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• how do i create a stereo mp3 file with latest version of ffmpeg ?

  17 juin 2016, par Sean

  I’m updating my code from the older version of ffmpeg (53) to the newer (54/55). Code that did work has now been deprecated or removed so i’m having problems updating it.

  Previously I could create a stereo MP3 file using a sample format called :

  SAMPLE_FMT_S16

  That matched up perfectly with my source stream. This has now been replace with

  AV_SAMPLE_FMT_S16

  Which works fine for mono recordings but when I try to create a stereo MP3 file it bugs out at avcodec_open2 with :

  "Specified sample_fmt is not supported."

  Through trial and error I’ve found that using

  AV_SAMPLE_FMT_S16P

  ...is accepted by avcodec_open2 but when I get through and create the MP3 file the sound is very distorted - it sounds about 2 octaves lower than usual with a massive hum in the background - here’s an example recording :

  http://hosting.ispyconnect.com/example.mp3

  I’ve been told by the ffmpeg guys that this is because I now need to manually deinterleave my byte stream before calling :

  avcodec_fill_audio_frame

  How do I do that ? I’ve tried using the swrescale library without success and i’ve tried manually feeding in L/R data into avcodec_fill_audio_frame but the results i’m getting are sounding exactly the same as without interleaving.

  Here is my code for encoding :

  void add_audio_sample( AudioWriterPrivateData^ data, BYTE* soundBuffer, int soundBufferSize)
  
  {
  
      libffmpeg::AVCodecContext* c = data->AudioStream->codec;
  
      memcpy(data->AudioBuffer + data->AudioBufferSizeCurrent,  soundBuffer, soundBufferSize);
  
      data->AudioBufferSizeCurrent += soundBufferSize;
  
      uint8_t* pSoundBuffer = (uint8_t *)data->AudioBuffer;
  
      DWORD nCurrentSize    = data->AudioBufferSizeCurrent;
  
  
  
      libffmpeg::AVFrame *frame;
  
  
  
      int got_packet;
  
      int ret;
  
      int size = libffmpeg::av_samples_get_buffer_size(NULL, c->channels,
  
                                                data->AudioInputSampleSize,
  
                                                c->sample_fmt, 1);
  
  
  
      while( nCurrentSize >= size)    {
  
  
  
          frame=libffmpeg::avcodec_alloc_frame();
  
          libffmpeg::avcodec_get_frame_defaults(frame);
  
  
  
          frame->nb_samples = data->AudioInputSampleSize;
  
  
  
          ret = libffmpeg::avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, pSoundBuffer, size, 1);
  
          if (ret<0)
  
          {
  
              throw gcnew System::IO::IOException("error filling audio");
  
          }
  
          //audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
  
  
  
          libffmpeg::AVPacket pkt = { 0 };
  
          libffmpeg::av_init_packet(&pkt);
  
  
  
          ret = libffmpeg::avcodec_encode_audio2(c, &pkt, frame, &got_packet);
  
  
  
          if (ret<0)
  
                  throw gcnew System::IO::IOException("error encoding audio");
  
          if (got_packet) {
  
              pkt.stream_index = data->AudioStream->index;
  
  
  
              if (pkt.pts != AV_NOPTS_VALUE)
  
                  pkt.pts = libffmpeg::av_rescale_q(pkt.pts, c->time_base, c->time_base);
  
              if (pkt.duration > 0)
  
                  pkt.duration = av_rescale_q(pkt.duration, c->time_base, c->time_base);
  
  
  
              pkt.flags |= AV_PKT_FLAG_KEY;
  
  
  
              if (libffmpeg::av_interleaved_write_frame(data->FormatContext, &pkt) != 0)
  
                      throw gcnew System::IO::IOException("unable to write audio frame.");
  
  
  
  
  
          }
  
          nCurrentSize -= size;  
  
          pSoundBuffer += size;   
  
      }
  
      memcpy(data->AudioBuffer, data->AudioBuffer + data->AudioBufferSizeCurrent - nCurrentSize, nCurrentSize);
  
      data->AudioBufferSizeCurrent = nCurrentSize; 
  
  
  
  }

  Would love to hear any ideas - I’ve been trying to get this working for 3 days now :(

• ffmpeg video streaming deep understanding

  14 novembre 2012, par Stefan Alexandru

  The main question is if it is possible to somehow go around the frame index checking that ffmpeg does when writing the frame to a file.

  Now I will explain my exact problem so you can understand better what I need or maybe think of an alternative solution.

  Problem n0.1 : I am getting video stream from two independent cameras and for some reason I want to save it in the same video file. First the frames from the first camera and then the frames from the second. When writing the frames from the second camera av_write_frame would return the error code -22 and will fail to add the frame. That's because the writing context is expecting a frame index following the index of the previously written frame (the last frame from camera 1) but he receives a frame with the index 0, the first frame from the second camera.

