Remove webrtc/base/scoped_ptr.h

webrtc/modules/video_coding/test/rtp_player.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/video_coding/test/rtp_player.h"
 
 #include <stdio.h>
 
+#include <cstdlib>
 #include <map>
 #include <memory>
 
 #include "webrtc/base/constructormagic.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/video_coding/internal_defines.h"
 #include "webrtc/modules/video_coding/test/test_util.h"
@@ skipping 310 matching lines @@
         lost_packets_(clock, rtt_ms),
         resend_packet_count_(0),
         no_loss_startup_(100),
         end_of_file_(false),
         reordering_(false),
         reorder_buffer_() {
     assert(clock);
     assert(packet_source);
     assert(packet_source->get());
     packet_source_.swap(*packet_source);
-    srand(321);
+    std::srand(321);
   }
 
   virtual ~RtpPlayerImpl() {}
 
   virtual int NextPacket(int64_t time_now) {
     // Send any packets ready to be resent.
     for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
          packet != NULL; packet = lost_packets_.NextPacketToResend(time_now)) {
       int ret = SendPacket(packet->data(), packet->length());
       if (ret > 0) {
@@ skipping 72 matching lines @@
         return -1;
       }
       uint32_t ssrc = header.ssrc;
       if (ssrc_handlers_.RegisterSsrc(ssrc, &lost_packets_, clock_) < 0) {
         DEBUG_LOG1("Unable to register ssrc: %d", ssrc);
         return -1;
       }
 
       if (no_loss_startup_ > 0) {
         no_loss_startup_--;
-      } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) {  // NOLINT
+      } else if ((std::rand() + 1.0) / (RAND_MAX + 1.0) <
+                 loss_rate_) {  // NOLINT
         uint16_t seq_num = header.sequenceNumber;
         lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
         DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
         return 0;
       }
     }
 
     ssrc_handlers_.IncomingPacket(data, length);
     return 1;
   }
@@ skipping 35 matching lines @@
     }
   }
 
   std::unique_ptr<RtpPlayerImpl> impl(
       new RtpPlayerImpl(payload_sink_factory, payload_types, clock,
                         &packet_source, loss_rate, rtt_ms, reordering));
   return impl.release();
 }
 }  // namespace rtpplayer
 }  // namespace webrtc