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

webrtc/modules/audio_coding/test/delay_test.cc

 /*
  *  Copyright (c) 2013 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 <assert.h>
 #include <math.h>
 
 #include <iostream>
+#include <memory>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
@@ skipping 187 matching lines @@
 
       if (num_frames > 10)
         average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
 
       ++num_frames;
       ++in_file_frames;
     }
     out_file_b_.Close();
   }
 
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
 
   Channel* channel_a2b_;
 
   PCMFile in_file_a_;
   PCMFile out_file_b_;
   int test_cntr_;
   int encoding_sample_rate_hz_;
 };
 
 }  // namespace webrtc
@@ skipping 18 matching lines @@
   test_setting.codec.num_channels = FLAGS_num_channels;
   test_setting.acm.dtx = FLAGS_dtx;
   test_setting.acm.fec = FLAGS_fec;
   test_setting.packet_loss = FLAGS_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }