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

talk/app/webrtc/peerconnectionfactory.h

 /*
  * libjingle
  * Copyright 2011 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 61 matching lines @@
       const MediaConstraintsInterface* constraints) override;
 
   rtc::scoped_refptr<VideoTrackInterface>
       CreateVideoTrack(const std::string& id,
                        VideoSourceInterface* video_source) override;
 
   rtc::scoped_refptr<AudioTrackInterface>
       CreateAudioTrack(const std::string& id,
                        AudioSourceInterface* audio_source) override;
 
-  bool StartAecDump(rtc::PlatformFile file) override;
+  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
   bool StartRtcEventLog(rtc::PlatformFile file) override;
   void StopRtcEventLog() override;
 
   virtual webrtc::MediaControllerInterface* CreateMediaController() const;
   virtual rtc::Thread* signaling_thread();
   virtual rtc::Thread* worker_thread();
   const Options& options() const { return options_; }
 
  protected:
   PeerConnectionFactory();
@@ skipping 25 matching lines @@
   // injected any. In that case, video engine will use the internal SW decoder.
   rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
       video_decoder_factory_;
 
   rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_