Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  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.
  */
 
@@ skipping 220 matching lines @@
                                size_t length,
                                const webrtc::PacketTime& packet_time) override;
 
   webrtc::Call::Stats GetStats() const override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(webrtc::NetworkState state) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
+  bool StartEventLog(rtc::PlatformFile log_file,
+                     int64_t max_size_bytes) override;
+  void StopEventLog() override;
+
   webrtc::Call::Config config_;
   webrtc::NetworkState network_state_;
   rtc::SentPacket last_sent_packet_;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_