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Diffstat (limited to 'third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc')
| -rw-r--r-- | third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc | 702 |
1 files changed, 702 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc new file mode 100644 index 0000000000..b09cb22f49 --- /dev/null +++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc @@ -0,0 +1,702 @@ +/* + * Copyright (c) 2012 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/goog_cc/send_side_bandwidth_estimation.h" + +#include <algorithm> +#include <cstdint> +#include <cstdio> +#include <limits> +#include <memory> +#include <string> +#include <utility> + +#include "absl/strings/match.h" +#include "absl/types/optional.h" +#include "api/field_trials_view.h" +#include "api/network_state_predictor.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/transport/network_types.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" +#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h" +#include "modules/remote_bitrate_estimator/include/bwe_defines.h" +#include "rtc_base/checks.h" +#include "rtc_base/experiments/field_trial_parser.h" +#include "rtc_base/logging.h" +#include "system_wrappers/include/field_trial.h" +#include "system_wrappers/include/metrics.h" + +namespace webrtc { +namespace { +constexpr TimeDelta kBweIncreaseInterval = TimeDelta::Millis(1000); +constexpr TimeDelta kBweDecreaseInterval = TimeDelta::Millis(300); +constexpr TimeDelta kStartPhase = TimeDelta::Millis(2000); +constexpr TimeDelta kBweConverganceTime = TimeDelta::Millis(20000); +constexpr int kLimitNumPackets = 20; +constexpr DataRate kDefaultMaxBitrate = DataRate::BitsPerSec(1000000000); +constexpr TimeDelta kLowBitrateLogPeriod = TimeDelta::Millis(10000); +constexpr TimeDelta kRtcEventLogPeriod = TimeDelta::Millis(5000); +// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals. +constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000); + +constexpr float kDefaultLowLossThreshold = 0.02f; +constexpr float kDefaultHighLossThreshold = 0.1f; +constexpr DataRate kDefaultBitrateThreshold = DataRate::Zero(); + +struct UmaRampUpMetric { + const char* metric_name; + int bitrate_kbps; +}; + +const UmaRampUpMetric kUmaRampupMetrics[] = { + {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500}, + {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000}, + {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}}; +const size_t kNumUmaRampupMetrics = + sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]); + +const char kBweLosExperiment[] = "WebRTC-BweLossExperiment"; + +bool BweLossExperimentIsEnabled() { + std::string experiment_string = + webrtc::field_trial::FindFullName(kBweLosExperiment); + // The experiment is enabled iff the field trial string begins with "Enabled". + return absl::StartsWith(experiment_string, "Enabled"); +} + +bool ReadBweLossExperimentParameters(float* low_loss_threshold, + float* high_loss_threshold, + uint32_t* bitrate_threshold_kbps) { + RTC_DCHECK(low_loss_threshold); + RTC_DCHECK(high_loss_threshold); + RTC_DCHECK(bitrate_threshold_kbps); + std::string experiment_string = + webrtc::field_trial::FindFullName(kBweLosExperiment); + int parsed_values = + sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold, + high_loss_threshold, bitrate_threshold_kbps); + if (parsed_values == 3) { + RTC_CHECK_GT(*low_loss_threshold, 0.0f) + << "Loss threshold must be greater than 0."; + RTC_CHECK_LE(*low_loss_threshold, 1.0f) + << "Loss threshold must be less than or equal to 1."; + RTC_CHECK_GT(*high_loss_threshold, 0.0f) + << "Loss threshold must be greater than 0."; + RTC_CHECK_LE(*high_loss_threshold, 