| 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
|
|
|