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 87b6c9056081e4e5d9d8ffb689c7a955517986c3..816960cbe008ceaebf88e4373c403be4eb6230e4 100644 |
--- a/webrtc/modules/video_coding/test/rtp_player.cc |
+++ b/webrtc/modules/video_coding/test/rtp_player.cc |
@@ -12,7 +12,6 @@ |
#include <stdio.h> |
-#include <cstdlib> |
#include <map> |
#include <memory> |
@@ -343,7 +342,7 @@ |
assert(packet_source); |
assert(packet_source->get()); |
packet_source_.swap(*packet_source); |
- std::srand(321); |
+ srand(321); |
} |
virtual ~RtpPlayerImpl() {} |
@@ -436,8 +435,7 @@ |
if (no_loss_startup_ > 0) { |
no_loss_startup_--; |
- } else if ((std::rand() + 1.0) / (RAND_MAX + 1.0) < |
- loss_rate_) { // NOLINT |
+ } else if ((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); |