Index: webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc |
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc |
index b77550f12120ef55a3309c9f0e71d4c3542440a3..aceb3e979f9bf32a89d524eb93eb0caae49a47bd 100644 |
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc |
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc |
@@ -60,7 +60,7 @@ int64_t RtpStream::GenerateFrame(int64_t time_now_us, PacketList* packets) { |
if (time_now_us < next_rtp_time_) { |
return next_rtp_time_; |
} |
- assert(packets != NULL); |
+ assert(packets != nullptr); |
size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_; |
size_t n_packets = |
std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u); |
@@ -86,7 +86,7 @@ int64_t RtpStream::next_rtp_time() const { |
// Generates an RTCP packet. |
RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) { |
if (time_now_us < next_rtcp_time_) { |
- return NULL; |
+ return nullptr; |
} |
RtcpPacket* rtcp = new RtcpPacket; |
int64_t send_time_us = time_now_us + kSendSideOffsetUs; |
@@ -171,7 +171,7 @@ void StreamGenerator::set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset) { |
// it possible to simulate different types of channels. |
int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets, |
int64_t time_now_us) { |
- assert(packets != NULL); |
+ assert(packets != nullptr); |
assert(packets->empty()); |
assert(capacity_ > 0); |
StreamMap::iterator it = std::min_element(streams_.begin(), streams_.end(), |