| 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(),
|
|
|