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

webrtc/video/call_stats.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_CALL_STATS_H_
 #define WEBRTC_VIDEO_CALL_STATS_H_
 
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 class CallStatsObserver;
-class CriticalSectionWrapper;
 class RtcpRttStats;
 
 // CallStats keeps track of statistics for a call.
 class CallStats : public Module {
  public:
   friend class RtcpObserver;
 
   explicit CallStats(Clock* clock);
   ~CallStats();
 
@@ skipping 18 matching lines @@
   };
 
  protected:
   void OnRttUpdate(int64_t rtt);
 
   int64_t avg_rtt_ms() const;
 
  private:
   Clock* const clock_;
   // Protecting all members.
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
   // Observer receiving statistics updates.
   rtc::scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
   // The last time 'Process' resulted in statistic update.
   int64_t last_process_time_;
   // The last RTT in the statistics update (zero if there is no valid estimate).
   int64_t max_rtt_ms_;
   int64_t avg_rtt_ms_;
 
   // All Rtt reports within valid time interval, oldest first.
   std::list<RttTime> reports_;
 
   // Observers getting stats reports.
   std::list<CallStatsObserver*> observers_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_CALL_STATS_H_