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• Headless Selenium not Recording Video

  12 mai 2015, par TIMBERings

  I’m using the Headless gem to create a headless session using xvfb and to record the session using ffmpeg. I’m not able to save the video created by the headless gem. I’m including the relevant snippets of my code and the output.

  Code

  $LOGGER.info("----------Starting test: #{@test_name}----------")
  
  
  
  if Utilities.linux? && ENV['DRIVER'] != 'sauce'
  
    @headless = Headless.new ({dimension: '1600x1200x16',
  
                      pid_file_path: "#{Utilities.get_root_directory}/tmp/headless_ffmpeg_#{@display}.pid",
  
                      tmp_file_path: "#{Utilities.get_root_directory}/tmp/headless_ffmpeg_#{@display}.mov",
  
                      log_file_path: "#{Utilities.get_root_directory}/tmp/headless_ffmpeg_#{@display}.txt"})
  
    @headless.start
  
    @headless.video.start_capture
  
    $LOGGER.info 'Creating headless session.'
  
  end
  
  $browser = DefaultWatir.initiate_browser(@test_name)
  
  
  
  if Utilities.linux? && ENV['DRIVER'] != 'sauce'
  
    $LOGGER.info "Starting video recording"
  
    @headless.video.start_capture
  
  end
  
  
  
  .........          .........
  
  ......... TEST RUN .........
  
  .........          .........
  
  
  
  $LOGGER.info "----------Ending test: #{@test_name}----------"
  
   puts "\n\n"
  
  
  
  $browser.close if $browser
  
  save_video if @headless
  
  
  
  def video_path
  
    return "#{Utilities.get_root_directory}/video/#{@test_name}_#{Time.now.to_s.gsub(' ', '-')}"
  
  end
  
  
  
  def save_video
  
    if Utilities.linux?  && ENV['DRIVER'] != 'sauce'
  
      unless self.instance_variable_get(:@exception).nil? && !$has_errors
  
        saved_video_path = video_path
  
        Dir.mkdir("#{Utilities.get_root_directory}/video") unless Dir.exists?("#{Utilities.get_root_directory}/video")
  
        $LOGGER.info "Video path = #{saved_video_path}"
  
        begin
  
          $LOGGER.info 'before save'
  
          @headless.video.stop_and_save(saved_video_path)
  
          $LOGGER.info 'after save'
  
        rescue ex
  
          $LOGGER.info 'Video save had exception.'
  
          puts ex.message
  
          puts ex.backtrace.join("\n\t")
  
        end
  
        $LOGGER.info "Video saved at: #{saved_video_path}"
  
      else
  
        @headless.video.stop_and_discard
  
      end
  
      @headless.destroy
  
    end
  
  end

  Output

  [INFO - 2015-03-06 21:58:37 +0000] - ----------Starting test : Example Test----------

  [INFO - 2015-03-06 21:58:37 +0000] - Creating headless session.

  [INFO - 2015-03-06 21:58:40 +0000] - Starting video recording

  [INFO - 2015-03-06 22:02:04 +0000] - ----------Ending test : Example Test----------

  INFO - 2015-03-06 22:02:05 +0000] - Video path = /var/lib/jenkins/workspace/end_to_end_firefox/selenium-tests/video/PoolPlayToSingleEliminationOneDivision_2015-03-06-22:02:05-+0000

  [INFO - 2015-03-06 22:02:05 +0000] - before save

  [INFO - 2015-03-06 22:02:05 +0000] - after save

  [INFO - 2015-03-06 22:02:05 +0000] - Video saved at : /var/lib/jenkins/workspace/end_to_end_firefox/selenium-tests/video/PoolPlayToSingleEliminationOneDivision_2015-03-06-22:02:05-+0000

• Set RTSP/UDP buffer size in FFmpeg

  16 mars 2015, par chuckleplant

  Note : I’m aware ffmpeg and libav are different libraries. This is a problem common to both.

  Disclaimer : Duplicate of SO question marked as answered but actually didn’t give a proper solution.

  ---

  Insufficient UDP buffer size causes broken streams for several high resolution video streams. In LibAV/FFMPEG it’s possible to set the udp buffer size for udp urls (udp ://...) by appending some options to it.

