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_; |
}; |