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 1dc15dff49571700e9a5ad8bdd94774c4812f824..16f17b1340a42f1e03c9af3615d1840c26b1e791 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 |
@@ -8,6 +8,8 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
+#include <memory> |
+ |
#include "webrtc/modules/include/module_common_types.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
#include "webrtc/system_wrappers/include/atomic32.h" |
@@ -83,7 +85,7 @@ class ExtensionVerifyTransport : public webrtc::Transport { |
kPacketsExpected = 10, |
kSleepIntervalMs = 10 |
}; |
- rtc::scoped_ptr<webrtc::RtpHeaderParser> parser_; |
+ std::unique_ptr<webrtc::RtpHeaderParser> parser_; |
webrtc::Atomic32 received_packets_; |
webrtc::Atomic32 bad_packets_; |
int audio_level_id_; |