Replace Atomic32 with webrtc/base/atomicops.h.

webrtc/video_engine/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_ENGINE_PAYLOAD_ROUTER_H_
 #define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
 
 #include <list>
 #include <vector>
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/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.
@@ skipping 24 matching lines @@
                     const RTPVideoHeader* rtp_video_hdr);
 
   // Configures current target bitrate per module. 'stream_bitrates' is assumed
   // to be in the same order as 'SetSendingRtpModules'.
   void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
 
   // 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; } }
+  void AddRef() { rtc::AtomicOps::Increment(&ref_count_); }
+  void Release() {
+    if (rtc::AtomicOps::Decrement(&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_;
 
   // Active sending RTP modules, in layer order.
   std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
   bool active_ GUARDED_BY(crit_.get());
 
-  Atomic32 ref_count_;
+  volatile int ref_count_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_