Index: webrtc/modules/audio_coding/neteq/time_stretch.cc |
diff --git a/webrtc/modules/audio_coding/neteq/time_stretch.cc b/webrtc/modules/audio_coding/neteq/time_stretch.cc |
index 6ae81e6e9662ed4a09a0c1917b03d2143581ced7..6a91ea487b59fe076a951c0621996450e145a8cf 100644 |
--- a/webrtc/modules/audio_coding/neteq/time_stretch.cc |
+++ b/webrtc/modules/audio_coding/neteq/time_stretch.cc |
@@ -11,9 +11,9 @@ |
#include "webrtc/modules/audio_coding/neteq/time_stretch.h" |
#include <algorithm> // min, max |
+#include <memory> |
#include "webrtc/base/safe_conversions.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
#include "webrtc/modules/audio_coding/neteq/background_noise.h" |
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" |
@@ -30,7 +30,7 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input, |
static_cast<size_t>(fs_mult_ * 120); // Corresponds to 15 ms. |
const int16_t* signal; |
- rtc::scoped_ptr<int16_t[]> signal_array; |
+ std::unique_ptr<int16_t[]> signal_array; |
size_t signal_len; |
if (num_channels_ == 1) { |
signal = input; |