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

talk/app/webrtc/peerconnectioninterface.h

 /*
  * libjingle
  * Copyright 2012 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 601 matching lines @@
       CreateAudioTrack(const std::string& label,
                        AudioSourceInterface* source) = 0;
 
   // Starts AEC dump using existing file. Takes ownership of |file| and passes
   // it on to VoiceEngine (via other objects) immediately, which will take
   // the ownerhip. If the operation fails, the file will be closed.
   // TODO(grunell): Remove when Chromium has started to use AEC in each source.
   // http://crbug.com/264611.
   virtual bool StartAecDump(rtc::PlatformFile file) = 0;
 
+  // Starts RtcEventLog using existing file. Takes ownership of |file| and
+  // passes it on to VoiceEngine, which will take the ownership. If the
+  // operation fails the file will be closed. The logging will stop
+  // automatically after 10 minutes have passed, or when the StopRtcEventLog
+  // function is called.
+  // This function as well as the StopRtcEventLog don't really belong on this
+  // interface, this is a temporary solution until we move the logging object
+  // from inside voice engine to webrtc::Call, which will happen when the VoE
+  // restructuring effort is further along.
+  // TODO(ivoc): Move this into being:
+  //             PeerConnection => MediaController => webrtc::Call.
+  virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
+
+  // Stops logging the RtcEventLog.
+  virtual void StopRtcEventLog() = 0;
+
  protected:
   // Dtor and ctor protected as objects shouldn't be created or deleted via
   // this interface.
   PeerConnectionFactoryInterface() {}
   ~PeerConnectionFactoryInterface() {} // NOLINT
 };
 
 // Create a new instance of PeerConnectionFactoryInterface.
 rtc::scoped_refptr<PeerConnectionFactoryInterface>
 CreatePeerConnectionFactory();
 
 // Create a new instance of PeerConnectionFactoryInterface.
 // Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
 // |decoder_factory| transferred to the returned factory.
 rtc::scoped_refptr<PeerConnectionFactoryInterface>
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_