Unify newapi::RtcpMode and RTCPMethod.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.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 179 matching lines @@
     configuration.outgoing_transport = &transport_;
     rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
 
     rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
 
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
         &fake_clock, &receiver_, rtp_feedback_.get(),
         &rtp_payload_registry_));
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
-    rtp_rtcp_module_->SetRTCPStatus(kRtcpCompound);
+    rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_receiver_->SetNACKStatus(kNackRtcp);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
     rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
     rtp_rtcp_module_->SetStartTimestamp(111111);
 
     transport_.SetSendModule(rtp_rtcp_module_, &rtp_payload_registry_,
                              rtp_receiver_.get());
 
     VideoCodec video_codec;
@@ skipping 131 matching lines @@
 
 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
       *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }