Recherche avancée

Sur d’autres sites (6311)

• Decoding and playing audio with ffmpeg and XAudio2 - frequency raito wrong

  12 juillet 2016, par Brent de Carteret

  I’m using ffmpeg to decode audio and output it using the XAudio2 API, it works and plays synced with the video output using the pts. But it’s high pitched (i.e. sounds like chipmunks).

  Setting breakpoints I can see it has sets the correct sample rate from the audio codec in CreateSourceVoice. I’m stumped.

  Any help would be much appreciated.

  #include "DVDAudioDevice.h"
  
  
  
  HANDLE m_hBufferEndEvent;
  
  
  
  CDVDAudio::CDVDAudio()
  
  {
  
      m_pXAudio2 = NULL;
  
      m_pMasteringVoice = NULL;
  
      m_pSourceVoice = NULL;
  
      m_pWfx  = NULL;
  
  
  
      m_VoiceCallback = NULL;
  
  
  
      m_hBufferEndEvent = CreateEvent(NULL, false, false, "Buffer end event");
  
  }
  
  
  
  CDVDAudio::~CDVDAudio()
  
  {
  
      m_pXAudio2 = NULL;
  
      m_pMasteringVoice = NULL;
  
      m_pSourceVoice = NULL;
  
      m_pWfx  = NULL;
  
  
  
      m_VoiceCallback = NULL;
  
  
  
      CloseHandle(m_hBufferEndEvent);
  
      m_hBufferEndEvent = NULL;
  
  }
  
  
  
  bool CDVDAudio::Create(int iChannels, int iBitrate, int iBitsPerSample, bool bPasstrough)
  
  {
  
      CoInitializeEx(NULL, COINIT_MULTITHREADED);
  
  
  
      HRESULT hr = XAudio2Create( &m_pXAudio2, 0, XAUDIO2_DEFAULT_PROCESSOR);
  
  
  
      if (SUCCEEDED(hr))
  
      {
  
          m_pXAudio2->CreateMasteringVoice( &m_pMasteringVoice );
  
      }
  
  
  
      // Create source voice
  
      WAVEFORMATEXTENSIBLE wfx;
  
      memset(&wfx, 0, sizeof(WAVEFORMATEXTENSIBLE));
  
  
  
      wfx.Format.wFormatTag           = WAVE_FORMAT_PCM;
  
      wfx.Format.nSamplesPerSec       = iBitrate;//pFFMpegData->pAudioCodecCtx->sample_rate;//48000 by default
  
      wfx.Format.nChannels            = iChannels;//pFFMpegData->pAudioCodecCtx->channels;
  
      wfx.Format.wBitsPerSample       = 16;
  
      wfx.Format.nBlockAlign          = wfx.Format.nChannels*16/8;
  
      wfx.Format.nAvgBytesPerSec      = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign;
  
      wfx.Format.cbSize               = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
  
      wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample;
  
  
  
      if(wfx.Format.nChannels == 1)
  
      {
  
          wfx.dwChannelMask = SPEAKER_MONO;
  
      }
  
      else if(wfx.Format.nChannels == 2)
  
      {
  
          wfx.dwChannelMask = SPEAKER_STEREO;
  
      }
  
      else if(wfx.Format.nChannels == 5)
  
      {
  
          wfx.dwChannelMask = SPEAKER_5POINT1;
  
      }
  
  
  
      wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
  
  
  
      unsigned int flags = 0;//XAUDIO2_VOICE_NOSRC;// | XAUDIO2_VOICE_NOPITCH;
  
  
  
      //Source voice
  
  
  
      m_VoiceCallback = new StreamingVoiceCallback(this);
  
  
  
      hr = m_pXAudio2->CreateSourceVoice(&m_pSourceVoice,(WAVEFORMATEX*)&wfx, 0 , 1.0f, m_VoiceCallback);
  
  
  
      if(!SUCCEEDED(hr))
  
          return false;
  
  
  
      // Start sound
  
      hr = m_pSourceVoice->Start(0);
  
  
  
      if(!SUCCEEDED(hr))
  
          return false;
  
  
  
      return true;
  
  }
  
  
  
  DWORD CDVDAudio::AddPackets(unsigned char* data, DWORD len)
  
  {  
  
          memset(&m_SoundBuffer,0,sizeof(XAUDIO2_BUFFER));
  
  
  
          m_SoundBuffer.AudioBytes = len;
  
          m_SoundBuffer.pAudioData = data;
  
          m_SoundBuffer.pContext = NULL;//(VOID*)data;
  
  
  
          XAUDIO2_VOICE_STATE state;
  
  
  
          while(m_pSourceVoice->GetState( &state ), state.BuffersQueued > 60)
  
          {
  
              WaitForSingleObject( m_hBufferEndEvent, INFINITE );
  
          }
  
  
  
          m_pSourceVoice->SubmitSourceBuffer( &m_SoundBuffer );
  
  
  
      return 0;
  
  }
  
  
  
  void CDVDAudio::Destroy()
  
  {
  
      m_pMasteringVoice->DestroyVoice();
  
      m_pXAudio2->Release();
  
  
  
      m_pSourceVoice->DestroyVoice();
  
  
  
      delete m_VoiceCallback;
  
      m_VoiceCallback = NULL;
  
  }

  #include "DVDAudioCodecFFmpeg.h"
  
  #include "Log.h"
  
  
  
  CDVDAudioCodecFFmpeg::CDVDAudioCodecFFmpeg() : CDVDAudioCodec()
  
  {
  
      m_iBufferSize = 0;
  
      m_pCodecContext = NULL;
  
      m_bOpenedCodec = false;
  
  }
  
  
  
  CDVDAudioCodecFFmpeg::~CDVDAudioCodecFFmpeg()
  
  {
  
      Dispose();
  
