Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/video/video_send_stream_tests.cc

Index: webrtc/video/video_send_stream_tests.cc
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index e6aabb9c89d9cfdc62d936bf8a0cbc1df2666674..f3ecf1f337316df100c4c3d11caf1b55c673fd4a 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -60,7 +60,9 @@ class VideoSendStreamTest : public test::CallTest {
 };
 
 TEST_F(VideoSendStreamTest, CanStartStartedStream) {
+  webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config;
+  call_config.event_log = &event_log;
   CreateSenderCall(call_config);
 
   test::NullTransport transport;
@@ -72,7 +74,9 @@ TEST_F(VideoSendStreamTest, CanStartStartedStream) {
 }
 
 TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
+  webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config;
+  call_config.event_log = &event_log;
   CreateSenderCall(call_config);
 
   test::NullTransport transport;
@@ -738,6 +742,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       Call::Config config;
       const int kMinBitrateBps = 30000;
       config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
+      config.event_log = &event_log_;
       return config;
     }
 
@@ -769,6 +774,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
     }
 
+    webrtc::RtcEventLogNullImpl event_log_;
     std::unique_ptr<internal::TransportAdapter> transport_adapter_;
     test::ConfigurableFrameSizeEncoder encoder_;
 
@@ -1438,7 +1444,10 @@ TEST_F(VideoSendStreamTest,
     int last_initialized_frame_height_ GUARDED_BY(&crit_);
   };
 
-  CreateSenderCall(Call::Config());
+  webrtc::RtcEventLogNullImpl event_log;
+  Call::Config call_config;
+  call_config.event_log = &event_log;
+  CreateSenderCall(call_config);
   test::NullTransport transport;
   CreateSendConfig(1, 0, &transport);
   EncoderObserver encoder;
@@ -1494,7 +1503,10 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
     int start_bitrate_kbps_ GUARDED_BY(crit_);
   };
 
-  CreateSenderCall(Call::Config());
+  webrtc::RtcEventLogNullImpl event_log;
+  Call::Config call_config;
+  call_config.event_log = &event_log;
+  CreateSenderCall(call_config);
 
   test::NullTransport transport;
   CreateSendConfig(1, 0, &transport);
@@ -1575,7 +1587,10 @@ TEST_F(VideoSendStreamTest, VideoSendStreamStopSetEncoderRateToZero) {
     int bitrate_kbps_ GUARDED_BY(crit_);
   };
 
-  CreateSenderCall(Call::Config());
+  webrtc::RtcEventLogNullImpl event_log;
+  Call::Config call_config;
+  call_config.event_log = &event_log;
+  CreateSenderCall(call_config);
 
   test::NullTransport transport;
   CreateSendConfig(1, 0, &transport);
@@ -1630,7 +1645,10 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
   };
 
   // Initialize send stream.
-  CreateSenderCall(Call::Config());
+  webrtc::RtcEventLogNullImpl event_log;
+  Call::Config call_config;
+  call_config.event_log = &event_log;
+  CreateSenderCall(call_config);
 
   test::NullTransport transport;
   CreateSendConfig(1, 0, &transport);
@@ -2270,6 +2288,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
 
     Call::Config GetSenderCallConfig() override {
       Call::Config config;
+      config.event_log = &event_log_;
       config.bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
       config.bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
       config.bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
@@ -2351,6 +2370,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
       WaitForSetRates(kIncreasedStartBitrateKbps);
     }
 
+    webrtc::RtcEventLogNullImpl event_log_;
     rtc::Event init_encode_event_;
     rtc::Event bitrate_changed_event_;
     rtc::CriticalSection crit_;