Remove webrtc::ScopedVector

webrtc/modules/audio_processing/audio_buffer.cc

 /*
  *  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.
  */
 
@@ skipping 57 matching lines @@
   assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);
 
   if (input_num_frames_ != proc_num_frames_ ||
       output_num_frames_ != proc_num_frames_) {
     // Create an intermediate buffer for resampling.
     process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
                                                    num_proc_channels_));
 
     if (input_num_frames_ != proc_num_frames_) {
       for (size_t i = 0; i < num_proc_channels_; ++i) {
-        input_resamplers_.push_back(
-            new PushSincResampler(input_num_frames_,
-                                  proc_num_frames_));
+        input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
+            new PushSincResampler(input_num_frames_, proc_num_frames_)));
       }
     }
 
     if (output_num_frames_ != proc_num_frames_) {
       for (size_t i = 0; i < num_proc_channels_; ++i) {
-        output_resamplers_.push_back(
-            new PushSincResampler(proc_num_frames_,
-                                  output_num_frames_));
+        output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
+            new PushSincResampler(proc_num_frames_, output_num_frames_)));
       }
     }
   }
 
   if (num_bands_ > 1) {
     split_data_.reset(new IFChannelBuffer(proc_num_frames_,
                                           num_proc_channels_,
                                           num_bands_));
     splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
                                                 num_bands_,
@@ skipping 359 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc