Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtp_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.
  */
 
 /*
  * This file includes unit tests for the RTPSender.
  */
 
 #include <list>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_proxy.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/stl_util.h"
 #include "webrtc/test/mock_transport.h"
@@ skipping 91 matching lines @@
     : public TransportSequenceNumberAllocator {
  public:
   MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
 };
 
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_rtc_event_log_(),
+        event_log_proxy(&mock_rtc_event_log_),
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
-        kMarkerBit(true) {
-  }
+        kMarkerBit(true) {}
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                     pacer ? &mock_paced_sender_ : nullptr,
                                     &seq_num_allocator_, nullptr, nullptr,
-                                    nullptr, nullptr, &mock_rtc_event_log_));
+                                    nullptr, nullptr, &event_log_proxy));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtcEventLog mock_rtc_event_log_;
+  RtcEventLogProxy event_log_proxy;
   MockRtpPacketSender mock_paced_sender_;
   MockTransportSequenceNumberAllocator seq_num_allocator_;
   rtc::scoped_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
   uint8_t packet_[kMaxPacketLength];
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
     VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
@@ skipping 783 matching lines @@
   EXPECT_EQ(timestamp, rtp_header.timestamp);
   // Verify transmission time offset. This packet is sent without delay.
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
-  rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, nullptr, &mock_paced_sender_, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_));
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
+                                  &mock_paced_sender_, nullptr, nullptr,
+                                  nullptr, nullptr, nullptr, &event_log_proxy));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
@@ skipping 557 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc