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• Revision cdb322dd72 : Adapt ARNR filter length and strength. Adjust the filter length and strength fo

  18 mars 2013, par Paul Wilkins

  Changed Paths : Modify /vp9/encoder/vp9_firstpass.c Modify /vp9/encoder/vp9_onyx_if.c Modify /vp9/encoder/vp9_onyx_int.h Modify /vp9/encoder/vp9_rdopt.c Modify /vp9/encoder/vp9_temporal_filter.c Adapt ARNR filter length and strength. Adjust the filter length and strength for each ARF group (...)

• FFMPEG for iPhone recorded video encoding

  16 décembre 2011, par Sarah

  Hi found lots and lots of links for video encoding through ffmpeg 1,2,3,4 etc but they all start with using terminal commands but when i try to implement any on terminal like :

  git clone git ://github.com/lajos/iFrameExtractor.gitit says that-bash : git : command not found.

  Also as per my knowledge it is not possible to use terminal command on iPhone. Can anybody point out how to encode a video recorded through ffmpeg in mp4 format and also to reduce the size of the video ?Thanks in advance.

  EDIT :
  I am already implementing this method to resize my video and it successfully takes place and I am able to send the video on server but then on server side it's giving problem in retrieving the data and to use it.

  - (void)imagePickerController:(UIImagePickerController *)picker 
  
  didFinishPickingMediaWithInfo:(NSDictionary *)info
  
  {
  
      [self convertVideoToLowQuailtyWithInputURL:videoURL1 outputURL:[NSURL fileURLWithPath:videoStoragePath] handler:^(AVAssetExportSession *exportSession)
  
       {
  
           if (exportSession.status == AVAssetExportSessionStatusCompleted)
  
           {
  
               NSLog(@"%@",exportSession.error);
  
               printf("completed\n");
  
           }
  
           else
  
           {
  
               NSLog(@"%@",exportSession.error);
  
               printf("error\n");
  
           }
  
       }];
  
  }
  
  
  
  - (void)convertVideoToLowQuailtyWithInputURL:(NSURL*)inputURL 
  
                                     outputURL:(NSURL*)outputURL 
  
                                       handler:(void (^)(AVAssetExportSession*))handler
  
  {
  
      [[NSFileManager defaultManager] removeItemAtURL:outputURL error:nil];
  
      AVURLAsset *asset = [AVURLAsset URLAssetWithURL:inputURL options:nil];
  
      AVAssetExportSession *exportSession = [[AVAssetExportSession alloc] initWithAsset:asset presetName:AVAssetExportPresetLowQuality];
  
      exportSession.outputURL = outputURL;
  
      exportSession.outputFileType = AVFileTypeQuickTimeMovie;
  
      [exportSession exportAsynchronouslyWithCompletionHandler:^(void) 
  
       {
  
           handler(exportSession);
  
           [exportSession release];
  
       }];
  
  }

• 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.