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Médias (3)

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Sur d’autres sites (4608)

  • 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 !

  • 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    Unfortunately, the tutorial is out of date (Stephen Dranger explaained that himself to me) and also uses sdl which im not doing - this is for Blackberry 10 application.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() &lt;&lt; "\n start loadstream";
       w = new QWaitCondition();
       w2 = new QWaitCondition();
       context = avformat_alloc_context();
       inst->threadStarted = true;
       cout &lt;&lt; "start of decoding thread";
       cout.flush();


       av_register_all();
       avcodec_register_all();
       avformat_network_init();
       av_log_set_callback(&amp;log_callback);
       AVInputFormat   *pFormat;
       //const char      device[]     = "/dev/video0";
       const char      formatName[] = "mp4";
       cout &lt;&lt; "2start of decoding thread";
       cout.flush();



       if (!(pFormat = av_find_input_format(formatName))) {
           printf("can&#39;t find input format %s\n", formatName);
           //return void*;
       }
       //open rtsp
       if(avformat_open_input(&amp;context, inst->capturedUrl.data(), pFormat,NULL) != 0){
           // return ;
           cout &lt;&lt; "error opening of decoding thread: " &lt;&lt; inst->capturedUrl.data();
           cout.flush();
       }

       cout &lt;&lt; "3start of decoding thread";
       cout.flush();
       // av_dump_format(context, 0, inst->capturedUrl.data(), 0);
       /*   if(avformat_find_stream_info(context,NULL) &lt; 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 &lt;&lt; "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 &lt;&lt; "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 &lt;&lt; "audio codec not found\n"; //fflush( stdout );
           exit(1);
       }

       if (avcodec_open2(inst->audioc, audio_codec, NULL) &lt; 0) {
           cout &lt;&lt; "could not open codec\n"; //fflush( stdout );
           exit(1);
       }

       if (avcodec_open2(ccontext, codec, &amp;optionsDict) &lt; 0) exit(1);

       cout &lt;&lt; "5start of decoding thread";
       cout.flush();
       inst->pic = avcodec_alloc_frame();

       av_init_packet(inst->packet);

       while(av_read_frame(context,inst->packet) >= 0 &amp;&amp; &amp;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, &amp;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 &lt;&lt; "fmt != 420!!!: " &lt;&lt; ccontext->pix_fmt &lt;&lt; std::endl;//
           // return (EXIT_SUCCESS);//-1;

       }


       if (inst->createForeignWindow(inst->myForeignWindow->windowGroup(),
               "HelloForeignWindowAppIDqq", 0,
               0, inst->newVideoWidth,
               inst->newVideoHeight)) {

       } else {
           qDebug() &lt;&lt; "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(&amp;audioq);

       packet_queue_init(&amp;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) &lt; 0 &amp;&amp; keepGoing)
           {
               printf("bufferframe Could not read a frame from stream.\n");
               fflush( stdout );


           }else {



               if(packet2->stream_index == 0) {
                   packet_queue_put(&amp;videoq, packet2);
               } else if(packet2->stream_index == 1) {
                   packet_queue_put(&amp;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 = &amp;pkt1;
       int len1, frameFinished;

       double pts;
       pic = avcodec_alloc_frame();

       for(;;) {
           if(packet_queue_get(&amp;videoq, packet, 1) &lt; 0) {
               // means we quit getting packets
               break;
           }

           pts = 0;

           global_video_pkt_pts2 = packet->pts;
           // Decode video frame
           len1 =  avcodec_decode_video2(ccontext, pic, &amp;frameFinished, packet);
           if(packet->dts == AV_NOPTS_VALUE
                   &amp;&amp; pic->opaque &amp;&amp; *(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 = &amp;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 &lt;&lt; "could not create buffers";
           cout.flush();
           fflush( stdout );
           return 0;
       }
       alGenBuffers(NUM_BUFFERS, buffers);
       alGenSources(1, &amp;source);
       if(alGetError() != AL_NO_ERROR) {
           cout &lt;&lt; "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 &lt; NUM_BUFFERS; i++)
       {
           if(packet_queue_get(&amp;audioq, packet, 1) &lt; 0) {
               // means we quit getting packets
               break;
           }
           cout &lt;&lt; "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 &lt;&lt; "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, &amp;gotFrame, packet);
           if(len &lt; 0) {
               cout &lt;&lt; "Error while decoding.\n";
               cout.flush(  );

               return -3;
           }
           if(len &lt; 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 &lt;&lt; "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 &lt;&lt; "OpenAL can&#39;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 &lt;&lt; "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 &lt;&lt; "before quoe buffers \n";
       cout.flush();
       alSourceQueueBuffers(source, NUM_BUFFERS, buffers);
       cout &lt;&lt; "before play.\n";
       cout.flush();
       alSourcePlay(source);
       cout &lt;&lt; "after play.\n";
       cout.flush();
       if((alError = alGetError()) != AL_NO_ERROR) {
           cout &lt;&lt; "error strating stream.\n";
           cout.flush();
           printAlError(alError);
           return 0;
       }


