Revert of Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/api/peerconnectioninterface.h

Index: webrtc/api/peerconnectioninterface.h
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index e57435559e25c6a1254ad46b801afeb91e73f6ff..0ac37e1564e935584913bd0bebe8a1f720e2f6e1 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -489,17 +489,6 @@
   virtual IceConnectionState ice_connection_state() = 0;
   virtual IceGatheringState ice_gathering_state() = 0;
 
-  // Starts RtcEventLog using existing file. Takes ownership of |file| and
-  // passes it on to Call, 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.
-  virtual bool StartRtcEventLog(rtc::PlatformFile file,
-                                int64_t max_size_bytes) = 0;
-
-  // Stops logging the RtcEventLog.
-  virtual void StopRtcEventLog() = 0;
-
   // Terminates all media and closes the transport.
   virtual void Close() = 0;
 
@@ -666,19 +655,25 @@
   // Stops logging the AEC dump.
   virtual void StopAecDump() = 0;
 
-  // This function is deprecated and will be removed when Chrome is updated to
-  // use the equivalent function on PeerConnectionInterface.
-  // TODO(ivoc) Remove after Chrome is updated.
+  // 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. A maximum filesize in bytes can be set, the logging
+  // will be stopped before exceeding this limit. If max_size_bytes is set to a
+  // value <= 0, no limit will be used.
+  // 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,
                                 int64_t max_size_bytes) = 0;
-  // This function is deprecated and will be removed when Chrome is updated to
-  // use the equivalent function on PeerConnectionInterface.
-  // TODO(ivoc) Remove after Chrome is updated.
+  // Deprecated, use the version above.
   virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
 
-  // This function is deprecated and will be removed when Chrome is updated to
-  // use the equivalent function on PeerConnectionInterface.
-  // TODO(ivoc) Remove after Chrome is updated.
+  // Stops logging the RtcEventLog.
   virtual void StopRtcEventLog() = 0;
 
  protected: