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

webrtc/modules/audio_processing/test/debug_dump_test.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.
  */
 
 #include <stddef.h>  // size_t
+
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 
-void MaybeResetBuffer(rtc::scoped_ptr<ChannelBuffer<float>>* buffer,
+void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
                       const StreamConfig& config) {
   auto& buffer_ref = *buffer;
   if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
       buffer_ref->num_channels() != config.num_channels()) {
     buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
                                              config.num_channels()));
   }
 }
 
 class DebugDumpGenerator {
@@ skipping 52 matching lines @@
   const std::string input_file_name_;
   ResampleInputAudioFile input_audio_;
   const int input_file_channels_;
 
   // Reverse file format.
   const std::string reverse_file_name_;
   ResampleInputAudioFile reverse_audio_;
   const int reverse_file_channels_;
 
   // Buffer for APM input/output.
-  rtc::scoped_ptr<ChannelBuffer<float>> input_;
-  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
-  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+  std::unique_ptr<ChannelBuffer<float>> input_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_;
+  std::unique_ptr<ChannelBuffer<float>> output_;
 
-  rtc::scoped_ptr<AudioProcessing> apm_;
+  std::unique_ptr<AudioProcessing> apm_;
 
   const std::string dump_file_name_;
 };
 
 DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name,
                                        int input_rate_hz,
                                        int input_channels,
                                        const std::string& reverse_file_name,
                                        int reverse_rate_hz,
                                        int reverse_channels,
@@ skipping 124 matching lines @@
   // Following functions are facilities for replaying debug dumps.
   void OnInitEvent(const audioproc::Init& msg);
   void OnStreamEvent(const audioproc::Stream& msg);
   void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
   void OnConfigEvent(const audioproc::Config& msg);
 
   void MaybeRecreateApm(const audioproc::Config& msg);
   void ConfigureApm(const audioproc::Config& msg);
 
   // Buffer for APM input/output.
-  rtc::scoped_ptr<ChannelBuffer<float>> input_;
-  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
-  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+  std::unique_ptr<ChannelBuffer<float>> input_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_;
+  std::unique_ptr<ChannelBuffer<float>> output_;
 
-  rtc::scoped_ptr<AudioProcessing> apm_;
+  std::unique_ptr<AudioProcessing> apm_;
 
   StreamConfig input_config_;
   StreamConfig reverse_config_;
   StreamConfig output_config_;
 };
 
 DebugDumpTest::DebugDumpTest()
     : input_(nullptr),  // will be created upon usage.
       reverse_(nullptr),
       output_(nullptr),
@@ skipping 335 matching lines @@
   config.Set<ExperimentalNs>(new ExperimentalNs(true));
   DebugDumpGenerator generator(config);
   generator.StartRecording();
   generator.Process(100);
   generator.StopRecording();
   VerifyDebugDump(generator.dump_file_name());
 }
 
 }  // namespace test
 }  // namespace webrtc