  }
  
  
  
  bool CDVDAudioCodecFFmpeg::Open(AVCodecID codecID, int iChannels, int iSampleRate)
  
  {
  
      AVCodec* pCodec;
  
      m_bOpenedCodec = false;
  
  
  
      av_register_all();
  
  
  
      pCodec = avcodec_find_decoder(codecID);
  
  
  
      m_pCodecContext = avcodec_alloc_context3(pCodec);//avcodec_alloc_context();
  
      avcodec_get_context_defaults3(m_pCodecContext, pCodec);
  
  
  
      if (!pCodec)
  
      {
  
          CLog::Log(LOGERROR, "CDVDAudioCodecFFmpeg::Open() Unable to find codec");
  
          return false;
  
      }
  
  
  
      m_pCodecContext->debug_mv = 0;
  
      m_pCodecContext->debug = 0;
  
      m_pCodecContext->workaround_bugs = 1;
  
  
  
      if (pCodec->capabilities & CODEC_CAP_TRUNCATED)
  
          m_pCodecContext->flags |= CODEC_FLAG_TRUNCATED;
  
  
  
      m_pCodecContext->channels = iChannels;
  
      m_pCodecContext->sample_rate = iSampleRate;
  
      //m_pCodecContext->bits_per_sample = 24;
  
  
  
  /* //FIXME BRENT
  
    if( ExtraData && ExtraSize > 0 )
  
    {
  
      m_pCodecContext->extradata_size = ExtraSize;
  
      m_pCodecContext->extradata = m_dllAvCodec.av_mallocz(ExtraSize + FF_INPUT_BUFFER_PADDING_SIZE);
  
      memcpy(m_pCodecContext->extradata, ExtraData, ExtraSize);
  
    }
  
  */
  
      // set acceleration
  
      //m_pCodecContext->dsp_mask = FF_MM_FORCE | FF_MM_MMX | FF_MM_MMXEXT | FF_MM_SSE; //BRENT
  
  
  
      if (avcodec_open2(m_pCodecContext, pCodec, NULL) < 0)
  
      {
  
          CLog::Log(LOGERROR, "CDVDAudioCodecFFmpeg::Open() Unable to open codec");
  
          Dispose();
  
          return false;
  
      }
  
  
  
      m_bOpenedCodec = true;
  
      return true;
  
  }
  
  
  
  void CDVDAudioCodecFFmpeg::Dispose()
  
  {
  
      if (m_pCodecContext)
  
      {
  
          if (m_bOpenedCodec) avcodec_close(m_pCodecContext);
  
          m_bOpenedCodec = false;
  
          av_free(m_pCodecContext);
  
          m_pCodecContext = NULL;
  
      }
  
      m_iBufferSize = 0;
  
  }
  
  
  
  int CDVDAudioCodecFFmpeg::Decode(BYTE* pData, int iSize)
  
  {
  
      int iBytesUsed;
  
      if (!m_pCodecContext) return -1;
  
  
  
      //Copy into a FFMpeg AVPAcket again
  
      AVPacket packet;
  
      av_init_packet(&packet);
  
  
  
      packet.data=pData;
  
      packet.size=iSize;
  
  
  
      int iOutputSize = AVCODEC_MAX_AUDIO_FRAME_SIZE; //BRENT
  
  
  
      iBytesUsed = avcodec_decode_audio3(m_pCodecContext, (int16_t *)m_buffer, &iOutputSize/*m_iBufferSize*/, &packet);
  
  
  
      m_iBufferSize = iOutputSize;//BRENT
  
  
  
      return iBytesUsed;
  
  }
  
  
  
  int CDVDAudioCodecFFmpeg::GetData(BYTE** dst)
  
  {
  
      *dst = m_buffer;
  
      return m_iBufferSize;
  
  }
  
  
  
  void CDVDAudioCodecFFmpeg::Reset()
  
  {
  
      if (m_pCodecContext) avcodec_flush_buffers(m_pCodecContext);
  
  }
  
  
  
  int CDVDAudioCodecFFmpeg::GetChannels()
  
  {
  
      if (m_pCodecContext) return m_pCodecContext->channels;
  
      return 0;
  
  }
  
  
  
  int CDVDAudioCodecFFmpeg::GetSampleRate()
  
  {
  
      if (m_pCodecContext) return m_pCodecContext->sample_rate;
  
      return 0;
  
  }
  
  
  
  int CDVDAudioCodecFFmpeg::GetBitsPerSample()
  
  {
  
      if (m_pCodecContext) return 16;
  
      return 0;
  
  }
  #include "DVDPlayerAudio.h"
  
  #include "DVDDemuxUtils.h"
  
  #include "Log.h"
  
  
  
  #include 
  
  #include "DVDAudioCodecFFmpeg.h" //FIXME Move to a codec factory!!
  
  
  
  CDVDPlayerAudio::CDVDPlayerAudio(CDVDClock* pClock) : CThread()
  
  {
  
      m_pClock = pClock;
  
      m_pAudioCodec = NULL;
  
      m_bInitializedOutputDevice = false;
  
      m_iSourceChannels = 0;
  
      m_audioClock = 0;
  
  
  
  //  m_currentPTSItem.pts = DVD_NOPTS_VALUE;
  
  //  m_currentPTSItem.timestamp = 0;
  
  
  
      SetSpeed(DVD_PLAYSPEED_NORMAL);
  
  
  
      InitializeCriticalSection(&m_critCodecSection);
  
      m_messageQueue.SetMaxDataSize(10 * 16 * 1024);
  
  //  g_dvdPerformanceCounter.EnableAudioQueue(&m_packetQueue);
  
  }
  
  
  
  CDVDPlayerAudio::~CDVDPlayerAudio()
  
