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

webrtc/modules/audio_coding/test/TestAllCodecs.h

 /*
  *  Copyright (c) 2012 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_TESTALLCODECS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Config;
 
 class TestPack : public AudioPacketizationCallback {
@@ skipping 37 matching lines @@
   // This is useful for codecs which support several sampling frequency.
   // Note! Only mono mode is tested in this test.
   void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
                          int rate, int packet_size, size_t extra_byte);
 
   void Run(TestPack* channel);
   void OpenOutFile(int test_number);
   void DisplaySendReceiveCodec();
 
   int test_mode_;
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
   TestPack* channel_a_to_b_;
   PCMFile infile_a_;
   PCMFile outfile_b_;
   int test_count_;
   int packet_size_samples_;
   size_t packet_size_bytes_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_