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• How to capture a layered window with transparency background properly ? (using BitBlt)

  25 octobre 2016, par Mitra M

  I want to capture a WPF window (WPF layered window) with transparency background.

  To do that I tried FFmpeg, But :

  1 - If I set AllowTransparency (this is a property of WPF window) to false ,I can capture the window by gdigrab (this is an ffmpeg device), but output has black background.(I don’t want black background)

  2 - If I set AllowTransparency to true then gdigrab won’t work. (get black frame only)

  I have read David’s nice article, he has said :

  if you use BitBlt to do this, you could “or in” the CAPTUREBLT flag if
  you wanted to capture windows that are layered

  The gdigrab uses BitBlt, this is gdigrab.c code snippet :

  /* Blit screen grab */
  
      if (!BitBlt(dest_hdc, 0, 0,
  
                  clip_rect.right - clip_rect.left,
  
                  clip_rect.bottom - clip_rect.top,
  
                  source_hdc,
  
                  clip_rect.left, clip_rect.top, SRCCOPY | CAPTUREBLT)) {
  
          WIN32_API_ERROR("Failed to capture image");
  
          return AVERROR(EIO);
  
      }

  You can see the flags . (SRCCOPY | CAPTUREBLT).

  Please tell me :

  1- Why the gdigrab can not capture a WPF window properly ?

  2 - What changes in this code should be done to do this ?

  (Sorry for my English, I used translate.google)

  Thanks

• Live video encoding using...?

  27 décembre 2013, par Basic

  I'm attempting to write a fairly simplistic application that will stream video/audio from a webcam to someone else across the internet (ala Skypebut with more control).

  There seems to be very little useful/relevant information on the subjectand what I can find is largely outdated. From my research so far, x264 seems to be the way to go as it offers an ultrafast option which is designed for this situation

  I'm able to turn on the webcam and receive a stream of images. I can also listen on an audio device and get samples.

  Where I'm failing is encoding that information in such a way as to be able to stream with a minimum of latency (from what I've read, 200ms delay is the goal for no obvious lag, including network latency - so let's aim for 100-150ms)

  Things I've tried

  ffmpeg

  This seems to be the most widely used option for encoding. I've had two real issues using it. Firstly, even using x264 with no look-aheads and the bare minimum buffers for stability, the delay seems to be on the order of 700ms using image2pipe. Secondly, it requires ffmpeg to be installed - being able to do this without an external dependency would be nice.

  VLC

  As with ffmpeg this requires an external program which is a negative. Even worse, I can't seem to get a latency of under 2 seconds which seems to increase over time. I've also only been able to get VLC to capture the camera itself rather than take a stream of images which means I don't get a chance to pre-process them.

  DirectShow

  I've seen a number of sites recommending using the windows direct show encoders but I haven't been able to find one that works at anything like real time. In fact, the only one I've managed to get going reliably is a Windows Media codec that has a massive latency and fairly large size.

  Other considerations

  None of the above address the problem of adding an audio stream to the video. I'm not sure if I should attempt to encode them together or send a separate stream alongside the video.

  In short, I've been Googling for a week or so now and haven't found a decent way to do this. Can someone please point me at a decent example/guide ?

• Confusion about PTS in video files and media streams

  11 novembre 2014, par user2452253

  Is it possible that the PTS of a particular frame in a file is different with the PTS of the same frame in the same file while it is being streamed ?

  When I read a frame using av_read_frame I store the video stream in an AVStream. After I decode the frame with avcodec_decode_video2, I store the time stamp of that frame in an int64_t using av_frame_get_best_effort_timestamp. Now if the program is getting its input from a file I get a different timestamp from when I stream the input (from the same file) to the program.

  To change the input type I simply change the argv argument from "/path/to/file.mp4" to something like "udp ://localhost:1234", then I stream the file with ffmpeg in command line : "ffmpeg -re -i /path/to/file.mp4 -f mpegts udp ://localhost:1234". Can it be because the "-f mpegts" arguments change some characteristics of the media ?

  Below is my code (simplified). By reading the ffmpeg mailing list archives I realized that the time_base that I’m looking for is in the AVStream and not the AVCodecContext. Instead of using av_frame_get_best_effort_timestamp I have also tried using the packet.pts but the results don’t change.
  I need the time stamps to have a notion of frame number in a streaming video that is being received.
  I would really appreciate any sort of help.

  //..
  
  //argv[1]="/file.mp4";
  
  argv[1]="udp://localhost:7777";
  
  // define AVFormatContext, AVFrame, etc.
  
  // register av, avcodec, avformat_network_init(), etc.
  
  avformat_open_input(&pFormatCtx, argv, NULL, NULL);
  
  avformat_find_stream_info(pFormatCtx, NULL);
  
  // find the video stream...
  
  // pointer to the codec context...
  
  // open codec...
  
  pFrame=av_frame_alloc();
  
  while(av_read_frame(pFormatCtx, &packet)>=0) {
  
          AVStream *strem = pFormatCtx->streams[videoStream];
  
          if(packet.stream_index==videoStream) {
  
              avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
  
              if(frameFinished) {
  
                  int64_t perts = av_frame_get_best_effort_timestamp(pFrame);
  
                  if (isMyFrame(pFrame)){
  
                       cout << perts*av_q2d(strem->time_base) << "\n";
  
                  }
  
               }
  
  }
  
  //free allocated space
  
  }
  
  //..