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 988f2cb482151aaa87288cf27ef119c7e5f38527..fab52b96a1b645c3524b63e1ed4dbdb1ce78a2cb 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.h |
+++ b/webrtc/tools/event_log_visualizer/analyzer.h |
@@ -100,6 +100,9 @@ class EventLogAnalyzer { |
void CreateAudioEncoderEnableFecGraph(Plot* plot); |
void CreateAudioEncoderEnableDtxGraph(Plot* plot); |
void CreateAudioEncoderNumChannelsGraph(Plot* plot); |
+ void CreateAudioJitterBufferGraph(const std::string& replacement_file_name, |
+ int file_sample_rate_hz, |
+ Plot* plot); |
// Returns a vector of capture and arrival timestamps for the video frames |
// of the stream with the most number of frames. |
@@ -163,6 +166,13 @@ class EventLogAnalyzer { |
std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_; |
+ // Maps an SSRC to the timestamps of parsed audio playout events. |
+ std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_; |
+ |
+ // Stores the timestamps for all log segments, in the form of associated start |
+ // and end events. |
+ std::vector<std::pair<uint64_t, uint64_t>> log_segments_; |
+ |
// A list of all updates from the send-side loss-based bandwidth estimator. |
std::vector<LossBasedBweUpdate> bwe_loss_updates_; |