Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/

webrtc/voice_engine/output_mixer.h

Index: webrtc/voice_engine/output_mixer.h
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index 91387e6256ec4160d173ac287dc5255f2c6e3b14..3287860f7f34b46751a2f15797afcc2b62f3b3f3 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
 #define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
@@ -23,7 +24,6 @@
 namespace webrtc {
 
 class AudioProcessing;
-class CriticalSectionWrapper;
 class FileWrapper;
 class VoEMediaProcess;
 
@@ -108,10 +108,9 @@ private:
     Statistics* _engineStatisticsPtr;
     AudioProcessing* _audioProcessingModulePtr;
 
-    // owns
-    CriticalSectionWrapper& _callbackCritSect;
+    rtc::CriticalSection _callbackCritSect;
     // protect the _outputFileRecorderPtr and _outputFileRecording
-    CriticalSectionWrapper& _fileCritSect;
+    rtc::CriticalSection _fileCritSect;
     AudioConferenceMixer& _mixerModule;
     AudioFrame _audioFrame;
     // Converts mixed audio to the audio device output rate.