Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/api/statscollector_unittest.cc

 /*
  *  Copyright 2014 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 11 matching lines @@
 #include "webrtc/api/peerconnectionfactory.h"
 #include "webrtc/api/test/fakedatachannelprovider.h"
 #include "webrtc/api/test/fakevideotracksource.h"
 #include "webrtc/api/test/mock_peerconnection.h"
 #include "webrtc/api/test/mock_webrtcsession.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/network.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/test/mock_mediachannel.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::DoAll;
 using testing::Field;
@@ skipping 444 matching lines @@
   StatsCollectorTest()
       : worker_thread_(rtc::Thread::Current()),
         network_thread_(rtc::Thread::Current()),
         media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(new cricket::ChannelManager(media_engine_,
                                                      worker_thread_,
                                                      network_thread_)),
         media_controller_(
             webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
                                                      worker_thread_,
-                                                     channel_manager_.get())),
+                                                     channel_manager_.get(),
+                                                     &event_log_)),
         session_(media_controller_.get()) {
     // By default, we ignore session GetStats calls.
     EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false));
     // Add default returns for mock classes.
     EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
     EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
     EXPECT_CALL(pc_, session()).WillRepeatedly(Return(&session_));
     EXPECT_CALL(pc_, sctp_data_channels())
         .WillRepeatedly(ReturnRef(data_channels_));
   }
@@ skipping 237 matching lines @@
     EXPECT_EQ(rtc::SSLStreamAdapter::SslCipherSuiteToName(
                   internal::TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA),
               dtls_cipher_suite);
     std::string srtp_crypto_suite =
         ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
                           StatsReport::kStatsValueNameSrtpCipher);
     EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80),
               srtp_crypto_suite);
   }
 
+  webrtc::RtcEventLogNullImpl event_log_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
   cricket::FakeMediaEngine* media_engine_;
   std::unique_ptr<cricket::ChannelManager> channel_manager_;
   std::unique_ptr<webrtc::MediaControllerInterface> media_controller_;
   MockWebRtcSession session_;
   MockPeerConnection pc_;
   FakeDataChannelProvider data_channel_provider_;
   SessionStats session_stats_;
   rtc::scoped_refptr<webrtc::MediaStream> stream_;
@@ skipping 1118 matching lines @@
   cricket::VoiceSenderInfo new_voice_sender_info;
   InitVoiceSenderInfo(&new_voice_sender_info);
   cricket::VoiceMediaInfo new_stats_read;
   reports.clear();
   SetupAndVerifyAudioTrackStats(
       new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName,
       media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
 }
 
 }  // namespace webrtc