mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 23:30:48 +01:00

Remove deprecated accessors returning time as raw int Add setters for all fields to simplify usage of this class in tests Remove unused min/max RTT fields Bug: webrtc:13757 Change-Id: Ia8966975c15b9a930f54b4db0fc75f7002dcffe1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304461 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40013}
42 lines
1.5 KiB
C++
42 lines
1.5 KiB
C++
/*
|
|
* 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) {
|
|
last_rtt_ = rtt;
|
|
sum_rtt_ += rtt;
|
|
++num_rtts_;
|
|
}
|
|
|
|
} // namespace webrtc
|