  {
  
  //  g_dvdPerformanceCounter.DisableAudioQueue();
  
  
  
      // close the stream, and don't wait for the audio to be finished
  
      CloseStream(true);
  
      DeleteCriticalSection(&m_critCodecSection);
  
  }
  
  
  
  bool CDVDPlayerAudio::OpenStream( CDemuxStreamAudio *pDemuxStream )                                 
  
  {
  
      // should alway's be NULL!!!!, it will probably crash anyway when deleting m_pAudioCodec here.
  
      if (m_pAudioCodec)
  
      {
  
          CLog::Log(LOGFATAL, "CDVDPlayerAudio::OpenStream() m_pAudioCodec != NULL");
  
          return false;
  
      }
  
  
  
      AVCodecID codecID = pDemuxStream->codec;
  
  
  
      CLog::Log(LOGNOTICE, "Finding audio codec for: %i", codecID);
  
      //m_pAudioCodec = CDVDFactoryCodec::CreateAudioCodec( pDemuxStream ); 
  
      m_pAudioCodec = new CDVDAudioCodecFFmpeg; //FIXME BRENT Codec Factory needed!
  
  
  
      if (!m_pAudioCodec->Open(pDemuxStream->codec, pDemuxStream->iChannels, pDemuxStream->iSampleRate))
  
      {
  
          m_pAudioCodec->Dispose();
  
          delete m_pAudioCodec;
  
          m_pAudioCodec = NULL;
  
          return false;
  
      }
  
  
  
      if( !m_pAudioCodec )
  
      {
  
          CLog::Log(LOGERROR, "Unsupported audio codec");
  
          return false;
  
      }
  
  
  
      m_codec = pDemuxStream->codec;
  
      m_iSourceChannels = pDemuxStream->iChannels;
  
  
  
      m_messageQueue.Init();
  
  
  
      CLog::Log(LOGNOTICE, "Creating audio thread");
  
      Create();
  
  
  
      return true;
  
  }
  
  
  
  void CDVDPlayerAudio::CloseStream(bool bWaitForBuffers)
  
  {
  
      // wait until buffers are empty
  
      if (bWaitForBuffers) m_messageQueue.WaitUntilEmpty();
  
  
  
      // send abort message to the audio queue
  
      m_messageQueue.Abort();
  
  
  
      CLog::Log(LOGNOTICE, "waiting for audio thread to exit");
  
  
  
      // shut down the adio_decode thread and wait for it
  
      StopThread(); // will set this->m_bStop to true
  
      this->WaitForThreadExit(INFINITE);
  
  
  
      // uninit queue
  
      m_messageQueue.End();
  
  
  
      CLog::Log(LOGNOTICE, "Deleting audio codec");
  
      if (m_pAudioCodec)
  
      {
  
          m_pAudioCodec->Dispose();
  
          delete m_pAudioCodec;
  
          m_pAudioCodec = NULL;
  
      }
  
  
  
      // flush any remaining pts values
  
      //FlushPTSQueue(); //FIXME BRENT
  
  }
  
  
  
  void CDVDPlayerAudio::OnStartup()
  
  {
  
      CThread::SetName("CDVDPlayerAudio");
  
      pAudioPacket = NULL;
  
      m_audioClock = 0;
  
      audio_pkt_data = NULL;
  
      audio_pkt_size = 0;
  
  
  
  //  g_dvdPerformanceCounter.EnableAudioDecodePerformance(ThreadHandle());
  
  }
  
  
  
  void CDVDPlayerAudio::Process()
  
  {
  
      CLog::Log(LOGNOTICE, "running thread: CDVDPlayerAudio::Process()");
  
  
  
      int result;
  
  
  
      // silence data
  
      BYTE silence[1024];
  
      memset(silence, 0, 1024);
  
  
  
      DVDAudioFrame audioframe;
  
  
  
      __int64 iClockDiff=0;
  
      while (!m_bStop)
  
      {
  
          //Don't let anybody mess with our global variables
  
          EnterCriticalSection(&m_critCodecSection);
  
          result = DecodeFrame(audioframe, m_speed != DVD_PLAYSPEED_NORMAL); // blocks if no audio is available, but leaves critical section before doing so
  
          LeaveCriticalSection(&m_critCodecSection);
  
  
  
          if( result & DECODE_FLAG_ERROR ) 
  
          {      
  
              CLog::Log(LOGERROR, "CDVDPlayerAudio::Process - Decode Error. Skipping audio frame");
  
              continue;
  
          }
  
  
  
          if( result & DECODE_FLAG_ABORT )
  
          {
  
              CLog::Log(LOGDEBUG, "CDVDPlayerAudio::Process - Abort recieved, exiting thread");
  
              break;
  
          }
  
  
  
          if( result & DECODE_FLAG_DROP ) //FIXME BRENT
  
          {
  
  /*          //frame should be dropped. Don't let audio move ahead of the current time thou
  
              //we need to be able to start playing at any time
  
              //when playing backwords, we try to keep as small buffers as possible
  
  
  
              // set the time at this delay
  
              AddPTSQueue(audioframe.pts, m_dvdAudio.GetDelay());
  
  */
  
              if (m_speed > 0)
  
              {
  
                  __int64 timestamp = m_pClock->GetAbsoluteClock() + (audioframe.duration * DVD_PLAYSPEED_NORMAL) / m_speed;
  
                  while( !m_bStop && timestamp > m_pClock->GetAbsoluteClock() ) Sleep(1);
  
              }
  
              continue;
  
          }
  
  
  
          if( audioframe.size > 0 ) 
  
          {
  
              // we have succesfully decoded an audio frame, openup the audio device if not already done
  
              if (!m_bInitializedOutputDevice)
  
              {
  
                  m_bInitializedOutputDevice = InitializeOutputDevice();
  
              }
  
  
  
