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• How can I really pick up the correct format when storing incoming stream ?

  3 février 2016, par Nyaruko

  I want to re-mux a incoming h264 stream. But how could I pick the correct AVOutputFormat for the AVFormatContext ?

  Currently I used :

  AVOutputFormat* fmt = av_guess_format(NULL, "xxx.avi", NULL);
  
  
  
  // Open the context
  
  //---------------------------------------------------------------------
  
  outFormatCtx = ffmpeg::avformat_alloc_context();
  
  
  
  //Set the output format
  
  //----------------------------------------------------------------------
  
  outFormatCtx->oformat = fmt;

  And everything works fine.
  However, if I change the first line to :
  av_guess_format("h264",NULL, NULL) ;
  the recorded stream cannot be played because of bad header/tailer.

  Is there a more smart way of picking the correct AVOutputFormat to be used ?

• Build Live Audio Stream Player

  5 décembre 2011, par Kurt

  For an internship project i've been trying to develop a simple audio player for audio live stream.

  Currently i'm using a homemade three buffering (of 1/3 s each) solution played by QAudioOutput, which recall himself after finished his reading.

  void VideoServer::getBuf(QBuffer * p_buf)
  
  {
  
      audio_chunk*    ac = NULL;
  
      std::vector v;
  
  
  
      v.clear();
  
      for (int i = 0; i < 20;)
  
      {
  
          ac = _audioPreviewSharedData->deQueueAudio();
  
          if (ac)
  
          {
  
              v.insert(v.end(), ac->v_buf.begin(), ac->v_buf.end());
  
              i++;
  
              delete ac;
  
          }
  
          else
  
              usleep(50000);
  
      }
  
      p_buf->close();
  
      p_buf->setData((const char *)(&v[0]), v.size()*2);
  
      p_buf->open(QIODevice::ReadOnly);
  
  }

  -

  void VideoServer::slot_launchAudioPreviewBuffering()
  
  {
  
      getBuf(_buf1);
  
      getBuf(_buf2);
  
      _state = 2;
  
      connect(_audioPreviewTimer, SIGNAL(timeout()), this, SLOT(slot_audioPreviewBuffering()));
  
      _audioPreviewTimer->start(0);
  
      connect(_audioOut, SIGNAL(stateChanged(QAudio::State)), this, SLOT(finishedPlaying(QAudio::State)));
  
  }

  -

  void VideoServer::finishedPlaying(QAudio::State state)
  
  {
  
      if(state == QAudio::IdleState) {
  
          slot_audioPreviewBuffering();
  
      }
  
  }

  -

  void VideoServer::slot_audioPreviewBuffering()
  
  {
  
      switch (_state) {
  
      case 0:
  
          {
  
              _audioOut->start(_buf2);
  
              getBuf(_buf1);
  
              _state = 1;
  
              break;
  
          }
  
      case 1:
  
          {
  
              _audioOut->start(_buf3);
  
              getBuf(_buf2);
  
              _state = 2;
  
              break;
  
          }
  
      case 2:
  
          {
  
              _audioOut->start(_buf1);
  
              getBuf(_buf3);
  
              _state = 0;
  
              break;
  
          }
  
      }
  
  }

  But i'm suffering of choppy sound (little interruption between audio chunk).

  How to play this flux without interruption () and with a reasonable delay between audio and video (less 1s) ? Is there a best way ? Am i doing wrong ?

  Thank you !

• avcodec/wavpack : Fix runtime error : shift exponent 32 is too large for 32-bit type...

  1er juin 2017, par Michael Niedermayer

  avcodec/wavpack : Fix runtime error : shift exponent 32 is too large for 32-bit type 'int'
  Fixes : 1967/clusterfuzz-testcase-minimized-5757031199801344
  Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/wavpack.c