Add NetEq delay plotting to event_log_visualizer

webrtc/tools/event_log_visualizer/analyzer.h

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 82 matching lines @@
 
   void CreateNetworkDelayFeedbackGraph(Plot* plot);
   void CreateTimestampGraph(Plot* plot);
 
   void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
   void CreateAudioEncoderFrameLengthGraph(Plot* plot);
   void CreateAudioEncoderUplinkPacketLossFractionGraph(Plot* plot);
   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.
   std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
 
  private:
   class StreamId {
    public:
     StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
         : ssrc_(ssrc), direction_(direction) {}
@@ skipping 43 matching lines @@
   std::set<StreamId> video_ssrcs_;
   std::set<StreamId> audio_ssrcs_;
 
   // Maps a stream identifier consisting of ssrc and direction to the parsed
   // RTP headers in that stream. Header extensions are parsed if the stream
   // has been configured.
   std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
 
   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_;
 
   std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
 
   std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent>
       bwe_probe_cluster_created_events_;
 
   std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
 
@@ skipping 11 matching lines @@
   uint64_t end_time_;
 
   // Duration (in seconds) of log file.
   float call_duration_s_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_