blob: 2998cc6ceed51ca87013f0c1f604f40f6c2199bb [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"
#include "rtc_base/checks.h"
namespace webrtc {
TimeDelta ReportBlockData::jitter(int rtp_clock_rate_hz) const {
RTC_DCHECK_GT(rtp_clock_rate_hz, 0);
// Conversion to TimeDelta and division are swapped to avoid conversion
// to/from floating point types.
return TimeDelta::Seconds(jitter()) / rtp_clock_rate_hz;
}
void ReportBlockData::SetReportBlock(uint32_t sender_ssrc,
const rtcp::ReportBlock& report_block,
Timestamp report_block_timestamp_utc) {
sender_ssrc_ = sender_ssrc;
source_ssrc_ = report_block.source_ssrc();
fraction_lost_raw_ = report_block.fraction_lost();
cumulative_lost_ = report_block.cumulative_lost();
extended_highest_sequence_number_ = report_block.extended_high_seq_num();
jitter_ = report_block.jitter();
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