  Problem no.2 : Consider the following problem independently to the first one.
  I am trying to save a video stream to a file but the frame rate is double the real speed. So because I couldn't find any working solution to speed down the frame rate i thought to write every frame twice in the video file. But it won't make any difference to the frame rate.
  I also tried a different approach on the frame rate problem but it also failed(question here).

  Any kind of working solution would be highly appreciated.

  Also it's important that I can't use console commands, I need C code, as I need to integrate those functionalities in an Android application that is automated.

• Trying to sync audio/visual using FFMpeg and openAL

  22 août 2013, par user1379811

  hI have been studying dranger ffmpeg tutorial which explains how to sync audio and visual once you have the frames displayed and audio playing which is where im at.

  Unfortunately, the tutorial is out of date (Stephen Dranger explaained that himself to me) and also uses sdl which im not doing - this is for Blackberry 10 application.

  I just cannot make the video frames display at the correct speed (they are just playing very fast) and I have been trying for over a week now - seriously !

  I have 3 threads happening - one to read from stream into audio and video queues and then 2 threads for audio and video.

  If somebody could explain whats happening after scanning my relevent code you would be a lifesaver.

  The delay (what I pass to usleep(testDelay) seems to be going up (incrementing) which doesn't seem right to me.

  count = 1;
  
      MyApp* inst = worker->app;//(VideoUploadFacebook*)arg;
  
      qDebug() << "\n start loadstream";
  
      w = new QWaitCondition();
  
      w2 = new QWaitCondition();
  
      context = avformat_alloc_context();
  
      inst->threadStarted = true;
  
      cout << "start of decoding thread";
  
      cout.flush();
  
  
  
  
  
      av_register_all();
  
      avcodec_register_all();
  
      avformat_network_init();
  
      av_log_set_callback(&log_callback);
  
      AVInputFormat   *pFormat;
  
      //const char      device[]     = "/dev/video0";
  
      const char      formatName[] = "mp4";
  
      cout << "2start of decoding thread";
  
      cout.flush();
  
  
  
  
  
  
  
      if (!(pFormat = av_find_input_format(formatName))) {
  
          printf("can't find input format %s\n", formatName);
  
          //return void*;
  
      }
  
      //open rtsp
  
      if(avformat_open_input(&context, inst->capturedUrl.data(), pFormat,NULL) != 0){
  
          // return ;
  
          cout << "error opening of decoding thread: " << inst->capturedUrl.data();
  
          cout.flush();
  
      }
  
  
  
      cout << "3start of decoding thread";
  
      cout.flush();
  
      // av_dump_format(context, 0, inst->capturedUrl.data(), 0);
  
      /*   if(avformat_find_stream_info(context,NULL) < 0){
  
          return EXIT_FAILURE;
  
      }
  
       */
  
      //search video stream
  
      for(int i =0;inb_streams;i++){
  
          if(context->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
  
              inst->video_stream_index = i;
  
      }
  
      cout << "3z start of decoding thread";
  
      cout.flush();
  
      AVFormatContext* oc = avformat_alloc_context();
  
      av_read_play(context);//play RTSP
  
      AVDictionary *optionsDict = NULL;
  
      ccontext = context->streams[inst->video_stream_index]->codec;
  
  
  
      inst->audioc = context->streams[1]->codec;
  
  
  
      cout << "4start of decoding thread";
  
      cout.flush();
  
      codec = avcodec_find_decoder(ccontext->codec_id);
  
      ccontext->pix_fmt = PIX_FMT_YUV420P;
  
  
  
      AVCodec* audio_codec = avcodec_find_decoder(inst->audioc->codec_id);
  
      inst->packet = new AVPacket();
  
      if (!audio_codec) {
  
          cout << "audio codec not found\n"; //fflush( stdout );
  
          exit(1);
  
      }
  
  
  
      if (avcodec_open2(inst->audioc, audio_codec, NULL) < 0) {
  
          cout << "could not open codec\n"; //fflush( stdout );
  
          exit(1);
  
      }
  
  
  
      if (avcodec_open2(ccontext, codec, &optionsDict) < 0) exit(1);
  
  
  
      cout << "5start of decoding thread";
  
      cout.flush();
  
      inst->pic = avcodec_alloc_frame();
  
  
  
      av_init_packet(inst->packet);
  
  
  
      while(av_read_frame(context,inst->packet) >= 0 && &inst->keepGoing)
  
      {
  
  
  
          if(inst->packet->stream_index == 0){//packet is video
  
  
  
              int check = 0;
  
  
  
  
  
  
  
              // av_init_packet(inst->packet);
  
              int result = avcodec_decode_video2(ccontext, inst->pic, &check, inst->packet);
  
