883f470d60
This class is completely new and tracks data transfer in a slightly different way compared to the BBR implementation in QUIC. The fundamental change is that receive time is used rather than packet index to identify the packets over which data rates should be calculated. This is part of a series of CLs adding a network controller based on the BBR congestion control method. The code is based on the QUIC BBR implementation in Chromium. Bug: webrtc:8415 Change-Id: I9d1f12634073ac89c4d542f965e3677a89a1526c Reviewed-on: https://webrtc-review.googlesource.com/63680 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22589}
86 lines
2.8 KiB
C++
86 lines
2.8 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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#include "rtc_base/logging.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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if (!samples_.empty()) {
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RTC_DCHECK_GE(send_time, samples_.back().send_time);
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RTC_DCHECK_GE(ack_time, samples_.back().ack_time);
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}
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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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