Video/Screenshare loopback tool.

webrtc/rtc_tools/event_log_visualizer/main.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
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   }
   if (FLAG_plot_audio_encoder_fec) {
     analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_audio_encoder_dtx) {
     analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_audio_encoder_num_channels) {
     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
   }
+  analyzer.CreateQueueDelayGraph(collection->AppendNewPlot());
   if (FLAG_plot_audio_jitter_buffer) {
     std::string wav_path;
     if (FLAG_wav_filename[0] != '\0') {
       wav_path = FLAG_wav_filename;
     } else {
       wav_path = webrtc::test::ResourcePath(
           "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
     }
     analyzer.CreateAudioJitterBufferGraph(wav_path, 48000,
                                           collection->AppendNewPlot());
@@ skipping 25 matching lines @@
   FLAG_plot_fraction_loss_feedback = setting;
   FLAG_plot_timestamps = setting;
   FLAG_plot_audio_encoder_bitrate_bps = setting;
   FLAG_plot_audio_encoder_frame_length_ms = setting;
   FLAG_plot_audio_encoder_packet_loss = setting;
   FLAG_plot_audio_encoder_fec = setting;
   FLAG_plot_audio_encoder_dtx = setting;
   FLAG_plot_audio_encoder_num_channels = setting;
   FLAG_plot_audio_jitter_buffer = setting;
 }