Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log.proto

 syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.rtclog;
 
+import "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto";
+
 enum MediaType {
   ANY = 0;
   AUDIO = 1;
   VIDEO = 2;
   DATA = 3;
 }
 
 // This is the main message to dump to a file, it can contain multiple event
 // messages, but it is possible to append multiple EventStreams (each with a
 // single event) to a file.
@@ skipping 15 matching lines @@
     LOG_END = 2;
     RTP_EVENT = 3;
     RTCP_EVENT = 4;
     AUDIO_PLAYOUT_EVENT = 5;
     BWE_PACKET_LOSS_EVENT = 6;
     BWE_PACKET_DELAY_EVENT = 7;
     VIDEO_RECEIVER_CONFIG_EVENT = 8;
     VIDEO_SENDER_CONFIG_EVENT = 9;
     AUDIO_RECEIVER_CONFIG_EVENT = 10;
     AUDIO_SENDER_CONFIG_EVENT = 11;
+    AUDIO_NETWORK_ADAPTOR_EVENT = 12;
   }
 
   // required - Indicates the type of this event
   optional EventType type = 2;
 
   // optional - but required if type == RTP_EVENT
   optional RtpPacket rtp_packet = 3;
 
   // optional - but required if type == RTCP_EVENT
   optional RtcpPacket rtcp_packet = 4;
 
   // optional - but required if type == AUDIO_PLAYOUT_EVENT
   optional AudioPlayoutEvent audio_playout_event = 5;
 
   // optional - but required if type == BWE_PACKET_LOSS_EVENT
   optional BwePacketLossEvent bwe_packet_loss_event = 6;
 
   // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
   optional VideoReceiveConfig video_receiver_config = 8;
 
   // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
   optional VideoSendConfig video_sender_config = 9;
 
   // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
   optional AudioReceiveConfig audio_receiver_config = 10;
 
   // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
   optional AudioSendConfig audio_sender_config = 11;
+
+  // optional - but required if type == AUDIO_NETWORK_ADAPTOR_EVENT
+  optional webrtc.audio_network_adaptor.debug_dump.EncoderRuntimeConfig
+      encoder_runtime_config = 12;
 }
 
 message RtpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
 
   // required
   optional MediaType type = 2;
 
   // required - The size of the packet including both payload and header.
@@ skipping 142 matching lines @@
   repeated RtpHeaderExtension header_extensions = 3;
 }
 
 message AudioSendConfig {
   // required - Synchronization source (stream identifier) for outgoing stream.
   optional uint32 ssrc = 1;
 
   // RTP header extensions used for the outgoing audio stream.
   repeated RtpHeaderExtension header_extensions = 2;
 }