Set RtcpSender transport at construction.

webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.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 212 matching lines @@
  protected:
   RtcpSenderTest()
       : clock_(1335900000),
         receive_statistics_(ReceiveStatistics::Create(&clock_)) {
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &clock_;
     configuration.outgoing_transport = &test_transport_;
 
     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
-    rtcp_sender_.reset(
-        new RTCPSender(false, &clock_, receive_statistics_.get(), nullptr));
+    rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                      nullptr, &test_transport_));
     rtcp_sender_->SetSSRC(kSenderSsrc);
     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-    EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
   }
 
   void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
     RTPHeader header;
     header.ssrc = remote_ssrc;
     header.sequenceNumber = seq_num;
     header.timestamp = 12345;
     header.headerLength = 12;
     size_t kPacketLength = 100;
     receive_statistics_->IncomingPacket(header, kPacketLength, false);
@@ skipping 412 matching lines @@
     EXPECT_TRUE(rtcp_sender_->SendTimeOfXrRrReport(mid_ntp, &time_ms));
     EXPECT_EQ(clock_.CurrentNtpInMilliseconds(), time_ms);
     clock_.AdvanceTimeMilliseconds(1000);
   }
   // The first report should no longer be stored.
   EXPECT_FALSE(rtcp_sender_->SendTimeOfXrRrReport(initial_mid_ntp, &time_ms));
 }
 
 TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
   RtcpPacketTypeCounterObserverImpl observer;
-  rtcp_sender_.reset(
-      new RTCPSender(false, &clock_, receive_statistics_.get(), &observer));
+  rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                    &observer, &test_transport_));
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-  EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
   rtcp_sender_->SetRTCPStatus(kRtcpNonCompound);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
   EXPECT_EQ(1, parser()->pli()->num_packets());
   EXPECT_EQ(kRemoteSsrc, observer.ssrc_);
   EXPECT_EQ(1U, observer.counter_.pli_packets);
   EXPECT_EQ(clock_.TimeInMilliseconds(),
             observer.counter_.first_packet_time_ms);
 }
 
 TEST_F(RtcpSenderTest, SendTmmbr) {
@@ skipping 71 matching lines @@
   rtcp_sender_->SetREMBData(kBitrate, ssrcs);
   std::set<RTCPPacketType> packet_types;
   packet_types.insert(kRtcpRemb);
   packet_types.insert(kRtcpPli);
   EXPECT_EQ(0, rtcp_sender_->SendCompoundRTCP(feedback_state(), packet_types));
   EXPECT_EQ(1, parser()->remb_item()->num_packets());
   EXPECT_EQ(1, parser()->pli()->num_packets());
 }
 
 }  // namespace webrtc