Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/

webrtc/modules/audio_processing/transient/transient_suppressor.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
 
 #include <deque>
+#include <memory>
 #include <set>
 
 #include "webrtc/base/gtest_prod_util.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class TransientDetector;
 
 // Detects transients in an audio stream and suppress them using a simple
 // restoration algorithm that attenuates unexpected spikes in the spectrum.
 class TransientSuppressor {
  public:
@@ skipping 35 matching lines @@
   void Suppress(float* in_ptr, float* spectral_mean, float* out_ptr);
 
   void UpdateKeypress(bool key_pressed);
   void UpdateRestoration(float voice_probability);
 
   void UpdateBuffers(float* data);
 
   void HardRestoration(float* spectral_mean);
   void SoftRestoration(float* spectral_mean);
 
-  rtc::scoped_ptr<TransientDetector> detector_;
+  std::unique_ptr<TransientDetector> detector_;
 
   size_t data_length_;
   size_t detection_length_;
   size_t analysis_length_;
   size_t buffer_delay_;
   size_t complex_analysis_length_;
   int num_channels_;
   // Input buffer where the original samples are stored.
-  rtc::scoped_ptr<float[]> in_buffer_;
-  rtc::scoped_ptr<float[]> detection_buffer_;
+  std::unique_ptr<float[]> in_buffer_;
+  std::unique_ptr<float[]> detection_buffer_;
   // Output buffer where the restored samples are stored.
-  rtc::scoped_ptr<float[]> out_buffer_;
+  std::unique_ptr<float[]> out_buffer_;
 
   // Arrays for fft.
-  rtc::scoped_ptr<size_t[]> ip_;
-  rtc::scoped_ptr<float[]> wfft_;
+  std::unique_ptr<size_t[]> ip_;
+  std::unique_ptr<float[]> wfft_;
 
-  rtc::scoped_ptr<float[]> spectral_mean_;
+  std::unique_ptr<float[]> spectral_mean_;
 
   // Stores the data for the fft.
-  rtc::scoped_ptr<float[]> fft_buffer_;
+  std::unique_ptr<float[]> fft_buffer_;
 
-  rtc::scoped_ptr<float[]> magnitudes_;
+  std::unique_ptr<float[]> magnitudes_;
 
   const float* window_;
 
-  rtc::scoped_ptr<float[]> mean_factor_;
+  std::unique_ptr<float[]> mean_factor_;
 
   float detector_smoothed_;
 
   int keypress_counter_;
   int chunks_since_keypress_;
   bool detection_enabled_;
   bool suppression_enabled_;
 
   bool use_hard_restoration_;
   int chunks_since_voice_change_;
 
   uint32_t seed_;
 
   bool using_reference_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_