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

webrtc/modules/audio_processing/transient/transient_suppression_test.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 "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 
 #include <stdlib.h>
 #include <stdio.h>
+
+#include <memory>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
 DEFINE_string(detection_file_name,
               "",
               "PCM file that contains the detection signal.");
@@ skipping 46 matching lines @@
 // [-1,1]. Return true iff all the buffers were filled completely.
 bool ReadBuffers(FILE* in_file,
                  size_t audio_buffer_size,
                  int num_channels,
                  int16_t* audio_buffer,
                  FILE* detection_file,
                  size_t detection_buffer_size,
                  float* detection_buffer,
                  FILE* reference_file,
                  float* reference_buffer) {
-  rtc::scoped_ptr<int16_t[]> tmpbuf;
+  std::unique_ptr<int16_t[]> tmpbuf;
   int16_t* read_ptr = audio_buffer;
   if (num_channels > 1) {
     tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
     read_ptr = tmpbuf.get();
   }
   if (fread(read_ptr,
             sizeof(*read_ptr),
             num_channels * audio_buffer_size,
             in_file) != num_channels * audio_buffer_size) {
     return false;
   }
   // De-interleave.
   if (num_channels > 1) {
     for (int i = 0; i < num_channels; ++i) {
       for (size_t j = 0; j < audio_buffer_size; ++j) {
         audio_buffer[i * audio_buffer_size + j] =
             read_ptr[i + j * num_channels];
       }
     }
   }
   if (detection_file) {
-    rtc::scoped_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
+    std::unique_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
     if (fread(ibuf.get(), sizeof(ibuf[0]), detection_buffer_size,
               detection_file) != detection_buffer_size)
       return false;
     for (size_t i = 0; i < detection_buffer_size; ++i)
       detection_buffer[i] = ibuf[i];
   }
   if (reference_file) {
-    rtc::scoped_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
+    std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
     if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file)
         != audio_buffer_size)
       return false;
     S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
   }
   return true;
 }
 
 // Write a number of samples to an open signed 16-bit host-endian PCM file.
 static void WritePCM(FILE* f,
                      size_t num_samples,
                      int num_channels,
                      const float* buffer) {
-  rtc::scoped_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
+  std::unique_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
   // Interleave.
   for (int i = 0; i < num_channels; ++i) {
     for (size_t j = 0; j < num_samples; ++j) {
       ibuf[i + j * num_channels] = FloatS16ToS16(buffer[i * num_samples + j]);
     }
   }
   fwrite(ibuf.get(), sizeof(ibuf[0]), num_channels * num_samples, f);
 }
 
 // This application tests the transient suppression by providing a processed
@@ skipping 34 matching lines @@
   TransientSuppressor suppressor;
   suppressor.Initialize(
       FLAGS_sample_rate_hz, detection_rate_hz, FLAGS_num_channels);
 
   const size_t audio_buffer_size =
       FLAGS_chunk_size_ms * FLAGS_sample_rate_hz / 1000;
   const size_t detection_buffer_size =
       FLAGS_chunk_size_ms * detection_rate_hz / 1000;
 
   // int16 and float variants of the same data.
-  rtc::scoped_ptr<int16_t[]> audio_buffer_i(
+  std::unique_ptr<int16_t[]> audio_buffer_i(
       new int16_t[FLAGS_num_channels * audio_buffer_size]);
-  rtc::scoped_ptr<float[]> audio_buffer_f(
+  std::unique_ptr<float[]> audio_buffer_f(
       new float[FLAGS_num_channels * audio_buffer_size]);
 
-  rtc::scoped_ptr<float[]> detection_buffer, reference_buffer;
+  std::unique_ptr<float[]> detection_buffer, reference_buffer;
 
   if (detection_file)
     detection_buffer.reset(new float[detection_buffer_size]);
   if (reference_file)
     reference_buffer.reset(new float[audio_buffer_size]);
 
   while (ReadBuffers(in_file,
                      audio_buffer_size,
                      FLAGS_num_channels,
                      audio_buffer_i.get(),
@@ skipping 40 matching lines @@
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   google::SetUsageMessage(webrtc::kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
   webrtc::void_main();
   return 0;
 }