d3eddff30c
These accessors would allow to deprecated report_block() accessor and then would allow to remove redundant RTCPReportBlock and ReportBlock types converging on single ReportBlockData type to pass that information across WebRTC components helpers like fraction_lost() and jitter() would also allow to unify conversion of the rtp specific format into more common way of represent such information Bug: None Change-Id: I3c97f96affcf83b529095899bd63af007f8b4014 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303880 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39975}
52 lines
1.8 KiB
C++
52 lines
1.8 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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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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report_block_.sender_ssrc = sender_ssrc;
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report_block_.source_ssrc = report_block.source_ssrc();
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report_block_.fraction_lost = report_block.fraction_lost();
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report_block_.packets_lost = report_block.cumulative_lost();
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report_block_.extended_highest_sequence_number =
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report_block.extended_high_seq_num();
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report_block_.jitter = report_block.jitter();
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report_block_.delay_since_last_sender_report =
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report_block.delay_since_last_sr();
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report_block_.last_sender_report_timestamp = report_block.last_sr();
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report_block_timestamp_utc_ = report_block_timestamp_utc;
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}
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void ReportBlockData::AddRoundTripTimeSample(TimeDelta rtt) {
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if (rtt > max_rtt_)
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max_rtt_ = rtt;
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if (num_rtts_ == 0 || rtt < min_rtt_)
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min_rtt_ = 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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