Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/

webrtc/modules/audio_coding/neteq/audio_classifier_unittest.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/audio_classifier.h"
 
 #include <math.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 static const size_t kFrameSize = 960;
 
 TEST(AudioClassifierTest, AllZeroInput) {
   int16_t in_mono[kFrameSize] = {0};
 
   // Test all-zero vectors and let the classifier converge from its default
   // to the expected value.
   AudioClassifier zero_classifier;
   for (int i = 0; i < 100; ++i) {
     zero_classifier.Analysis(in_mono, kFrameSize, 1);
   }
   EXPECT_TRUE(zero_classifier.is_music());
 }
 
 void RunAnalysisTest(const std::string& audio_filename,
                      const std::string& data_filename,
                      size_t channels) {
   AudioClassifier classifier;
-  rtc::scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
+  std::unique_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
   bool is_music_ref;
 
   FILE* audio_file = fopen(audio_filename.c_str(), "rb");
   ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
                                   << std::endl;
   FILE* data_file = fopen(data_filename.c_str(), "rb");
   ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
                                   << std::endl;
   while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) ==
          channels * kFrameSize) {
@@ skipping 19 matching lines @@
 #endif // WEBRTC_ARCH_ARM
 }
 
 TEST(AudioClassifierTest, DoAnalysisStereo) {
   RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"),
                   test::ResourcePath("short_mixed_stereo_48", "dat"),
                   2);
 }
 
 }  // namespace webrtc