Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/call/rtc_event_log.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 32 matching lines @@
   virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
 
   // Starts logging for the specified duration to the specified file.
   // The logging will stop automatically after the specified duration.
   // If the file already exists it will be overwritten.
   // If the file cannot be opened, the RtcEventLog will not start logging.
   virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
 
   // Starts logging until either the 10 minute timer runs out or the StopLogging
   // function is called. The RtcEventLog takes ownership of the supplied
-  // rtc::PlatformFile.
+  // rtc::PlatformFile. The max_size_bytes argument is ignored, it is added here
+  // for future use.
+  virtual bool StartLogging(rtc::PlatformFile log_file,
+                            int64_t max_size_bytes) {
+    return StartLogging(log_file);
+  }
+
   virtual bool StartLogging(rtc::PlatformFile log_file) = 0;
 
   virtual void StopLogging() = 0;
 
   // Logs configuration information for webrtc::VideoReceiveStream
   virtual void LogVideoReceiveStreamConfig(
       const webrtc::VideoReceiveStream::Config& config) = 0;
 
   // Logs configuration information for webrtc::VideoSendStream
   virtual void LogVideoSendStreamConfig(
@@ skipping 22 matching lines @@
 
   // Reads an RtcEventLog file and returns true when reading was successful.
   // The result is stored in the given EventStream object.
   static bool ParseRtcEventLog(const std::string& file_name,
                                rtclog::EventStream* result);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_