Added StopAecDump function to PeerConnectionFactory.

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;
 
+  // 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:
@@ skipping 21 matching lines @@
 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_