blob: 81f1b0108a4fd8b1d3b8600ee9c10e3ba5a0f24a [file] [log] [blame]
/*
* Copyright 2019 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/include/report_block_data.h"
namespace webrtc {
TimeDelta ReportBlockData::AvgRtt() const {
return num_rtts_ > 0 ? sum_rtt_ / num_rtts_ : TimeDelta::Zero();
}
void ReportBlockData::SetReportBlock(uint32_t sender_ssrc,
const rtcp::ReportBlock& report_block,
Timestamp report_block_timestamp_utc) {
report_block_.sender_ssrc = sender_ssrc;
report_block_.source_ssrc = report_block.source_ssrc();
report_block_.fraction_lost = report_block.fraction_lost();
report_block_.packets_lost = report_block.cumulative_lost_signed();
report_block_.extended_highest_sequence_number =
report_block.extended_high_seq_num();
report_block_.jitter = report_block.jitter();
report_block_.delay_since_last_sender_report =
report_block.delay_since_last_sr();
report_block_.last_sender_report_timestamp = report_block.last_sr();
report_block_timestamp_utc_ = report_block_timestamp_utc;
}
void ReportBlockData::AddRoundTripTimeSample(TimeDelta rtt) {
if (rtt > max_rtt_)
max_rtt_ = rtt;
if (num_rtts_ == 0 || rtt < min_rtt_)
min_rtt_ = rtt;
last_rtt_ = rtt;
sum_rtt_ += rtt;
++num_rtts_;
}
} // namespace webrtc