/* * 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 #include #include #include #include #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")), subtract_additional_backoff_term_(!key_value_config.IsDisabled( "WebRTC-Bwe-SubtractAdditionalBackoffTerm")) { 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& 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 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 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) && subtract_additional_backoff_term_) { 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(), 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 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