Index: webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc |
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc |
index 24ecc694ffba4052086111592a01448f432e9641..a0f4e0e0192226cc4978f2e3f107793584112e0f 100644 |
--- a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc |
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc |
@@ -11,9 +11,9 @@ |
#include "webrtc/modules/audio_coding/acm2/acm_receiver.h" |
#include <algorithm> // std::min |
+#include <memory> |
#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h" |
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" |
@@ -153,9 +153,9 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback, |
return 0; |
} |
- rtc::scoped_ptr<AcmReceiver> receiver_; |
+ std::unique_ptr<AcmReceiver> receiver_; |
rtc::ArrayView<const CodecInst> codecs_; |
- rtc::scoped_ptr<AudioCodingModule> acm_; |
+ std::unique_ptr<AudioCodingModule> acm_; |
WebRtcRTPHeader rtp_header_; |
uint32_t timestamp_; |
bool packet_sent_; // Set when SendData is called reset when inserting audio. |