Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  Copyright (c) 2015 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 90 matching lines @@
                              size_t length) = 0;
 
   // Logs an audio playout event.
   virtual void LogAudioPlayout(uint32_t ssrc) = 0;
 
   // Logs a bitrate update from the bandwidth estimator based on packet loss.
   virtual void LogBwePacketLossEvent(int32_t bitrate,
                                      uint8_t fraction_loss,
                                      int32_t total_packets) = 0;
 
+  // Logs audio network adapror decisions.
+  virtual void LogAnaDecisionEvent(
+      rtc::Optional<int> bitrate_bps,
+      rtc::Optional<int> frame_length_ms,
+      rtc::Optional<float> uplink_packet_loss_fraction,
+      rtc::Optional<bool> enable_fec,
+      rtc::Optional<bool> enable_dtx,
+      rtc::Optional<size_t> num_channels) = 0;
+
   // Reads an RtcEventLog file and returns true when reading was successful.
   // The result is stored in the given EventStream object.
   // The order of the events in the EventStream is implementation defined.
   // The current implementation writes a LOG_START event, then the old
   // configurations, then the remaining events in timestamp order and finally
   // a LOG_END event. However, this might change without further notice.
   // TODO(terelius): Change result type to a vector?
   static bool ParseRtcEventLog(const std::string& file_name,
                                rtclog::EventStream* result);
 };
@@ skipping 21 matching lines @@
                     const uint8_t* header,
                     size_t packet_length) override {}
   void LogRtcpPacket(PacketDirection direction,
                      MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override {}
   void LogAudioPlayout(uint32_t ssrc) override {}
   void LogBwePacketLossEvent(int32_t bitrate,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {}
+  void LogAnaDecisionEvent(rtc::Optional<int> bitrate_bps,
+                           rtc::Optional<int> frame_length_ms,
+                           rtc::Optional<float> uplink_packet_loss_fraction,
+                           rtc::Optional<bool> enable_fec,
+                           rtc::Optional<bool> enable_dtx,
+                           rtc::Optional<size_t> num_channels) override{};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_