       // AVPacket *pkt = &amp;is->audio_pkt;

       while(keepGoing)
       {
           while(packet_queue_get(&amp;audioq, packet, 1)  >= 0) {
               // means we quit getting packets

               do {
                   alGetSourcei(source, AL_BUFFERS_PROCESSED, &amp;val2);
                   usleep(SLEEP_BUFFERING);
               } while(val2 &lt;= 0);
               if(alGetError() != AL_NO_ERROR)
               {
                   fprintf(stderr, "Error gettingsource :(\n");
                   return 1;
               }

               while(val2--)
               {



                   ALuint buffer;
                   alSourceUnqueueBuffers(source, 1, &amp;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 &lt;&lt; "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, &amp;gotFrame, packet);
                   if(len &lt; 0) {
                       cout &lt;&lt; "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 &lt;= 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, &amp;buffer);
                       if(alGetError() != AL_NO_ERROR)
                       {
                           fprintf(stderr, "Error queueing buffers :(\n");
                           return 1;
                       }
                   }





               }

               alGetSourcei(source, AL_SOURCE_STATE, &amp;val2);
               if(val2 != AL_PLAYING)
                   alSourcePlay(source);

           }


           //pic = avcodec_alloc_frame();
       }
       qDebug() &lt;&lt; "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, &amp;picrgb->data[0], &amp;picrgb->linesize[0]);

           printf("rescaled frame %d, %d\n", newVideoWidth, newVideoHeight);
           fflush( stdout );
           //av_free_packet(packet);
           //av_init_packet(packet);

           qDebug() &lt;&lt; "waking audio as video finished";
           ////mutex.unlock();
           //mutex2.lock();
           doingVideoFrame = false;
           //doingAudioFrame = false;
           ////mutex2.unlock();


           //mutex2.unlock();
           //w2->wakeAll();
           //w->wakeAll();
           qDebug() &lt;&lt; "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", &amp;srcy);
           fflush( stdout );
           unsigned char *ptr = NULL;
           screen_get_buffer_property_pv(mScreenPixelBuffer, SCREEN_PROPERTY_POINTER, (void**) &amp;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 &lt; newVideoHeight; i++)
           {
               int doff = i * mStride;
               int soff = i * picrgb->linesize[0];
               memcpy(&amp;y[doff], &amp;srcy[soff], newVideoWidth);
           }

           for ( i = 0; i &lt; newVideoHeight / 2; i++)
           {
               int doff = i * mStride / 2;
               int soff = i * picrgb->linesize[1];
               memcpy(&amp;u[doff], &amp;srcu[soff], newVideoWidth / 2);
           }

           for ( i = 0; i &lt; newVideoHeight / 2; i++)
           {
               int doff = i * mStride / 2;
               int soff = i * picrgb->linesize[2];
               memcpy(&amp;v[doff], &amp;srcv[soff], newVideoWidth / 2);
           }
           printf("before posttoscreen \n");
           fflush( stdout );

           video_refresh_timer();
           qDebug() &lt;&lt; "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**) &amp;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 = &amp;is->pictq[is->pictq_rindex];

               delay = actualPts - is->frame_last_pts; /* the pts from last time */
               if(delay &lt;= 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&#39;t "know if this is the best guess." */
               sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay : AV_SYNC_THRESHOLD;
               if(fabs(diff) &lt; AV_NOSYNC_THRESHOLD) {
                   if(diff &lt;= -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 &lt; 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() &lt;&lt; "start schedule refresh timer" &lt;&lt; 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() &lt;&lt; "MAINTHREAD" &lt;&lt; testDelay;


           w->wait(&amp;mutex);
           mutex.unlock();
           qDebug() &lt;&lt; "MAINTHREAD past wait";

           if(!keepGoing)
           {
               break;
           }
           if(testDelay > 0 &amp;&amp; typeOfEvent == FF_REFRESH_EVENT2)
           {
               usleep(testDelay);
               refreshNeededSlot2();
           }
           else   if(testDelay > 0 &amp;&amp; typeOfEvent == FF_QUIT_EVENT2)
           {
               keepGoing = false;
               exit(0);
               break;
               // usleep(testDelay);
               // refreshNeededSlot2();
           }
           qDebug() &lt;&lt; "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

  • Why does FFMPEG always make large WebM files ?

    2 avril 2013, par Student of Hogwarts

    I'm trying to encode my movies into WebM :

    ffmpeg -i input.MOV -codec:v libvpx -quality good -cpu-used 0 -b:v 10k
    -qmin 10 -qmax 42 -maxrate 10k -bufsize 20k -threads 8 -vf scale=-1:1080
    -codec:a libvorbis -b:a 192k
    output.webm

    I want to encode at a couple of different bit rates (video and audio combined) :

    • 2192 kbps
    • 1692 kbps
    • 1000 kbps

    The problem is that no matter which bit rates I enter, I always get a file with a bit rate higher than 1900 kbps. (1914 kbps with the code example above.)

    What am I doing wrong ?