              //Add any packets play
  
              m_dvdAudio.AddPackets(audioframe.data, audioframe.size);
  
  
  
              // store the delay for this pts value so we can calculate the current playing
  
              //AddPTSQueue(audioframe.pts, m_dvdAudio.GetDelay() - audioframe.duration);//BRENT
  
          }
  
  
  
          // if we where asked to resync on this packet, do so here
  
          if( result & DECODE_FLAG_RESYNC )
  
          {      
  
              CLog::Log(LOGDEBUG, "CDVDPlayerAudio::Process - Resync recieved.");
  
              //while (!m_bStop && (unsigned int)m_dvdAudio.GetDelay() > audioframe.duration ) Sleep(5); //BRENT
  
              m_pClock->Discontinuity(CLOCK_DISC_NORMAL, audioframe.pts);
  
          }
  
  
  
  #ifdef USEOLDSYNC
  
          //Clock should be calculated after packets have been added as m_audioClock points to the 
  
          //time after they have been played
  
  
  
          const __int64 iCurrDiff = (m_audioClock - m_dvdAudio.GetDelay()) - m_pClock->GetClock();
  
          const __int64 iAvDiff = (iClockDiff + iCurrDiff)/2;
  
  
  
          //Check for discontinuity in the stream, use a moving average to
  
          //eliminate highfreq fluctuations of large packet sizes
  
          if( ABS(iAvDiff) > 5000 ) // sync clock if average diff is bigger than 5 msec 
  
          {
  
              //Wait untill only the new audio frame wich triggered the discontinuity is left
  
              //then set disc state
  
              while (!m_bStop && (unsigned int)m_dvdAudio.GetBytesInBuffer() > audioframe.size ) Sleep(5);
  
  
  
              m_pClock->Discontinuity(CLOCK_DISC_NORMAL, m_audioClock - m_dvdAudio.GetDelay());
  
              CLog::("CDVDPlayer:: Detected Audio Discontinuity, syncing clock. diff was: %I64d, %I64d, av: %I64d", iClockDiff, iCurrDiff, iAvDiff);
  
              iClockDiff = 0;
  
          }
  
          else
  
          {      
  
              //Do gradual adjustments (not working yet)
  
              //m_pClock->AdjustSpeedToMatch(iClock + iAvDiff);
  
              iClockDiff = iCurrDiff;
  
          }
  
  #endif
  
      }
  
  }
  
  
  
  void CDVDPlayerAudio::OnExit()
  
  {
  
      //g_dvdPerformanceCounter.DisableAudioDecodePerformance();
  
  
  
      // destroy audio device
  
      CLog::Log(LOGNOTICE, "Closing audio device");
  
      m_dvdAudio.Destroy();
  
      m_bInitializedOutputDevice = false;
  
  
  
      CLog::Log(LOGNOTICE, "thread end: CDVDPlayerAudio::OnExit()");
  
  }
  
  
  
  // decode one audio frame and returns its uncompressed size
  
  int CDVDPlayerAudio::DecodeFrame(DVDAudioFrame &audioframe, bool bDropPacket)
  
  {
  
      CDVDDemux::DemuxPacket* pPacket = pAudioPacket;
  
      int n=48000*2*16/8, len;
  
  
  
      //Store amount left at this point, and what last pts was
  
      unsigned __int64 first_pkt_pts = 0;
  
      int first_pkt_size = 0; 
  
      int first_pkt_used = 0;
  
      int result = 0;
  
  
  
      // make sure the sent frame is clean
  
      memset(&audioframe, 0, sizeof(DVDAudioFrame));
  
  
  
      if (pPacket)
  
      {
  
          first_pkt_pts = pPacket->pts;
  
          first_pkt_size = pPacket->iSize;
  
          first_pkt_used = first_pkt_size - audio_pkt_size;
  
      }
  
  
  
      for (;;)
  
      {
  
          /* NOTE: the audio packet can contain several frames */
  
          while (audio_pkt_size > 0)
  
          {
  
              len = m_pAudioCodec->Decode(audio_pkt_data, audio_pkt_size);
  
              if (len < 0)
  
              {
  
                  /* if error, we skip the frame */
  
                  audio_pkt_size=0;
  
                  m_pAudioCodec->Reset();
  
                  break;
  
              }
  
  
  
              // fix for fucked up decoders //FIXME BRENT
  
              if( len > audio_pkt_size )
  
              {        
  
                  CLog::Log(LOGERROR, "CDVDPlayerAudio:DecodeFrame - Codec tried to consume more data than available. Potential memory corruption");        
  
                  audio_pkt_size=0;
  
                  m_pAudioCodec->Reset();
  
                  assert(0);
  
              }
  
  
  
              // get decoded data and the size of it
  
              audioframe.size = m_pAudioCodec->GetData(&audioframe.data);
  
  
  
              audio_pkt_data += len;
  
              audio_pkt_size -= len;
  
  
  
              if (audioframe.size <= 0) continue;
  
  
  
              audioframe.pts = m_audioClock;
  
  
  
              // compute duration.
  
              n = m_pAudioCodec->GetChannels() * m_pAudioCodec->GetBitsPerSample() / 8 * m_pAudioCodec->GetSampleRate();
  
              if (n > 0)
  
              {
  
                  // safety check, if channels == 0, n will result in 0, and that will result in a nice devide exception
  
                  audioframe.duration = (unsigned int)(((__int64)audioframe.size * DVD_TIME_BASE) / n);
  
  
  
                  // increase audioclock to after the packet
  
                  m_audioClock += audioframe.duration;
  
              }
  
  
  
              //If we are asked to drop this packet, return a size of zero. then it won't be played
  
              //we currently still decode the audio.. this is needed since we still need to know it's 
  
              //duration to make sure clock is updated correctly.
  
              if( bDropPacket )
  
              {
  
                  result |= DECODE_FLAG_DROP;
  
              }
  
              return result;
  
          }
  
  
  
          // free the current packet
  
          if (pPacket)
  
          {
  
              CDVDDemuxUtils::FreeDemuxPacket(pPacket); //BRENT FIXME
  
              pPacket = NULL;
  
              pAudioPacket = NULL;
  
          }
  
  
  
          if (m_messageQueue.RecievedAbortRequest()) return DECODE_FLAG_ABORT;
  
  
  
          // read next packet and return -1 on error
  
          LeaveCriticalSection(&m_critCodecSection); //Leave here as this might stall a while
  
  
  
          CDVDMsg* pMsg;
  
          MsgQueueReturnCode ret = m_messageQueue.Get(&pMsg, INFINITE);
  
  
  
          EnterCriticalSection(&m_critCodecSection);
  
          if (MSGQ_IS_ERROR(ret) || ret == MSGQ_ABORT) return DECODE_FLAG_ABORT;
  
  
  
          if (pMsg->IsType(CDVDMsg::DEMUXER_PACKET))
  
          {
  
              CDVDMsgDemuxerPacket* pMsgDemuxerPacket = (CDVDMsgDemuxerPacket*)pMsg;
  
              pPacket = pMsgDemuxerPacket->GetPacket();
  
              pMsgDemuxerPacket->m_pPacket = NULL; // XXX, test
  
              pAudioPacket = pPacket;
  
              audio_pkt_data = pPacket->pData;
  
              audio_pkt_size = pPacket->iSize;
  
          }
  
          else
  
          {
  
              // other data is not used here, free if
  
              // msg itself will still be available
  
              pMsg->Release();
  
          }
  
  
  
          // if update the audio clock with the pts
  
          if (pMsg->IsType(CDVDMsg::DEMUXER_PACKET) || pMsg->IsType(CDVDMsg::GENERAL_RESYNC))
  
          {
  
              if (pMsg->IsType(CDVDMsg::GENERAL_RESYNC))
  
              { 
  
                  //player asked us to sync on this package
  
                  CDVDMsgGeneralResync* pMsgGeneralResync = (CDVDMsgGeneralResync*)pMsg;
  
                  result |= DECODE_FLAG_RESYNC;
  
                  m_audioClock = pMsgGeneralResync->GetPts();
  
              }
  
              else if (pPacket->pts != DVD_NOPTS_VALUE) // CDVDMsg::DEMUXER_PACKET, pPacket is already set above
  
              {
  
                  if (first_pkt_size == 0) 
  
                  { 
  
                      //first package
  
                      m_audioClock = pPacket->pts;        
  
                  }
  
                  else if (first_pkt_pts > pPacket->pts)
  
                  { 
  
                      //okey first packet in this continous stream, make sure we use the time here        
  
                      m_audioClock = pPacket->pts;        
  
                  }
  
                  else if((unsigned __int64)m_audioClock < pPacket->pts || (unsigned __int64)m_audioClock > pPacket->pts)
  
                  {
  
                      //crap, moved outsided correct pts
  
                      //Use pts from current packet, untill we find a better value for it.
  
                      //Should be ok after a couple of frames, as soon as it starts clean on a packet
  
                      m_audioClock = pPacket->pts;
  
                  }
  
                  else if(first_pkt_size == first_pkt_used)
  
                  {
  
                      //Nice starting up freshly on the start of a packet, use pts from it
  
                      m_audioClock = pPacket->pts;
  
                  }
  
              }
  
          }
  
          pMsg->Release();
  
      }
  
  }
  
  
  
  void CDVDPlayerAudio::SetSpeed(int speed)
  
  { 
  
      m_speed = speed;
  
  
  
      //if (m_speed == DVD_PLAYSPEED_PAUSE) m_dvdAudio.Pause(); //BRENT FIXME
  
      //else m_dvdAudio.Resume();
  
  }
  
  
  
  bool CDVDPlayerAudio::InitializeOutputDevice()
  
  {
  
      int iChannels = m_pAudioCodec->GetChannels();
  
      int iSampleRate = m_pAudioCodec->GetSampleRate();
  
      int iBitsPerSample = m_pAudioCodec->GetBitsPerSample();
  
      //bool bPasstrough = m_pAudioCodec->NeedPasstrough(); //BRENT
  
  
  
      if (iChannels == 0 || iSampleRate == 0 || iBitsPerSample == 0)
  
      {
  
          CLog::Log(LOGERROR, "Unable to create audio device, (iChannels == 0 || iSampleRate == 0 || iBitsPerSample == 0)");
  
          return false;
  
      }
  
  
  
      CLog::Log(LOGNOTICE, "Creating audio device with codec id: %i, channels: %i, sample rate: %i", m_codec, iChannels, iSampleRate);
  
      if (m_dvdAudio.Create(iChannels, iSampleRate, iBitsPerSample, /*bPasstrough*/0)) // always 16 bit with ffmpeg ? //BRENT Passthrough needed?
  
      {
  
          return true;
  
      }
  
  
  
      CLog::Log(LOGERROR, "Failed Creating audio device with codec id: %i, channels: %i, sample rate: %i", m_codec, iChannels, iSampleRate);
  
      return false;
  
  }

• HTTP Livestreaming with ffmpeg

  25 août 2016, par Hugo14453

  Some context : I have an MKV file, I am attempting to stream it to http://localhost:8090/test.flv as an flv file.

  The stream begins and then immediately ends.

  The command I am using is :

  sudo ffmpeg -re -i input.mkv -c:v libx264 -maxrate 1000k -bufsize 2000k -an -bsf:v h264_mp4toannexb -g 50 http://localhost:8090/test.flv

  A breakdown of what I believe these options do incase this post becomes useful for someone else :

  sudo

  Run as root

  ffmpeg

  The stream command thingy

  -re

  Stream in real-time

  -i input.mkv

  Input option and path to input file

  -c:v libx264

  Use codec libx264 for conversion

  -maxrate 1000k -bufsize 2000k

  No idea, some options for conversion, seems to help

  -an -bsf:v h264_mp4toannexb

  Audio options I think, not sure really. Also seems to help

  -g 50

  Still no idea, maybe frame rateframerateframerateframerate ?

  http://localhost:8090/test.flv

  Output using http protocol to localhost on port 8090 as a file called test.flv

  Anyway the actual issue I have is that it begins to stream for about a second and then immediately ends.

  The mpeg command result :

  ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  
    configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 48.101 / 57. 48.101
  
    libavformat    57. 41.100 / 57. 41.100
  
    libavdevice    57.  0.102 / 57.  0.102
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libavresample   3.  0.  0 /  3.  0.  0
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, matroska,webm, from 'input.mkv':
  
    Metadata:
  
      encoder         : libebml v1.3.0 + libmatroska v1.4.0
  
      creation_time   : 1970-01-01 00:00:02
  
    Duration: 00:01:32.26, start: 0.000000, bitrate: 4432 kb/s
  
      Stream #0:0(eng): Video: h264 (High 10), yuv420p10le, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
  
      Stream #0:1(nor): Audio: flac, 48000 Hz, stereo, s16 (default)
  
  [libx264 @ 0x2e1c380] using SAR=1/1
  
  [libx264 @ 0x2e1c380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x2e1c380] profile High, level 4.0
  
  [libx264 @ 0x2e1c380] 264 - core 148 r2643 5c65704 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=50 keyint_min=5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1000 vbv_bufsize=2000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
  
  [flv @ 0x2e3f0a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
  
  Output #0, flv, to 'http://localhost:8090/test.flv':
  
    Metadata:
  
      encoder         : Lavf57.41.100
  
      Stream #0:0(eng): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 1k tbn, 23.98 tbc (default)
  
      Metadata:
  
        encoder         : Lavc57.48.101 libx264
  
      Side data:
  
        cpb: bitrate max/min/avg: 1000000/0/0 buffer size: 2000000 vbv_delay: -1
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  
  Press [q] to stop, [?] for help
  
  Killed   26 fps= 26 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x  

  The ffserver outputs :

  Sat Aug 20 12:40:11 2016 File '/test.flv' not found
  
  Sat Aug 20 12:40:11 2016 [SERVER IP] - - [POST] "/test.flv HTTP/1.1" 404 189

  The config file is :

  #Sample ffserver configuration file
  
  
  
  # Port on which the server is listening. You must select a different
  
  # port from your standard HTTP web server if it is running on the same
  
  # computer.
  
  Port 8090
  
  
  
  # Address on which the server is bound. Only useful if you have
  
  # several network interfaces.
  
  BindAddress 0.0.0.0
  
  
  
  # Number of simultaneous HTTP connections that can be handled. It has
  
  # to be defined *before* the MaxClients parameter, since it defines the
  
  # MaxClients maximum limit.
  
  MaxHTTPConnections 2000
  
  
  
  # Number of simultaneous requests that can be handled. Since FFServer
  
  # is very fast, it is more likely that you will want to leave this high
  
  # and use MaxBandwidth, below.
  
  MaxClients 1000
  
  
  
  # This the maximum amount of kbit/sec that you are prepared to
  
  # consume when streaming to clients.
  
  MaxBandwidth 1000
  
  
  
  # Access log file (uses standard Apache log file format)
  
  # '-' is the standard output.
  
  CustomLog -
  
  
  
  # Suppress that if you want to launch ffserver as a daemon.
  
  #NoDaemon
  
  
  
  
  
  ##################################################################
  
  # Definition of the live feeds. Each live feed contains one video
  
  # and/or audio sequence coming from an ffmpeg encoder or another
  
  # ffserver. This sequence may be encoded simultaneously with several
  
  # codecs at several resolutions.
  
  
  
  <feed>
  
  
  
  ACL allow 192.168.0.0 192.168.255.255
  
  
  
  # You must use 'ffmpeg' to send a live feed to ffserver. In this
  
  # example, you can type:
  
  #
  
  #ffmpeg http://localhost:8090/test.ffm
  
  
  
  # ffserver can also do time shifting. It means that it can stream any
  
  # previously recorded live stream. The request should contain:
  
  # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
  
  # a path where the feed is stored on disk. You also specify the
  
  # maximum size of the feed, where zero means unlimited. Default:
  
  # File=/tmp/feed_name.ffm FileMaxSize=5M
  
  File /tmp/feed1.ffm
  
  FileMaxSize 200m
  
  
  
  # You could specify
  
  # ReadOnlyFile /saved/specialvideo.ffm
  
  # This marks the file as readonly and it will not be deleted or updated.
  
  
  
  # Specify launch in order to start ffmpeg automatically.
  
  # First ffmpeg must be defined with an appropriate path if needed,
  
  # after that options can follow, but avoid adding the http:// field
  
  #Launch ffmpeg
  
  
  
  # Only allow connections from localhost to the feed.
  
      ACL allow 127.0.0.1
  
  
  
  </feed>
  
  
  
  
  
  ##################################################################
  
  # Now you can define each stream which will be generated from the
  
  # original audio and video stream. Each format has a filename (here
  
  # 'test1.mpg'). FFServer will send this stream when answering a
  
  # request containing this filename.
  
  
  
  <stream>
  
  
  
  # coming from live feed 'feed1'
  
  Feed feed1.ffm
  
  
  
