Replace scoped_ptr with unique_ptr in webrtc/modules/remote_bitrate_estimator/

webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc

 /*
  *  Copyright (c) 2014 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 <stdio.h>
 
+#include <memory>
+
 #include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 class Observer : public webrtc::RemoteBitrateObserver {
  public:
   explicit Observer(webrtc::Clock* clock) : clock_(clock) {}
 
@@ skipping 16 matching lines @@
   webrtc::test::RtpFileReader* reader;
   webrtc::RemoteBitrateEstimator* estimator;
   webrtc::RtpHeaderParser* parser;
   std::string estimator_used;
   webrtc::SimulatedClock clock(0);
   Observer observer(&clock);
   if (!ParseArgsAndSetupEstimator(argc, argv, &clock, &observer, &reader,
                                   &parser, &estimator, &estimator_used)) {
     return -1;
   }
-  rtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
-  rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
-  rtc::scoped_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator);
+  std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
+  std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
+  std::unique_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator);
 
   // Process the file.
   int packet_counter = 0;
   int64_t next_rtp_time_ms = 0;
   int64_t first_rtp_time_ms = -1;
   int abs_send_time_count = 0;
   int ts_offset_count = 0;
   webrtc::test::RtpPacket packet;
   if (!rtp_reader->NextPacket(&packet)) {
     printf("No RTP packet found\n");
@@ skipping 40 matching lines @@
          clock.TimeInMilliseconds());
   printf("Estimator used: %s\n", estimator_used.c_str());
   printf("Packets with absolute send time: %d\n",
          abs_send_time_count);
   printf("Packets with timestamp offset: %d\n",
          ts_offset_count);
   printf("Packets with no extension: %d\n",
          packet_counter - ts_offset_count - abs_send_time_count);
   return 0;
 }