Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/common_audio/audio_converter.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_COMMON_AUDIO_AUDIO_CONVERTER_H_
 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
 
+#include <memory>
+
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 // Format conversion (remixing and resampling) for audio. Only simple remixing
 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
 // upmix from mono (i.e. |src_channels == 1|).
 //
 // The source and destination chunks have the same duration in time; specifying
 // the number of frames is equivalent to specifying the sample rates.
 class AudioConverter {
  public:
   // Returns a new AudioConverter, which will use the supplied format for its
   // lifetime. Caller is responsible for the memory.
-  static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
+  static std::unique_ptr<AudioConverter> Create(size_t src_channels,
                                                 size_t src_frames,
                                                 size_t dst_channels,
                                                 size_t dst_frames);
   virtual ~AudioConverter() {};
 
   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
   // capacity of |dst_capacity|. Both point to a series of buffers containing
   // the samples for each channel. The sizes must correspond to the format
   // passed to Create().
   virtual void Convert(const float* const* src, size_t src_size,
@@ skipping 17 matching lines @@
   const size_t src_frames_;
   const size_t dst_channels_;
   const size_t dst_frames_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_