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• FFmpeg and types of samples

  23 juillet 2015, par Kyborg2011

  Decoded audio using FFmpeg (function avcodec_decode_audio3) and try to reproduce it through Audiotrack in Android. When playing I hear some growling. No music. On the forums advised that the problem with the samples. Question : how to solve this problem, if it is associated with a mismatch of types of samples ? Code decoding :

          while (av_read_frame(pFormatCtx, &packet)>= 0) {
  
              if (aCodecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                          int data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 2;
  
                          int size=packet.size;
  
   int decoded = 0;
  
                          while(size > 0) {
  
                                  int len = avcodec_decode_audio3(aCodecCtx, (uint16_t *) pAudioBuffer, &data_size, &packet);
  
  
  
  
  
                                  jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
                                  memcpy(bytes + decoded, (uint16_t *) pAudioBuffer, len);
  
                                  (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
  
  
                                  (*env)->CallStaticVoidMethod(env, cls, mid, array);
  
  
  
  
  
                                  size -= len;
  
                                  decoded += len;
  
                                  }
  
              }
  
  
  
       }

  Java code :

      protected void onCreate(Bundle savedInstanceState) {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.main);
  
  
  
           int bufSize = AudioTrack.getMinBufferSize(44100,                                AudioFormat.CHANNEL_CONFIGURATION_MONO, 
  
                          AudioFormat.ENCODING_PCM_16BIT);
  
          track = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_MONO, 
  
                      AudioFormat.ENCODING_PCM_16BIT, bufSize, AudioTrack.MODE_STREAM);
  
          track.play();
  
  
  
              bytes = new byte[bufSize];
  
              int res = main(2, "/sdcard/muzika_iz_reklami_bmw_5_series_-_bmw_5_series.mp3", bytes);
  
  
  
  
  
              System.out.println(res);
  
      }
  
      private static void play(byte[] play) {
  
  
  
          track.write(play, 0, play.length);
  
      }

  How to solve this problem ?
  P.S. The file itself is checked - the standard player plays. Format : mp3.

• VLC empty edit issue

  12 août 2013, par user2675209

  We have recorded a live stream. The time stamps are seconds from epoch. We have used FFMPEG for muxing and provided the same time stamps as in live stream.
  I have two observations :

  1. If initial timestamp is 0. The file plays properly using VLC.
  2. But we require to preserve the timestamps while muliplexing.

  In this case when initial time stamp is non-zero, VLC shows 2 tracks for 1 elst box , It waits for the first track to get over, which is actually an empty edit list with media time=-1.

  Can you suggest any change while multiplexing using FFMPEG to play the file normally.
  PS : ffplay plays the file properly.

  Thanking in anticipation

• Struggling with where to start with creating a x264 .Net Wrapper [closed]

  3 août 2013, par Rob ElCalvo Perry

  I have a compiled libx264-129.dll for Windows and its functions are clearly visible in DLL Viewer. However, I havent got a clue where to start to create a .NET wrapper for it. The ultimate aim is to create a piece of screen recording software with x264 as the codec..

  Can anyone shed some light on where to start (I understand about P/Invoke etc) What I'm looking for is the fundamentals needed to wrap the library and encode bitmaps from .NET..

  I know that x264_param_t plays a part in creating the encoder object but with no knowledge of C, I'm totally stuck really :-s