  However, for RTSP urls this is not supported. These are the only solutions I’ve found :

  • Rebuilding ffmpeg/libav changing the UDP_MAX_PKT_SIZE in the udp.c source file.
  • Using a nasty hack to find the required value.
  • Using a different decoding library (proposed solution to aforementioned related SO question).

  None of these is actually a solution. From what I found it should be possible to use the API’s AVOptions to find and set this value. Or else, the AVDictionary.

  It’s very difficult to find how to set these throughout the documentation of either libav or ffmpeg.

• RTP packets detected as UDP

  8 juillet 2024, par fritz

  Here is what I am trying to do :

  



  WebRTC endpoint > RTP Endpoint > ffmpeg > RTMP server.


  



  This is what my SDP file looks like.

  



  var cm_offer = "v=0\n" +
              "o=- 3641290734 3641290734 IN IP4 127.0.0.1\n" +
              "s=nginx\n" +
              "c=IN IP4 127.0.0.1\n" +
              "t=0 0\n" +
              "m=audio 60820 RTP/AVP 0\n" +
              "a=rtpmap:0 PCMU/8000\n" +
              "a=recvonly\n" +
              "m=video 59618 RTP/AVP 101\n" +
              "a=rtpmap:101 H264/90000\n" +
              "a=recvonly\n";


  



  What's happening is that wireshark can detect the incoming packets at port 59618, but not as RTP packets but UDP packets. I am trying to capture the packets using ffmpeg with the following command :

  



  ubuntu@ip-132-31-40-100:~$ ffmpeg -i udp://127.0.0.1:59618 -vcodec copy stream.mp4
ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
  libavutil      55. 44.100 / 55. 44.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 63.100 / 57. 63.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 69.100 /  6. 69.100
  libavresample   3.  2.  0 /  3.  2.  0
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100 


  



  All I get is a blinking cursor and The stream.mp4 file is not written to disk after I exit (ctrl+c).

  



  So can you help me figure out :

  



  

  1. why wireshark cannot detect the packets as RTP (I suspect it has something to do with SDP)

  


  2. How to handle SDP answer when the RTP endpoint is pushing to ffmpeg which doesn't send an answer back.

  


  



  Here is the entire code (hello world tutorial modified)

  



  /*
     * (C) Copyright 2014-2015 Kurento (http://kurento.org/)
     *
     * Licensed under the Apache License, Version 2.0 (the "License");
     * you may not use this file except in compliance with the License.
     * You may obtain a copy of the License at
     *
     *   http://www.apache.org/licenses/LICENSE-2.0
     *
     * Unless required by applicable law or agreed to in writing, software
     * distributed under the License is distributed on an "AS IS" BASIS,
     * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     * See the License for the specific language governing permissions and
     * limitations under the License.
     */

    function getopts(args, opts)
    {
      var result = opts.default || {};
      args.replace(
          new RegExp("([^?=&]+)(=([^&]*))?", "g"),
          function($0, $1, $2, $3) { result[$1] = decodeURI($3); });

      return result;
    };

    var args = getopts(location.search,
    {
      default:
      {
        ws_uri: 'wss://' + location.hostname + ':8433/kurento',
        ice_servers: undefined
      }
    });

    function setIceCandidateCallbacks(webRtcPeer, webRtcEp, onerror)
    {
      webRtcPeer.on('icecandidate', function(candidate) {
        console.log("Local candidate:",candidate);

        candidate = kurentoClient.getComplexType('IceCandidate')(candidate);

        webRtcEp.addIceCandidate(candidate, onerror)
      });

      webRtcEp.on('OnIceCandidate', function(event) {
        var candidate = event.candidate;

        console.log("Remote candidate:",candidate);

        webRtcPeer.addIceCandidate(candidate, onerror);
      });
    }


    function setIceCandidateCallbacks2(webRtcPeer, rtpEp, onerror)
    {
      webRtcPeer.on('icecandidate', function(candidate) {
        console.log("Localr candidate:",candidate);

        candidate = kurentoClient.getComplexType('IceCandidate')(candidate);

