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

webrtc/modules/audio_coding/test/PacketLossTest.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/test/PacketLossTest.h"
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 ReceiverWithPacketLoss::ReceiverWithPacketLoss()
     : loss_rate_(0),
       burst_length_(1),
       packet_counter_(0),
@@ skipping 96 matching lines @@
       receiver_(new ReceiverWithPacketLoss),
       expected_loss_rate_(expected_loss_rate),
       actual_loss_rate_(actual_loss_rate),
       burst_length_(burst_length) {
 }
 
 void PacketLossTest::Perform() {
 #ifndef WEBRTC_CODEC_OPUS
   return;
 #else
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
 
   int codec_id = acm->Codec("opus", 48000, channels_);
 
   RTPFile rtpFile;
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "packet_loss_test");
 
   // Encode to file
   rtpFile.Open(fileName.c_str(), "wb+");
   rtpFile.WriteHeader();
@@ skipping 18 matching lines @@
 
   receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
                    actual_loss_rate_, burst_length_);
   receiver_->Run();
   receiver_->Teardown();
   rtpFile.Close();
 #endif
 }
 
 }  // namespace webrtc