Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call/rampup_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.
  */
 
@@ skipping 50 matching lines @@
     for (size_t i = 0; i < video_ssrcs_.size(); ++i)
       rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
   }
 }
 
 RampUpTester::~RampUpTester() {
   event_.Set();
 }
 
 Call::Config RampUpTester::GetSenderCallConfig() {
-  Call::Config call_config;
+  Call::Config call_config(&event_log_);
   if (start_bitrate_bps_ != 0) {
     call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
   }
   call_config.bitrate_config.min_bitrate_bps = 10000;
   return call_config;
 }
 
 void RampUpTester::OnVideoStreamsCreated(
     VideoSendStream* send_stream,
     const std::vector<VideoReceiveStream*>& receive_streams) {
@@ skipping 293 matching lines @@
     // An audio send stream doesn't have bitrate stats, so the call send BW is
     // currently used instead.
     int transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps;
     EvolveTestState(transmit_bitrate_bps, false);
   }
 
   return !event_.Wait(kPollIntervalMs);
 }
 
 Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
-  Call::Config config;
+  Call::Config config(&event_log_);
   config.bitrate_config.min_bitrate_bps = 10000;
   return config;
 }
 
 std::string RampUpDownUpTester::GetModifierString() const {
   std::string str("_");
   if (num_video_streams_ > 0) {
     std::ostringstream s;
     s << num_video_streams_;
     str += s.str();
@@ skipping 224 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumberUri, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc