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

webrtc/modules/audio_coding/test/PacketLossTest.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_TEST_PACKETLOSSTEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
+#include <memory>
 #include <string>
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
 
 class ReceiverWithPacketLoss : public Receiver {
  public:
   ReceiverWithPacketLoss();
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
              std::string out_file_name, int channels, int loss_rate,
              int burst_length);
@@ skipping 22 matching lines @@
 
 class PacketLossTest : public ACMTest {
  public:
   PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
                  int burst_length);
   void Perform();
  protected:
   int channels_;
   std::string in_file_name_;
   int sample_rate_hz_;
-  rtc::scoped_ptr<SenderWithFEC> sender_;
-  rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
+  std::unique_ptr<SenderWithFEC> sender_;
+  std::unique_ptr<ReceiverWithPacketLoss> receiver_;
   int expected_loss_rate_;
   int actual_loss_rate_;
   int burst_length_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_