Index: webrtc/tools/event_log_visualizer/analyzer.h |
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h |
index 9b69ff12630d9186ef8c964fc5fb00b3d670a5f2..0b92c10e1086decd181b8b4ac63e621401fce8ad 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.h |
+++ b/webrtc/tools/event_log_visualizer/analyzer.h |
@@ -67,6 +67,13 @@ class EventLogAnalyzer { |
RTPHeader header; |
}; |
+ struct BwePacketLossEvent { |
+ uint64_t timestamp; |
+ int32_t new_bitrate; |
+ uint8_t fraction_loss; |
+ int32_t expected_packets; |
+ }; |
+ |
const ParsedRtcEventLog& parsed_log_; |
// A list of SSRCs we are interested in analysing. |
@@ -78,6 +85,9 @@ class EventLogAnalyzer { |
// if the stream has been configured. |
std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; |
+ // A list of all updates from the send-side loss-based bandwidth estimator. |
+ std::vector<BwePacketLossEvent> bwe_loss_updates_; |
+ |
// Window and step size used for calculating moving averages, e.g. bitrate. |
// The generated data points will be |step_| microseconds apart. |
// Only events occuring at most |window_duration_| microseconds before the |