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• Choppy sound when using ffmpeg

  24 novembre 2011, par Kurt

  I suffered some choppy audio when i try to capture audio from a live stream.
  Another essential problem which could explain the problem is that the Wav file created is twice longer than the capture time.

  The audio is perfect when i play the avs input file with ffplay, so the avs is ok, the problem is after whether in the capture or in the Wav writing.

  To capture :

  av_read_frame(pFormatCtx, &packet)
  
  
  
  if(packet.stream_index == mAudioStream)
  
  {
  
      int buff_size = sizeof(mAudioBuffer);
  
      std::cout << "Buff_size " << buff_size << std::endl;
  
      len = avcodec_decode_audio3(pAudioCodecCtx,(int16_t*)mAudioBuffer, &buff_size,&packet);
  
      if(len < 0){
  
          qDebug("Extractor - Audio isEnd = -1;");
  
          mAudioBufferSize = 0;
  
          isEnd = ERROR_;
  
          return isEnd;
  
      }
  
  
  
      // Set packet result type
  
      mFrameType = AUDIO_PKT;
  
      mAudioBufferSize = buff_size;
  
      //store audio synchronization informations:
  
      if(packet.pts != AV_NOPTS_VALUE) {
  
           mAudioPts_ = av_q2d(pFormatCtx->streams[mAudioStream]->time_base);
  
           mAudioPts_ *= packet.pts;
  
      }
  
  }
  
  
  
          // store a copy of current audio frame in _frame
  
          _frame.audioFrame = new decoded_frame_t::audio_frame_t();
  
          _frame.audioFrame->sampleRate = mediaInfos.audioSampleRate;
  
          _frame.audioFrame->sampleSize = mediaInfos.audioSampleSize;
  
          _frame.audioFrame->nbChannels = mediaInfos.audioNbChannels;
  
          _frame.audioFrame->nbSamples = mAudioBufferSize / ((mediaInfos.audioSampleSize/8) * mediaInfos.audioNbChannels);
  
          _frame.audioFrame->buf.resize(mAudioBufferSize);
  
          memcpy(&_frame.audioFrame->buf[0],mAudioBuffer,mAudioBufferSize);

  Then i store in a Wav File using libsndfile :

  SNDFILE*            fd;
  
  SF_INFO             sfInf;
  
  
  
  sfInf.frames = 0;
  
  sfInf.channels = p_capt->ui_nbChannels;
  
  sfInf.samplerate = p_capt->ui_sampleRate;
  
  sfInf.format = SF_FORMAT_WAV | SF_FORMAT_PCM_U8;
  
  sfInf.sections = 0;
  
  sfInf.seekable = 0;
  
  
  
  if (sf_format_check(&sfInf) == FALSE)
  
      std::cout << "Format parameter are uncorrect ! Exit saving !" << std::endl;
  
  else
  
  {
  
      fd = sf_open(fileName.toStdString().c_str(), SFM_WRITE, &sfInf);
  
      if (fd == NULL)
  
      {
  
          std::cout << "Unable to open the file " << fileName.toStdString() << std::endl;
  
          return GRAB_ST_NOK;
  
      }
  
  
  
      //little trick because v_buf is a uint8_t vector
  
      sf_count_t l = sf_write_short(fd, (const short *)(&(p_capt->v_buf[0])), p_capt->v_buf.size()/2);
  
  
  
      if (l != p_capt->v_buf.size()/2)
  
      {
  
         std::cout << "sf_write didn't write the right amoung of bits " << l << " != " << p_capt->v_buf.size()/2 << std::endl;
  
         ret = GRAB_ST_NOK;
  
      }
  
      else
  
      {
  
          sf_write_sync(fd);
  
          sf_close(fd);
  
          ret = GRAB_ST_OK;
  
      }
  
  }

  I hope it's understandable. Waiting for remarks.

  Kurt

• Why encoding G723 sound clicks ?

  19 septembre 2012, par Stanislav Litvinenko

  I need to encode G723_1
  To do this, I want to use avcodec.
  But when listening to the encoded file only sound clicks.
  To use an example to test api-example.c

  Example encoded file : test.g723

  Why this can happen ? And how it can be solved ?

• flv from vlc to ffmpeg live video error when no sound temporarily

  14 septembre 2012, par dvch

  When we get live stream from vlc to ffmpeg , wherever there is sometimes 5-6 second no sound part of video , then ffmpeg is dead with this log

  flv @ 0x8b426d0]illegal ac vlc code at 4x6

  [flv @ 0x8b426d0]Error at MB : 142

  [flv @ 0x8b426d0]concealing 257 DC, 257 AC, 257 MV errors

  [mpegts @ 0x8b44e50]dts < pcr, TS is invalid

  Is there anyway to avoid this problem ?