Visualize events related to probing in the total bitrate graph.

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 459 matching lines @@
         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: {
+        bwe_probe_cluster_created_events_.push_back(
+            parsed_log_.GetBweProbeClusterCreated(i));
         break;
       }
       case ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
+        bwe_probe_result_events_.push_back(parsed_log_.GetBweProbeResult(i));
         break;
       }
       case ParsedRtcEventLog::UNKNOWN_EVENT: {
         break;
       }
     }
   }
 
   if (last_timestamp < first_timestamp) {
     // No useful events in the log.
@@ skipping 438 matching lines @@
   // Overlay the send-side bandwidth estimate over the outgoing bitrate.
   if (desired_direction == kOutgoingPacket) {
     TimeSeries* time_series =
         plot->AddTimeSeries("Loss-based estimate", LINE_STEP_GRAPH);
     for (auto& bwe_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);
     }
+
+    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);
+    }
+
+    TimeSeries* result_series =
+        plot->AddTimeSeries("Probing results.", DOT_GRAPH);
+    for (auto& result : bwe_probe_result_events_) {
+      if (result.bitrate_bps) {
+        float x = static_cast<float>(result.timestamp - begin_time_) / 1000000;
+        float y = static_cast<float>(*result.bitrate_bps) / 1000;
+        result_series->points.emplace_back(x, y);
+      }
+    }
   }
+
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
   if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
     plot->SetTitle("Incoming RTP bitrate");
   } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
     plot->SetTitle("Outgoing RTP bitrate");
   }
 }
 
 // For each SSRC, plot the bandwidth used by that stream.
@@ skipping 402 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