Move RtcEventLog object from inside VoiceEngine to Call.

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 220 matching lines @@
   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, nullptr, &test_transport_));
+                                      nullptr, &test_transport_));
     rtcp_sender_->SetSSRC(kSenderSsrc);
     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
   }
 
   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;
@@ skipping 387 matching lines @@
   EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
   rtcp_sender_->SendRtcpXrReceiverReferenceTime(false);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
   EXPECT_EQ(0, parser()->xr_header()->num_packets());
   EXPECT_EQ(0, parser()->rrtr()->num_packets());
 }
 
 TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
   RtcpPacketTypeCounterObserverImpl observer;
   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
-                                    &observer, nullptr, &test_transport_));
+                                    &observer, &test_transport_));
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   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);
 }
 
@@ skipping 99 matching lines @@
         EXPECT_TRUE(is_last_packet) <<
           "Bye packet should be last in a compound RTCP packet.";
       }
     }
 
     return true;
   }));
 
   // Re-configure rtcp_sender_ with mock_transport_
   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
-                                    nullptr, nullptr, &mock_transport));
+                                    nullptr, &mock_transport));
   rtcp_sender_->SetSSRC(kSenderSsrc);
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
 
   // Set up XR VoIP metric to be included with BYE
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   RTCPVoIPMetric metric;
   EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
 }
 
 }  // namespace webrtc