how to enable transport wide congestion control in gstreamer

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• How to open GStreamer pipeline in OpenCV

  I'm a software engineer in South Korea.

  I'm trying to open webm video using GStreamer pipeline in opencv program But I can't find any solution to figure out it.

  I'm using OpenCV 3.4.1 in Visual Studio 19 Community IDE.

  Below is my code.

  #include <opencv2/opencv.hpp>

  int main()

  {

  std::string pipeline = "playbin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm";
  std::cout << "Using pipeline: \n" << pipeline << "\n";
  
  cv::VideoCapture cap(pipeline, cv::CAP_GSTREAMER);
  
  if (!cap.isOpened()) {
      std::cout << "Failed to open camera." << std::endl;
      return (-1);
  }
  
  cv::namedWindow("CSI Camera", cv::WINDOW_AUTOSIZE);
  cv::Mat img;
  
  std::cout << "Hit ESC to exit" << "\n";
  while (true)
  {
      if (!cap.read(img)) {
          std::cout << "Capture read error" << std::endl;
          break;
      }
  
      cv::imshow("CSI Camera", img);
      int keycode = cv::waitKey(10) & 0xff;
      if (keycode == 27) break;
  }
  
  cap.release();
  cv::destroyAllWindows();
  return 0;
  

  }

  ---

  It is very simple code like Tutorial. But I can't open VieoCapure cap...

  Anybody have tried this project or figured out?

  Best regard
• Gstreamer: Is there way to convert h264 video-stream from byte-stream to avc format

  I would need the above mentioned method to convert h264 stream with stream format byte-stream to avc (sort of packetized format) in order for it to be fed into matroskamux

  I have used C codes to program my pipeline and I have tried to use h264parse element before feeding to the matroskamux but it just simply parse it without changing the stream format.

  The result is as below

  However when I tried to use command line interfaces to script out the static pipeline that I use to mimic the same effects. Miraculously, the result is what I need and there is virtually no other h264parse properties that i use. I can obtain what I need to feed the matroskamux

  The result

  I tried to figure out what is the property to set for h264parse but none of which seem to be able to explicitly convert to another format. Is there something else to packetize the stream bytes into avc? Or it is I must force the pad?

  I am using vpuh264_enc on ARM platform and as such would have limited choice of plugins available but would still be happy to hear what I could do about the problem.

  Thanks

  ==========================================================

  More information:

  How i have gotten the video pad of splitmuxsink

  1. first i get the pad template

     video_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (splitmuxsink), "video");

  2. Next I get the pad using request pad

     pad = tee_pad = gst_element_request_pad (splitmuxsink,
                                     video_pad_template,
                                     NULL,
                                     NULL);
     

  3. After which i joined as per normal

     'if (gst_pad_link (src_pad, pad) != GST_PAD_LINK_OK ) { g_printerr (" could not be linked.\n"); gst_object_unref (pipeline); return -1; }'

  ============================================================

  The CLI command (which I use)

    gst-launch-1.0 \
      v4l2src \
          ! video/x-raw,width=640,height=480,framerate=30/1,is-live=true \
          ! videorate \
          ! video/x-raw,framerate=25/1 \
          ! tee name=tv  \
         tv. \
          ! queue \
          ! vpuenc_h264 \
          ! tee name=tv2 \
         tv2. \
          ! queue \
          ! h264parse \
          ! queue \
          ! m.video  \
         tv2. \
          ! queue \
          ! rtph264pay pt=96 \
          ! udpsink host="192.168.50.3" port=2534 \
       pulsesrc  \
          ! audioconvert \
          ! audioresample \
          ! audio/x-raw,rate=8000,channels=1,depth=8,format=S16LE \
          ! tee name=ta  \
          ! queue \
          ! mulawenc \
          ! tee name=ta2  \
         ta2. \
          ! queue \
          ! rtppcmupay pt=97 \
          ! udpsink host="192.168.50.3" port=2536 \
         ta2. \
          ! queue \
          ! m.audio_0  splitmuxsink location=log%02d.mkv muxer=matroskamux max-size-time=60000000000 name=m 
  
• Gstreamer rtsp server with different urls for payloaders

  I am working on streaming device with CSI camera input. I want to duplicate the incomming stream with tee and subsequently access each of these streams with different url using gst-rtsp-server. I can have only one consumer on my camera so it is impossible to have two standalone pipelines. Is this possible? See the pseudo pipeline below.

  source -> tee name=t -> rtsp with url0 .t -> rtsp with url1
  

  Thanks!
• Corrupted H264 video when streaming via RTP/UDP

  I'm trying to stream a video encoded in H264 over RTP/UDP.

  Sending:

  gst-launch-1.0 \
    videotestsrc ! \
    video/x-raw,format=RGBx,width=960,height=540,framerate=25/1 ! \
    videoconvert ! \
    x264enc bitrate=2000 ! \
    rtph264pay config-interval=1 pt=96 ! \
    udpsink port=5000
  

  Receive:

  gst-launch-1.0 \
    udpsrc port=5000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" ! \
    rtph264depay ! \
    decodebin ! \
    videoconvert ! \
    ximagesink
  

  If I start receiving the video before sending it, then everything works as intended.

  However, if I start receiving video after the start of sending, then the image "breaks".

  An example of a corrupted image

  How to fix this problem?
• How to properly close a pjmedia_transport element of an AudioStream?

  I'm using pjsip pjproject and i want to establish 2 rtp streams connections customizable (in some field as ipaddress,port,codec,ecc) from the the user throught a web interface at runtime. Whenever the user edit some rtp stream, my backend application save the new data, tear down the old streams (closing the rtp e rtpc transport/socket) and then it will create the new streams selected. My actual results seem to indicate that pjmedia_transport_close(), that i used to close the pjmedia_transport associated to the stream, doesn't work cause at the creation of the new streams (if i mantain the same port for example) it log this error message:

  [CRITICAL] create pjmedia1 uc transport udp failed: 120098 -> Address already in use
  

  So i want to recreate rtp streams, with the same transport element, without error freeing the busied socket. Is there some another properly procedure to close the pjmedia_transport? I need to add some other lines code?
• how to get send time or captured time of the RTP buffer at receiver in gstreamer

  I am sending a webrtc stream from browser to gstreamer. at receiver I want to get time at which a particular buffer was captured at sender to be sent.
• RTCP package's Cumulative number of packets lost is different from NACK message's FCI description

  in first rtcp SR package, it show Cumulative number of packets lost is 2, but in NACK part it shows 3 frames lost, just curious about this, i though this two part should be same.