Remove webrtc/base/scoped_ptr.h

webrtc/modules/video_coding/test/rtp_player.cc

Index: webrtc/modules/video_coding/test/rtp_player.cc
diff --git a/webrtc/modules/video_coding/test/rtp_player.cc b/webrtc/modules/video_coding/test/rtp_player.cc
index 41cd360d133d0b137f09a28e9fc48e407051fa39..d5fa9ae936c02130e738f026d96be23b4de07629 100644
--- a/webrtc/modules/video_coding/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/test/rtp_player.cc
@@ -12,6 +12,7 @@
 
 #include <stdio.h>
 
+#include <cstdlib>
 #include <map>
 #include <memory>
 
@@ -341,7 +342,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
     assert(packet_source);
     assert(packet_source->get());
     packet_source_.swap(*packet_source);
-    srand(321);
+    std::srand(321);
   }
 
   virtual ~RtpPlayerImpl() {}
@@ -434,7 +435,8 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
       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);