Added functions on libjingle API to start and stop the recording of an RtcEventLog.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 90 matching lines @@
 
   VoEWrapper* voe() { return voe_wrapper_.get(); }
   int GetLastEngineError();
 
   // Set the external ADM. This can only be called before Init.
   bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
 
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
 
+  // Starts recording an RtcEventLog using an existing file until 10 minutes
+  // pass or the StopRtcEventLog function is called.
+  bool StartRtcEventLog(rtc::PlatformFile file);
+
+  // Stops recording the RtcEventLog.
+  void StopRtcEventLog();
+
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
 
  private:
   void Construct();
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
@@ skipping 223 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_