Add ana config to event log visualiser

webrtc/tools/event_log_visualizer/analyzer.h

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 bb7667fbb8adb9bacfed5a0ca2e0f52550e09d89..48de89a5bbf656c9bfde0ef0647c1bd64ecaea16 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.h
+++ b/webrtc/tools/event_log_visualizer/analyzer.h
@@ -52,6 +52,11 @@ struct BwePacketLossEvent {
   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
@@ -87,6 +92,13 @@ class EventLogAnalyzer {
   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;
@@ -152,6 +164,8 @@ class EventLogAnalyzer {
   // 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