Compare serialized Configs.

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 158 matching lines @@
   std::list<ProcessingComponent*> component_list_;
   CriticalSectionWrapper* crit_;
   rtc::scoped_ptr<AudioBuffer> render_audio_;
   rtc::scoped_ptr<AudioBuffer> capture_audio_;
   rtc::scoped_ptr<AudioConverter> render_converter_;
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
   int WriteMessageToDebugFile();
   int WriteInitMessage();
-  int WriteConfigMessage();
-
-  // Updates |config_| and returns true if any change is made.
-  bool UpdateCurrentConfig();
+  int WriteConfigMessageIfChanged();
 
   rtc::scoped_ptr<FileWrapper> debug_file_;
   rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
   std::string event_str_;  // Memory for protobuf serialization.
   rtc::scoped_ptr<audioproc::Config> config_;
+  std::string last_serialized_config_;
 #endif
 
   // Format of processing streams at input/output call sites.
   ProcessingConfig api_format_;
 
   // Only the rate and samples fields of fwd_proc_format_ are used because the
   // forward processing number of channels is mutable and is tracked by the
   // capture_audio_.
   StreamConfig fwd_proc_format_;
   StreamConfig rev_proc_format_;
@@ skipping 22 matching lines @@
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   bool intelligibility_enabled_;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_