Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.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 <stdlib.h>
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.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/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
@@ skipping 96 matching lines @@
     return padding_bytes_in_packet + header_length;
   }
 
   virtual void TearDown() {
     delete video_module_;
     delete transport_;
     delete receiver_;
   }
 
   int test_id_;
-  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* video_module_;
   LoopBackTransport* transport_;
   TestRtpReceiver* receiver_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
   uint8_t  video_frame_[65000];
   size_t payload_data_length_;
   SimulatedClock fake_clock;
 };
@@ skipping 20 matching lines @@
                                                      codec.plType,
                                                      90000,
                                                      0,
                                                      codec.maxBitrate));
   for (int frame_idx = 0; frame_idx < 10; ++frame_idx) {
     for (int packet_idx = 0; packet_idx < 5; ++packet_idx) {
       size_t packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
                                          kPadSize);
       ++seq_num;
       RTPHeader header;
-      rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+      std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
       EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header));
       PayloadUnion payload_specific;
       EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
                                                             &payload_specific));
       const uint8_t* payload = padding_packet + header.headerLength;
       const size_t payload_length = packet_size - header.headerLength;
       EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, payload,
                                                    payload_length,
                                                    payload_specific, true));
       EXPECT_EQ(0u, receiver_->payload_size());
       EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
     }
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
   }
 }
 
 }  // namespace webrtc