Replace Atomic32 with webrtc/base/atomicops.h.

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

Index: webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 4f92e0670d20382304492d81fb7515db098d9f4e..f37b198682fbca71864d094037ec5721961b0e1e 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -8,9 +8,9 @@
  *  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"
 
@@ -43,11 +43,11 @@ class ExtensionVerifyTransport : public webrtc::Transport {
       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);
   }
 
@@ -68,12 +68,12 @@ class ExtensionVerifyTransport : public webrtc::Transport {
 
   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:
@@ -82,8 +82,8 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     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_;
 };