Source code
Revision control
Copy as Markdown
Other Tools
/*
* Copyright (c) 2014 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/remote_bitrate_estimator/aimd_rate_control.h"
#include <inttypes.h>
#include <algorithm>
#include <cmath>
#include <cstdio>
#include <string>
#include "absl/strings/match.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace {
constexpr TimeDelta kDefaultRtt = TimeDelta::Millis(200);
constexpr double kDefaultBackoffFactor = 0.85;
constexpr char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
double ReadBackoffFactor(const FieldTrialsView& key_value_config) {
std::string experiment_string =
key_value_config.Lookup(kBweBackOffFactorExperiment);
double backoff_factor;
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%lf", &backoff_factor);
if (parsed_values == 1) {
if (backoff_factor >= 1.0) {
RTC_LOG(LS_WARNING) << "Back-off factor must be less than 1.";
} else if (backoff_factor <= 0.0) {
RTC_LOG(LS_WARNING) << "Back-off factor must be greater than 0.";
} else {
return backoff_factor;
}
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for AimdRateControl "
"experiment from field trial string. Using default.";
return kDefaultBackoffFactor;
}
} // namespace
AimdRateControl::AimdRateControl(const FieldTrialsView& key_value_config)
: AimdRateControl(key_value_config, /* send_side =*/false) {}
AimdRateControl::AimdRateControl(const FieldTrialsView& key_value_config,
bool send_side)
: min_configured_bitrate_(kCongestionControllerMinBitrate),
max_configured_bitrate_(DataRate::KilobitsPerSec(30000)),
current_bitrate_(max_configured_bitrate_),
latest_estimated_throughput_(current_bitrate_),
link_capacity_(),
rate_control_state_(RateControlState::kRcHold),
time_last_bitrate_change_(Timestamp::MinusInfinity()),
time_last_bitrate_decrease_(Timestamp::MinusInfinity()),
time_first_throughput_estimate_(Timestamp::MinusInfinity()),
bitrate_is_initialized_(false),
beta_(key_value_config.IsEnabled(kBweBackOffFactorExperiment)
? ReadBackoffFactor(key_value_config)
: kDefaultBackoffFactor),
in_alr_(false),
rtt_(kDefaultRtt),
send_side_(send_side),
no_bitrate_increase_in_alr_(
key_value_config.IsEnabled("WebRTC-DontIncreaseDelayBasedBweInAlr")) {
ParseFieldTrial(
{&disable_estimate_bounded_increase_,
&use_current_estimate_as_min_upper_bound_},
key_value_config.Lookup("WebRTC-Bwe-EstimateBoundedIncrease"));
RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
}
AimdRateControl::~AimdRateControl() {}
void AimdRateControl::SetStartBitrate(DataRate start_bitrate) {
current_bitrate_ = start_bitrate;
latest_estimated_throughput_ = current_bitrate_;
bitrate_is_initialized_ = true;
}
void AimdRateControl::SetMinBitrate(DataRate min_bitrate) {
min_configured_bitrate_ = min_bitrate;
current_bitrate_ = std::max(min_bitrate, current_bitrate_);
}
bool AimdRateControl::ValidEstimate() const {
return bitrate_is_initialized_;
}
TimeDelta AimdRateControl::GetFeedbackInterval() const {
// Estimate how often we can send RTCP if we allocate up to 5% of bandwidth
// to feedback.
const DataSize kRtcpSize = DataSize::Bytes(80);
const DataRate rtcp_bitrate = current_bitrate_ * 0.05;
const TimeDelta interval = kRtcpSize / rtcp_bitrate;
const TimeDelta kMinFeedbackInterval = TimeDelta::Millis(200);
const TimeDelta kMaxFeedbackInterval = TimeDelta::Millis(1000);
return interval.Clamped(kMinFeedbackInterval, kMaxFeedbackInterval);
}
bool AimdRateControl::TimeToReduceFurther(Timestamp at_time,
DataRate estimated_throughput) const {
const TimeDelta bitrate_reduction_interval =
rtt_.Clamped(TimeDelta::Millis(10), TimeDelta::Millis(200));
if (at_time - time_last_bitrate_change_ >= bitrate_reduction_interval) {
return true;
}
if (ValidEstimate()) {
// TODO(terelius/holmer): Investigate consequences of increasing
// the threshold to 0.95 * LatestEstimate().
const DataRate threshold = 0.5 * LatestEstimate();
return estimated_throughput < threshold;
}
return false;
}
bool AimdRateControl::InitialTimeToReduceFurther(Timestamp at_time) const {
return ValidEstimate() &&
TimeToReduceFurther(at_time,
LatestEstimate() / 2 - DataRate::BitsPerSec(1));
}
DataRate AimdRateControl::LatestEstimate() const {
return current_bitrate_;
}
void AimdRateControl::SetRtt(TimeDelta rtt) {
rtt_ = rtt;
}
DataRate AimdRateControl::Update(const RateControlInput& input,
Timestamp at_time) {
// Set the initial bit rate value to what we're receiving the first half
// second.
// TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
if (!bitrate_is_initialized_) {
const TimeDelta kInitializationTime = TimeDelta::Seconds(5);
RTC_DCHECK_LE(kBitrateWindow, kInitializationTime);
if (time_first_throughput_estimate_.IsInfinite()) {
if (input.estimated_throughput)
time_first_throughput_estimate_ = at_time;
} else if (at_time - time_first_throughput_estimate_ >
kInitializationTime &&
input.estimated_throughput) {
current_bitrate_ = *input.estimated_throughput;
bitrate_is_initialized_ = true;
}
}
ChangeBitrate(input, at_time);
return current_bitrate_;
}
void AimdRateControl::SetInApplicationLimitedRegion(bool in_alr) {
in_alr_ = in_alr;
}
void AimdRateControl::SetEstimate(DataRate bitrate, Timestamp at_time) {
bitrate_is_initialized_ = true;
DataRate prev_bitrate = current_bitrate_;
current_bitrate_ = ClampBitrate(bitrate);
time_last_bitrate_change_ = at_time;
if (current_bitrate_ < prev_bitrate) {
time_last_bitrate_decrease_ = at_time;
}
}
void AimdRateControl::SetNetworkStateEstimate(
const absl::optional<NetworkStateEstimate>& estimate) {
network_estimate_ = estimate;
}
double AimdRateControl::GetNearMaxIncreaseRateBpsPerSecond() const {
RTC_DCHECK(!current_bitrate_.IsZero());
const TimeDelta kFrameInterval = TimeDelta::Seconds(1) / 30;
DataSize frame_size = current_bitrate_ * kFrameInterval;
const DataSize kPacketSize = DataSize::Bytes(1200);
double packets_per_frame = std::ceil(frame_size / kPacketSize);
DataSize avg_packet_size = frame_size / packets_per_frame;
// Approximate the over-use estimator delay to 100 ms.
TimeDelta response_time = rtt_ + TimeDelta::Millis(100);
response_time = response_time * 2;
double increase_rate_bps_per_second =
(avg_packet_size / response_time).bps<double>();
double kMinIncreaseRateBpsPerSecond = 4000;
return std::max(kMinIncreaseRateBpsPerSecond, increase_rate_bps_per_second);
}
TimeDelta AimdRateControl::GetExpectedBandwidthPeriod() const {
const TimeDelta kMinPeriod = TimeDelta::Seconds(2);
const TimeDelta kDefaultPeriod = TimeDelta::Seconds(3);
const TimeDelta kMaxPeriod = TimeDelta::Seconds(50);
double increase_rate_bps_per_second = GetNearMaxIncreaseRateBpsPerSecond();
if (!last_decrease_)
return kDefaultPeriod;
double time_to_recover_decrease_seconds =
last_decrease_->bps() / increase_rate_bps_per_second;
TimeDelta period = TimeDelta::Seconds(time_to_recover_decrease_seconds);
return period.Clamped(kMinPeriod, kMaxPeriod);
}
void AimdRateControl::ChangeBitrate(const RateControlInput& input,
Timestamp at_time) {
absl::optional<DataRate> new_bitrate;
DataRate estimated_throughput =
input.estimated_throughput.value_or(latest_estimated_throughput_);
if (input.estimated_throughput)
latest_estimated_throughput_ = *input.estimated_throughput;
// An over-use should always trigger us to reduce the bitrate, even though
// we have not yet established our first estimate. By acting on the over-use,
// we will end up with a valid estimate.
if (!bitrate_is_initialized_ &&
input.bw_state != BandwidthUsage::kBwOverusing)
return;
ChangeState(input, at_time);
switch (rate_control_state_) {
case RateControlState::kRcHold:
break;
case RateControlState::kRcIncrease: {
if (estimated_throughput > link_capacity_.UpperBound())
link_capacity_.Reset();
// We limit the new bitrate based on the troughput to avoid unlimited
// bitrate increases. We allow a bit more lag at very low rates to not too
// easily get stuck if the encoder produces uneven outputs.
DataRate increase_limit =
1.5 * estimated_throughput + DataRate::KilobitsPerSec(10);
if (send_side_ && in_alr_ && no_bitrate_increase_in_alr_) {
// Do not increase the delay based estimate in alr since the estimator
// will not be able to get transport feedback necessary to detect if
// the new estimate is correct.
// If we have previously increased above the limit (for instance due to
// probing), we don't allow further changes.
increase_limit = current_bitrate_;
}
if (current_bitrate_ < increase_limit) {
DataRate increased_bitrate = DataRate::MinusInfinity();
if (link_capacity_.has_estimate()) {
// The link_capacity estimate is reset if the measured throughput
// is too far from the estimate. We can therefore assume that our
// target rate is reasonably close to link capacity and use additive
// increase.
DataRate additive_increase =
AdditiveRateIncrease(at_time, time_last_bitrate_change_);
increased_bitrate = current_bitrate_ + additive_increase;
} else {
// If we don't have an estimate of the link capacity, use faster ramp
// up to discover the capacity.
DataRate multiplicative_increase = MultiplicativeRateIncrease(
at_time, time_last_bitrate_change_, current_bitrate_);
increased_bitrate = current_bitrate_ + multiplicative_increase;
}
new_bitrate = std::min(increased_bitrate, increase_limit);
}
time_last_bitrate_change_ = at_time;
break;
}
case RateControlState::kRcDecrease: {
DataRate decreased_bitrate = DataRate::PlusInfinity();
// Set bit rate to something slightly lower than the measured throughput
// to get rid of any self-induced delay.
decreased_bitrate = estimated_throughput * beta_;
if (decreased_bitrate > DataRate::KilobitsPerSec(5)) {
decreased_bitrate -= DataRate::KilobitsPerSec(5);
}
if (decreased_bitrate > current_bitrate_) {
// TODO(terelius): The link_capacity estimate may be based on old
// throughput measurements. Relying on them may lead to unnecessary
// BWE drops.
if (link_capacity_.has_estimate()) {
decreased_bitrate = beta_ * link_capacity_.estimate();
}
}
// Avoid increasing the rate when over-using.
if (decreased_bitrate < current_bitrate_) {
new_bitrate = decreased_bitrate;
}
if (bitrate_is_initialized_ && estimated_throughput < current_bitrate_) {
if (!new_bitrate.has_value()) {
last_decrease_ = DataRate::Zero();
} else {
last_decrease_ = current_bitrate_ - *new_bitrate;
}
}
if (estimated_throughput < link_capacity_.LowerBound()) {
// The current throughput is far from the estimated link capacity. Clear
// the estimate to allow an immediate update in OnOveruseDetected.
link_capacity_.Reset();
}
bitrate_is_initialized_ = true;
link_capacity_.OnOveruseDetected(estimated_throughput);
// Stay on hold until the pipes are cleared.
rate_control_state_ = RateControlState::kRcHold;
time_last_bitrate_change_ = at_time;
time_last_bitrate_decrease_ = at_time;
break;
}
default:
RTC_DCHECK_NOTREACHED();
}
current_bitrate_ = ClampBitrate(new_bitrate.value_or(current_bitrate_));
}
DataRate AimdRateControl::ClampBitrate(DataRate new_bitrate) const {
if (!disable_estimate_bounded_increase_ && network_estimate_ &&
network_estimate_->link_capacity_upper.IsFinite()) {
DataRate upper_bound =
use_current_estimate_as_min_upper_bound_
? std::max(network_estimate_->link_capacity_upper, current_bitrate_)
: network_estimate_->link_capacity_upper;
new_bitrate = std::min(upper_bound, new_bitrate);
}
if (network_estimate_ && network_estimate_->link_capacity_lower.IsFinite() &&
new_bitrate < current_bitrate_) {
new_bitrate = std::min(
current_bitrate_,
std::max(new_bitrate, network_estimate_->link_capacity_lower * beta_));
}
new_bitrate = std::max(new_bitrate, min_configured_bitrate_);
return new_bitrate;
}
DataRate AimdRateControl::MultiplicativeRateIncrease(
Timestamp at_time,
Timestamp last_time,
DataRate current_bitrate) const {
double alpha = 1.08;
if (last_time.IsFinite()) {
auto time_since_last_update = at_time - last_time;
alpha = pow(alpha, std::min(time_since_last_update.seconds<double>(), 1.0));
}
DataRate multiplicative_increase =
std::max(current_bitrate * (alpha - 1.0), DataRate::BitsPerSec(1000));
return multiplicative_increase;
}
DataRate AimdRateControl::AdditiveRateIncrease(Timestamp at_time,
Timestamp last_time) const {
double time_period_seconds = (at_time - last_time).seconds<double>();
double data_rate_increase_bps =
GetNearMaxIncreaseRateBpsPerSecond() * time_period_seconds;
return DataRate::BitsPerSec(data_rate_increase_bps);
}
void AimdRateControl::ChangeState(const RateControlInput& input,
Timestamp at_time) {
switch (input.bw_state) {
case BandwidthUsage::kBwNormal:
if (rate_control_state_ == RateControlState::kRcHold) {
time_last_bitrate_change_ = at_time;
rate_control_state_ = RateControlState::kRcIncrease;
}
break;
case BandwidthUsage::kBwOverusing:
if (rate_control_state_ != RateControlState::kRcDecrease) {
rate_control_state_ = RateControlState::kRcDecrease;
}
break;
case BandwidthUsage::kBwUnderusing:
rate_control_state_ = RateControlState::kRcHold;
break;
default:
RTC_DCHECK_NOTREACHED();
}
}
} // namespace webrtc