| 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_;
|
|
|
|
|