Added StopAecDump function to PeerConnectionFactory.

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);
 
+  // Stops AEC dump.
+  void StopAecDump();
+
   // 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();
 
@@ skipping 14 matching lines @@
 
   // webrtc::VoiceEngineObserver:
   void CallbackOnError(int channel_id, int errCode) override;
 
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
       bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
 
   void StartAecDump(const std::string& filename);
-  void StopAecDump();
   int CreateVoiceChannel(VoEWrapper* voe);
 
   static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_;
   int log_filter_;
@@ skipping 191 matching lines @@
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_