Index: webrtc/tools/event_log_visualizer/analyzer.cc |
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc |
index ba95c2565d7da640696c44dffc43829e2831a0a4..0d2483e24dcd082013037d3724774bb4c20c9670 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.cc |
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc |
@@ -467,6 +467,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
break; |
} |
case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: { |
+ bwe_delay_updates_.push_back(parsed_log_.GetDelayBasedBweUpdate(i)); |
break; |
} |
case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: { |
@@ -933,13 +934,22 @@ void EventLogAnalyzer::CreateTotalBitrateGraph( |
// 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 = |