Recherche avancée

Sur d’autres sites (5415)

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  Unfortunately, the tutorial is out of date (Stephen Dranger explaained that himself to me) and also uses sdl which im not doing - this is for Blackberry 10 application.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

• split video (avi/h264) on keyframe

  30 novembre 2012, par m.sr

  Hallo.

  I have a big video file. ffmpeg, tcprobe and other tool say, it is an h264-stream in an AVI-container.

  Now i'd like to cut out small chunks form the video.

  1. Problem : The index of the video seam corrupted/destroyed. I kind of fixed this via mplayer -forceidx -saveidx <indexfile> <bigvideofile></bigvideofile></indexfile>. The Problem here is, that I'm now stuck with mplayer/mencoder which can use this index file via -loadidx <indexfile></indexfile>. I have tried correcting the index like described in man aviindex (mplayer -frames 0 -saveidx mpidx broken.avi ; aviindex -i mpidx -o tcindex ; avimerge -x tcindex -i broken.avi -o fixed.avi), but this didn't fix my video - meaning that most tools i've tested couldn't search in the video file.

  2. Problem : I cut out parts of the video via following command : mencoder -loadidx in.idx -ss 8578 -endpos 20 -oac faac -ovc x264 -sws 9 -lavfopts format=mp4 -x264encopts <lotsofopts> -of lavf -vf scale=800:-10,harddup in.avi -o out.mp4</lotsofopts>. Now here the problem is, that some videos are corrupted at the beginning. I think this is because the fact, that i do not necessarily cut at keyframe.

  Questions :

  1. What is the best way to fix the index of an avi "inline" so that every tool can again work as expected with it ?

  2. How can i split at the keyframes ? Is there an mencoder-option for this ?

  3. Are Keyframes coming in a frequency ? How to find out this frequency ? (So with a bit of math it should be possible to calculate the next keyframe and cut there)

  4. Is ther perhaps some completely other way to split this movie ? Doing it by hand is no option, i've to cut out 1000+ chunks ...

  Thanks a lot !

• Transcoding a video from GoToMeeting WMV to MP4 using FFMPEG on Windows locks up dos window [migrated]

  22 août 2011, par Ryan

  I've been tasked with converting some pre-recorded GoToMeeting videos from wmv to mp4 format. I've got the GoToMeeting transcoder app available from Citrix (g2mtranscoder.exe) that takes the source wmv and makes it so that ffmpeg can work with the wmv. The problem I have is when I then try and take that wmv and transcode it to mp4.

  I'm using the latest pre-compiled static version of ffmpeg for Windows from here http://ffmpeg.zeranoe.com/builds/

  Here is the command line I'm using to begin the conversion :

  "C:\ffmpeg\ffmpeg.exe" -i "999_1366_768_g2m_transcoded.wmv" -f "mp4" -b "1000k" -r "30" -ab "128k" -ar "22050" -s "1366x768" -strict "experimental" -y "999.mp4"

  Here is FFMPEG's output. The last line is where the dos window seems to stop executing/locks up (when I say lock up I mean just the dos window is locked up, not the entire machine) :

  ffmpeg version N-31932-g41bf67d, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 16 2011 18:54:12 with gcc 4.6.1
  
  configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil    51. 12. 0 / 51. 12. 0
  
    libavcodec   53. 10. 0 / 53. 10. 0
  
    libavformat  53.  7. 0 / 53.  7. 0
  
    libavdevice  53.  3. 0 / 53.  3. 0
  
    libavfilter   2. 31. 1 /  2. 31. 1
  
    libswscale    2.  0. 0 /  2.  0. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  [asf @ 01BBB600] max_analyze_duration 5000000 reached at 5194000
  
  
  
  Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 59.92 (719/12)
  
  Input #0, asf, from 'D:\Inetpub\g2m\Transcoding\999_1366_768_g2m_transcoded.wmv':
  
    Metadata:
  
      WMFSDKVersion   : 10.00.00.4007
  
      WMFSDKNeeded    : 0.0.0.0000
  
      IsVBR           : 1
  
      VBR Peak        : 4403
  
      Buffer Average  : 1470
  
      WM/ToolVersion  : 4.8 Build 723
  
      WM/ToolName     : GoToMeeting
  
      BitRateFrom the writer: 1553
  
      Audio samples   : 6221
  
      Video samples   : 3455
  
      recording time  : Mon, 22 Aug 2011 11:48:42 Eastern Daylight Time
  
    Duration: 00:38:30.71, start: 0.000000, bitrate: 160 kb/s
  
      Stream #0.0: Audio: wmav2, 44100 Hz, 1 channels, s16, 48 kb/s
  
      Stream #0.1: Video: wmv3 (Main), yuv420p, 1366x768, 2154 kb/s, 59.92 tbr, 1k tbn, 1k tbc
  
  [buffer @ 01BBC120] w:1366 h:768 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param:
  
  Output #0, mp4, to '\\nas4\FMS_APPLICATIONS\SVC\sk12\webinars\999.mp4':
  
    Metadata:
  
      WMFSDKVersion   : 10.00.00.4007
  
      WMFSDKNeeded    : 0.0.0.0000
  
      IsVBR           : 1
  
      VBR Peak        : 4403
  
      Buffer Average  : 1470
  
      WM/ToolVersion  : 4.8 Build 723
  
      WM/ToolName     : GoToMeeting
  
      BitRateFrom the writer: 1553
  
      Audio samples   : 6221
  
      Video samples   : 3455
  
      recording time  : Mon, 22 Aug 2011 11:48:42 Eastern Daylight Time
  
      encoder         : Lavf53.7.0
  
      Stream #0.0: Video: mpeg4, yuv420p, 1366x768, q=2-31, 1000 kb/s, 30 tbn, 30 tbc
  
      Stream #0.1: Audio: aac, 22050 Hz, 1 channels, s16, 128 kb/s
  
  Stream mapping:
  
    Stream #0.1 -> #0.0
  
    Stream #0.0 -> #0.1
  
  Press [q] to stop, [?] for help
  
  frame=  149 fps= 63 q=23.5 size=    1103kB time=00:00:04.96 bitrate=1818.6kbits/s dup=144 drop=0

  Does anyone have any idea why it's locking up the dos window and essentially failing to convert the file to mp4 ? I had a friend test it on a linux box and he was able to convert the video just fine.

  Thanks