Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/common_audio/audio_converter.cc

 /*
  *  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.
  */
 
@@ skipping 118 matching lines @@
     }
     converters_.back()->Convert(buffers_.back()->channels(),
                                 buffers_.back()->size(), dst, dst_capacity);
   }
 
  private:
   ScopedVector<AudioConverter> converters_;
   ScopedVector<ChannelBuffer<float>> buffers_;
 };
 
-rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
+std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
                                                        size_t src_frames,
                                                        size_t dst_channels,
                                                        size_t dst_frames) {
-  rtc::scoped_ptr<AudioConverter> sp;
+  std::unique_ptr<AudioConverter> sp;
   if (src_channels > dst_channels) {
     if (src_frames != dst_frames) {
       ScopedVector<AudioConverter> converters;
       converters.push_back(new DownmixConverter(src_channels, src_frames,
                                                 dst_channels, src_frames));
       converters.push_back(new ResampleConverter(dst_channels, src_frames,
                                                  dst_channels, dst_frames));
       sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
@@ skipping 38 matching lines @@
   RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
             src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
   RTC_CHECK_EQ(src_size, src_channels() * src_frames());
   RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc