Video/Screenshare loopback tool.

webrtc/rtc_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 87 matching lines @@
   void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
   void CreateAudioEncoderFrameLengthGraph(Plot* plot);
   void CreateAudioEncoderPacketLossGraph(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);
 
+  void CreateQueueDelayGraph(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) {}
     bool operator<(const StreamId& other) const {
@@ skipping 61 matching lines @@
 
   std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
 
   std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent>
       bwe_probe_cluster_created_events_;
 
   std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
 
   std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
 
+  std::vector<ParsedRtcEventLog::BweAckedBitrateEvent> acked_bitrate_events_;
+
+  std::vector<ParsedRtcEventLog::AlrStateEvent> alr_state_events_;
+
+  std::vector<ParsedRtcEventLog::PacketQueueTime> packet_queue_time_events_;
+
   // 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
   // current data point will be part of the average.
   uint64_t window_duration_;
   uint64_t step_;
 
   // First and last events of the log.
   uint64_t begin_time_;
   uint64_t end_time_;
 
   // Duration (in seconds) of log file.
   float call_duration_s_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_