| 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);
|
|
|