Remove mutable from rtc::CriticalSection members.

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 221 matching lines @@
                                      ApmDebugDumpThreadState* debug_state);
   int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
   // Writes Config message. If not |forced|, only writes the current config if
   // it is different from the last saved one; if |forced|, writes the config
   // regardless of the last saved.
   int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
       EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   // Critical section.
-  mutable rtc::CriticalSection crit_debug_;
+  rtc::CriticalSection crit_debug_;
 
   // Debug dump state.
   ApmDebugDumpState debug_dump_;
 #endif
 
   // Critical sections.
-  mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
-  mutable rtc::CriticalSection crit_capture_;
+  rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection crit_capture_;
 
   // Structs containing the pointers to the submodules.
   rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
   rtc::scoped_ptr<ApmPrivateSubmodules> private_submodules_
       GUARDED_BY(crit_capture_);
 
   // State that is written to while holding both the render and capture locks
   // but can be read without any lock being held.
   // As this is only accessed internally of APM, and all internal methods in APM
   // either are holding the render or capture locks, this construct is safe as
@@ skipping 77 matching lines @@
 
   struct ApmRenderState {
     rtc::scoped_ptr<AudioConverter> render_converter;
     rtc::scoped_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_