Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/

webrtc/modules/audio_processing/agc/histogram.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_HISTOGRAM_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_HISTOGRAM_H_
 
 #include <string.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // This class implements the histogram of loudness with circular buffers so that
 // the histogram tracks the last T seconds of the loudness.
 class Histogram {
  public:
   // Create a non-sliding Histogram.
   static Histogram* Create();
@@ skipping 39 matching lines @@
   // Audio content, this should be equal to the sum of the components of
   // |bin_count_q10_|.
   int64_t audio_content_q10_;
 
   // Histogram of input RMS in Q10 with |kHistSize_| bins. In each 'Update(),'
   // we increment the associated histogram-bin with the given probability. The
   // increment is implemented in Q10 to avoid rounding errors.
   int64_t bin_count_q10_[kHistSize];
 
   // Circular buffer for probabilities
-  rtc::scoped_ptr<int[]> activity_probability_;
+  std::unique_ptr<int[]> activity_probability_;
   // Circular buffer for histogram-indices of probabilities.
-  rtc::scoped_ptr<int[]> hist_bin_index_;
+  std::unique_ptr<int[]> hist_bin_index_;
   // Current index of circular buffer, where the newest data will be written to,
   // therefore, pointing to the oldest data if buffer is full.
   int buffer_index_;
   // Indicating if buffer is full and we had a wrap around.
   int buffer_is_full_;
   // Size of circular buffer.
   int len_circular_buffer_;
   int len_high_activity_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_HISTOGRAM_H_