Unify newapi::RtcpMode and RTCPMethod.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc

 /*
  *  Copyright (c) 2012 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 35 matching lines @@
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     configuration.outgoing_transport = transport_;
 
     video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
         &fake_clock, receiver_, NULL, &rtp_payload_registry_));
 
-    video_module_->SetRTCPStatus(kRtcpCompound);
+    video_module_->SetRTCPStatus(RtcpMode::kCompound);
     video_module_->SetSSRC(test_ssrc_);
     rtp_receiver_->SetNACKStatus(kNackRtcp);
     video_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
 
     transport_->SetSendModule(video_module_, &rtp_payload_registry_,
                               rtp_receiver_.get(), receive_statistics_.get());
 
     VideoCodec video_codec;
     memset(&video_codec, 0, sizeof(video_codec));
@@ skipping 115 matching lines @@
                                                    payload_specific, true));
       EXPECT_EQ(0u, receiver_->payload_size());
       EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
     }
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
   }
 }
 
 }  // namespace webrtc