Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/common_audio/channel_buffer.h

 /*
  *  Copyright (c) 2014 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_CHANNEL_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
 
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gtest_prod_util.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 
 namespace webrtc {
 
 // Helper to encapsulate a contiguous data buffer, full or split into frequency
 // bands, with access to a pointer arrays of the deinterleaved channels and
 // bands. The buffer is zero initialized at creation.
 //
 // The buffer structure is showed below for a 2 channel and 2 bands case:
 //
@@ skipping 89 matching lines @@
   size_t num_channels() const { return num_channels_; }
   size_t num_bands() const { return num_bands_; }
   size_t size() const {return num_frames_ * num_channels_; }
 
   void SetDataForTesting(const T* data, size_t size) {
     RTC_CHECK_EQ(size, this->size());
     memcpy(data_.get(), data, size * sizeof(*data));
   }
 
  private:
-  rtc::scoped_ptr<T[]> data_;
-  rtc::scoped_ptr<T* []> channels_;
-  rtc::scoped_ptr<T* []> bands_;
+  std::unique_ptr<T[]> data_;
+  std::unique_ptr<T* []> channels_;
+  std::unique_ptr<T* []> bands_;
   const size_t num_frames_;
   const size_t num_frames_per_band_;
   const size_t num_channels_;
   const size_t num_bands_;
 };
 
 // One int16_t and one float ChannelBuffer that are kept in sync. The sync is
 // broken when someone requests write access to either ChannelBuffer, and
 // reestablished when someone requests the outdated ChannelBuffer. It is
 // therefore safe to use the return value of ibuf_const() and fbuf_const()
@@ skipping 19 matching lines @@
 
   mutable bool ivalid_;
   mutable ChannelBuffer<int16_t> ibuf_;
   mutable bool fvalid_;
   mutable ChannelBuffer<float> fbuf_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_