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• Xuggler Encoding video of Desktop With Audio - audio has gaps

  2 novembre 2012, par Chris

  I am using Xuggler to convert images captured from the java Robot class and sound read from TargetDataLine class and encoding this into a video. I am then attempting to http stream this video data (after writing my header) to a flash client via http (Socket OutputStream) but it plays and stutters (never just playing smoothly) no matter what buffer value I use on the client side.

  I am asking for help and showing my java code because I suspect it might be to do with how I am encoding the video or something about sending data via http socket which i am not getting..

  ByteArrayURLHandler ba = new ByteArrayURLHandler();
  
  final IRational FRAME_RATE = IRational.make(30); 
  
  final int SECONDS_TO_RUN_FOR = 20; 
  
  final Robot robot = new Robot(); 
  
  final Toolkit toolkit = Toolkit.getDefaultToolkit(); 
  
  final Rectangle screenBounds = new Rectangle(toolkit.getScreenSize()); 
  
  IMediaWriter writer;
  
  
  
  writer = ToolFactory.makeWriter(
  
      XugglerIO.map( 
  
          XugglerIO.generateUniqueName(out, ".flv"), 
  
          out 
  
      )); 
  
  
  
  writer.addListener(new MediaListenerAdapter() { 
  
      public void onAddStream(IAddStreamEvent event) { 
  
          event.getSource().getContainer().setInputBufferLength(1000);
  
          IStreamCoder coder = event.getSource().getContainer().getStream(event.getStreamIndex()).getStreamCoder();
  
          if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_AUDIO) { 
  
              coder.setFlag(IStreamCoder.Flags.FLAG_QSCALE, false);   
  
              coder.setBitRate(32000); 
  
              System.out.println("onaddstream"+ coder.getPropertyNames().toString());
  
          } 
  
          if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_VIDEO) { 
  
              // coder.setBitRate(64000); 
  
              // coder.setBitRateTolerance(64000); 
  
          } 
  
      } 
  
  });
  
  
  
  writer.addVideoStream(videoStreamIndex, videoStreamId, 1024, 768);
  
  final int channelCount = 1;       
  
  
  
  int audionumber =   writer.addAudioStream(audioStreamIndex, audioStreamId,1, 44100);
  
  int bufferSize = (int)audioFormat.getSampleRate()   *audioFormat.getFrameSize();//*6;///6;
  
  byte[] audioBuf;// = new byte[bufferSize]; 
  
  
  
  int i = 0;
  
  
  
  final int audioStreamIndex = 1;
  
  final int audioStreamId = 1;
  
  BufferedImage screen, bgrScreen;
  
  long startTime = System.nanoTime();
  
  while(keepGoing)
  
  { 
  
  
  
      audioBuf = new byte[bufferSize]; 
  
      i++;
  
  
  
      screen = robot.createScreenCapture(screenBounds); 
  
  
  
      bgrScreen = convertToType(screen, BufferedImage.TYPE_3BYTE_BGR); 
  
      long nanoTs = System.nanoTime()-startTime; 
  
      writer.encodeVideo(0, bgrScreen, nanoTs, TimeUnit.NANOSECONDS);
  
      audioBuf = new byte[line.available()];
  
      int nBytesRead = line.read(audioBuf, 0, audioBuf.length); 
  
  
  
      IBuffer iBuf = IBuffer.make(null, audioBuf, 0, nBytesRead);
  
  
  
      IAudioSamples smp = IAudioSamples.make(iBuf,1,IAudioSamples.Format.FMT_S16);
  
      if (smp == null) {
  
          return;
  
      }
  
  
  
      long numSample = audioBuf.length / smp.getSampleSize();
  
  
  
      smp.setComplete(true, numSample,(int)
  
      audioFormat.getSampleRate(), audioFormat.getChannels(),
  
      IAudioSamples.Format.FMT_S16, nanoTs/1000);
  
  
  
      writer.encodeAudio(1, smp);
  
  
  
      writer.flush();
  
  }

• x264 rate control

  31 août 2012, par Craig

  We are using the x264 encoder in a video conferencing project, we have the basic streaming video working, however, we are having trouble understanding how the various rate control settings determine the final bitrate.
  We set the following params :

  x264_param_t params;
  
  x264_param_default_preset(&params, "ultrafast", "zerolatency");
  
  params.i_threads = 1;
  
  params.i_width = width;
  
  params.i_height = height;
  
  params.i_fps_num = fps;
  
  params.i_keyint_max = fps;
  
  params.b_intra_refresh = 1;
  
  params.b_repeat_headers = 1;
  
  params.b_annexb = 1;
  
  
  
  //Set rate control stuff here
  
  
  
  x264_param_apply_profile(&params, "baseline");

  If we only set the params.rc.i_bitrate param, the encoder seems to massively overshoot the bitrate. If we set the i_vbv_max_bitrate & i_vbv_buffer_size params we see a bitrate which peaks (and sometimes overshoots) the i_vbv_max_bitrate setting. Obviously having tight control over the biterate is important for video conferencing, but the documentation is kind of opaque. Is anybody else using x264 for video conferencing ? How are you setting the encoder ? Any help appreciated, thanks in advance.

• oggenc : Set the right AVOption size for the pref_duration option

  6 juin 2014, par Martin Storsjö

  oggenc : Set the right AVOption size for the pref_duration option
  On big endian machines, the default value set via the faulty
  
  AVOption ended up as 2^32 times too big.
  This fixes the fate-lavf-ogg test which currently is broken on
  
  big endian machines, broken since 3831362. Since that commit,
  
  a final zero-sized packet is written to the ogg muxer in that test,
  
  which caused different flushing behaviour on little and big endian
  
  depending on whether the pref_duration option was handled as it
  
  should or not.
  CC : libav-stable@libav.org
  
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavformat/oggenc.c