Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/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 68 matching lines @@
   optional BweProbeCluster probe_cluster = 17;
 
   // optional - but required if type == BWE_PROBE_RESULT_EVENT
   optional BweProbeResult probe_result = 18;
 }
 
 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;
+  optional MediaType type = 2 [deprecated = true];
 
   // required - The size of the packet including both payload and header.
   optional uint32 packet_length = 3;
 
   // required - The RTP header only.
   optional bytes header = 4;
 
   // optional - The probe cluster id.
   optional uint32 probe_cluster_id = 5;
 
   // Do not add code to log user payload data without a privacy review!
 }
 
 message RtcpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
 
-  // required
-  optional MediaType type = 2;
+  optional MediaType type = 2 [deprecated = true];
 
   // required - The whole packet including both payload and header.
   optional bytes packet_data = 3;
 }
 
 message AudioPlayoutEvent {
   // TODO(ivoc): Rename, we currently use the "remote" ssrc, i.e. identifying
   // the receive stream, while local_ssrc identifies the send stream, if any.
   // required - The SSRC of the audio stream associated with the playout event.
   optional uint32 local_ssrc = 2;
@@ skipping 187 matching lines @@
     INVALID_SEND_RECEIVE_RATIO = 2;
     TIMEOUT = 3;
   }
 
   // required - The result of this probing attempt.
   optional ResultType result = 2;
 
   // optional - but required if result == SUCCESS. The resulting bitrate in bps.
   optional uint64 bitrate_bps = 3;
 }