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

webrtc/modules/audio_coding/neteq/normal_unittest.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.
  */
 
 // Unit tests for Normal class.
 
 #include "webrtc/modules/audio_coding/neteq/normal.h"
 
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
@@ skipping 21 matching lines @@
   size_t channels = 1;
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
   StatisticsCalculator statistics;
   MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
                     channels);
   Normal normal(fs, &db, bgn, &expand);
 
   int16_t input[1000] = {0};
-  rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
   for (size_t i = 0; i < channels; ++i) {
     mute_factor_array[i] = 16384;
   }
   AudioMultiVector output(channels);
 
   // Zero input length.
   EXPECT_EQ(
       0,
       normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
   EXPECT_EQ(0u, output.Size());
@@ skipping 25 matching lines @@
   size_t channels = 2;
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(channels, 1000);
   RandomVector random_vector;
   StatisticsCalculator statistics;
   MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
                     channels);
   Normal normal(fs, &db, bgn, &expand);
 
   int16_t input[1000] = {0};
-  rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
   for (size_t i = 0; i < channels; ++i) {
     mute_factor_array[i] = 16384;
   }
   AudioMultiVector output(channels);
 
   // Let the number of samples be one sample less than 80 samples per channel.
   size_t input_len = 80 * channels - 1;
   EXPECT_EQ(
       0,
       normal.Process(
           input, input_len, kModeExpand, mute_factor_array.get(), &output));
   EXPECT_EQ(0u, output.Size());
 
   EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
   EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
 }
 
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc