| /* |
| * 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 <cassert> |
| #include <cmath> |
| #include <cstdio> |
| #include <string> |
| |
| #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"; |
| |
| bool IsEnabled(const WebRtcKeyValueConfig& field_trials, |
| absl::string_view key) { |
| return field_trials.Lookup(key).find("Enabled") == 0; |
| } |
| |
| bool IsNotDisabled(const WebRtcKeyValueConfig& field_trials, |
| absl::string_view key) { |
| return field_trials.Lookup(key).find("Disabled") != 0; |
| } |
| |
| double ReadBackoffFactor(const WebRtcKeyValueConfig& 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(WARNING) << "Back-off factor must be less than 1."; |
| } else if (backoff_factor <= 0.0) { |
| RTC_LOG(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 WebRtcKeyValueConfig* key_value_config) |
| : AimdRateControl(key_value_config, /* send_side =*/false) {} |
| |
| AimdRateControl::AimdRateControl(const WebRtcKeyValueConfig* key_value_config, |
| bool send_side) |
| : min_configured_bitrate_(congestion_controller::GetMinBitrate()), |
| max_configured_bitrate_(DataRate::kbps(30000)), |
| current_bitrate_(max_configured_bitrate_), |
| latest_estimated_throughput_(current_bitrate_), |
| link_capacity_(), |
| rate_control_state_(kRcHold), |
| time_last_bitrate_change_(Timestamp::MinusInfinity()), |
| time_last_bitrate_decrease_(Timestamp::MinusInfinity()), |
| time_first_throughput_estimate_(Timestamp::MinusInfinity()), |
| bitrate_is_initialized_(false), |
| beta_(IsEnabled(*key_value_config, kBweBackOffFactorExperiment) |
| ? ReadBackoffFactor(*key_value_config) |
| : kDefaultBackoffFactor), |
| in_alr_(false), |
| rtt_(kDefaultRtt), |
| send_side_(send_side), |
| in_experiment_(!AdaptiveThresholdExperimentIsDisabled(*key_value_config)), |
| no_bitrate_increase_in_alr_( |
| IsEnabled(*key_value_config, |
| "WebRTC-DontIncreaseDelayBasedBweInAlr")), |
| smoothing_experiment_( |
| IsEnabled(*key_value_config, "WebRTC-Audio-BandwidthSmoothing")), |
| estimate_bounded_backoff_( |
| IsNotDisabled(*key_value_config, |
| "WebRTC-Bwe-EstimateBoundedBackoff")), |
| estimate_bounded_increase_( |
| IsNotDisabled(*key_value_config, |
| "WebRTC-Bwe-EstimateBoundedIncrease")), |
| initial_backoff_interval_("initial_backoff_interval"), |
| low_throughput_threshold_("low_throughput", DataRate::Zero()), |
| link_capacity_fix_("link_capacity_fix") { |
| // E.g |
| // WebRTC-BweAimdRateControlConfig/initial_backoff_interval:100ms, |
| // low_throughput:50kbps/ |
| ParseFieldTrial({&initial_backoff_interval_, &low_throughput_threshold_, |
| &link_capacity_fix_}, |
| key_value_config->Lookup("WebRTC-BweAimdRateControlConfig")); |
| if (initial_backoff_interval_) { |
| RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval" |
| " " |
| << ToString(*initial_backoff_interval_) << "."; |
| } |
| 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 { |
| if (!initial_backoff_interval_) { |
| return ValidEstimate() && |
| TimeToReduceFurther(at_time, |
| LatestEstimate() / 2 - DataRate::bps(1)); |
| } |
| // TODO(terelius): We could use the RTT (clamped to suitable limits) instead |
| // of a fixed bitrate_reduction_interval. |
| if (time_last_bitrate_decrease_.IsInfinite() || |
| at_time - time_last_bitrate_decrease_ >= *initial_backoff_interval_) { |
| return true; |
| } |
| return false; |
| } |
| |
| DataRate AimdRateControl::LatestEstimate() const { |
| return current_bitrate_; |
| } |
| |
| void AimdRateControl::SetRtt(TimeDelta rtt) { |
| rtt_ = rtt; |
| } |
| |
| DataRate AimdRateControl::Update(const RateControlInput* input, |
| Timestamp at_time) { |
| RTC_CHECK(input); |
| |
| // 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(kBitrateWindowMs, kInitializationTime.ms()); |
| 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; |
| } |
| } |
| |
| current_bitrate_ = ChangeBitrate(current_bitrate_, *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, 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); |
| if (in_experiment_) |
| 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 = |
| smoothing_experiment_ ? TimeDelta::Millis(500) : 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 smoothing_experiment_ ? kMinPeriod : 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); |
| } |
| |
| DataRate AimdRateControl::ChangeBitrate(DataRate new_bitrate, |
| const RateControlInput& input, |
| Timestamp at_time) { |
| 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 current_bitrate_; |
| |
| ChangeState(input, at_time); |
| |
| switch (rate_control_state_) { |
| case kRcHold: |
| break; |
| |
| case kRcIncrease: |
| if (estimated_throughput > link_capacity_.UpperBound()) |
| link_capacity_.Reset(); |
| |
| // 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 (!(send_side_ && in_alr_ && no_bitrate_increase_in_alr_)) { |
| 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_); |
| new_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_, new_bitrate); |
| new_bitrate += multiplicative_increase; |
| } |
| } |
| |
| time_last_bitrate_change_ = at_time; |
| break; |
| |
| case kRcDecrease: |
| if (estimated_throughput > low_throughput_threshold_) { |
| // Set bit rate to something slightly lower than the measured throughput |
| // to get rid of any self-induced delay. |
| new_bitrate = estimated_throughput * beta_; |
| if (new_bitrate > current_bitrate_ && !link_capacity_fix_) { |
| // 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()) { |
| new_bitrate = beta_ * link_capacity_.estimate(); |
| } |
| } |
| if (estimate_bounded_backoff_ && network_estimate_) { |
| new_bitrate = std::max( |
| new_bitrate, network_estimate_->link_capacity_lower * beta_); |
| } |
| } else { |
| new_bitrate = estimated_throughput; |
| if (link_capacity_.has_estimate()) { |
| new_bitrate = std::max(new_bitrate, link_capacity_.estimate()); |
| } |
| new_bitrate = std::min(new_bitrate, low_throughput_threshold_.Get()); |
| } |
| // Avoid increasing the rate when over-using. |
| new_bitrate = std::min(new_bitrate, current_bitrate_); |
| |
| if (bitrate_is_initialized_ && estimated_throughput < current_bitrate_) { |
| constexpr double kDegradationFactor = 0.9; |
| if (smoothing_experiment_ && |
| new_bitrate < kDegradationFactor * beta_ * current_bitrate_) { |
| // If bitrate decreases more than a normal back off after overuse, it |
| // indicates a real network degradation. We do not let such a decrease |
| // to determine the bandwidth estimation period. |
| last_decrease_ = absl::nullopt; |
| } 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_ = kRcHold; |
| time_last_bitrate_change_ = at_time; |
| time_last_bitrate_decrease_ = at_time; |
| break; |
| |
| default: |
| assert(false); |
| } |
| return ClampBitrate(new_bitrate, estimated_throughput); |
| } |
| |
| DataRate AimdRateControl::ClampBitrate(DataRate new_bitrate, |
| DataRate estimated_throughput) const { |
| // Allow the estimate to increase as long as alr is not detected to ensure |
| // that there is no BWE values that can make the estimate stuck at a too |
| // low bitrate. If an encoder can not produce the bitrate necessary to |
| // fully use the capacity, alr will sooner or later trigger. |
| if (!(send_side_ && no_bitrate_increase_in_alr_)) { |
| // Don't change the bit rate if the send side is too far off. |
| // We allow a bit more lag at very low rates to not too easily get stuck if |
| // the encoder produces uneven outputs. |
| const DataRate max_bitrate = |
| 1.5 * estimated_throughput + DataRate::kbps(10); |
| if (new_bitrate > current_bitrate_ && new_bitrate > max_bitrate) { |
| new_bitrate = std::max(current_bitrate_, max_bitrate); |
| } |
| } |
| |
| if (estimate_bounded_increase_ && network_estimate_) { |
| DataRate upper_bound = network_estimate_->link_capacity_upper; |
| new_bitrate = std::min(new_bitrate, upper_bound); |
| } |
| 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::bps(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::bps(data_rate_increase_bps); |
| } |
| |
| void AimdRateControl::ChangeState(const RateControlInput& input, |
| Timestamp at_time) { |
| switch (input.bw_state) { |
| case BandwidthUsage::kBwNormal: |
| if (rate_control_state_ == kRcHold) { |
| time_last_bitrate_change_ = at_time; |
| rate_control_state_ = kRcIncrease; |
| } |
| break; |
| case BandwidthUsage::kBwOverusing: |
| if (rate_control_state_ != kRcDecrease) { |
| rate_control_state_ = kRcDecrease; |
| } |
| break; |
| case BandwidthUsage::kBwUnderusing: |
| rate_control_state_ = kRcHold; |
| break; |
| default: |
| assert(false); |
| } |
| } |
| |
| } // namespace webrtc |