1.0f) + << "Loss threshold must be less than or equal to 1."; + RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold) + << "The low loss threshold must be less than or equal to the high loss " + "threshold."; + RTC_CHECK_GE(*bitrate_threshold_kbps, 0) + << "Bitrate threshold can't be negative."; + RTC_CHECK_LT(*bitrate_threshold_kbps, + std::numeric_limits<int>::max() / 1000) + << "Bitrate must be smaller enough to avoid overflows."; + return true; + } + RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment " + "experiment from field trial string. Using default."; + *low_loss_threshold = kDefaultLowLossThreshold; + *high_loss_threshold = kDefaultHighLossThreshold; + *bitrate_threshold_kbps = kDefaultBitrateThreshold.kbps(); + return false; +} +} // namespace + +LinkCapacityTracker::LinkCapacityTracker() + : tracking_rate("rate", TimeDelta::Seconds(10)) { + ParseFieldTrial({&tracking_rate}, + field_trial::FindFullName("WebRTC-Bwe-LinkCapacity")); +} + +LinkCapacityTracker::~LinkCapacityTracker() {} + +void LinkCapacityTracker::UpdateDelayBasedEstimate( + Timestamp at_time, + DataRate delay_based_bitrate) { + if (delay_based_bitrate < last_delay_based_estimate_) { + capacity_estimate_bps_ = + std::min(capacity_estimate_bps_, delay_based_bitrate.bps<double>()); + last_link_capacity_update_ = at_time; + } + last_delay_based_estimate_ = delay_based_bitrate; +} + +void LinkCapacityTracker::OnStartingRate(DataRate start_rate) { + if (last_link_capacity_update_.IsInfinite()) + capacity_estimate_bps_ = start_rate.bps<double>(); +} + +void LinkCapacityTracker::OnRateUpdate(absl::optional<DataRate> acknowledged, + DataRate target, + Timestamp at_time) { + if (!acknowledged) + return; + DataRate acknowledged_target = std::min(*acknowledged, target); + if (acknowledged_target.bps() > capacity_estimate_bps_) { + TimeDelta delta = at_time - last_link_capacity_update_; + double alpha = delta.IsFinite() ? exp(-(delta / tracking_rate.Get())) : 0; + capacity_estimate_bps_ = alpha * capacity_estimate_bps_ + + (1 - alpha) * acknowledged_target.bps<double>(); + } + last_link_capacity_update_ = at_time; +} + +void LinkCapacityTracker::OnRttBackoff(DataRate backoff_rate, + Timestamp at_time) { + capacity_estimate_bps_ = + std::min(capacity_estimate_bps_, backoff_rate.bps<double>()); + last_link_capacity_update_ = at_time; +} + +DataRate LinkCapacityTracker::estimate() const { + return DataRate::BitsPerSec(capacity_estimate_bps_); +} + +RttBasedBackoff::RttBasedBackoff(const FieldTrialsView* key_value_config) + : disabled_("Disabled"), + configured_limit_("limit", TimeDelta::Seconds(3)), + drop_fraction_("fraction", 0.8), + drop_interval_("interval", TimeDelta::Seconds(1)), + bandwidth_floor_("floor", DataRate::KilobitsPerSec(5)), + rtt_limit_(TimeDelta::PlusInfinity()), + // By initializing this to plus infinity, we make sure that we never + // trigger rtt backoff unless packet feedback is enabled. + last_propagation_rtt_update_(Timestamp::PlusInfinity()), + last_propagation_rtt_(TimeDelta::Zero()), + last_packet_sent_(Timestamp::MinusInfinity()) { + ParseFieldTrial({&disabled_, &configured_limit_, &drop_fraction_, + &drop_interval_, &bandwidth_floor_}, + key_value_config->Lookup("WebRTC-Bwe-MaxRttLimit")); + if (!disabled_) { + rtt_limit_ = configured_limit_.Get(); + } +} + +void RttBasedBackoff::UpdatePropagationRtt(Timestamp at_time, + TimeDelta propagation_rtt) { + last_propagation_rtt_update_ = at_time; + last_propagation_rtt_ = propagation_rtt; +} + +bool RttBasedBackoff::IsRttAboveLimit() const { + return CorrectedRtt() > rtt_limit_; +} + +TimeDelta RttBasedBackoff::CorrectedRtt() const { + // Avoid timeout when no packets