Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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_VIDEO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
 
 #include <list>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/onetimeevent.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.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/modules/rtp_rtcp/source/video_codec_information.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class CriticalSectionWrapper;
 
 class RTPSenderVideo {
  public:
   RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender);
   virtual ~RTPSenderVideo();
 
   virtual RtpVideoCodecTypes VideoCodecType() const;
 
   size_t FECPacketOverhead() const;
 
@@ skipping 49 matching lines @@
                             const size_t rtpHeaderLength,
                             uint16_t video_seq_num,
                             const uint32_t capture_timestamp,
                             int64_t capture_time_ms,
                             StorageType media_packet_storage,
                             bool protect);
 
   RTPSenderInterface& _rtpSender;
 
   // Should never be held when calling out of this class.
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  const rtc::CriticalSection crit_;
 
   RtpVideoCodecTypes _videoType;
   int32_t _retransmissionSettings GUARDED_BY(crit_);
 
   // FEC
   ForwardErrorCorrection fec_;
   bool fec_enabled_ GUARDED_BY(crit_);
   int8_t red_payload_type_ GUARDED_BY(crit_);
   int8_t fec_payload_type_ GUARDED_BY(crit_);
   FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
   FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
   ProducerFec producer_fec_ GUARDED_BY(crit_);
 
   // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
   // and any padding overhead.
   Bitrate _fecOverheadRate;
   // Bitrate used for video payload and RTP headers
   Bitrate _videoBitrate;
   OneTimeEvent first_frame_sent_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_