Unify newapi::RtcpMode and RTCPMethod.

webrtc/call/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 94 matching lines @@
 namespace {
 // The functions in this namespace convert enums from the runtime format
 // that the rest of the WebRtc project can use, to the corresponding
 // serialized enum which is defined by the protobuf.
 
 // Do not add default return values to the conversion functions in this
 // unnamed namespace. The intention is to make the compiler warn if anyone
 // adds unhandled new events/modes/etc.
 
 rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(
-    newapi::RtcpMode rtcp_mode) {
+    RtcpMode rtcp_mode) {
   switch (rtcp_mode) {
-    case newapi::kRtcpCompound:
+    case RtcpMode::COMPOUND:
       return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
-    case newapi::kRtcpReducedSize:
+    case RtcpMode::REDUCED_SIZE:
       return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
+    case RtcpMode::OFF:
+      RTC_NOTREACHED();
+      return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
   }
   RTC_NOTREACHED();
   return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
 }
 
 rtclog::MediaType ConvertMediaType(MediaType media_type) {
   switch (media_type) {
     case MediaType::ANY:
       return rtclog::MediaType::ANY;
     case MediaType::AUDIO:
@@ skipping 265 matching lines @@
   return result->ParseFromString(dump_buffer);
 }
 
 #endif  // ENABLE_RTC_EVENT_LOG
 
 // RtcEventLog member functions.
 rtc::scoped_ptr<RtcEventLog> RtcEventLog::Create() {
   return rtc::scoped_ptr<RtcEventLog>(new RtcEventLogImpl());
 }
 }  // namespace webrtc