| Index: webrtc/voice_engine/transmit_mixer.h
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| diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
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| index 0aee1062316ce930c18a6bd59c09d4f8eb721b60..e0246cf4ceaa9595f6bf77a4d908dd20ab42e132 100644
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| --- a/webrtc/voice_engine/transmit_mixer.h
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| +++ b/webrtc/voice_engine/transmit_mixer.h
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| @@ -11,6 +11,7 @@
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|  #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
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|  #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
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|  
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| +#include "webrtc/base/criticalsection.h"
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|  #include "webrtc/base/scoped_ptr.h"
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|  #include "webrtc/common_audio/resampler/include/push_resampler.h"
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|  #include "webrtc/common_types.h"
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| @@ -210,8 +211,8 @@ private:
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|      bool _fileCallRecording;
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|      voe::AudioLevel _audioLevel;
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|      // protect file instances and their variables in MixedParticipants()
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| -    CriticalSectionWrapper& _critSect;
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| -    CriticalSectionWrapper& _callbackCritSect;
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| +    rtc::CriticalSection _critSect;
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| +    rtc::CriticalSection _callbackCritSect;
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|  
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|  #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
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|      webrtc::TypingDetection _typingDetection;
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| 
 |