Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/

webrtc/modules/rtp_rtcp/test/testAPI/test_api.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 "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "webrtc/test/null_transport.h"
 
 namespace webrtc {
 
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
                                       ReceiveStatistics* receive_statistics) {
@@ skipping 10 matching lines @@
 bool LoopBackTransport::SendRtp(const uint8_t* data,
                                 size_t len,
                                 const PacketOptions& options) {
   count_++;
   if (packet_loss_ > 0) {
     if ((count_ % packet_loss_) == 0) {
       return true;
     }
   }
   RTPHeader header;
-  rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
   if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
     return false;
   }
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
     return false;
   }
   receive_statistics_->IncomingPacket(header, len, false);
   if (!rtp_receiver_->IncomingRtpPacket(header,
@@ skipping 38 matching lines @@
     configuration.audio = true;
     configuration.clock = &fake_clock_;
     configuration.outgoing_transport = &null_transport_;
     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
     rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
         &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
   }
 
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
-  rtc::scoped_ptr<RtpRtcp> module_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpRtcp> module_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
   std::vector<uint32_t> test_csrcs_;
   SimulatedClock fake_clock_;
   test::NullTransport null_transport_;
 };
 
 TEST_F(RtpRtcpAPITest, Basic) {
   module_->SetSequenceNumber(test_sequence_number_);
@@ skipping 65 matching lines @@
   rtx_header.payloadType = kRtxPayloadType;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = 0;
   EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
 
 }  // namespace webrtc