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

webrtc/modules/audio_coding/test/Channel.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_CODING_TEST_CHANNEL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
 
 #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
 
 // TODO(turajs): Write constructor for this structure.
 struct ACMTestFrameSizeStats {
   uint16_t frameSizeSample;
   size_t maxPayloadLen;
   uint32_t numPackets;
   uint64_t totalPayloadLenByte;
   uint64_t totalEncodedSamples;
@@ skipping 60 matching lines @@
   }
 
  private:
   void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
 
   AudioCodingModule* _receiverACM;
   uint16_t _seqNo;
   // 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;
   ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
   bool _isStereo;
   WebRtcRTPHeader _rtpInfo;
   bool _leftChannel;
   uint32_t _lastInTimestamp;
   bool _useLastFrameSize;
   uint32_t _lastFrameSizeSample;
   // FEC Test variables
   int16_t _packetLoss;
   bool _useFECTestWithPacketLoss;
   uint64_t _beginTime;
   uint64_t _totalBytes;
 
   // External timing info, defaulted to -1. Only used if they are
   // non-negative.
   int64_t external_send_timestamp_;
   int32_t external_sequence_number_;
   int num_packets_to_drop_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_