Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 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.
  */
 
 #include <memory>
 #include <string>
 #include <utility>
 
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/fakemediacontroller.h"
 #include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/rtpreceiver.h"
 #include "webrtc/api/rtpsender.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/test/fakevideotracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/api/videotracksource.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/engine/fakewebrtccall.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 
 using ::testing::_;
 using ::testing::Exactly;
@@ skipping 12 matching lines @@
 
 class RtpSenderReceiverTest : public testing::Test {
  public:
   RtpSenderReceiverTest()
       :  // Create fake media engine/etc. so we can create channels to use to
         // test RtpSenders/RtpReceivers.
         media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(media_engine_,
                          rtc::Thread::Current(),
                          rtc::Thread::Current()),
-        fake_call_(webrtc::Call::Config()),
+        fake_call_(Call::Config(&event_log_)),
         fake_media_controller_(&channel_manager_, &fake_call_),
         stream_(MediaStream::Create(kStreamLabel1)) {
     // Create channels to be used by the RtpSenders and RtpReceivers.
     channel_manager_.Init();
     voice_channel_ = channel_manager_.CreateVoiceChannel(
         &fake_media_controller_, &fake_transport_controller_, cricket::CN_AUDIO,
         nullptr, false, cricket::AudioOptions());
     video_channel_ = channel_manager_.CreateVideoChannel(
         &fake_media_controller_, &fake_transport_controller_, cricket::CN_VIDEO,
         nullptr, false, cricket::VideoOptions());
@@ skipping 141 matching lines @@
     EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
     EXPECT_EQ(0, volume);
   }
 
   void VerifyVideoChannelNoOutput() {
     // Verify that the media channel's sink is reset.
     EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc));
   }
 
  protected:
+  webrtc::RtcEventLogNullImpl event_log_;
   cricket::FakeMediaEngine* media_engine_;
   cricket::FakeTransportController fake_transport_controller_;
   cricket::ChannelManager channel_manager_;
   cricket::FakeCall fake_call_;
   cricket::FakeMediaController fake_media_controller_;
   cricket::VoiceChannel* voice_channel_;
   cricket::VideoChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_media_channel_;
   cricket::FakeVideoMediaChannel* video_media_channel_;
   rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
@@ skipping 380 matching lines @@
   CreateVideoRtpReceiver();
 
   RtpParameters params = video_rtp_receiver_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
   EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
 
   DestroyVideoRtpReceiver();
 }
 
 }  // namespace webrtc