Delay based logging.

webrtc/tools/event_log_visualizer/analyzer.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 449 matching lines @@
       case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
         LossBasedBweUpdate bwe_update;
         bwe_update.timestamp = parsed_log_.GetTimestamp(i);
         parsed_log_.GetLossBasedBweUpdate(i, &bwe_update.new_bitrate,
                                           &bwe_update.fraction_loss,
                                           &bwe_update.expected_packets);
         bwe_loss_updates_.push_back(bwe_update);
         break;
       }
       case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
+        bwe_delay_updates_.push_back(parsed_log_.GetDelayBasedBweUpdate(i));
         break;
       }
       case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
         AudioNetworkAdaptationEvent ana_event;
         ana_event.timestamp = parsed_log_.GetTimestamp(i);
         parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
         audio_network_adaptation_events_.push_back(ana_event);
         break;
       }
       case ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
@@ skipping 446 matching lines @@
     }
     float window_duration_in_seconds =
         static_cast<float>(window_duration_) / 1000000;
     float x = static_cast<float>(time - begin_time_) / 1000000;
     float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
     time_series->points.emplace_back(x, y);
   }
 
   // Overlay the send-side bandwidth estimate over the outgoing bitrate.
   if (desired_direction == kOutgoingPacket) {
-    TimeSeries* time_series =
+    TimeSeries* loss_series =
         plot->AddTimeSeries("Loss-based estimate", LINE_STEP_GRAPH);
-    for (auto& bwe_update : bwe_loss_updates_) {
+    for (auto& loss_update : bwe_loss_updates_) {
       float x =
-          static_cast<float>(bwe_update.timestamp - begin_time_) / 1000000;
-      float y = static_cast<float>(bwe_update.new_bitrate) / 1000;
-      time_series->points.emplace_back(x, y);
+          static_cast<float>(loss_update.timestamp - begin_time_) / 1000000;
+      float y = static_cast<float>(loss_update.new_bitrate) / 1000;
+      loss_series->points.emplace_back(x, y);
+    }
+
+    TimeSeries* delay_series =
+        plot->AddTimeSeries("Delay-based estimate", LINE_STEP_GRAPH);
+    for (auto& delay_update : bwe_delay_updates_) {
+      float x =
+          static_cast<float>(delay_update.timestamp - begin_time_) / 1000000;
+      float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
+      delay_series->points.emplace_back(x, y);
     }
 
     TimeSeries* created_series =
         plot->AddTimeSeries("Probe cluster created.", DOT_GRAPH);
     for (auto& cluster : bwe_probe_cluster_created_events_) {
       float x = static_cast<float>(cluster.timestamp - begin_time_) / 1000000;
       float y = static_cast<float>(cluster.bitrate_bps) / 1000;
       created_series->points.emplace_back(x, y);
     }
 
@@ skipping 423 matching lines @@
         return rtc::Optional<float>();
       },
       audio_network_adaptation_events_, begin_time_, time_series);
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
 }  // namespace plotting
 }  // namespace webrtc