Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  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.
  */
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
@@ skipping 106 matching lines @@
     delete module2;
     delete transport1;
     delete transport2;
     delete data_receiver1;
     delete data_receiver2;
     delete rtp_callback;
   }
 
   RtpRtcp* module1;
   RtpRtcp* module2;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver1_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver2_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics1_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics2_;
+  std::unique_ptr<RtpReceiver> rtp_receiver1_;
+  std::unique_ptr<RtpReceiver> rtp_receiver2_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
   RTPCallback* rtp_callback;
   uint32_t test_ssrc;
   uint32_t test_timestamp;
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
@@ skipping 184 matching lines @@
   for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc