Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 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 411 matching lines @@
   // Returns the current SignalingState.
   virtual SignalingState signaling_state() = 0;
 
   // TODO(bemasc): Remove ice_state when callers are changed to
   // IceConnection/GatheringState.
   // Returns the current IceState.
   virtual IceState ice_state() = 0;
   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) = 0;
+
+  // Stops logging the RtcEventLog.
+  virtual void StopRtcEventLog() = 0;
+
   // Terminates all media and closes the transport.
   virtual void Close() = 0;
 
  protected:
   // Dtor protected as objects shouldn't be deleted via this interface.
   ~PeerConnectionInterface() {}
 };
 
 // PeerConnection callback interface. Application should implement these
 // methods.
@@ skipping 113 matching lines @@
   // the ownerhip. If the operation fails, the file will be closed.
   // A maximum file size in bytes can be specified. When the file size limit is
   // reached, logging is stopped automatically. If max_size_bytes is set to a
   // value <= 0, no limit will be used, and logging will continue until the
   // StopAecDump function is called.
   virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
 
   // Stops logging the AEC dump.
   virtual void StopAecDump() = 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.
+  // 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.
   virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
 
-  // Stops logging the RtcEventLog.
+  // 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.
   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  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_