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• Compressed mp4 video is taking too long time to play (exoplayer)

  10 août 2018, par User

  Video(mp4) is recorded from android camera and sent to backend, here I am using ffmpeg wrapper to compress the video[44mb video to 5.76mb]. compression is working well, But when I send the video for play in android(exo player), is taking too long time to start.

  below is my code to compress :

   FFmpegBuilder builder = new FFmpegBuilder()
  
                     .setInput("D:/dummyVideos/myvideo.mp4")     // Filename, or a FFmpegProbeResult
  
                     .overrideOutputFiles(true) // Override the output if it exists
  
                     .addOutput("D:/dummyVideos/myvideo_ffmpeg.mp4")   // Filename for the destination
  
                     .setFormat("mp4")       // Format is inferred from filename, or can be set
  
                     .disableSubtitle()       // No subtiles
  
                     .setAudioChannels(1)         // Mono audio
  
                     .setAudioCodec("aac")       // using the aac codec
  
                     .setAudioSampleRate(48_000) // at 48KHz
  
                     .setAudioBitRate(32768)     // at 32 kbit/s
  
                     .setVideoCodec("libx264")     // Video using x264
  
                     .setVideoFrameRate(24, 1)     // at 24 frames per second 
  
                     .setVideoResolution(1280, 720) // at 640x480 resolution
  
                     .setVideoBitRate(762800)
  
                     .setStrict(FFmpegBuilder.Strict.EXPERIMENTAL) // Allow FFmpeg to use experimental specs
  
                     .done();

  Can anyone tell me why video is taking too long time to play in exo player ? Is anything wrong in the compression ?

• wrong play audio samples

  8 août 2014, par Ivan Lisovich

  I have a problem with a ffmpeg and NAudio libs.
  I worked with the old ffmpeg library and there the audio plays correctly.
  read video in manage c++

  // Read frames and save to list audio frames
  
  while(av_read_frame(pFormatCtx, &packet) >= 0) 
  
  {
  
      if(packet.stream_index == videoStream) 
  
      {
  
          // reade image
  
      }
  
      else if(packet.stream_index == audioStream) 
  
      {
  
          int b = av_dup_packet(&packet);
  
          if(b >= 0) {
  
              int audio_pkt_size = packet.size;
  
              libffmpeg::uint8_t* audio_pkt_data = packet.data;
  
              while(audio_pkt_size > 0) 
  
              {
  
                  int got_frame = 0;
  
                  int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
  
                  if(len1 < 0)
  
                  {
  
                      /* if error, skip frame */
  
                      audio_pkt_size = 0;
  
                      break;
  
                  }
  
                  audio_pkt_data += len1;
  
                  audio_pkt_size -= len1;
  
                  if (got_frame)
  
                  {
  
                      int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
  
                      array<byte>^ managedBuf = gcnew array<byte>(data_size);
  
                      System::IntPtr iptr = System::IntPtr( frame.data[0] );
  
                      System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
  
                      audioData->Add(managedBuf);
  
                  }
  
              }
  
          }
  
      }
  
      // Free the packet that was allocated by av_read_frame
  
      libffmpeg::av_free_packet(&amp;packet);
  
  }
  
  </byte></byte>

  I return audioData to c# code and play in NAudio library
  play in c#

  var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
  
  var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
  
  var waveOut = new DirectSoundOut();
  
  waveOut.Init(waveProvider);
  
  waveOut.Play();
  
  foreach (byte[] data in audioData)
  
  {
  
      waveProvider.AddSamples(data, 0, data.Length);
  
  }

  but audio not playing.
  what am I doing wrong ?

• wrong play audio sasamples

  7 août 2014, par Ivan Lisovich

  I have a problem with a ffmpeg and NAudio libs.
  I worked with the old ffmpeg library and there the audio plays correctly.
  read video in manage c++

  // Read frames and save to list audio frames
  
  while(av_read_frame(pFormatCtx, &packet) >= 0) 
  
  {
  
      if(packet.stream_index == videoStream) 
  
      {
  
          // reade image
  
      }
  
      else if(packet.stream_index == audioStream) 
  
      {
  
          int b = av_dup_packet(&packet);
  
          if(b >= 0) {
  
              int audio_pkt_size = packet.size;
  
              libffmpeg::uint8_t* audio_pkt_data = packet.data;
  
              while(audio_pkt_size > 0) 
  
              {
  
                  int got_frame = 0;
  
                  int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
  
                  if(len1 < 0)
  
                  {
  
                      /* if error, skip frame */
  
                      audio_pkt_size = 0;
  
                      break;
  
                  }
  
                  audio_pkt_data += len1;
  
                  audio_pkt_size -= len1;
  
                  if (got_frame)
  
                  {
  
                      int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
  
                      array<byte>^ managedBuf = gcnew array<byte>(data_size);
  
                      System::IntPtr iptr = System::IntPtr( frame.data[0] );
  
                      System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
  
                      audioData->Add(managedBuf);
  
                  }
  
              }
  
          }
  
      }
  
      // Free the packet that was allocated by av_read_frame
  
      libffmpeg::av_free_packet(&amp;packet);
  
  }
  
  </byte></byte>

  I return audioData to c# code and play in NAudio library
  play in c#

  var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
  
  var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
  
  var waveOut = new DirectSoundOut();
  
  waveOut.Init(waveProvider);
  
  waveOut.Play();
  
  foreach (byte[] data in audioData)
  
  {
  
      waveProvider.AddSamples(data, 0, data.Length);
  
  }

  but audio not playing.
  what am I doing wrong ?