Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

Index: webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
index ce55a4fb39b740a422eda7b874a8a000f88e4601..3b808b2e43d451fac6915f3aa2a891c5685a2160 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -79,6 +79,8 @@ ParsedRtcEventLog::EventType GetRuntimeEventType(
       return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
+    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
   }
   RTC_NOTREACHED();
   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
@@ -454,4 +456,29 @@ void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
   }
 }
 
+void ParsedRtcEventLog::GetAudioNetworkAdaptation(
+    size_t index,
+    AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
+  RTC_CHECK(event.has_audio_network_adaptation());
+  const rtclog::AudioNetworkAdaptation& ana_event =
+      event.audio_network_adaptation();
+  if (ana_event.has_bitrate_bps())
+    config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
+  if (ana_event.has_enable_fec())
+    config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
+  if (ana_event.has_enable_dtx())
+    config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
+  if (ana_event.has_frame_length_ms())
+    config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
+  if (ana_event.has_num_channels())
+    config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
+  if (ana_event.has_uplink_packet_loss_fraction())
+    config->uplink_packet_loss_fraction =
+        rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
+}
+
 }  // namespace webrtc