Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call/call_unittest.cc

Index: webrtc/call/call_unittest.cc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 2fe4fdec4372b41a8f4a966e3372cad815f4afde..2eaaeb0f17fbfc65d9f9e4a09507f385fb768213 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -13,6 +13,7 @@
 
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ -26,6 +27,7 @@ struct CallHelper {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     webrtc::Call::Config config;
+    config.event_log = &event_log_;
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
     call_.reset(webrtc::Call::Create(config));
   }
@@ -34,6 +36,7 @@ struct CallHelper {
 
  private:
   testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
+  webrtc::RtcEventLogNullImpl event_log_;
   std::unique_ptr<webrtc::Call> call_;
 };
 }  // namespace