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

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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.
  */
 
 /*
  * This file includes unit tests for NetEQ.
  */
 
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>  // memset
 
 #include <algorithm>
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
 #else
@@ skipping 56 matching lines @@
     return;
   ASSERT_EQ(static_cast<size_t>(size),
             fwrite(message.data(), sizeof(char), size, file));
 }
 
 void ReadMessage(FILE* file, std::string* message) {
   int32_t size;
   ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
   if (size <= 0)
     return;
-  rtc::scoped_ptr<char[]> buffer(new char[size]);
+  std::unique_ptr<char[]> buffer(new char[size]);
   ASSERT_EQ(static_cast<size_t>(size),
             fread(buffer.get(), sizeof(char), size, file));
   message->assign(buffer.get(), size);
 }
 #endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 
 }  // namespace
 
 namespace webrtc {
 
@@ skipping 197 matching lines @@
                              bool pull_audio_during_freeze,
                              int delay_tolerance_ms,
                              int max_time_to_speech_ms);
 
   void DuplicateCng();
 
   uint32_t PlayoutTimestamp();
 
   NetEq* neteq_;
   NetEq::Config config_;
-  rtc::scoped_ptr<test::RtpFileSource> rtp_source_;
-  rtc::scoped_ptr<test::Packet> packet_;
+  std::unique_ptr<test::RtpFileSource> rtp_source_;
+  std::unique_ptr<test::Packet> packet_;
   unsigned int sim_clock_;
   int16_t out_data_[kMaxBlockSize];
   int output_sample_rate_;
   int algorithmic_delay_ms_;
 };
 
 // Allocating the static const so that it can be passed by reference.
 const int NetEqDecodingTest::kTimeStepMs;
 const size_t NetEqDecodingTest::kBlockSize8kHz;
 const size_t NetEqDecodingTest::kBlockSize16kHz;
@@ skipping 1322 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc