Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.

webrtc/video/vie_remb.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_VIDEO_VIE_REMB_H_
 #define WEBRTC_VIDEO_VIE_REMB_H_
 
 #include <list>
 #include <utility>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
 class ProcessThread;
 class RtpRtcp;
 
 class VieRemb : public RemoteBitrateObserver {
  public:
   explicit VieRemb(Clock* clock);
   ~VieRemb();
 
   // Called to add a receive channel to include in the REMB packet.
   void AddReceiveChannel(RtpRtcp* rtp_rtcp);
@@ skipping 15 matching lines @@
   // estimate has decreased or if no RTCP REMB packet has been sent for
   // a certain time interval.
   // Implements RtpReceiveBitrateUpdate.
   virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
                                        unsigned int bitrate);
 
  private:
   typedef std::list<RtpRtcp*> RtpModules;
 
   Clock* const clock_;
-  rtc::scoped_ptr<CriticalSectionWrapper> list_crit_;
+  mutable rtc::CriticalSection list_crit_;
 
   // The last time a REMB was sent.
   int64_t last_remb_time_;
   unsigned int last_send_bitrate_;
 
   // All RtpRtcp modules to include in the REMB packet.
   RtpModules receive_modules_;
 
   // All modules that can send REMB RTCP.
   RtpModules rtcp_sender_;
 
   // The last bitrate update.
   unsigned int bitrate_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_REMB_H_