Add simulation of receive-side bandwidth estimate to event_log_analyzer

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.
  */
 
@@ skipping 53 matching lines @@
             "Plot the total bitrate used by all incoming streams.");
 DEFINE_bool(plot_outgoing_bitrate,
             true,
             "Plot the total bitrate used by all outgoing streams.");
 DEFINE_bool(plot_incoming_stream_bitrate,
             true,
             "Plot the bitrate used by each incoming stream.");
 DEFINE_bool(plot_outgoing_stream_bitrate,
             true,
             "Plot the bitrate used by each outgoing stream.");
+DEFINE_bool(plot_simulated_receiveside_bwe,
+            false,
+            "Run the receive-side bandwidth estimator with the incoming rtp "
+            "packets and plot the resulting estimate.");
 DEFINE_bool(plot_simulated_sendside_bwe,
             false,
             "Run the send-side bandwidth estimator with the outgoing rtp and "
             "incoming rtcp and plot the resulting estimate.");
 DEFINE_bool(plot_network_delay_feedback,
             true,
             "Compute network delay based on sent packets and the received "
             "transport feedback.");
 DEFINE_bool(plot_fraction_loss_feedback,
             true,
@@ skipping 136 matching lines @@
                                      FLAG_show_detector_state);
   }
   if (FLAG_plot_incoming_stream_bitrate) {
     analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
                                       collection->AppendNewPlot());
   }
   if (FLAG_plot_outgoing_stream_bitrate) {
     analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
                                       collection->AppendNewPlot());
   }
+  if (FLAG_plot_simulated_receiveside_bwe) {
+    analyzer.CreateReceiveSideBweSimulationGraph(collection->AppendNewPlot());
+  }
   if (FLAG_plot_simulated_sendside_bwe) {
-    analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
+    analyzer.CreateSendSideBweSimulationGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_network_delay_feedback) {
     analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_fraction_loss_feedback) {
     analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_timestamps) {
     analyzer.CreateTimestampGraph(collection->AppendNewPlot());
   }
@@ skipping 41 matching lines @@
   FLAG_plot_audio_playout = setting;
   FLAG_plot_audio_level = setting;
   FLAG_plot_incoming_sequence_number_delta = setting;
   FLAG_plot_incoming_delay_delta = setting;
   FLAG_plot_incoming_delay = setting;
   FLAG_plot_incoming_loss_rate = setting;
   FLAG_plot_incoming_bitrate = setting;
   FLAG_plot_outgoing_bitrate = setting;
   FLAG_plot_incoming_stream_bitrate = setting;
   FLAG_plot_outgoing_stream_bitrate = setting;
+  FLAG_plot_simulated_receiveside_bwe = setting;
   FLAG_plot_simulated_sendside_bwe = setting;
   FLAG_plot_network_delay_feedback = setting;
   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;
 }