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

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 599 matching lines @@
                        VideoSourceInterface* source) = 0;
 
   // Creates an new AudioTrack. At the moment |source| can be NULL.
   virtual rtc::scoped_refptr<AudioTrackInterface>
       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;
+  // 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;
 
   // 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.
@@ skipping 22 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_