Remove CriticalSectionWrapper from audio conference mixer.

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
 #include <memory>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class AudioProcessing;
-class CriticalSectionWrapper;
 
 struct FrameAndMuteInfo {
   FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
   AudioFrame* frame;
   bool muted;
 };
 
 typedef std::list<FrameAndMuteInfo> AudioFrameList;
 typedef std::list<MixerParticipant*> MixerParticipantList;
 
@@ skipping 104 matching lines @@
                         const AudioFrameList& audioFrameList) const;
 
     // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
     // record will be kept of this mix (e.g. the corresponding MixerParticipants
     // will not be marked as IsMixed()
     int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
                                    const AudioFrameList& audioFrameList) const;
 
     bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
-    std::unique_ptr<CriticalSectionWrapper> _crit;
-    std::unique_ptr<CriticalSectionWrapper> _cbCrit;
+    const rtc::CriticalSection _crit;
+    const rtc::CriticalSection _cbCrit;
 
     int32_t _id;
 
     Frequency _minimumMixingFreq;
 
     // Mix result callback
     AudioMixerOutputReceiver* _mixReceiver;
 
     // The current sample frequency and sample size when mixing.
     Frequency _outputFrequency;
@@ skipping 20 matching lines @@
     // Counter keeping track of concurrent calls to process.
     // Note: should never be higher than 1 or lower than 0.
     int16_t _processCalls;
 
     // Used for inhibiting saturation in mixing.
     std::unique_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_