Event log cleanup in tests.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/video/video_quality_test.h"
@@ skipping 1566 matching lines @@
 
   FILE* graph_data_output_file = nullptr;
   if (!params_.analyzer.graph_data_output_filename.empty()) {
     graph_data_output_file =
         fopen(params_.analyzer.graph_data_output_filename.c_str(), "w");
     RTC_CHECK(graph_data_output_file)
         << "Can't open the file " << params_.analyzer.graph_data_output_filename
         << "!";
   }
 
-  webrtc::RtcEventLogNullImpl event_log;
-  Call::Config call_config(&event_log_);
+  Call::Config call_config(event_log_.get());
   call_config.bitrate_config = params.call.call_bitrate_config;
   CreateCalls(call_config, call_config);
 
   test::LayerFilteringTransport send_transport(
       params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params_.video.selected_tl, params_.ss.selected_sl);
   test::DirectTransport recv_transport(
       params_.pipe, receiver_call_.get(), MediaType::VIDEO);
 
   std::string graph_title = params_.analyzer.graph_title;
@@ skipping 130 matching lines @@
 
   *audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
 }
 
 void VideoQualityTest::RunWithRenderers(const Params& params) {
   params_ = params;
   CheckParams();
 
   // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
   // match the full stack tests.
-  webrtc::RtcEventLogNullImpl event_log;
-  Call::Config call_config(&event_log_);
+  Call::Config call_config(event_log_.get());
   call_config.bitrate_config = params_.call.call_bitrate_config;
 
   ::VoiceEngineState voe;
   if (params_.audio.enabled) {
     CreateVoiceEngine(&voe, decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
@@ skipping 133 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc