Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call/call_perf_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 <algorithm>
 #include <limits>
 #include <memory>
 #include <string>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/config.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
@@ skipping 126 matching lines @@
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int recv_channel_id = voe_base->CreateChannel();
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
-  Call::Config sender_config;
+  Call::Config sender_config(&event_log_);
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
-  Call::Config receiver_config;
+  Call::Config receiver_config(&event_log_);
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
   // Helper class to ensure we deliver correct media_type to the receiving call.
   class MediaTypePacketReceiver : public PacketReceiver {
    public:
     MediaTypePacketReceiver(PacketReceiver* packet_receiver,
@@ skipping 497 matching lines @@
       last_set_bitrate_ = new_target_bitrate_kbps;
       if (encoder_inits_ == 2 &&
           new_target_bitrate_kbps > kReconfigureThresholdKbps) {
         time_to_reconfigure_.Set();
       }
       return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
     }
 
     Call::Config GetSenderCallConfig() override {
       Call::Config config = EndToEndTest::GetSenderCallConfig();
+      config.event_log = &event_log_;
       config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
       return config;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config->video_stream_factory =
@@ skipping 29 matching lines @@
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc