Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.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 <algorithm>
 #include <iterator>
 #include <list>
+#include <memory>
 #include <set>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
@@ skipping 70 matching lines @@
                const PacketOptions& options) override {
     count_++;
     const unsigned char* ptr = static_cast<const unsigned char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
     if (ssrc == rtx_ssrc_)
       count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
     size_t packet_length = len;
     uint8_t restored_packet[1500];
     RTPHeader header;
-    rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
       return false;
     }
 
     if (!rtp_payload_registry_->IsRtx(header)) {
       // Don't store retransmitted packets since we compare it to the list
       // created by the receiver.
       expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
                                         sequence_number);
     }
@@ skipping 153 matching lines @@
       fake_clock.AdvanceTimeMilliseconds(28);  //  33ms - 5ms delay.
       rtp_rtcp_module_->Process();
       // Prepare next frame.
       timestamp += 3000;
     }
     receiver_.sequence_numbers_.sort();
   }
 
   void TearDown() override { delete rtp_rtcp_module_; }
 
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTPPayloadRegistry rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* rtp_rtcp_module_;
-  rtc::scoped_ptr<TestRtpFeedback> rtp_feedback_;
+  std::unique_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
   VerifyingRtxReceiver receiver_;
   uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
 };
 
 TEST_F(RtpRtcpRtxNackTest, LongNackList) {
   const int kNumPacketsToDrop = 900;
   const int kNumRequiredRtcp = 4;
@@ skipping 36 matching lines @@
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
             *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
 
 }  // namespace webrtc