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..888eb182f41f31e22e44f8dd731ea99cd9f8e834 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.h |
+++ b/webrtc/tools/event_log_visualizer/analyzer.h |
@@ -67,6 +67,21 @@ class EventLogAnalyzer { |
RTPHeader header; |
}; |
+ struct BwePacketLossEvent { |
+ BwePacketLossEvent(uint64_t timestamp, |
+ int32_t new_bitrate, |
+ uint8_t fraction_loss, |
+ int expected_packets) |
+ : timestamp(timestamp), |
+ new_bitrate(new_bitrate), |
+ fraction_loss(fraction_loss), |
+ expected_packets(expected_packets) {} |
+ 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 +93,9 @@ 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 |