Replace Atomic32 with webrtc/base/atomicops.h.

webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h

 /*
  *  Copyright (c) 2011 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 SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
 #define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
 
 #include <deque>
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
 
 class TestErrorObserver;
 
 class LoopBackTransport : public webrtc::Transport {
  public:
@@ skipping 18 matching lines @@
 
   int SendRTCPPacket(const void* data, size_t len) override {
     StorePacket(Packet::Rtcp, data, len);
     return static_cast<int>(len);
   }
 
   void WaitForTransmittedPackets(int32_t packet_count) {
     enum {
       kSleepIntervalMs = 10
     };
-    int32_t limit = transmitted_packets_.Value() + packet_count;
-    while (transmitted_packets_.Value() < limit) {
+    int32_t limit =
+        rtc::AtomicOps::AcquireLoad(&transmitted_packets_) + packet_count;
+    while (rtc::AtomicOps::AcquireLoad(&transmitted_packets_) < limit) {
       webrtc::SleepMs(kSleepIntervalMs);
     }
   }
 
   void AddChannel(uint32_t ssrc, int channel) {
     webrtc::CriticalSectionScoped lock(crit_.get());
     channels_[ssrc] = channel;
   }
 
  private:
@@ skipping 61 matching lines @@
 
       switch (p.type) {
         case Packet::Rtp:
           voe_network_->ReceivedRTPPacket(channel, p.data, p.len,
                                           webrtc::PacketTime());
           break;
         case Packet::Rtcp:
           voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
           break;
       }
-      ++transmitted_packets_;
+      rtc::AtomicOps::Increment(&transmitted_packets_);
     }
     return true;
   }
 
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
   std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
   const int channel_;
   std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
   webrtc::VoENetwork* const voe_network_;
-  webrtc::Atomic32 transmitted_packets_;
+  volatile int transmitted_packets_;
 };
 
 // This fixture initializes the voice engine in addition to the work
 // done by the before-initialization fixture. It also registers an error
 // observer which will fail tests on error callbacks. This fixture is
 // useful to tests that want to run before we have started any form of
 // streaming through the voice engine.
 class AfterInitializationFixture : public BeforeInitializationFixture {
  public:
   AfterInitializationFixture();
   virtual ~AfterInitializationFixture();
 
  protected:
   rtc::scoped_ptr<TestErrorObserver> error_observer_;
 };
 
 #endif  // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_