Index: webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
index a304e8240a2719a159264b17cffbfd3e5bce7670..0a85466db0165e73bff4cc302a0b4fbba1f9c2c8 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
@@ -19,13 +19,13 @@ |
#include <string.h> // memset |
#include <algorithm> |
+#include <memory> |
#include <set> |
#include <string> |
#include <vector> |
#include "gflags/gflags.h" |
#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" |
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" |
@@ -102,7 +102,7 @@ void ReadMessage(FILE* file, std::string* message) { |
ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file)); |
if (size <= 0) |
return; |
- rtc::scoped_ptr<char[]> buffer(new char[size]); |
+ std::unique_ptr<char[]> buffer(new char[size]); |
ASSERT_EQ(static_cast<size_t>(size), |
fread(buffer.get(), sizeof(char), size, file)); |
message->assign(buffer.get(), size); |
@@ -320,8 +320,8 @@ class NetEqDecodingTest : public ::testing::Test { |
NetEq* neteq_; |
NetEq::Config config_; |
- rtc::scoped_ptr<test::RtpFileSource> rtp_source_; |
- rtc::scoped_ptr<test::Packet> packet_; |
+ std::unique_ptr<test::RtpFileSource> rtp_source_; |
+ std::unique_ptr<test::Packet> packet_; |
unsigned int sim_clock_; |
int16_t out_data_[kMaxBlockSize]; |
int output_sample_rate_; |