. However, it had quite a few compile errors. Turns out it was written for the mono compiler, I tried compiling it with mono but couldn't figure out how. I then started to manually fix the compiler errors myself, but came across a few scary ones and thought I'd better leave those alone. So I gave up on ffmpeg-sharp.

  



  Then I looked at SharpFFmpeg and it looks like what I want, all the functions P/Invoked for me. However its GPL ? Both the AVCodec.cs and AVFormat.cs files look like ports of avcodec.c and avformat.c which I reckon I could port myself ? Then not have to worry about licencing.

  



  But I want to get this right before I go ahead and start coding. Should I :

  



  

  1. Write my own C++ library for interacting with ffmpeg, then have my C# program talk to the C++ library in order to play/convert videos etc.

  


  



  OR

  



  

  2. Port avcodec.h and avformat.h (is that all i need ?) to c# by using a whole lot of DllImports and write it entirely in C# ?

  


  



  First of all consider that I'm not great at C++ as I rarely use it but I know enough to get around. The reason I'm thinking #1 might be the better option is that most FFmpeg tutorials are in C++ and I'd also have more control over memory management than if I was to do it in c#.

  



  What do you think ?
Also would you happen to have any useful links (perhaps a tutorial) for using FFmpeg ?

  


• How to change metadata with ffmpeg/avconv without creating a new file ?

  16 mars, par Stephan Kulla

  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.

  


• how to reencode with ffmpeg (with limited x264)

  6 septembre 2017, par Sarfraz

  Until now I used this script to reencode my rips for my box (tv decoder) :

  ^_^ ( ~ ) -> cat ~/++/src/convert.sh 
  
  #! /bin/bash
  
  
  
  name=$(path -r "$1") # it gives the file name without the extension
  
  
  
  [ "$1" = *.mp4 ] && ffmpeg -i "$name".mp4 -vcodec copy -acodec copy "$name".mkv
  
  x264 --preset veryfast --tune animation --crf 18 --vf resize:720,576,16:15 -o "$name".tmp.mkv "$name".mkv
  
  mkvmerge -o "$name [freeplayer sd]".mkv "$name".tmp.mkv --no-video "$1"
  
  rm -rf "$name".tmp.mkv
  
  [ "$1" = *.mp4 ] && rm -rf "$name".mkv
  
  exit 0
  
  
  
  #EOF

  It works on my ubuntu and archlinux laptops. But it doesn’t on my desktop witch runs fedora.
  Google says that the x264 package shiped by rpmfusion doesn,t support lavf and ffms2.
  And I cannot unistall it because smplayer (witch i like) needs it.

  Ok, so I have to compile it. Google then says "you have to build ffmpeg, ffms2 tnen x264 ensuring that the flags are correctly refered." Well, didn’t work (ffms2 cannot find LIBAV - even when I am telling where - and x264 does’t configure with lavf...)

  My question is : can I use ffmpeg alone to do what my script does.
  I have ffmpeg version 0.8.11, x264 0.116.2048 59cb2eb and gcc : 4.6.1 20110804 (Red Hat 4.6.1-7)

  EDIT : Ok, I found that : ffmpeg -i input file -acodec copy -vcodec libx264 -preset veryfast -tune animation [that part I don’t have] output