Add ana config to event log visualiser

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 34 matching lines @@
   std::unique_ptr<rtcp::RtcpPacket> packet;
 };
 
 struct BwePacketLossEvent {
   uint64_t timestamp;
   int32_t new_bitrate;
   uint8_t fraction_loss;
   int32_t expected_packets;
 };
 
+struct AudioNetworkAdaptationEvent {
+  uint64_t timestamp;
+  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+};
+
 class EventLogAnalyzer {
  public:
   // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
   // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
 
   void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
 
   void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
@@ skipping 15 matching lines @@
 
   void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot);
 
   void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
 
   void CreateBweSimulationGraph(Plot* plot);
 
   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);
+
   // 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 45 matching lines @@
   // 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_;
 
   // A list of all updates from the send-side loss-based bandwidth estimator.
   std::vector<BwePacketLossEvent> bwe_loss_updates_;
 
+  std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_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_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_