Added option to specify a maximum file size when recording an AEC dump.

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 76 matching lines @@
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
   bool AdjustAgcLevel(int delta);
 
   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);
+  // Starts AEC dump using an existing file. A maximum file size in bytes can be
+  // specified. When the maximum file size is reached, logging is stopped and
+  // the file is closed. If max_size_bytes is set to <= 0, no limit will be
+  // used.
+  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // 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();
@@ skipping 171 matching lines @@
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_