Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class RTPSenderAudio : public DTMFqueue {
  public:
@@ skipping 41 matching lines @@
       uint32_t dtmfTimeStamp,
       uint16_t duration,
       bool markerBit);  // set on first packet in talk burst
 
   bool MarkerBit(const FrameType frameType, const int8_t payloadType);
 
  private:
   Clock* const _clock;
   RTPSender* const _rtpSender;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+  rtc::CriticalSection _sendAudioCritsect;
 
   uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
   // DTMF
   bool _dtmfEventIsOn;
   bool _dtmfEventFirstPacketSent;
   int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
   uint32_t _dtmfTimestamp;
   uint8_t _dtmfKey;
   uint32_t _dtmfLengthSamples;
@@ skipping 12 matching lines @@
   int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
 
   // Audio level indication
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
   OneTimeEvent first_packet_sent_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_