Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log.proto

Index: webrtc/logging/rtc_event_log/rtc_event_log.proto
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto
index a6d169579659e496379394de86cb9b4fac220b13..3577c43715e299eddb5a5140e3e763e6c7d332ae 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.proto
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto
@@ -2,6 +2,8 @@ syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.rtclog;
 
+import "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto";

[terelius, 2016/12/22 16:04:24]

I'd prefer to keep all relevant event types in this file. It will make it easier
to see what types of data are stored (e.g. reviewing log from a privacy
perspective).

[minyue-webrtc, 2016/12/29 20:28:51]

Acknowledged.

+
 enum MediaType {
   ANY = 0;
   AUDIO = 1;
@@ -37,6 +39,7 @@ message Event {
     VIDEO_SENDER_CONFIG_EVENT = 9;
     AUDIO_RECEIVER_CONFIG_EVENT = 10;
     AUDIO_SENDER_CONFIG_EVENT = 11;
+    AUDIO_ENCODER_CONFIG_EVENT = 12;

[terelius, 2016/12/22 16:04:24]

Your use of "CONFIG" is quite different from the other configs above. The send
and receive configs are construction-time settings that remain unchanged
throughout the call, and they are handled separately from all other events in
the log. Not sure it is worth changing the name, but have you considered
something like AUDIO_NETWORK_ADAPTATION_EVENT?

[minyue-webrtc, 2016/12/29 20:28:50]

Sure, I like your suggestion. 

   }
 
   // required - Indicates the type of this event
@@ -65,6 +68,10 @@ message Event {
 
   // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
   optional AudioSendConfig audio_sender_config = 11;
+
+  // optional - but required if type == AUDIO_ENCODER_CONFIG_EVENT
+  optional webrtc.audio_network_adaptor.debug_dump.EncoderRuntimeConfig
+      encoder_config = 12;

[terelius, 2016/12/22 16:04:24]

Field tag numbers in the range [1,15] require one byte less in the serialized
stream. I would therefore like to reserve them for the most frequent event
types.

[minyue-webrtc, 2016/12/29 20:28:50]

ok, that is nice. I'd pick 16 then.

 }
 
 message RtpPacket {