| 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;
|
|
|