are being sent. + TimeDelta timeout_correction = std::max( + last_packet_sent_ - last_propagation_rtt_update_, TimeDelta::Zero()); + return timeout_correction + last_propagation_rtt_; +} + +RttBasedBackoff::~RttBasedBackoff() = default; + +SendSideBandwidthEstimation::SendSideBandwidthEstimation( + const FieldTrialsView* key_value_config, + RtcEventLog* event_log) + : rtt_backoff_(key_value_config), + lost_packets_since_last_loss_update_(0), + expected_packets_since_last_loss_update_(0), + current_target_(DataRate::Zero()), + last_logged_target_(DataRate::Zero()), + min_bitrate_configured_(kCongestionControllerMinBitrate), + max_bitrate_configured_(kDefaultMaxBitrate), + last_low_bitrate_log_(Timestamp::MinusInfinity()), + has_decreased_since_last_fraction_loss_(false), + last_loss_feedback_(Timestamp::MinusInfinity()), + last_loss_packet_report_(Timestamp::MinusInfinity()), + last_fraction_loss_(0), + last_logged_fraction_loss_(0), + last_round_trip_time_(TimeDelta::Zero()), + receiver_limit_(DataRate::PlusInfinity()), + delay_based_limit_(DataRate::PlusInfinity()), + time_last_decrease_(Timestamp::MinusInfinity()), + first_report_time_(Timestamp::MinusInfinity()), + initially_lost_packets_(0), + bitrate_at_2_seconds_(DataRate::Zero()), + uma_update_state_(kNoUpdate), + uma_rtt_state_(kNoUpdate), + rampup_uma_stats_updated_(kNumUmaRampupMetrics, false), + event_log_(event_log), + last_rtc_event_log_(Timestamp::MinusInfinity()), + low_loss_threshold_(kDefaultLowLossThreshold), + high_loss_threshold_(kDefaultHighLossThreshold), + bitrate_threshold_(kDefaultBitrateThreshold), + loss_based_bandwidth_estimator_v1_(key_value_config), + loss_based_bandwidth_estimator_v2_(key_value_config), + loss_based_state_(LossBasedState::kDelayBasedEstimate), + disable_receiver_limit_caps_only_("Disabled") { + RTC_DCHECK(event_log); + if (BweLossExperimentIsEnabled()) { + uint32_t bitrate_threshold_kbps; + if (ReadBweLossExperimentParameters(&low_loss_threshold_, + &high_loss_threshold_, + &bitrate_threshold_kbps)) { + RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters " + << low_loss_threshold_ << ", " << high_loss_threshold_ + << ", " << bitrate_threshold_kbps; + bitrate_threshold_ = DataRate::KilobitsPerSec(bitrate_threshold_kbps); + } + } + ParseFieldTrial({&disable_receiver_limit_caps_only_}, + key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly")); + if (LossBasedBandwidthEstimatorV2Enabled()) { + loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate( + min_bitrate_configured_, max_bitrate_configured_); + } +} + +SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {} + +void SendSideBandwidthEstimation::OnRouteChange() { + lost_packets_since_last_loss_update_ = 0; + expected_packets_since_last_loss_update_ = 0; + current_target_ = DataRate::Zero(); + min_bitrate_configured_ = kCongestionControllerMinBitrate; + max_bitrate_configured_ = kDefaultMaxBitrate; + last_low_bitrate_log_ = Timestamp::MinusInfinity(); + has_decreased_since_last_fraction_loss_ = false; + last_loss_feedback_ = Timestamp::MinusInfinity(); + last_loss_packet_report_ = Timestamp::MinusInfinity(); + last_fraction_loss_ = 0; + last_logged_fraction_loss_ = 0; + last_round_trip_time_ = TimeDelta::Zero(); + receiver_limit_ = DataRate::PlusInfinity(); + delay_based_limit_ = DataRate::PlusInfinity(); + time_last_decrease_ = Timestamp::MinusInfinity(); + first_report_time_ = Timestamp::MinusInfinity(); + initially_lost_packets_ = 0; + bitrate_at_2_seconds_ = DataRate::Zero(); + uma_update_state_ = kNoUpdate; + uma_rtt_state_ = kNoUpdate; + last_rtc_event_log_ = Timestamp::MinusInfinity(); +} + +void SendSideBandwidthEstimation::SetBitrates( + absl::optional<DataRate> send_bitrate, + DataRate