Index: webrtc/modules/audio_coding/test/PacketLossTest.h |
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h |
index f3570ae1ca4ee95e95d5339755ed369db96c9bde..705fe73ff509db672afeb2fa1c29840590b8c427 100644 |
--- a/webrtc/modules/audio_coding/test/PacketLossTest.h |
+++ b/webrtc/modules/audio_coding/test/PacketLossTest.h |
@@ -11,8 +11,8 @@ |
#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |
#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |
+#include <memory> |
#include <string> |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" |
namespace webrtc { |
@@ -55,8 +55,8 @@ class PacketLossTest : public ACMTest { |
int channels_; |
std::string in_file_name_; |
int sample_rate_hz_; |
- rtc::scoped_ptr<SenderWithFEC> sender_; |
- rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_; |
+ std::unique_ptr<SenderWithFEC> sender_; |
+ std::unique_ptr<ReceiverWithPacketLoss> receiver_; |
int expected_loss_rate_; |
int actual_loss_rate_; |
int burst_length_; |