Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

 /*
  *  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.
  */
 
 #include <cmath>
 #include <cstring>
+#include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace {
 
 // Almost all conversions have an RMS error of around -14 dbFS.
@@ skipping 37 matching lines @@
 };
 
 void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
   const size_t input_samples = static_cast<size_t>(input_rate_ / 100);
   const size_t output_samples = static_cast<size_t>(output_rate_ / 100);
   const int kResampleIterations = 500000;
 
   // Source for data to be resampled.
   ZeroSource resampler_source;
 
-  rtc::scoped_ptr<float[]> resampled_destination(new float[output_samples]);
-  rtc::scoped_ptr<float[]> source(new float[input_samples]);
-  rtc::scoped_ptr<int16_t[]> source_int(new int16_t[input_samples]);
-  rtc::scoped_ptr<int16_t[]> destination_int(new int16_t[output_samples]);
+  std::unique_ptr<float[]> resampled_destination(new float[output_samples]);
+  std::unique_ptr<float[]> source(new float[input_samples]);
+  std::unique_ptr<int16_t[]> source_int(new int16_t[input_samples]);
+  std::unique_ptr<int16_t[]> destination_int(new int16_t[output_samples]);
 
   resampler_source.Run(input_samples, source.get());
   for (size_t i = 0; i < input_samples; ++i) {
     source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
   }
 
   printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n",
          kResampleIterations, input_rate_, output_rate_);
   const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
   SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
@@ skipping 58 matching lines @@
   const double input_nyquist_freq = 0.5 * input_rate_;
 
   // Source for data to be resampled.
   SinusoidalLinearChirpSource resampler_source(
       input_rate_, input_samples, input_nyquist_freq, 0);
 
   PushSincResampler resampler(input_block_size, output_block_size);
 
   // TODO(dalecurtis): If we switch to AVX/SSE optimization, we'll need to
   // allocate these on 32-byte boundaries and ensure they're sized % 32 bytes.
-  rtc::scoped_ptr<float[]> resampled_destination(new float[output_samples]);
-  rtc::scoped_ptr<float[]> pure_destination(new float[output_samples]);
-  rtc::scoped_ptr<float[]> source(new float[input_samples]);
-  rtc::scoped_ptr<int16_t[]> source_int(new int16_t[input_block_size]);
-  rtc::scoped_ptr<int16_t[]> destination_int(new int16_t[output_block_size]);
+  std::unique_ptr<float[]> resampled_destination(new float[output_samples]);
+  std::unique_ptr<float[]> pure_destination(new float[output_samples]);
+  std::unique_ptr<float[]> source(new float[input_samples]);
+  std::unique_ptr<int16_t[]> source_int(new int16_t[input_block_size]);
+  std::unique_ptr<int16_t[]> destination_int(new int16_t[output_block_size]);
 
   // The sinc resampler has an implicit delay of approximately half the kernel
   // size at the input sample rate. By moving to a push model, this delay
   // becomes explicit and is managed by zero-stuffing in PushSincResampler. We
   // deal with it in the test by delaying the "pure" source to match. It must be
   // checked before the first call to Resample(), because ChunkSize() will
   // change afterwards.
   const size_t output_delay_samples = output_block_size -
       resampler.get_resampler_for_testing()->ChunkSize();
 
@@ skipping 155 matching lines @@
         // To 32 kHz
         ::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
         ::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
         ::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
         ::testing::make_tuple(44100, 32000, -16.44, -51.10),
         ::testing::make_tuple(48000, 32000, -16.90, -44.03),
         ::testing::make_tuple(96000, 32000, -19.61, -18.04),
         ::testing::make_tuple(192000, 32000, -21.02, -10.94)));
 
 }  // namespace webrtc