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• 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 ?

• CoreAudio : how to retrieve actual sampling rate of raw data ?

  13 août 2014, par jyavenard

  When attempting to play AAC-HE content in an mp4 container, the reported sampling rate found in the mp4 container appears to be half of the actual sampling rate.

  E.g it appears as 24kHz instead of 48kHz.

  Using the FFmpeg AAC decoder, retrieving the actual sampling rate can be done by simply decoding an audio packet using

  avcodec_decode_audio4

  And looking at AVCodecContext::sample_rate which will be updated appropriately. From that it’s easy to adapt the output.

  With CoreAudio decoder, I would use a AudioConverterRef set the input and output AudioStreamBasicDescription
  and call AudioConverterFillComplexBuffer

  As the converter performs all the required internal conversion including resampling it’s fine. But it plays the content after resampling it to 24kHz (as that’s what the input AudioStreamBasicDescription contains.

  Would there be a way to retrieve the actual sampling rate as found be the decoder (rather than the demuxer) in a similar fashion as one can with FFmpeg ?

  Would prefer to avoid losing audio quality if at all possible, and not downmix data

  Thanks

• FFmpeg AVI to MP4 transcode not playing fully in Quicktime

  11 août 2014, par Joel Kennedy

  I’m trying to transcode an AVI video to an MP4 stream on the fly using FFmpeg which can be watched on a webpage with a HTML5 video tag. I already know that MP4 isn’t designed for streaming, however it is possible if segmented so I’m still trying to use MP4 as it’s the method that I’ve found is best supported by multiple browsers.

  My stream plays fine in many browsers except for Safari on OS X, which is using Quicktime to play the HTML5 video.

  

  You can see in the image above that Quicktime only plays the video for a second, and then stops. I’ve tested the same video source in Google Chrome on OS X, and it continues playing fine even after a second has elapsed.

  I believe this is something to do with the set Duration in the tkhd and mdhd atoms ? User vbence described that "FFMpeg’s muxer will set the Duration in both tkhd and mdhd atoms to 0xffffffff for each track. This causes problems in some players (for example Quicktime will not play such files). You should find a tool and change it to zero (0x00000000)" in an answer here.

  Is there a way of getting FFmpeg to set a longer duration on the video to get Quicktime to continue playing it instead of stopping ? Because I’m transcoding on the fly, I wouldn’t want to use a separate tool apart from FFmpeg as in my situation I can’t transcode and save the video file, I have to transcode and pipe it to the client on request.

  The command that I’m using with FFmpeg to transcode the video is the following :

  ffmpeg -re -i
  http://www.example.com/video/stream?path=videopathhere -g 52
  -reset_timestamps 1 -vsync 1 -flags global_header -vcodec libx264 -strict -2 -f mp4 -b:v 512k -s 1280x720 -movflags frag_keyframe+empty_moov+faststart pipe :

  Any help or pointers in the right direction would be greatly appreciated.