3e70781361
This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
84 lines
2.7 KiB
C++
84 lines
2.7 KiB
C++
/*
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* Copyright (c) 2018 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/congestion_controller/bbr/data_transfer_tracker.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace bbr {
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DataTransferTracker::DataTransferTracker() {}
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DataTransferTracker::~DataTransferTracker() {}
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void DataTransferTracker::AddSample(DataSize size_delta,
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Timestamp send_time,
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Timestamp ack_time) {
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size_sum_ += size_delta;
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RTC_DCHECK(samples_.empty() || ack_time >= samples_.back().ack_time);
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if (!samples_.empty() && ack_time == samples_.back().ack_time) {
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samples_.back().send_time = send_time;
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samples_.back().size_sum = size_sum_;
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} else {
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Sample new_sample;
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new_sample.ack_time = ack_time;
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new_sample.send_time = send_time;
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new_sample.size_delta = size_delta;
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new_sample.size_sum = size_sum_;
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samples_.push_back(new_sample);
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}
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}
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void DataTransferTracker::ClearOldSamples(Timestamp excluding_end) {
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while (!samples_.empty() && samples_.front().ack_time < excluding_end) {
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samples_.pop_front();
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}
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}
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DataTransferTracker::Result DataTransferTracker::GetRatesByAckTime(
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Timestamp covered_start,
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Timestamp including_end) {
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Result res;
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// Last sample before covered_start.
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const Sample* window_begin = nullptr;
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// Sample at end time or first sample after end time-
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const Sample* window_end = nullptr;
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// To handle the case when the first sample is after covered_start.
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if (samples_.front().ack_time < including_end)
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window_begin = &samples_.front();
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// To handle the case when the last sample is before including_end.
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if (samples_.back().ack_time > covered_start)
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window_end = &samples_.back();
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for (const auto& sample : samples_) {
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if (sample.ack_time < covered_start) {
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window_begin = &sample;
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} else if (sample.ack_time >= including_end) {
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window_end = &sample;
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break;
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}
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}
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if (window_begin != nullptr && window_end != nullptr) {
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res.acked_data = window_end->size_sum - window_begin->size_sum;
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res.send_timespan = window_end->send_time - window_begin->send_time;
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res.ack_timespan = window_end->ack_time - window_begin->ack_time;
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} else {
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res.acked_data = DataSize::Zero();
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res.ack_timespan = including_end - covered_start;
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res.send_timespan = TimeDelta::Zero();
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}
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return res;
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}
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} // namespace bbr
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} // namespace webrtc
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