Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/

webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.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 "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 
 #include <algorithm>  // std::min
+#include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 namespace acm2 {
@@ skipping 119 matching lines @@
     if (ret_val < 0) {
       assert(false);
       return -1;
     }
     rtp_header_.header.sequenceNumber++;
     packet_sent_ = true;
     last_frame_type_ = frame_type;
     return 0;
   }
 
-  rtc::scoped_ptr<AcmReceiver> receiver_;
+  std::unique_ptr<AcmReceiver> receiver_;
   rtc::ArrayView<const CodecInst> codecs_;
-  rtc::scoped_ptr<AudioCodingModule> acm_;
+  std::unique_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_header_;
   uint32_t timestamp_;
   bool packet_sent_;  // Set when SendData is called reset when inserting audio.
   uint32_t last_packet_send_timestamp_;
   FrameType last_frame_type_;
 };
 
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
 #else
@@ skipping 220 matching lines @@
               receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
 #endif
 
 }  // namespace acm2
 
 }  // namespace webrtc