Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/

webrtc/modules/audio_coding/neteq/time_stretch.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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"
 
 namespace webrtc {
 
 TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
                                               size_t input_len,
                                               bool fast_mode,
                                               AudioMultiVector* output,
                                               size_t* length_change_samples) {
   // Pre-calculate common multiplication with |fs_mult_|.
   size_t fs_mult_120 =
       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;
     signal_len = input_len;
   } else {
     // We want |signal| to be only the first channel of |input|, which is
     // interleaved. Thus, we take the first sample, skip forward |num_channels|
     // samples, and continue like that.
     signal_len = input_len / num_channels_;
     signal_array.reset(new int16_t[signal_len]);
@@ skipping 164 matching lines @@
     int temp_scale = WebRtcSpl_NormW32(left_side);
     left_side = left_side << temp_scale;
     right_side = right_side >> (2 * scaling - temp_scale);
   } else {
     left_side = left_side << 2 * scaling;
   }
   return left_side > right_side;
 }
 
 }  // namespace webrtc