  
  
              if(check)
  
                  break;
  
          }
  
      }
  
  
  
  
  
  
  
      inst->originalVideoWidth = inst->pic->width;
  
      inst->originalVideoHeight = inst->pic->height;
  
      float aspect = (float)inst->originalVideoHeight / (float)inst->originalVideoWidth;
  
      inst->newVideoWidth = inst->originalVideoWidth;
  
      int newHeight = (int)(inst->newVideoWidth * aspect);
  
      inst->newVideoHeight = newHeight;//(int)inst->originalVideoHeight / inst->originalVideoWidth * inst->newVideoWidth;// = new height
  
      int size = avpicture_get_size(PIX_FMT_YUV420P, inst->originalVideoWidth, inst->originalVideoHeight);
  
      uint8_t* picture_buf = (uint8_t*)(av_malloc(size));
  
      avpicture_fill((AVPicture *) inst->pic, picture_buf, PIX_FMT_YUV420P, inst->originalVideoWidth, inst->originalVideoHeight);
  
  
  
      picrgb = avcodec_alloc_frame();
  
      int size2 = avpicture_get_size(PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight);
  
      uint8_t* picture_buf2 = (uint8_t*)(av_malloc(size2));
  
      avpicture_fill((AVPicture *) picrgb, picture_buf2, PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight);
  
  
  
  
  
  
  
      if(ccontext->pix_fmt != PIX_FMT_YUV420P)
  
      {
  
          std::cout << "fmt != 420!!!: " << ccontext->pix_fmt << std::endl;//
  
          // return (EXIT_SUCCESS);//-1;
  
  
  
      }
  
  
  
  
  
      if (inst->createForeignWindow(inst->myForeignWindow->windowGroup(),
  
              "HelloForeignWindowAppIDqq", 0,
  
              0, inst->newVideoWidth,
  
              inst->newVideoHeight)) {
  
  
  
      } else {
  
          qDebug() << "The ForeginWindow was not properly initialized";
  
      }
  
  
  
  
  
  
  
  
  
      inst->keepGoing = true;
  
  
  
      inst->img_convert_ctx = sws_getContext(inst->originalVideoWidth, inst->originalVideoHeight, PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight,
  
              PIX_FMT_YUV420P, SWS_BILINEAR, NULL, NULL, NULL);
  
  
  
      is = (VideoState*)av_mallocz(sizeof(VideoState));
  
      if (!is)
  
          return NULL;
  
  
  
      is->audioStream = 1;
  
      is->audio_st = context->streams[1];
  
      is->audio_buf_size = 0;
  
      is->audio_buf_index = 0;
  
      is->videoStream = 0;
  
      is->video_st = context->streams[0];
  
  
  
      is->frame_timer = (double)av_gettime() / 1000000.0;
  
      is->frame_last_delay = 40e-3;
  
  
  
      is->av_sync_type = DEFAULT_AV_SYNC_TYPE;
  
      //av_strlcpy(is->filename, filename, sizeof(is->filename));
  
      is->iformat = pFormat;
  
      is->ytop    = 0;
  
      is->xleft   = 0;
  
  
  
      /* start video display */
  
      is->pictq_mutex = new QMutex();
  
      is->pictq_cond  = new QWaitCondition();
  
  
  
      is->subpq_mutex = new QMutex();
  
      is->subpq_cond  = new QWaitCondition();
  
  
  
      is->video_current_pts_time = av_gettime();
  
  
  
  
  
      packet_queue_init(&audioq);
  
  
  
      packet_queue_init(&videoq);
  
      is->audioq = audioq;
  
      is->videoq = videoq;
  
      AVPacket* packet2  = new AVPacket();
  
  
  
      ccontext->get_buffer = our_get_buffer;
  
      ccontext->release_buffer = our_release_buffer;
  
  
  
  
  
      av_init_packet(packet2);
  
      while(inst->keepGoing)
  
      {
  
  
  
  
  
          if(av_read_frame(context,packet2) < 0 && keepGoing)
  
          {
  
              printf("bufferframe Could not read a frame from stream.\n");
  
              fflush( stdout );
  
  
  
  
  
          }else {
  
  
  
  
  
  
  
              if(packet2->stream_index == 0) {
  
                  packet_queue_put(&videoq, packet2);
  
              } else if(packet2->stream_index == 1) {
  
                  packet_queue_put(&audioq, packet2);
  
              } else {
  
                  av_free_packet(packet2);
  
              }
  
  
  
  
  
              if(!videoThreadStarted)
  
              {
  
                  videoThreadStarted = true;
  
                  QThread* thread = new QThread;
  
                  videoThread = new VideoStreamWorker(this);
  
  
  
                  // Give QThread ownership of Worker Object
  
                  videoThread->moveToThread(thread);
  
                  connect(videoThread, SIGNAL(error(QString)), this, SLOT(errorHandler(QString)));
  
