Unify newapi::RtcpMode and RTCPMethod.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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 424 matching lines @@
   }
 
   return rtp_states;
 }
 
 void VideoSendStream::SignalNetworkState(NetworkState state) {
   // When network goes up, enable RTCP status before setting transmission state.
   // When it goes down, disable RTCP afterwards. This ensures that any packets
   // sent due to the network state changed will not be dropped.
   if (state == kNetworkUp)
-    vie_channel_->SetRTCPMode(kRtcpCompound);
+    vie_channel_->SetRTCPMode(RtcpMode::kCompound);
   vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
   if (state == kNetworkDown)
-    vie_channel_->SetRTCPMode(kRtcpOff);
+    vie_channel_->SetRTCPMode(RtcpMode::kOff);
 }
 
 int64_t VideoSendStream::GetRtt() const {
   webrtc::RtcpStatistics rtcp_stats;
   uint16_t frac_lost;
   uint32_t cumulative_lost;
   uint32_t extended_max_sequence_number;
   uint32_t jitter;
   int64_t rtt_ms;
   if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
@@ skipping 43 matching lines @@
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc