Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/call.h

 /*
  *  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.
  */
 #ifndef WEBRTC_CALL_H_
 #define WEBRTC_CALL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 
 const char* Version();
 
@@ skipping 100 matching lines @@
   // of maximum for entire Call. This should be fixed along with the above.
   // Specifying a start bitrate (>0) will currently reset the current bitrate
   // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
+  virtual bool StartEventLog(rtc::PlatformFile log_file,
+                             int64_t max_size_bytes) = 0;
+

[the sun, 2016/03/23 10:26:45]

nit: remove blank line

[ivoc, 2016/03/23 14:27:26]

Done.

+  virtual void StopEventLog() = 0;
+
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_