                  QObject::connect(videoThread, SIGNAL(refreshNeeded()), this, SLOT(refreshNeededSlot()));
  
                  connect(thread, SIGNAL(started()), videoThread, SLOT(doWork()));
  
                  connect(videoThread, SIGNAL(finished()), thread, SLOT(quit()));
  
                  connect(videoThread, SIGNAL(finished()), videoThread, SLOT(deleteLater()));
  
                  connect(thread, SIGNAL(finished()), thread, SLOT(deleteLater()));
  
  
  
                  thread->start();
  
              }
  
  
  
              if(!audioThreadStarted)
  
              {
  
                  audioThreadStarted = true;
  
                  QThread* thread = new QThread;
  
                  AudioStreamWorker* videoThread = new AudioStreamWorker(this);
  
  
  
                  // Give QThread ownership of Worker Object
  
                  videoThread->moveToThread(thread);
  
  
  
                  // Connect videoThread error signal to this errorHandler SLOT.
  
                  connect(videoThread, SIGNAL(error(QString)), this, SLOT(errorHandler(QString)));
  
  
  
                  // Connects the thread’s started() signal to the process() slot in the videoThread, causing it to start.
  
                  connect(thread, SIGNAL(started()), videoThread, SLOT(doWork()));
  
                  connect(videoThread, SIGNAL(finished()), thread, SLOT(quit()));
  
                  connect(videoThread, SIGNAL(finished()), videoThread, SLOT(deleteLater()));
  
  
  
                  // Make sure the thread object is deleted after execution has finished.
  
                  connect(thread, SIGNAL(finished()), thread, SLOT(deleteLater()));
  
  
  
                  thread->start();
  
              }
  
  
  
          }
  
  
  
      } //finished main loop
  
  
  
      int MyApp::video_thread() {
  
      //VideoState *is = (VideoState *)arg;
  
      AVPacket pkt1, *packet = &pkt1;
  
      int len1, frameFinished;
  
  
  
      double pts;
  
      pic = avcodec_alloc_frame();
  
  
  
      for(;;) {
  
          if(packet_queue_get(&videoq, packet, 1) < 0) {
  
              // means we quit getting packets
  
              break;
  
          }
  
  
  
          pts = 0;
  
  
  
          global_video_pkt_pts2 = packet->pts;
  
          // Decode video frame
  
          len1 =  avcodec_decode_video2(ccontext, pic, &frameFinished, packet);
  
          if(packet->dts == AV_NOPTS_VALUE
  
                  && pic->opaque && *(uint64_t*)pic->opaque != AV_NOPTS_VALUE) {
  
              pts = *(uint64_t *)pic->opaque;
  
          } else if(packet->dts != AV_NOPTS_VALUE) {
  
              pts = packet->dts;
  
          } else {
  
              pts = 0;
  
          }
  
          pts *= av_q2d(is->video_st->time_base);
  
          // Did we get a video frame?
  
  
  
                  if(frameFinished) {
  
                      pts = synchronize_video(is, pic, pts);
  
                      actualPts = pts;
  
                      refreshSlot();
  
                  }
  
                  av_free_packet(packet);
  
      }
  
      av_free(pic);
  
      return 0;
  
  }
  
  
  
  
  
  int MyApp::audio_thread() {
  
      //VideoState *is = (VideoState *)arg;
  
      AVPacket pkt1, *packet = &pkt1;
  
      int len1, frameFinished;
  
      ALuint source;
  
      ALenum format = 0;
  
      //   ALuint frequency;
  
      ALenum alError;
  
      ALint val2;
  
      ALuint buffers[NUM_BUFFERS];
  
      int dataSize;
  
  
  
  
  
      ALCcontext *aContext;
  
      ALCdevice *device;
  
      if (!alutInit(NULL, NULL)) {
  
          // printf(stderr, "init alut error\n");
  
      }
  
      device = alcOpenDevice(NULL);
  
      if (device == NULL) {
  
          // printf(stderr, "device error\n");
  
      }
  
  
  
      //Create a context
  
      aContext = alcCreateContext(device, NULL);
  
      alcMakeContextCurrent(aContext);
  
      if(!(aContext)) {
  
          printf("Could not create the OpenAL context!\n");
  
          return 0;
  
      }
  
  
  
      alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f);
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
      //ALenum alError;
  
      if(alGetError() != AL_NO_ERROR) {
  
          cout << "could not create buffers";
  
          cout.flush();
  
          fflush( stdout );
  
          return 0;
  
      }
  
      alGenBuffers(NUM_BUFFERS, buffers);
  
      alGenSources(1, &source);
  
      if(alGetError() != AL_NO_ERROR) {
  
          cout << "after Could not create buffers or the source.\n";
  
          cout.flush(  );
  
          return 0;
  
      }
  
  
  
      int i;
  
      int indexOfPacket;
  
      double pts;
  
      //double pts;
  
      int n;
  
  
  
  
  
      for(i = 0; i < NUM_BUFFERS; i++)
  
      {
  
          if(packet_queue_get(&audioq, packet, 1) < 0) {
  
              // means we quit getting packets
  
              break;
  
          }
  
          cout << "streamindex=audio \n";
  
          cout.flush(  );
  
          //printf("before decode  audio\n");
  
          //fflush( stdout );
  
          // AVPacket *packet = new AVPacket();//malloc(sizeof(AVPacket*));
  
          AVFrame *decodedFrame = NULL;
  
          int gotFrame = 0;
  
          // AVFrame* decodedFrame;
  
  
  
          if(!decodedFrame) {
  
              if(!(decodedFrame = avcodec_alloc_frame())) {
  
                  cout << "Run out of memory, stop the streaming...\n";
  
                  fflush( stdout );
  
                  cout.flush();
  
  
  
  
  
                  return -2;
  
              }
  
          } else {
  
              avcodec_get_frame_defaults(decodedFrame);
  
          }
  
  
  
          int  len = avcodec_decode_audio4(audioc, decodedFrame, &gotFrame, packet);
  
          if(len < 0) {
  
              cout << "Error while decoding.\n";
  
              cout.flush(  );
  
  
  
              return -3;
  
          }
  
          if(len < 0) {
  
              /* if error, skip frame */
  
              is->audio_pkt_size = 0;
  
              //break;
  
          }
  
          is->audio_pkt_data += len;
  
          is->audio_pkt_size -= len;
  
  
  
          pts = is->audio_clock;
  
          // *pts_ptr = pts;
  
          n = 2 * is->audio_st->codec->channels;
  
          is->audio_clock += (double)packet->size/
  
                  (double)(n * is->audio_st->codec->sample_rate);
  
          if(gotFrame) {
  
              cout << "got audio frame.\n";
  
              cout.flush(  );
  
              // We have a buffer ready, send it
  
              dataSize = av_samples_get_buffer_size(NULL, audioc->channels,
  
                      decodedFrame->nb_samples, audioc->sample_fmt, 1);
  
  
  
              if(!format) {
  
                  if(audioc->sample_fmt == AV_SAMPLE_FMT_U8 ||
  
                          audioc->sample_fmt == AV_SAMPLE_FMT_U8P) {
  
                      if(audioc->channels == 1) {
  
                          format = AL_FORMAT_MONO8;
  
                      } else if(audioc->channels == 2) {
  
                          format = AL_FORMAT_STEREO8;
  
                      }
  
                  } else if(audioc->sample_fmt == AV_SAMPLE_FMT_S16 ||
  
                          audioc->sample_fmt == AV_SAMPLE_FMT_S16P) {
  
                      if(audioc->channels == 1) {
  
                          format = AL_FORMAT_MONO16;
  
                      } else if(audioc->channels == 2) {
  
                          format = AL_FORMAT_STEREO16;
  
                      }
  
                  }
  
  
  
                  if(!format) {
  
                      cout << "OpenAL can't open this format of sound.\n";
  
                      cout.flush(  );
  
  
  
                      return -4;
  
                  }
  
              }
  
              printf("albufferdata audio b4.\n");
  
              fflush( stdout );
  
              alBufferData(buffers[i], format, *decodedFrame->data, dataSize, decodedFrame->sample_rate);
  
              cout << "after albufferdata all buffers \n";
  
              cout.flush(  );
  
              av_free_packet(packet);
  
              //=av_free(packet);
  
              av_free(decodedFrame);
  
  
  
              if((alError = alGetError()) != AL_NO_ERROR) {
  
                  printf("Error while buffering.\n");
  
  
  
                  printAlError(alError);
  
                  return -6;
  
              }
  
          }
  
      }
  
  
  
  
  
      cout << "before quoe buffers \n";
  
      cout.flush();
  
      alSourceQueueBuffers(source, NUM_BUFFERS, buffers);
  
      cout << "before play.\n";
  
      cout.flush();
  
      alSourcePlay(source);
  
      cout << "after play.\n";
  
      cout.flush();
  
      if((alError = alGetError()) != AL_NO_ERROR) {
  
          cout << "error strating stream.\n";
  
          cout.flush();
  
          printAlError(alError);
  
          return 0;
  
      }
  
  
  
  
  
      // AVPacket *pkt = &is->audio_pkt;
  
  
  
      while(keepGoing)
  
      {
  
          while(packet_queue_get(&audioq, packet, 1)  >= 0) {
  
              // means we quit getting packets
  
  
  
              do {
  
                  alGetSourcei(source, AL_BUFFERS_PROCESSED, &val2);
  
                  usleep(SLEEP_BUFFERING);
  
