Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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/base/checks.h"
@@ skipping 364 matching lines @@
 const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
 const int CallTest::kNackRtpHistoryMs = 1000;
 
 BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
 }
 
 BaseTest::~BaseTest() {
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
-  return Call::Config();
+  Call::Config config;
+  config.event_log = &event_log_;
+  return config;
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
-  return Call::Config();
+  Call::Config config;
+  config.event_log = &event_log_;
+  return config;
 }
 
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
   return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
                              FakeNetworkPipe::Config());
 }
 
@@ skipping 44 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc