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

webrtc/video/payload_router.h

 /*
  *  Copyright (c) 2015 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_PAYLOAD_ROUTER_H_
 #define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
 
 #include <list>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
 class RTPFragmentationHeader;
 class RtpRtcp;
 struct RTPVideoHeader;
 
 // PayloadRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
 class PayloadRouter {
  public:
   PayloadRouter();
   ~PayloadRouter();
@@ skipping 26 matching lines @@
   // Returns the maximum allowed data payload length, given the configured MTU
   // and RTP headers.
   size_t MaxPayloadLength() const;
 
   void AddRef() { ++ref_count_; }
   void Release() { if (--ref_count_ == 0) { delete this; } }
 
  private:
   // TODO(mflodman): When the new video API has launched, remove crit_ and
   // assume rtp_modules_ will never change during a call.
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
 
   // Active sending RTP modules, in layer order.
-  std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
-  bool active_ GUARDED_BY(crit_.get());
+  std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_);
+  bool active_ GUARDED_BY(crit_);
 
   Atomic32 ref_count_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_