              } while(val2 <= 0);
  
              if(alGetError() != AL_NO_ERROR)
  
              {
  
                  fprintf(stderr, "Error gettingsource :(\n");
  
                  return 1;
  
              }
  
  
  
              while(val2--)
  
              {
  
  
  
  
  
  
  
                  ALuint buffer;
  
                  alSourceUnqueueBuffers(source, 1, &buffer);
  
                  if(alGetError() != AL_NO_ERROR)
  
                  {
  
                      fprintf(stderr, "Error unqueue buffers :(\n");
  
                      //  return 1;
  
                  }
  
                  AVFrame *decodedFrame = NULL;
  
                  int gotFrame = 0;
  
                  // AVFrame* decodedFrame;
  
  
  
                  if(!decodedFrame) {
  
                      if(!(decodedFrame = avcodec_alloc_frame())) {
  
                          cout << "Run out of memory, stop the streaming...\n";
  
                          //fflush( stdout );
  
                          cout.flush();
  
  
  
  
  
                          return -2;
  
                      }
  
                  } else {
  
                      avcodec_get_frame_defaults(decodedFrame);
  
                  }
  
  
  
                  int  len = avcodec_decode_audio4(audioc, decodedFrame, &gotFrame, packet);
  
                  if(len < 0) {
  
                      cout << "Error while decoding.\n";
  
                      cout.flush(  );
  
                      is->audio_pkt_size = 0;
  
                      return -3;
  
                  }
  
  
  
                  is->audio_pkt_data += len;
  
                  is->audio_pkt_size -= len;
  
                  if(packet->size <= 0) {
  
                      /* No data yet, get more frames */
  
                      //continue;
  
                  }
  
  
  
  
  
                  if(gotFrame) {
  
                      pts = is->audio_clock;
  
                      len = synchronize_audio(is, (int16_t *)is->audio_buf,
  
                              packet->size, pts);
  
                      is->audio_buf_size = packet->size;
  
                      pts = is->audio_clock;
  
                      // *pts_ptr = pts;
  
                      n = 2 * is->audio_st->codec->channels;
  
                      is->audio_clock += (double)packet->size /
  
                              (double)(n * is->audio_st->codec->sample_rate);
  
                      if(packet->pts != AV_NOPTS_VALUE) {
  
                          is->audio_clock = av_q2d(is->audio_st->time_base)*packet->pts;
  
                      }
  
                      len = av_samples_get_buffer_size(NULL, audioc->channels,
  
                              decodedFrame->nb_samples, audioc->sample_fmt, 1);
  
                      alBufferData(buffer, format, *decodedFrame->data, len, decodedFrame->sample_rate);
  
                      if(alGetError() != AL_NO_ERROR)
  
                      {
  
                          fprintf(stderr, "Error buffering :(\n");
  
                          return 1;
  
                      }
  
                      alSourceQueueBuffers(source, 1, &buffer);
  
                      if(alGetError() != AL_NO_ERROR)
  
                      {
  
                          fprintf(stderr, "Error queueing buffers :(\n");
  
                          return 1;
  
                      }
  
                  }
  
  
  
  
  
  
  
  
  
  
  
              }
  
  
  
              alGetSourcei(source, AL_SOURCE_STATE, &val2);
  
              if(val2 != AL_PLAYING)
  
                  alSourcePlay(source);
  
  
  
          }
  
  
  
  
  
          //pic = avcodec_alloc_frame();
  
      }
  
      qDebug() << "end audiothread";
  
      return 1;
  
  }
  
  
  
  void MyApp::refreshSlot()
  
  {
  
  
  
  
  
      if(true)
  
      {
  
  
  
          printf("got frame %d, %d\n", pic->width, ccontext->width);
  
          fflush( stdout );
  
  
  
          sws_scale(img_convert_ctx, (const uint8_t **)pic->data, pic->linesize,
  
                  0, originalVideoHeight, &picrgb->data[0], &picrgb->linesize[0]);
  
  
  
          printf("rescaled frame %d, %d\n", newVideoWidth, newVideoHeight);
  
          fflush( stdout );
  
          //av_free_packet(packet);
  
          //av_init_packet(packet);
  
  
  
          qDebug() << "waking audio as video finished";
  
          ////mutex.unlock();
  
          //mutex2.lock();
  
          doingVideoFrame = false;
  
          //doingAudioFrame = false;
  
          ////mutex2.unlock();
  
  
  
  
  
          //mutex2.unlock();
  
          //w2->wakeAll();
  
          //w->wakeAll();
  
          qDebug() << "now woke audio";
  
  
  
