blob: bbe5e42883be1d0e0fc658327f7be8129ed576eb [file] [log] [blame]
/*
* Copyright (c) 2018 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/congestion_controller/bbr/rtt_stats.h"
#include <algorithm>
#include <string>
#include <type_traits>
#include "rtc_base/logging.h"
namespace webrtc {
namespace bbr {
namespace {
// Default initial rtt used before any samples are received.
const int kInitialRttMs = 100;
const double kAlpha = 0.125;
const double kOneMinusAlpha = (1 - kAlpha);
const double kBeta = 0.25;
const double kOneMinusBeta = (1 - kBeta);
const int64_t kNumMicrosPerMilli = 1000;
} // namespace
RttStats::RttStats()
: latest_rtt_(TimeDelta::Zero()),
min_rtt_(TimeDelta::Zero()),
smoothed_rtt_(TimeDelta::Zero()),
previous_srtt_(TimeDelta::Zero()),
mean_deviation_(TimeDelta::Zero()),
initial_rtt_us_(kInitialRttMs * kNumMicrosPerMilli) {}
void RttStats::ExpireSmoothedMetrics() {
mean_deviation_ =
std::max(mean_deviation_, (smoothed_rtt_ - latest_rtt_).Abs());
smoothed_rtt_ = std::max(smoothed_rtt_, latest_rtt_);
}
// Updates the RTT based on a new sample.
void RttStats::UpdateRtt(TimeDelta send_delta,
TimeDelta ack_delay,
Timestamp now) {
if (send_delta.IsInfinite() || send_delta <= TimeDelta::Zero()) {
RTC_LOG(LS_WARNING) << "Ignoring measured send_delta, because it's is "
<< "either infinite, zero, or negative. send_delta = "
<< ToString(send_delta);
return;
}
// Update min_rtt_ first. min_rtt_ does not use an rtt_sample corrected for
// ack_delay but the raw observed send_delta, since poor clock granularity at
// the client may cause a high ack_delay to result in underestimation of the
// min_rtt_.
if (min_rtt_.IsZero() || min_rtt_ > send_delta) {
min_rtt_ = send_delta;
}
// Correct for ack_delay if information received from the peer results in a
// positive RTT sample. Otherwise, we use the send_delta as a reasonable
// measure for smoothed_rtt.
TimeDelta rtt_sample = send_delta;
previous_srtt_ = smoothed_rtt_;
if (rtt_sample > ack_delay) {
rtt_sample = rtt_sample - ack_delay;
}
latest_rtt_ = rtt_sample;
// First time call.
if (smoothed_rtt_.IsZero()) {
smoothed_rtt_ = rtt_sample;
mean_deviation_ = rtt_sample / 2;
} else {
mean_deviation_ = kOneMinusBeta * mean_deviation_ +
kBeta * (smoothed_rtt_ - rtt_sample).Abs();
smoothed_rtt_ = kOneMinusAlpha * smoothed_rtt_ + kAlpha * rtt_sample;
RTC_LOG(LS_VERBOSE) << " smoothed_rtt(us):" << smoothed_rtt_.us()
<< " mean_deviation(us):" << mean_deviation_.us();
}
}
void RttStats::OnConnectionMigration() {
latest_rtt_ = TimeDelta::Zero();
min_rtt_ = TimeDelta::Zero();
smoothed_rtt_ = TimeDelta::Zero();
mean_deviation_ = TimeDelta::Zero();
initial_rtt_us_ = kInitialRttMs * kNumMicrosPerMilli;
}
} // namespace bbr
} // namespace webrtc