Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/common_audio/resampler/push_sinc_resampler.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_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
 #define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
 
+#include <memory>
+
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // A thin wrapper over SincResampler to provide a push-based interface as
 // required by WebRTC. SincResampler uses a pull-based interface, and will
 // use SincResamplerCallback::Run() to request data upon a call to Resample().
 // These Run() calls will happen on the same thread Resample() is called on.
 class PushSincResampler : public SincResamplerCallback {
@@ skipping 23 matching lines @@
   }
 
  protected:
   // Implements SincResamplerCallback.
   void Run(size_t frames, float* destination) override;
 
  private:
   friend class PushSincResamplerTest;
   SincResampler* get_resampler_for_testing() { return resampler_.get(); }
 
-  rtc::scoped_ptr<SincResampler> resampler_;
-  rtc::scoped_ptr<float[]> float_buffer_;
+  std::unique_ptr<SincResampler> resampler_;
+  std::unique_ptr<float[]> float_buffer_;
   const float* source_ptr_;
   const int16_t* source_ptr_int_;
   const size_t destination_frames_;
 
   // True on the first call to Resample(), to prime the SincResampler buffer.
   bool first_pass_;
 
   // Used to assert we are only requested for as much data as is available.
   size_t source_available_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_