Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log.cc

Index: webrtc/logging/rtc_event_log/rtc_event_log.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 976ff2321c08106dcea319a19c63f50619ebb0af..d35bd72010e87a5c34fc7f11a4b99081c5dbc102 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -77,6 +77,12 @@ class RtcEventLogImpl final : public RtcEventLog {
   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;
 
  private:
   void StoreEvent(std::unique_ptr<rtclog::Event>* event);
@@ -429,7 +435,7 @@ void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
                                             int32_t total_packets) {
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(clock_->TimeInMicroseconds());
-  event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTOR_EVENT);
   auto bwe_event = event->mutable_bwe_packet_loss_event();
   bwe_event->set_bitrate(bitrate);
   bwe_event->set_fraction_loss(fraction_loss);
@@ -437,6 +443,32 @@ void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
   StoreEvent(&event);
 }
 
+void RtcEventLogImpl::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) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  auto ana_event = event->mutable_audio_network_adaptor_decition();
+  if (bitrate_bps)
+    ana_event->set_bitrate_bps(*bitrate_bps);
+  if (frame_length_ms)
+    ana_event->set_frame_length_ms(*frame_length_ms);
+  if (uplink_packet_loss_fraction)
+    ana_event->set_uplink_packet_loss_fraction(*uplink_packet_loss_fraction);
+  if (enable_fec)
+    ana_event->set_enable_fec(*enable_fec);
+  if (enable_dtx)
+    ana_event->set_enable_dtx(*enable_dtx);
+  if (num_channels)
+    ana_event->set_num_channels(*num_channels);
+  StoreEvent(&event);
+}
+
 void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
   if (!event_queue_.Insert(event)) {
     LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";