  # Format of the stream : you can choose among:
  
  # mpeg       : MPEG-1 multiplexed video and audio
  
  # mpegvideo  : only MPEG-1 video
  
  # mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
  
  # ogg        : Ogg format (Vorbis audio codec)
  
  # rm         : RealNetworks-compatible stream. Multiplexed audio and video.
  
  # ra         : RealNetworks-compatible stream. Audio only.
  
  # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
  
  # jpeg       : Generate a single JPEG image.
  
  # asf        : ASF compatible streaming (Windows Media Player format).
  
  # swf        : Macromedia Flash compatible stream
  
  # avi        : AVI format (MPEG-4 video, MPEG audio sound)
  
  Format mpeg
  
  
  
  # Bitrate for the audio stream. Codecs usually support only a few
  
  # different bitrates.
  
  AudioBitRate 32
  
  
  
  # Number of audio channels: 1 = mono, 2 = stereo
  
  AudioChannels 2
  
  
  
  # Sampling frequency for audio. When using low bitrates, you should
  
  # lower this frequency to 22050 or 11025. The supported frequencies
  
  # depend on the selected audio codec.
  
  AudioSampleRate 44100
  
  
  
  # Bitrate for the video stream
  
  VideoBitRate 64
  
  
  
  # Ratecontrol buffer size
  
  VideoBufferSize 40
  
  
  
  # Number of frames per second
  
  VideoFrameRate 3
  
  
  
  # Size of the video frame: WxH (default: 160x128)
  
  # The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
  
  # qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
  
  # wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
  
  # hd1080
  
  VideoSize hd1080
  
  
  
  # Transmit only intra frames (useful for low bitrates, but kills frame rate).
  
  #VideoIntraOnly
  
  
  
  # If non-intra only, an intra frame is transmitted every VideoGopSize
  
  # frames. Video synchronization can only begin at an intra frame.
  
  VideoGopSize 12
  
  
  
  # More MPEG-4 parameters
  
  # VideoHighQuality
  
  # Video4MotionVector
  
  
  
  # Choose your codecs:
  
  #AudioCodec mp2
  
  #VideoCodec mpeg1video
  
  
  
  # Suppress audio
  
  #NoAudio
  
  
  
  # Suppress video
  
  #NoVideo
  
  
  
  #VideoQMin 3
  
  #VideoQMax 31
  
  
  
  # Set this to the number of seconds backwards in time to start. Note that
  
  # most players will buffer 5-10 seconds of video, and also you need to allow
  
  # for a keyframe to appear in the data stream.
  
  #Preroll 15
  
  
  
  # ACL:
  
  
  
  # You can allow ranges of addresses (or single addresses)
  
  ACL ALLOW localhost
  
  
  
  # You can deny ranges of addresses (or single addresses)
  
  #ACL DENY <first address="address"> 
  
  
  
  # You can repeat the ACL allow/deny as often as you like. It is on a per
  
  # stream basis. The first match defines the action. If there are no matches,
  
  # then the default is the inverse of the last ACL statement.
  
  #
  
  # Thus 'ACL allow localhost' only allows access from localhost.
  
  # 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
  
  # allow everybody else.
  
  
  
  </first></stream>
  
  
  
  
  
  ##################################################################
  
  # Example streams
  
  
  
  
  
  # Multipart JPEG
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format mpjpeg
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  #NoAudio
  
  #Strict -1
  
  #</stream>
  
  
  
  
  
  # Single JPEG
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format jpeg
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  ##VideoSize 352x240
  
  #NoAudio
  
  #Strict -1
  
  #</stream>
  
  
  
  
  
  # Flash
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format swf
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  # ASF compatible
  
  
  
  <stream>
  
  Feed feed1.ffm
  
  Format asf
  
  VideoFrameRate 15
  
  VideoSize 352x240
  
  VideoBitRate 256
  
  VideoBufferSize 40
  
  VideoGopSize 30
  
  AudioBitRate 64
  
  StartSendOnKey
  
  </stream>
  
  
  
  
  
  # MP3 audio
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format mp2
  
  #AudioCodec mp3
  
  #AudioBitRate 64
  
  #AudioChannels 1
  
  #AudioSampleRate 44100
  
  #NoVideo
  
  #</stream>
  
  
  
  
  
  # Ogg Vorbis audio
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Title "Stream title"
  
  #AudioBitRate 64
  
  #AudioChannels 2
  
  #AudioSampleRate 44100
  
  #NoVideo
  
  #</stream>
  
  
  
  
  
  # Real with audio only at 32 kbits
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format rm
  
  #AudioBitRate 32
  
  #NoVideo
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  # Real with audio and video at 64 kbits
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format rm
  
  #AudioBitRate 32
  
  #VideoBitRate 128
  
  #VideoFrameRate 25
  
  #VideoGopSize 25
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # A stream coming from a file: you only need to set the input
  
  # filename and optionally a new format. Supported conversions:
  
  #    AVI -> ASF
  
  
  
  #<stream>
  
  #File "/usr/local/httpd/htdocs/tlive.rm"
  
  #NoAudio
  
  #</stream>
  
  
  
  #<stream>
  
  #File "/usr/local/httpd/htdocs/test.asf"
  
  #NoAudio
  
  #Author "Me"
  
  #Copyright "Super MegaCorp"
  
  #Title "Test stream from disk"
  
  #Comment "Test comment"
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # RTSP examples
  
  #
  
  # You can access this stream with the RTSP URL:
  
  #   rtsp://localhost:5454/test1-rtsp.mpg
  
  #
  
  # A non-standard RTSP redirector is also created. Its URL is:
  
  #   http://localhost:8090/test1-rtsp.rtsp
  
  
  
  #<stream>
  
  #Format rtp
  
  #File "/usr/local/httpd/htdocs/test1.mpg"
  
  #</stream>
  
  
  
  
  
