Index: webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h |
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h |
index fb7b3e5b1ec83fe0918f69fede84bca8de26afe5..3560eac3427f40bdd7c416e596956747dc75415f 100644 |
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h |
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h |
@@ -11,9 +11,9 @@ |
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ |
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ |
+#include <memory> |
#include <string> |
#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/typedefs.h" |
namespace webrtc { |
@@ -61,11 +61,11 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> { |
// Expected output number of samples-per-channel in a frame. |
size_t output_length_sample_; |
- rtc::scoped_ptr<int16_t[]> in_data_; |
- rtc::scoped_ptr<int16_t[]> out_data_; |
+ std::unique_ptr<int16_t[]> in_data_; |
+ std::unique_ptr<int16_t[]> out_data_; |
size_t data_pointer_; |
size_t loop_length_samples_; |
- rtc::scoped_ptr<uint8_t[]> bit_stream_; |
+ std::unique_ptr<uint8_t[]> bit_stream_; |
// Maximum number of bytes in output bitstream for a frame of audio. |
size_t max_bytes_; |