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

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 
 #include <fstream>
+#include <memory>
 #include <gflags/gflags.h>
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/typedefs.h"
 
 using google::RegisterFlagValidator;
 
 namespace webrtc {
 namespace test {
@@ skipping 23 matching lines @@
  public:
   GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
   bool Lost() override;
 
  private:
   // Prob. of losing current packet, when previous packet is lost.
   double prob_trans_11_;
   // Prob. of losing current packet, when previous packet is not lost.
   double prob_trans_01_;
   bool lost_last_;
-  rtc::scoped_ptr<UniformLoss> uniform_loss_model_;
+  std::unique_ptr<UniformLoss> uniform_loss_model_;
 };
 
 class NetEqQualityTest : public ::testing::Test {
  protected:
   NetEqQualityTest(int block_duration_ms,
                    int in_sampling_khz,
                    int out_sampling_khz,
                    NetEqDecoder decoder_type);
   virtual ~NetEqQualityTest();
 
@@ skipping 40 matching lines @@
 
   // Number of samples per channel in a frame.
   const size_t in_size_samples_;
 
   // Expected output number of samples per channel in a frame.
   const size_t out_size_samples_;
 
   size_t payload_size_bytes_;
   size_t max_payload_bytes_;
 
-  rtc::scoped_ptr<InputAudioFile> in_file_;
-  rtc::scoped_ptr<AudioSink> output_;
+  std::unique_ptr<InputAudioFile> in_file_;
+  std::unique_ptr<AudioSink> output_;
   std::ofstream log_file_;
 
-  rtc::scoped_ptr<RtpGenerator> rtp_generator_;
-  rtc::scoped_ptr<NetEq> neteq_;
-  rtc::scoped_ptr<LossModel> loss_model_;
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<LossModel> loss_model_;
 
-  rtc::scoped_ptr<int16_t[]> in_data_;
-  rtc::scoped_ptr<uint8_t[]> payload_;
-  rtc::scoped_ptr<int16_t[]> out_data_;
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<uint8_t[]> payload_;
+  std::unique_ptr<int16_t[]> out_data_;
   WebRtcRTPHeader rtp_header_;
 
   size_t total_payload_size_bytes_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_