  # Transcode an incoming live feed to another live feed,
  
  # using libx264 and video presets
  
  
  
  #<stream>
  
  #Format rtp
  
  #Feed feed1.ffm
  
  #VideoCodec libx264
  
  #VideoFrameRate 24
  
  #VideoBitRate 100
  
  #VideoSize 480x272
  
  #AVPresetVideo default
  
  #AVPresetVideo baseline
  
  #AVOptionVideo flags +global_header
  
  #
  
  #AudioCodec libfaac
  
  #AudioBitRate 32
  
  #AudioChannels 2
  
  #AudioSampleRate 22050
  
  #AVOptionAudio flags +global_header
  
  #</stream>
  
  
  
  ##################################################################
  
  # SDP/multicast examples
  
  #
  
  # If you want to send your stream in multicast, you must set the
  
  # multicast address with MulticastAddress. The port and the TTL can
  
  # also be set.
  
  #
  
  # An SDP file is automatically generated by ffserver by adding the
  
  # 'sdp' extension to the stream name (here
  
  # http://localhost:8090/test1-sdp.sdp). You should usually give this
  
  # file to your player to play the stream.
  
  #
  
  # The 'NoLoop' option can be used to avoid looping when the stream is
  
  # terminated.
  
  
  
  #<stream>
  
  #Format rtp
  
  #File "/usr/local/httpd/htdocs/test1.mpg"
  
  #MulticastAddress 224.124.0.1
  
  #MulticastPort 5000
  
  #MulticastTTL 16
  
  #NoLoop
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # Special streams
  
  
  
  # Server status
  
  
  
  <stream>
  
  Format status
  
  
  
  # Only allow local people to get the status
  
  ACL allow localhost
  
  ACL allow 192.168.0.0 192.168.255.255
  
  
  
  #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
  
  </stream>
  
  
  
  
  
  # Redirect index.html to the appropriate site
  
  
  
  <redirect>
  
  URL http://www.ffmpeg.org/
  
  </redirect>
  
  
  
  
  
  #http://www.ffmpeg.org/

  Any help is greatly appreciated, I will do my best draw a picture of the best answer based on their username.

• ffmpeg concat doesn't keep video speed/framerate ?

  28 août 2016, par Nick

  I have a bunch of small webm clips in directory vidchunks. These clips were generated using javascript MediaRecorder API.

  MediaRecorder code :

  var mRecorder;
  
  var chunks = [];
  
  navigator.mediaDevices.getUserMedia({
  
    audio: false,
  
    video: {
  
    width: 1280,
  
    height: 760,
  
    mozMediaSource: "screen",
  
    mediaSource: "screen"
  
  }
  
  }).then(function(stream) {
  
    mRecorder = new MediaRecorder(stream,{mimeType:"video/webm"});
  
    mRecorder.ondataavailable = function(event) {
  
      var blob = event.data;
  
      chunks.push(blob);
  
      var vidchunk = new Blob(chunks);
  
      // save to directory vid chunk.
  
    };
  
    mRecorder.start(5000);
  
  }).catch(function(error) {
  
    console.log(error.message);
  
  });

  Then, once I have a bunch of these 5 second clips, I merge them :

  ffmpeg -safe 0 \
  
  -f concat -i <(for f in `ls vidchunks/* | sort -V`; do echo file $f; done) \
  
  -c:v copy -r 30 -vsync drop test.webm

  When I play back these chunks individually, they play correctly. All 5 seconds of video are rendered smoothly.

  However, once I merge, some 5 second chunks are condensed to like 1 second. So the total video, which should be 50 seconds for 10 chunks, turns out to be like 38 seconds because three of the chunks got condensed to 1 second.

  Any ideas on how to fix this ?

  EDIT : I’ve tried :

  • without "-r 30 -vsync drop"
  • without "-c:v copy"

  No changes...

  EDIT2 (some ffprobe outputs) :

  ffprobe 1472343170-1.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343170-1.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343245-16.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343245-16.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343350-37.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343350-37.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  I also noticed that the merged video plays normal when the mouse is moving around on the screen, but when the mouse is not moving, it moves 5x faster.

  I know this because I have a timer on the screen that I’m recording, and I can see that the timer speeds up significantly when mouse isn’t moving but when mouse is moving, the timer moves at normal speed.

  EDIT3 :
  The DTS/PTS looks like resets after each file is merged :

  DTS -9223363446915256, next:6866667 st:0 invalid dropping
  
  PTS -9223363446915256, next:6866667 invalid dropping st:0
  
  DTS -9223363446915222, next:6900000 st:0 invalid dropping
  
  PTS -9223363446915222, next:6900000 invalid dropping st:0
  
  DTS -9223363446915188, next:6933333 st:0 invalid dropping
  
  PTS -9223363446915188, next:6933333 invalid dropping st:0
  
  [concat @ 0x1cde3e0] DTS -9223363446920184 < 5006 out of order
  
  DTS -9223363446920184, next:5033333 st:0 invalid dropping
  
  PTS -9223363446920184, next:5033333 invalid dropping st:0
  
  DTS -9223363446920056, next:5066667 st:0 invalid dropping
  
  PTS -9223363446920056, next:5066667 invalid dropping st:0
  
  DTS -9223363446919956, next:5100000 st:0 invalid dropping
  
  PTS -9223363446919956, next:5100000 invalid dropping st:0

  EDIT4 : Tried ffmpeg -i "$f" -y -c copy -fflags +genpts "$f" and then merge again.

  Works a lot better, but now other files are getting skipped.

  Here’s ffprobe from a file that’s skipped :

  Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:01.17, start: 0.000000, bitrate: 428 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1c01940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343225-12.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)

  Here’s ffprobe from a file not skipped :

  Input #0, matroska,webm, from '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:05.03, start: 0.000000, bitrate: 80 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1349940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)