Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/

webrtc/modules/audio_processing/test/test_utils.cc

 /*
  *  Copyright (c) 2015 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 15 matching lines @@
 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
 #error "Need to convert samples to little-endian when writing to PCM file"
 #endif
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
 void RawFile::WriteSamples(const float* samples, size_t num_samples) {
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
-ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
     : file_(std::move(file)) {}
 
 bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
   RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
   interleaved_.resize(buffer->size());
   if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
       interleaved_.size()) {
     return false;
   }
 
   FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
   Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
                buffer->channels());
   return true;
 }
 
-ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
     : file_(std::move(file)) {}
 
 void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
   RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
   interleaved_.resize(buffer.size());
   Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
              &interleaved_[0]);
   FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
   file_->WriteSamples(&interleaved_[0], interleaved_.size());
 }
 
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
                   RawFile* raw_file) {
   if (wav_file) {
     wav_file->WriteSamples(data, length);
   }
   if (raw_file) {
     raw_file->WriteSamples(data, length);
   }
 }
 
 void WriteFloatData(const float* const* data,
                     size_t samples_per_channel,
                     size_t num_channels,
                     WavWriter* wav_file,
                     RawFile* raw_file) {
   size_t length = num_channels * samples_per_channel;
-  rtc::scoped_ptr<float[]> buffer(new float[length]);
+  std::unique_ptr<float[]> buffer(new float[length]);
   Interleave(data, samples_per_channel, num_channels, buffer.get());
   if (raw_file) {
     raw_file->WriteSamples(buffer.get(), length);
   }
   // TODO(aluebs): Use ScaleToInt16Range() from audio_util
   for (size_t i = 0; i < length; ++i) {
     buffer[i] = buffer[i] > 0 ?
                 buffer[i] * std::numeric_limits<int16_t>::max() :
                 -buffer[i] * std::numeric_limits<int16_t>::min();
   }
@@ skipping 51 matching lines @@
 
 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
                                       size_t num_mics) {
   std::vector<Point> result = ParseArrayGeometry(mic_positions);
   RTC_CHECK_EQ(result.size(), num_mics)
       << "Could not parse mic_positions or incorrect number of points.";
   return result;
 }
 
 }  // namespace webrtc