Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
 #include <list>
 #include <memory>
 #include <string>
 
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ skipping 72 matching lines @@
  public:
   BitrateEstimatorTest() : mock_voice_engine_(decoder_factory_),
                            receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &mock_voice_engine_;
     Call::Config config;
+    config.event_log = &event_log_;
     config.audio_state = AudioState::Create(audio_state_config);
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
     send_transport_.reset(new test::DirectTransport(sender_call_.get()));
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
     video_send_config_ = VideoSendStream::Config(send_transport_.get());
@@ skipping 121 matching lines @@
     BitrateEstimatorTest* test_;
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
     std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
+  webrtc::RtcEventLogNullImpl event_log_;

[stefan-webrtc, 2016/10/05 07:35:22]

This is a CallTest and should already have the event log.

[skvlad, 2016/10/06 01:31:38]

Done.

   testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
   LogObserver receiver_log_;
   std::unique_ptr<test::DirectTransport> send_transport_;
   std::unique_ptr<test::DirectTransport> receive_transport_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
@@ skipping 59 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc