Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/call/mock/mock_rtc_event_log.h

 /*
  *  Copyright (c) 2016 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_MOCK_MOCK_RTC_EVENT_LOG_H_
 #define WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_
 
 #include <string>
 
 #include "testing/gmock/include/gmock/gmock.h"
 
 #include "webrtc/call/rtc_event_log.h"
 
 namespace webrtc {
 
 class MockRtcEventLog : public RtcEventLog {
  public:
   MOCK_METHOD1(SetBufferDuration, void(int64_t buffer_duration_us));
 
   MOCK_METHOD2(StartLogging,
                void(const std::string& file_name, int duration_ms));
 
-  MOCK_METHOD1(StartLogging, bool(rtc::PlatformFile log_file));
+  MOCK_METHOD2(StartLogging,
+               bool(rtc::PlatformFile log_file, int64_t max_size_bytes));
 
   MOCK_METHOD0(StopLogging, void());
 
   MOCK_METHOD1(LogVideoReceiveStreamConfig,
                void(const webrtc::VideoReceiveStream::Config& config));
 
   MOCK_METHOD1(LogVideoSendStreamConfig,
                void(const webrtc::VideoSendStream::Config& config));
 
   MOCK_METHOD4(LogRtpHeader,
@@ skipping 12 matching lines @@
 
   MOCK_METHOD3(LogBwePacketLossEvent,
                void(int32_t bitrate,
                     uint8_t fraction_loss,
                     int32_t total_packets));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_