fea3280c13
Some TODOs with an old from where added in https://webrtc-review.googlesource.com/c/src/+/363946. This CL updates the TODO comments to the current form. Bug: None Change-Id: Id61dca5a0f4d705f4dfe74f6523dae3e357d49ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365140 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#43209}
47 lines
1.7 KiB
C++
47 lines
1.7 KiB
C++
/*
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* Copyright 2019 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/include/report_block_data.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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TimeDelta ReportBlockData::jitter(int rtp_clock_rate_hz) const {
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RTC_DCHECK_GT(rtp_clock_rate_hz, 0);
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// Conversion to TimeDelta and division are swapped to avoid conversion
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// to/from floating point types.
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return TimeDelta::Seconds(jitter()) / rtp_clock_rate_hz;
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}
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// TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
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// issue is fixed.
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void ReportBlockData::SetReportBlock(uint32_t sender_ssrc,
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const rtcp::ReportBlock& report_block,
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Timestamp report_block_timestamp_utc,
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Timestamp report_block_timestamp) {
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sender_ssrc_ = sender_ssrc;
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source_ssrc_ = report_block.source_ssrc();
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fraction_lost_raw_ = report_block.fraction_lost();
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cumulative_lost_ = report_block.cumulative_lost();
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extended_highest_sequence_number_ = report_block.extended_high_seq_num();
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jitter_ = report_block.jitter();
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report_block_timestamp_utc_ = report_block_timestamp_utc;
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report_block_timestamp_ = report_block_timestamp;
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}
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void ReportBlockData::AddRoundTripTimeSample(TimeDelta rtt) {
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last_rtt_ = rtt;
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sum_rtt_ += rtt;
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++num_rtts_;
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}
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} // namespace webrtc
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