| 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_;
|
|
|