Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.

webrtc/modules/audio_coding/test/Channel.h

Index: webrtc/modules/audio_coding/test/Channel.h
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h
index b047aa9909f1bac676f2c6ac6749c7ed72c0e238..3dcd499c03cdc6bfe5cf5bb1cf27239bdc10f38c 100644
--- a/webrtc/modules/audio_coding/test/Channel.h
+++ b/webrtc/modules/audio_coding/test/Channel.h
@@ -13,14 +13,13 @@
 
 #include <stdio.h>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 #define MAX_NUM_PAYLOADS   50
 #define MAX_NUM_FRAMESIZES  6
 
@@ -101,7 +100,7 @@ class Channel : public AudioPacketizationCallback {
   // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
   uint8_t _payloadData[60 * 32 * 2 * 2];
 
-  CriticalSectionWrapper* _channelCritSect;
+  mutable rtc::CriticalSection _channelCritSect;
   FILE* _bitStreamFile;
   bool _saveBitStream;
   int16_t _lastPayloadType;