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• Add Filters to Video like Instagram or Snapchat via FFmpeg

  27 avril 2017, par Karandeep Atwal

  I am using FFmpeg in my android app. I have implemented following filters/effects successfully on video-

  1. Invert Color
  2. Black and white
  3. Sepia
  4. Vignette
  5. Gamma effect

  I followed FFmpeg Video Filter docs.

  There are similar questions asked but without any proper answer.

  I want to apply some Good Filters like on Instagram or Snapchat via FFmpeg . Below are few example filters taken from this link.

  Can these filters be achieved via ffmpeg ?

  

  

• Invalid pixel aspect ratio in ffmpeg

  26 septembre 2014, par dudeofea

  I want to scale a video to a certain size using ffmpeg, but I get the following error :

  ffmpeg -i Ad_939.avi -vf scale=1000:64 test_ad.avi       
  
  ffmpeg version 2.3.3 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Sep 26 2014 10:31:06 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: 
  
    libavutil      52. 92.100 / 52. 92.100
  
    libavcodec     55. 69.100 / 55. 69.100
  
    libavformat    55. 48.100 / 55. 48.100
  
    libavdevice    55. 13.102 / 55. 13.102
  
    libavfilter     4. 11.100 /  4. 11.100
  
    libswscale      2.  6.100 /  2.  6.100
  
    libswresample   0. 19.100 /  0. 19.100
  
  Input #0, avi, from 'Ad_939.avi':
  
    Metadata:
  
      encoder         : Lavf56.4.101
  
    Duration: 00:00:20.50, start: 0.000000, bitrate: 75 kb/s
  
      Stream #0:0: Video: wmv2 (WMV2 / 0x32564D57), yuv420p, 160x64, 68 kb/s, SAR 32:45 DAR 16:9, 24 fps, 24 tbr, 24 tbn, 24 tbc
  
  [mpeg4 @ 0x2305cc0] Invalid pixel aspect ratio 128/1125, limit is 255/255 reducing
  
  [mpeg4 @ 0x2305cc0] too many threads/slices (5), reducing to 4
  
  Output #0, avi, to 'test_ad.avi':
  
    Metadata:
  
      ISFT            : Lavf55.48.100
  
      Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1000x64 [SAR 19:167 DAR 2375:1336], q=2-31, 200 kb/s, SAR 128:1125 DAR 16:9, 24 fps, 24 tbn, 24 tbc
  
      Metadata:
  
        encoder         : Lavc55.69.100 mpeg4
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (wmv2 (native) -> mpeg4 (native))
  
  Press [q] to stop, [?] for help
  
  frame=  228 fps=0.0 q=2.0 size=     222kB time=00:00:09.50 bitrate= 191.6kbits/s
  
  frame=  468 fps=467 q=2.0 size=     435kB time=00:00:19.50 bitrate= 182.8kbits/s
  
  frame=  492 fps=456 q=2.0 Lsize=     458kB time=00:00:20.50 bitrate= 183.1kbits/s    
  
  video:441kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.981523%

  I’ve read around and it seems this was a past bug but that was 3 years ago. I’ve tried running this command with version 2.4.1, 2.3.3, 2.4.git (compiled from source Sept 16, 2014), and N-40951-g1ea7a3e (Newest static build) and they all fail.

  EDIT : When I say fail I mean that the video is not playing at the proper resolution. The file’s properties say that it is 1000x64 but VLC plays (as well as my website using a <video></video> tag) it with the original aspect ratio with the new width which is not what I want. I want a crunched up video the same size no matter what.

  Is there something I’m missing ?

• Error in converting audio file format from ogg to wav [on hold]

  9 juin 2014, par Sumit Bisht

  I am trying to convert an ogg format file that was created using webrtc (html5 usermedia content generated on firefox) and transferred and decoded on the server into a wav file through ffmpeg but am getting this error on cmmand line while trying to convert :

  $ ffmpeg -i 2014-6-5_16-17-54.ogg res1.wav
  
  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on May  1 2014 13:12:12 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
  
    configuration: --enable-gpl --enable-version3 --enable-shared --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, ogg, from '2014-6-5_16-17-54.ogg':
  
    Duration: 00:00:01.84, start: 0.000000, bitrate: 18 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono
  
      Metadata:
  
        ENCODER         : Mozilla29.0.1
  
  [graph 0 input from stream 0:0 @ 0x18dca20] Invalid sample format (null)
  
  Error opening filters!

  Although, I am able to play the file on server and using the same command, am able to convert .ogg files generated somewhere else. What might be I missing ?

  Edit :
  Here’s the source code that is used to write to the file :

  1) During startup - use the methods of getUserMedia API.

  navigator.getUserMedia({
  
          audio: true,
  
          video: false
  
      }, function(stream) {
  
          audioStream = RecordRTC(stream, {
  
              bufferSize: 16384
  
          });
  
          audioStream.startRecording();

  2) During stopping of the recording - extracting the recorded information.

  function(audioDataURL) {
  
       var audioFile = {};
  
       audioFile = {
  
         contents: audioDataURL
  
       **strong text**};

  On server end, the following code is creating a file from this data :

  dataURL = dataURL.split(',').pop(); // dataURL is the audioDataURL as defined above
  
  fileBuffer = new Buffer(dataURL, 'base64');
  
  fs.writeFileSync(filePath, fileBuffer);