min_bitrate, + DataRate max_bitrate, + Timestamp at_time) { + SetMinMaxBitrate(min_bitrate, max_bitrate); + if (send_bitrate) { + link_capacity_.OnStartingRate(*send_bitrate); + SetSendBitrate(*send_bitrate, at_time); + } +} + +void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate, + Timestamp at_time) { + RTC_DCHECK_GT(bitrate, DataRate::Zero()); + // Reset to avoid being capped by the estimate. + delay_based_limit_ = DataRate::PlusInfinity(); + UpdateTargetBitrate(bitrate, at_time); + // Clear last sent bitrate history so the new value can be used directly + // and not capped. + min_bitrate_history_.clear(); +} + +void SendSideBandwidthEstimation::SetMinMaxBitrate(DataRate min_bitrate, + DataRate max_bitrate) { + min_bitrate_configured_ = + std::max(min_bitrate, kCongestionControllerMinBitrate); + if (max_bitrate > DataRate::Zero() && max_bitrate.IsFinite()) { + max_bitrate_configured_ = std::max(min_bitrate_configured_, max_bitrate); + } else { + max_bitrate_configured_ = kDefaultMaxBitrate; + } + loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(min_bitrate_configured_, + max_bitrate_configured_); +} + +int SendSideBandwidthEstimation::GetMinBitrate() const { + return min_bitrate_configured_.bps<int>(); +} + +DataRate SendSideBandwidthEstimation::target_rate() const { + DataRate target = current_target_; + if (!disable_receiver_limit_caps_only_) + target = std::min(target, receiver_limit_); + return std::max(min_bitrate_configured_, target); +} + +LossBasedState SendSideBandwidthEstimation::loss_based_state() const { + return loss_based_state_; +} + +bool SendSideBandwidthEstimation::IsRttAboveLimit() const { + return rtt_backoff_.IsRttAboveLimit(); +} + +DataRate SendSideBandwidthEstimation::GetEstimatedLinkCapacity() const { + return link_capacity_.estimate(); +} + +void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time, + DataRate bandwidth) { + // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no + // limitation. + receiver_limit_ = bandwidth.IsZero() ? DataRate::PlusInfinity() : bandwidth; + ApplyTargetLimits(at_time); +} + +void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time, + DataRate bitrate) { + link_capacity_.UpdateDelayBasedEstimate(at_time, bitrate); + // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no + // limitation. + delay_based_limit_ = bitrate.IsZero() ? DataRate::PlusInfinity() : bitrate; + ApplyTargetLimits(at_time); +} + +void SendSideBandwidthEstimation::SetAcknowledgedRate( + absl::optional<DataRate> acknowledged_rate, + Timestamp at_time) { + acknowledged_rate_ = acknowledged_rate; + if (!acknowledged_rate.has_value()) { + return; + } + if (LossBasedBandwidthEstimatorV1Enabled()) { + loss_based_bandwidth_estimator_v1_.UpdateAcknowledgedBitrate( + *acknowledged_rate, at_time); + } + if (LossBasedBandwidthEstimatorV2Enabled()) { + loss_based_bandwidth_estimator_v2_.SetAcknowledgedBitrate( + *acknowledged_rate); + } +} + +void SendSideBandwidthEstimation::UpdateLossBasedEstimator( + const TransportPacketsFeedback& report, + BandwidthUsage delay_detector_state, + absl::optional<DataRate> probe_bitrate, + bool in_alr) { + if (LossBasedBandwidthEstimatorV1Enabled()) { + loss_based_bandwidth_estimator_v1_.UpdateLossStatistics( + report.packet_feedbacks, report.feedback_time); + } + if (LossBasedBandwidthEstimatorV2Enabled()) { + loss_based_bandwidth_estimator_v2_.UpdateBandwidthEstimate( + report.packet_feedbacks, delay_based_limit_, in_alr); + UpdateEstimate(report.feedback_time); + } +} + +void SendSideBandwidthEstimation::UpdatePacketsLost(int64_t packets_lost, + int64_t number_of_packets, + Timestamp at_time) { + last_loss_feedback_ = at_time; + if (first_report_time_.IsInfinite()) + first_report_time_ = at_time; + + // Check sequence number diff and weight loss report + if (number_of_packets > 0) { + int64_t expected = + expected_packets_since_last_loss_update_ + number_of_packets; + + // Don't generate a loss rate until it can be based on enough packets. + if (expected < kLimitNumPackets) { + // Accumulate reports. + expected_packets_since_last_loss_update_ = expected; + lost_packets_since_last_loss_update_ += packets_lost; + return; + } + + has_decreased_since_last_fraction_loss_ = false; + int64_t lost_q8 = + std::max<int64_t>(lost_packets_since_last_loss_update_ + packets_lost, + 0) + << 8; + last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255); + + // Reset accumulators. + lost_packets_since_last_loss_update_ = 0; + expected_packets_since_last_loss_update_ = 0; + last_loss_packet_report_ = at_time; + UpdateEstimate(at_time); + } + + UpdateUmaStatsPacketsLost(at_time, packets_lost); +} + +void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(Timestamp at_time, + int packets_lost) { + DataRate bitrate_kbps = + DataRate::KilobitsPerSec((current_target_.bps() + 500) / 1000); + for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) { + if (!rampup_uma_stats_updated_[i] && + bitrate_kbps.kbps() >= kUmaRampupMetrics[i].bitrate_kbps) { + RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name, + (at_time - first_report_time_).ms()); + rampup_uma_stats_updated_[i] = true; + } + } + if (IsInStartPhase(at_time)) { + initially_lost_packets_ += packets_lost; + } else if (uma_update_state_ == kNoUpdate) { + uma_update_state_ = kFirstDone; + bitrate_at_2_seconds_ = bitrate_kbps; + RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets", + initially_lost_packets_, 0, 100, 50); + RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate", + bitrate_at_2_seconds_.kbps(), 0, 2000, 50); + } else if (uma_update_state_ == kFirstDone && + at_time - first_report_time_ >= kBweConverganceTime) { + uma_update_state_ = kDone; + int bitrate_diff_kbps = std::max( + bitrate_at_2_seconds_.kbps<int>() - bitrate_kbps.kbps<int>(), 0); + RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, + 0, 2000, 50); + } +} + +void SendSideBandwidthEstimation::UpdateRtt(TimeDelta rtt, Timestamp at_time) { + // Update RTT if we were able to compute an RTT based on this RTCP. + // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT. + if (rtt > TimeDelta::Zero()) + last_round_trip_time_ = rtt; + + if (!IsInStartPhase(at_time) && uma_rtt_state_ == kNoUpdate) { + uma_rtt_state_ = kDone; + RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt.ms<int>(), 0, 2000, 50); + } +} + +void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) { + if (rtt_backoff_.IsRttAboveLimit()) { + if (at_time - time_last_decrease_ >= rtt_backoff_.drop_interval_ && + current_target_ > rtt_backoff_.bandwidth_floor_) { + time_last_decrease_ = at_time; + DataRate new_bitrate = + std::max(current_target_ * rtt_backoff_.drop_fraction_, + rtt_backoff_.bandwidth_floor_.Get()); + link_capacity_.OnRttBackoff(new_bitrate, at_time); + UpdateTargetBitrate(new_bitrate, at_time); + return; + } + // TODO(srte): This is likely redundant in most cases. + ApplyTargetLimits(at_time); + return; + } + + // We trust the REMB and/or delay-based estimate during the first 2 seconds if + // we haven't had any packet loss reported, to allow startup bitrate probing. + if (last_fraction_loss_ == 0 && IsInStartPhase(at_time) && + !loss_based_bandwidth_estimator_v2_.ReadyToUseInStartPhase()) { + DataRate new_bitrate = current_target_; + // TODO(srte): We should not allow the new_bitrate to be larger than the + // receiver limit here. + if (receiver_limit_.IsFinite()) + new_bitrate = std::max(receiver_limit_, new_bitrate); + if (delay_based_limit_.IsFinite()) + new_bitrate = std::max(delay_based_limit_, new_bitrate); + if (LossBasedBandwidthEstimatorV1Enabled()) { + loss_based_bandwidth_estimator_v1_.Initialize(new_bitrate); + } + + if (new_bitrate != current_target_) { + min_bitrate_history_.clear(); + if (LossBasedBandwidthEstimatorV1Enabled()) { + min_bitrate_history_.push_back(std::make_pair(at_time, new_bitrate)); + } else { + min_bitrate_history_.push_back( + std::make_pair(at_time, current_target_)); + } + UpdateTargetBitrate(new_bitrate, at_time); + return; + } + } + UpdateMinHistory(at_time); + if (last_loss_packet_report_.IsInfinite()) { + // No feedback received. + // TODO(srte): This is likely redundant in most cases. + ApplyTargetLimits(at_time); + return; + } + + if (LossBasedBandwidthEstimatorV1ReadyForUse()) { + DataRate new_bitrate = loss_based_bandwidth_estimator_v1_.Update( + at_time, min_bitrate_history_.front().second, delay_based_limit_, + last_round_trip_time_); + UpdateTargetBitrate(new_bitrate, at_time); + return; + } + + if (LossBasedBandwidthEstimatorV2ReadyForUse()) { + LossBasedBweV2::Result result = + loss_based_bandwidth_estimator_v2_.GetLossBasedResult(); + loss_based_state_ = result.state; + UpdateTargetBitrate(result.bandwidth_estimate, at_time); + return; + } + + TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_; + if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) { + // We only care about loss above a given bitrate threshold. + float loss = last_fraction_loss_ / 256.0f; + // We only make decisions based on loss when the bitrate is above a + // threshold. This is a crude way of handling loss which is uncorrelated + // to congestion. + if (current_target_ < bitrate_threshold_ || loss <= low_loss_threshold_) { + // Loss < 2%: Increase rate by 8% of the min bitrate in the last + // kBweIncreaseInterval. + // Note that by remembering the bitrate over the last second one can + // rampup up one second faster than if only allowed to start ramping + // at 8% per second rate now. E.g.: + // If sending a constant 100kbps it can rampup immediately to 108kbps + // whenever a receiver report is received with lower packet loss. + // If instead one would do: current_bitrate_ *= 1.08^(delta time), + // it would take over one second since the lower packet loss to achieve + // 108kbps. + DataRate new_bitrate = DataRate::BitsPerSec( + min_bitrate_history_.front().second.bps() * 1.08 + 0.5); + + // Add 1 kbps extra, just to make sure that we do not get stuck + // (gives a little extra increase at low rates, negligible at higher + // rates). + new_bitrate += DataRate::BitsPerSec(1000); + UpdateTargetBitrate(new_bitrate, at_time); + return; + } else if (current_target_ > bitrate_threshold_) { + if (loss <= high_loss_threshold_) { + // Loss between 2% - 10%: Do nothing. + } else { + // Loss > 10%: Limit the rate decreases to once a kBweDecreaseInterval + // + rtt. + if (!has_decreased_since_last_fraction_loss_ && + (at_time - time_last_decrease_) >= + (kBweDecreaseInterval + last_round_trip_time_)) { + time_last_decrease_ = at_time; + + // Reduce rate: + // newRate = rate * (1 - 0.5*lossRate); + // where packetLoss = 256*lossRate; + DataRate new_bitrate = DataRate::BitsPerSec( + (current_target_.bps() * + static_cast<double>(512 - last_fraction_loss_)) / + 512.0); + has_decreased_since_last_fraction_loss_ = true; + UpdateTargetBitrate(new_bitrate, at_time); + return; + } + } + } + } + // TODO(srte): This is likely redundant in most cases. + ApplyTargetLimits(at_time); +} + +void SendSideBandwidthEstimation::UpdatePropagationRtt( + Timestamp at_time, + TimeDelta propagation_rtt) { + rtt_backoff_.UpdatePropagationRtt(at_time, propagation_rtt); +} + +void SendSideBandwidthEstimation::OnSentPacket(const SentPacket& sent_packet) { + // Only feedback-triggering packets will be reported here. + rtt_backoff_.last_packet_sent_ = sent_packet.send_time; +} + +bool SendSideBandwidthEstimation::IsInStartPhase(Timestamp at_time) const { + return first_report_time_.IsInfinite() || + at_time - first_report_time_ < kStartPhase; +} + +void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) { + // Remove old data points from history. + // Since history precision is in ms, add one so it is able to increase + // bitrate if it is off by as little as 0.5ms. + while (!min_bitrate_history_.empty() && + at_time - min_bitrate_history_.front().first + TimeDelta::Millis(1) > + kBweIncreaseInterval) { + min_bitrate_history_.pop_front(); + } + + // Typical minimum sliding-window algorithm: Pop values higher than current + // bitrate before pushing it. + while (!min_bitrate_history_.empty() && + current_target_ <= min_bitrate_history_.back().second) { + min_bitrate_history_.pop_back(); + } + + min_bitrate_history_.push_back(std::make_pair(at_time, current_target_)); +} + +DataRate SendSideBandwidthEstimation::GetUpperLimit() const { + DataRate upper_limit = delay_based_limit_; + if (disable_receiver_limit_caps_only_) + upper_limit = std::min(upper_limit, receiver_limit_); + return std::min(upper_limit, max_bitrate_configured_); +} + +void SendSideBandwidthEstimation::MaybeLogLowBitrateWarning(DataRate bitrate, + Timestamp at_time) { + if (at_time - last_low_bitrate_log_ > kLowBitrateLogPeriod) { + RTC_LOG(LS_WARNING) << "Estimated available bandwidth " << ToString(bitrate) + << " is below configured min bitrate " + << ToString(min_bitrate_configured_) << "."; + last_low_bitrate_log_ = at_time; + } +} + +void SendSideBandwidthEstimation::MaybeLogLossBasedEvent(Timestamp at_time) { + if (current_target_ != last_logged_target_ || + last_fraction_loss_ != last_logged_fraction_loss_ || + at_time - last_rtc_event_log_ > kRtcEventLogPeriod) { + event_log_->Log(std::make_unique<RtcEventBweUpdateLossBased>( + current_target_.bps(), last_fraction_loss_, + expected_packets_since_last_loss_update_)); + last_logged_fraction_loss_ = last_fraction_loss_; + last_logged_target_ = current_target_; + last_rtc_event_log_ = at_time; + } +} + +void SendSideBandwidthEstimation::UpdateTargetBitrate(DataRate new_bitrate, + Timestamp at_time) { + new_bitrate = std::min(new_bitrate, GetUpperLimit()); + if (new_bitrate < min_bitrate_configured_) { + MaybeLogLowBitrateWarning(new_bitrate, at_time); + new_bitrate = min_bitrate_configured_; + } + current_target_ = new_bitrate; + MaybeLogLossBasedEvent(at_time); + link_capacity_.OnRateUpdate(acknowledged_rate_, current_target_, at_time); +} + +void SendSideBandwidthEstimation::ApplyTargetLimits(Timestamp at_time) { + UpdateTargetBitrate(current_target_, at_time); +} + +bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1Enabled() const { + return loss_based_bandwidth_estimator_v1_.Enabled() && + !LossBasedBandwidthEstimatorV2Enabled(); +} + +bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1ReadyForUse() + const { + return LossBasedBandwidthEstimatorV1Enabled() && + loss_based_bandwidth_estimator_v1_.InUse(); +} + +bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2Enabled() const { + return loss_based_bandwidth_estimator_v2_.IsEnabled(); +} + +bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse() + const { + return LossBasedBandwidthEstimatorV2Enabled() && + loss_based_bandwidth_estimator_v2_.IsReady(); +} + +} // namespace webrtc |