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}
98 lines
3.2 KiB
C++
98 lines
3.2 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/rtt_stats.h"
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#include <algorithm>
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#include <string>
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#include <type_traits>
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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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namespace {
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// Default initial rtt used before any samples are received.
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const int kInitialRttMs = 100;
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const double kAlpha = 0.125;
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const double kOneMinusAlpha = (1 - kAlpha);
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const double kBeta = 0.25;
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const double kOneMinusBeta = (1 - kBeta);
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const int64_t kNumMicrosPerMilli = 1000;
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} // namespace
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RttStats::RttStats()
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: latest_rtt_(TimeDelta::Zero()),
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min_rtt_(TimeDelta::Zero()),
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smoothed_rtt_(TimeDelta::Zero()),
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previous_srtt_(TimeDelta::Zero()),
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mean_deviation_(TimeDelta::Zero()),
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initial_rtt_us_(kInitialRttMs * kNumMicrosPerMilli) {}
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void RttStats::ExpireSmoothedMetrics() {
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mean_deviation_ =
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std::max(mean_deviation_, (smoothed_rtt_ - latest_rtt_).Abs());
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smoothed_rtt_ = std::max(smoothed_rtt_, latest_rtt_);
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}
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// Updates the RTT based on a new sample.
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void RttStats::UpdateRtt(TimeDelta send_delta,
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TimeDelta ack_delay,
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Timestamp now) {
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if (send_delta.IsInfinite() || send_delta <= TimeDelta::Zero()) {
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RTC_LOG(LS_WARNING) << "Ignoring measured send_delta, because it's is "
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<< "either infinite, zero, or negative. send_delta = "
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<< ToString(send_delta);
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return;
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}
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// Update min_rtt_ first. min_rtt_ does not use an rtt_sample corrected for
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// ack_delay but the raw observed send_delta, since poor clock granularity at
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// the client may cause a high ack_delay to result in underestimation of the
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// min_rtt_.
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if (min_rtt_.IsZero() || min_rtt_ > send_delta) {
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min_rtt_ = send_delta;
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}
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// Correct for ack_delay if information received from the peer results in a
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// positive RTT sample. Otherwise, we use the send_delta as a reasonable
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// measure for smoothed_rtt.
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TimeDelta rtt_sample = send_delta;
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previous_srtt_ = smoothed_rtt_;
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if (rtt_sample > ack_delay) {
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rtt_sample = rtt_sample - ack_delay;
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}
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latest_rtt_ = rtt_sample;
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// First time call.
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if (smoothed_rtt_.IsZero()) {
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smoothed_rtt_ = rtt_sample;
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mean_deviation_ = rtt_sample / 2;
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} else {
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mean_deviation_ = kOneMinusBeta * mean_deviation_ +
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kBeta * (smoothed_rtt_ - rtt_sample).Abs();
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smoothed_rtt_ = kOneMinusAlpha * smoothed_rtt_ + kAlpha * rtt_sample;
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RTC_LOG(LS_VERBOSE) << " smoothed_rtt(us):" << smoothed_rtt_.us()
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<< " mean_deviation(us):" << mean_deviation_.us();
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}
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}
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void RttStats::OnConnectionMigration() {
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latest_rtt_ = TimeDelta::Zero();
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min_rtt_ = TimeDelta::Zero();
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smoothed_rtt_ = TimeDelta::Zero();
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mean_deviation_ = TimeDelta::Zero();
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initial_rtt_us_ = kInitialRttMs * kNumMicrosPerMilli;
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}
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} // namespace bbr
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} // namespace webrtc
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