Replace Atomic32 with webrtc/base/atomicops.h.

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
 
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::Eq;
 using ::testing::Field;
 
 class ExtensionVerifyTransport : public webrtc::Transport {
  public:
@@ skipping 12 matching lines @@
           !header.extension.hasAudioLevel) {
         ok = false;
       }
       if (absolute_sender_time_id_ >= 0 &&
           !header.extension.hasAbsoluteSendTime) {
         ok = false;
       }
       if (!ok) {
         // bad_packets_ count packets we expected to have an extension but
         // didn't have one.
-        ++bad_packets_;
+        rtc::AtomicOps::Increment(&bad_packets_);
       }
     }
     // received_packets_ count all packets we receive.
-    ++received_packets_;
+    rtc::AtomicOps::Increment(&received_packets_);
     return static_cast<int>(len);
   }
 
   int SendRTCPPacket(const void* data, size_t len) override {
     return static_cast<int>(len);
   }
 
   void SetAudioLevelId(int id) {
     audio_level_id_ = id;
     parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, id);
   }
 
   void SetAbsoluteSenderTimeId(int id) {
     absolute_sender_time_id_ = id;
     parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAbsoluteSendTime,
                                         id);
   }
 
   bool Wait() {
     // Wait until we've received to specified number of packets.
-    while (received_packets_.Value() < kPacketsExpected) {
+    while (rtc::AtomicOps::AcquireLoad(&received_packets_) < kPacketsExpected) {
       webrtc::SleepMs(kSleepIntervalMs);
     }
     // Check whether any were 'bad' (didn't contain an extension when they
     // where supposed to).
-    return bad_packets_.Value() == 0;
+    return rtc::AtomicOps::AcquireLoad(&bad_packets_) == 0;
   }
 
  private:
   enum {
     kPacketsExpected = 10,
     kSleepIntervalMs = 10
   };
   rtc::scoped_ptr<webrtc::RtpHeaderParser> parser_;
-  webrtc::Atomic32 received_packets_;
-  webrtc::Atomic32 bad_packets_;
+  volatile int received_packets_;
+  volatile int bad_packets_;
   int audio_level_id_;
   int absolute_sender_time_id_;
 };
 
 class SendRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
  protected:
   void SetUp() override {
     EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
     EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_,
                                                          verifying_transport_));
@@ skipping 47 matching lines @@
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
                                                               3));
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
                                                                 9));
   verifying_transport_.SetAbsoluteSenderTimeId(3);
   // Don't register audio level with header parser - unknown extensions should
   // be ignored when parsing.
   ResumePlaying();
   EXPECT_TRUE(verifying_transport_.Wait());
 }