Added support for logging the SSRC corresponding to AudioPlayout events.

webrtc/video/rtc_event_log.proto

 syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.rtclog;
 
 
 enum MediaType {
   ANY = 0;
   AUDIO = 1;
   VIDEO = 2;
   DATA = 3;
@@ skipping 88 matching lines @@
   // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM.
   enum EventType {
     UNKNOWN_EVENT = 0;
     LOG_START = 1;
     LOG_END = 2;
     AUDIO_PLAYOUT = 3;
   }
 
   // required
   optional EventType type = 1;
+
+  // required if type == AUDIO_PLAYOUT
+  optional uint32 local_ssrc = 2;
 }
 
 
 // TODO(terelius): Video and audio streams could in principle share SSRC,
 // so identifying a stream based only on SSRC might not work.
 // It might be better to use a combination of SSRC and media type
 // or SSRC and port number, but for now we will rely on SSRC only.
 message VideoReceiveConfig {
   // required - Synchronization source (stream identifier) to be received.
   optional uint32 remote_ssrc = 1;
@@ skipping 100 matching lines @@
 
 
 message AudioReceiveConfig {
   // TODO(terelius): Add audio-receive config.
 }
 
 
 message AudioSendConfig {
   // TODO(terelius): Add audio-receive config.
 }