Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call/rampup_tests.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_CALL_RAMPUP_TESTS_H_
 #define WEBRTC_CALL_RAMPUP_TESTS_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/event.h"
 #include "webrtc/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/test/call_test.h"
 
 namespace webrtc {
 
 static const int kTransmissionTimeOffsetExtensionId = 6;
 static const int kAbsSendTimeExtensionId = 7;
 static const int kTransportSequenceNumberExtensionId = 8;
 static const unsigned int kSingleStreamTargetBps = 1000000;
 
 class Clock;
@@ skipping 20 matching lines @@
                        size_t* total_packets_sent,
                        size_t* total_sent,
                        size_t* padding_sent,
                        size_t* media_sent) const;
 
   void ReportResult(const std::string& measurement,
                     size_t value,
                     const std::string& units) const;
   void TriggerTestDone();
 
+  webrtc::RtcEventLogNullImpl event_log_;
   rtc::Event event_;
   Clock* const clock_;
   FakeNetworkPipe::Config forward_transport_config_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
   const bool rtx_;
   const bool red_;
   Call* sender_call_;
   VideoSendStream* send_stream_;
   test::PacketTransport* send_transport_;
@@ skipping 58 matching lines @@
   std::string GetModifierString() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_