          //pic = picrgb;
  
          uint8_t *srcy = picrgb->data[0];
  
          uint8_t *srcu = picrgb->data[1];
  
          uint8_t *srcv = picrgb->data[2];
  
          printf("got src yuv frame %d\n", &srcy);
  
          fflush( stdout );
  
          unsigned char *ptr = NULL;
  
          screen_get_buffer_property_pv(mScreenPixelBuffer, SCREEN_PROPERTY_POINTER, (void**) &ptr);
  
          unsigned char *y = ptr;
  
          unsigned char *u = y + (newVideoHeight * mStride) ;
  
          unsigned char *v = u + (newVideoHeight * mStride) / 4;
  
          int i = 0;
  
          printf("got buffer  picrgbwidth= %d \n", newVideoWidth);
  
          fflush( stdout );
  
          for ( i = 0; i < newVideoHeight; i++)
  
          {
  
              int doff = i * mStride;
  
              int soff = i * picrgb->linesize[0];
  
              memcpy(&y[doff], &srcy[soff], newVideoWidth);
  
          }
  
  
  
          for ( i = 0; i < newVideoHeight / 2; i++)
  
          {
  
              int doff = i * mStride / 2;
  
              int soff = i * picrgb->linesize[1];
  
              memcpy(&u[doff], &srcu[soff], newVideoWidth / 2);
  
          }
  
  
  
          for ( i = 0; i < newVideoHeight / 2; i++)
  
          {
  
              int doff = i * mStride / 2;
  
              int soff = i * picrgb->linesize[2];
  
              memcpy(&v[doff], &srcv[soff], newVideoWidth / 2);
  
          }
  
          printf("before posttoscreen \n");
  
          fflush( stdout );
  
  
  
          video_refresh_timer();
  
          qDebug() << "end refreshslot";
  
  
  
      }
  
      else
  
      {
  
  
  
      }
  
  
  
  
  
  
  
  
  
  
  
  }
  
  
  
  void  MyApp::refreshNeededSlot2()
  
      {
  
          printf("blitting to buffer");
  
          fflush(stdout);
  
  
  
          screen_buffer_t screen_buffer;
  
          screen_get_window_property_pv(mScreenWindow, SCREEN_PROPERTY_RENDER_BUFFERS, (void**) &screen_buffer);
  
          int attribs[] = { SCREEN_BLIT_SOURCE_WIDTH, newVideoWidth, SCREEN_BLIT_SOURCE_HEIGHT, newVideoHeight, SCREEN_BLIT_END };
  
          int res2 = screen_blit(mScreenCtx, screen_buffer, mScreenPixelBuffer, attribs);
  
          printf("dirty rectangles");
  
          fflush(stdout);
  
          int dirty_rects[] = { 0, 0, newVideoWidth, newVideoHeight };
  
          screen_post_window(mScreenWindow, screen_buffer, 1, dirty_rects, 0);
  
          printf("done screneposdtwindow");
  
          fflush(stdout);
  
  
  
      }
  
  
  
  void MyApp::video_refresh_timer() {
  
      testDelay = 0;
  
      //  VideoState *is = ( VideoState* )userdata;
  
      VideoPicture *vp;
  
      //double pts = 0    ;
  
      double actual_delay, delay, sync_threshold, ref_clock, diff;
  
  
  
      if(is->video_st) {
  
          if(false)////is->pictq_size == 0)
  
          {
  
              testDelay = 1;
  
              schedule_refresh(is, 1);
  
          } else {
  
              // vp = &is->pictq[is->pictq_rindex];
  
  
  
              delay = actualPts - is->frame_last_pts; /* the pts from last time */
  
              if(delay <= 0 || delay >= 1.0) {
  
                  /* if incorrect delay, use previous one */
  
                  delay = is->frame_last_delay;
  
              }
  
              /* save for next time */
  
              is->frame_last_delay = delay;
  
              is->frame_last_pts = actualPts;
  
  
  
              is->video_current_pts = actualPts;
  
              is->video_current_pts_time = av_gettime();
  
              /* update delay to sync to audio */
  
              ref_clock = get_audio_clock(is);
  
              diff = actualPts - ref_clock;
  
  
  
              /* Skip or repeat the frame. Take delay into account
  
       FFPlay still doesn't "know if this is the best guess." */
  
              sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay : AV_SYNC_THRESHOLD;
  
              if(fabs(diff) < AV_NOSYNC_THRESHOLD) {
  
                  if(diff <= -sync_threshold) {
  
                      delay = 0;
  
                  } else if(diff >= sync_threshold) {
  
                      delay = 2 * delay;
  
                  }
  
              }
  
              is->frame_timer += delay;
  
              /* computer the REAL delay */
  
              actual_delay = is->frame_timer - (av_gettime() / 1000000.0);
  
              if(actual_delay < 0.010) {
  
                  /* Really it should skip the picture instead */
  
                  actual_delay = 0.010;
  
              }
  
              testDelay = (int)(actual_delay * 1000 + 0.5);
  
              schedule_refresh(is, (int)(actual_delay * 1000 + 0.5));
  
              /* show the picture! */
  
              //video_display(is);
  
  
  
  
  
              // SDL_CondSignal(is->pictq_cond);
  
              // SDL_UnlockMutex(is->pictq_mutex);
  
          }
  
      } else {
  
          testDelay = 100;
  
          schedule_refresh(is, 100);
  
  
  
      }
  
  }
  
  
  
  void MyApp::schedule_refresh(VideoState *is, int delay) {
  
      qDebug() << "start schedule refresh timer" << delay;
  
      typeOfEvent = FF_REFRESH_EVENT2;
  
      w->wakeAll();
  
      //  SDL_AddTimer(delay,
  
  
  
  
  
  }

  I am currently waiting on data in a loop in the following way

  QMutex mutex;
  
      mutex.lock();
  
      while(keepGoing)
  
      {
  
  
  
  
  
  
  
          qDebug() << "MAINTHREAD" << testDelay;
  
  
  
  
  
          w->wait(&mutex);
  
          mutex.unlock();
  
          qDebug() << "MAINTHREAD past wait";
  
  
  
          if(!keepGoing)
  
          {
  
              break;
  
          }
  
          if(testDelay > 0 && typeOfEvent == FF_REFRESH_EVENT2)
  
          {
  
              usleep(testDelay);
  
              refreshNeededSlot2();
  
          }
  
          else   if(testDelay > 0 && typeOfEvent == FF_QUIT_EVENT2)
  
          {
  
              keepGoing = false;
  
              exit(0);
  
              break;
  
              // usleep(testDelay);
  
              // refreshNeededSlot2();
  
          }
  
          qDebug() << "MAINTHREADend";
  
          mutex.lock();
  
  
  
      }
  
      mutex.unlock();

  Please let me know if I need to provide any more relevent code. I'm sorry my code is untidy - I still learning c++ and have been modifying this code for over a week now as previously mentioned.

  Just added a sample of output I'm seeing from print outs I do to console - I can't get my head around it (it's almost too complicated for my level of expertise) but when you see the frames being played and audio playing it's very difficult to give up especially when it took me a couple of weeks to get to this stage.

  Please someone give me a hand if they spot the problem.

  MAINTHREAD past wait
  pts after syncvideo= 1073394046
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.66833
  frame lastpts = 1.63497
  start schedule refresh timer need to delay for 123

  pts after syncvideo= 1073429033
  got frame 640, 640
  MAINTHREAD loop delay before refresh = 123
  start video_refresh_timer
  actualpts = 1.7017
  frame lastpts = 1.66833
  start schedule refresh timer need to delay for 115

  MAINTHREAD past wait
  pts after syncvideo= 1073464021
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.73507
  frame lastpts = 1.7017
  start schedule refresh timer need to delay for 140

  MAINTHREAD loop delay before refresh = 140
  pts after syncvideo= 1073499008
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.76843
  frame lastpts = 1.73507
  start schedule refresh timer need to delay for 163

  MAINTHREAD past wait
  pts after syncvideo= 1073533996
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.8018
  frame lastpts = 1.76843
  start schedule refresh timer need to delay for 188

  MAINTHREAD loop delay before refresh = 188
  pts after syncvideo= 1073568983
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.83517
  frame lastpts = 1.8018
  start schedule refresh timer need to delay for 246

  MAINTHREAD past wait
  pts after syncvideo= 1073603971
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.86853
  frame lastpts = 1.83517
  start schedule refresh timer need to delay for 299

  MAINTHREAD loop delay before refresh = 299
  pts after syncvideo= 1073638958
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.9019
  frame lastpts = 1.86853
  start schedule refresh timer need to delay for 358

  MAINTHREAD past wait
  pts after syncvideo= 1073673946
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.93527
  frame lastpts = 1.9019
  start schedule refresh timer need to delay for 416

  MAINTHREAD loop delay before refresh = 416
  pts after syncvideo= 1073708933
  got frame 640, 640
  start video_refresh_timer
  actualpts = 1.96863
  frame lastpts = 1.93527
  start schedule refresh timer need to delay for 474

  MAINTHREAD past wait
  pts after syncvideo= 1073742872
  got frame 640, 640
  MAINTHREAD loop delay before refresh = 474
  start video_refresh_timer
  actualpts = 2.002
  frame lastpts = 1.96863
  start schedule refresh timer need to delay for 518

  MAINTHREAD past wait
  pts after syncvideo= 1073760366
  got frame 640, 640
  start video_refresh_timer
  actualpts = 2.03537
  frame lastpts = 2.002
  start schedule refresh timer need to delay for 575