        rtpEp.addIceCandidate(candidate, onerror)
      });
    }


    window.addEventListener('load', function()
    {
      console = new Console();

      var webRtcPeer;
      var pipeline;
      var webRtcEpt;

      var videoInput = document.getElementById('videoInput');
      var videoOutput = document.getElementById('videoOutput');

      var startButton = document.getElementById("start");
      var stopButton = document.getElementById("stop");

      startButton.addEventListener("click", function()
      {
        showSpinner(videoInput, videoOutput);

        var options = {
          localVideo: videoInput,
          remoteVideo: videoOutput
        };


        if (args.ice_servers) {
         console.log("Use ICE servers: " + args.ice_servers);
         options.configuration = {
           iceServers : JSON.parse(args.ice_servers)
         };
        } else {
         console.log("Use freeice")
        }

        webRtcPeer = kurentoUtils.WebRtcPeer.WebRtcPeerSendrecv(options, function(error)
        {
          if(error) return onError(error)

          this.generateOffer(onOffer)
        });

        function onOffer(error, sdpOffer)
        {
          if(error) return onError(error)

          kurentoClient(args.ws_uri, function(error, client)
          {
            if(error) return onError(error);

            client.create("MediaPipeline", function(error, _pipeline)
            {
              if(error) return onError(error);

              pipeline = _pipeline;

              pipeline.create("WebRtcEndpoint", function(error, webRtc){
                if(error) return onError(error);

                webRtcEpt = webRtc;

                setIceCandidateCallbacks(webRtcPeer, webRtc, onError)

                webRtc.processOffer(sdpOffer, function(error, sdpAnswer){
                  if(error) return onError(error);

                  webRtcPeer.processAnswer(sdpAnswer, onError);
                });
                webRtc.gatherCandidates(onError);

                webRtc.connect(webRtc, function(error){
                  if(error) return onError(error);

                  console.log("Loopback established");
                });
              });



            pipeline.create("RtpEndpoint", function(error, rtp){
                if(error) return onError(error);

                //setIceCandidateCallbacks2(webRtcPeer, rtp, onError)


                var cm_offer = "v=0\n" +
                      "o=- 3641290734 3641290734 IN IP4 127.0.0.1\n" +
                      "s=nginx\n" +
                      "c=IN IP4 127.0.0.1\n" +
                      "t=0 0\n" +
                      "m=audio 60820 RTP/AVP 0\n" +
                      "a=rtpmap:0 PCMU/8000\n" +
                      "a=recvonly\n" +
                      "m=video 59618 RTP/AVP 101\n" +
                      "a=rtpmap:101 H264/90000\n" +
                      "a=recvonly\n";



                rtp.processOffer(cm_offer, function(error, cm_sdpAnswer){
                  if(error) return onError(error);

                  //webRtcPeer.processAnswer(cm_sdpAnswer, onError);
                });
                //rtp.gatherCandidates(onError);

                webRtcEpt.connect(rtp, function(error){
                  if(error) return onError(error);

                  console.log("RTP endpoint connected to webRTC");
                });
              });









            });
          });
        }
      });
      stopButton.addEventListener("click", stop);


      function stop() {
        if (webRtcPeer) {
          webRtcPeer.dispose();
          webRtcPeer = null;
        }

        if(pipeline){
          pipeline.release();
          pipeline = null;
        }

        hideSpinner(videoInput, videoOutput);
      }

      function onError(error) {
        if(error)
        {
          console.error(error);
          stop();
        }
      }
    })


    function showSpinner() {
      for (var i = 0; i < arguments.length; i++) {
        arguments[i].poster = 'img/transparent-1px.png';
        arguments[i].style.background = "center transparent url('img/spinner.gif') no-repeat";
      }
    }

    function hideSpinner() {
      for (var i = 0; i < arguments.length; i++) {
        arguments[i].src = '';
        arguments[i].poster = 'img/webrtc.png';
        arguments[i].style.background = '';
      }
    }

    /**
     * Lightbox utility (to display media pipeline image in a modal dialog)
     */
    $(document).delegate('*[data-toggle="lightbox"]', 'click', function(event) {
      event.preventDefault();
      $(this).ekkoLightbox();
    });