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..e024d61ac93c6f7e2341c5c66f91eef0289a10cc 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -20,6 +20,7 @@
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
@@ -77,6 +78,8 @@ class RtcEventLogImpl final : public RtcEventLog {
   void LogBwePacketLossEvent(int32_t bitrate,
                              uint8_t fraction_loss,
                              int32_t total_packets) override;
+  void LogAudioEncoderConfig(
+      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
 
  private:
   void StoreEvent(std::unique_ptr<rtclog::Event>* event);
@@ -437,6 +440,16 @@ void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
   StoreEvent(&event);
 }
 
+void RtcEventLogImpl::LogAudioEncoderConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::AUDIO_ENCODER_CONFIG_EVENT);
+  DebugDumpWriter::ConvertEncoderConfigToDumpEntry(
+      config, event->mutable_encoder_config());
+  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.";