| /* |
| * 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/bitrate_controller/send_side_bandwidth_estimation.h" |
| |
| #include <algorithm> |
| #include <cstdio> |
| #include <limits> |
| #include <string> |
| |
| #include "absl/memory/memory.h" |
| #include "logging/rtc_event_log/events/rtc_event.h" |
| #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "modules/remote_bitrate_estimator/include/bwe_defines.h" |
| #include "rtc_base/checks.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 int kFeedbackTimeoutIntervals = 3; |
| constexpr TimeDelta kTimeoutInterval = TimeDelta::Millis<1000>(); |
| |
| 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 experiment_string.find("Enabled") == 0; |
| } |
| |
| 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 |
| |
| RttBasedBackoffConfig::RttBasedBackoffConfig() |
| : rtt_limit("limit", TimeDelta::PlusInfinity()), |
| drop_fraction("fraction", 0.5), |
| drop_interval("interval", TimeDelta::ms(300)) { |
| std::string trial_string = |
| field_trial::FindFullName("WebRTC-Bwe-MaxRttLimit"); |
| ParseFieldTrial({&rtt_limit, &drop_fraction, &drop_interval}, trial_string); |
| } |
| RttBasedBackoffConfig::RttBasedBackoffConfig(const RttBasedBackoffConfig&) = |
| default; |
| RttBasedBackoffConfig::~RttBasedBackoffConfig() = default; |
| |
| SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log) |
| : rtt_backoff_config_(RttBasedBackoffConfig()), |
| lost_packets_since_last_loss_update_(0), |
| expected_packets_since_last_loss_update_(0), |
| current_bitrate_(DataRate::Zero()), |
| min_bitrate_configured_( |
| DataRate::bps(congestion_controller::GetMinBitrateBps())), |
| 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_timeout_(Timestamp::MinusInfinity()), |
| last_fraction_loss_(0), |
| last_logged_fraction_loss_(0), |
| last_round_trip_time_(TimeDelta::Zero()), |
| // 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()), |
| bwe_incoming_(DataRate::Zero()), |
| delay_based_bitrate_(DataRate::Zero()), |
| 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()), |
| in_timeout_experiment_( |
| webrtc::field_trial::IsEnabled("WebRTC-FeedbackTimeout")), |
| low_loss_threshold_(kDefaultLowLossThreshold), |
| high_loss_threshold_(kDefaultHighLossThreshold), |
| bitrate_threshold_(kDefaultBitrateThreshold) { |
| 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::kbps(bitrate_threshold_kbps); |
| } |
| } |
| } |
| |
| SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {} |
| |
| 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) |
| SetSendBitrate(*send_bitrate, at_time); |
| } |
| |
| void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate, |
| Timestamp at_time) { |
| RTC_DCHECK(bitrate > DataRate::Zero()); |
| // Reset to avoid being capped by the estimate. |
| delay_based_bitrate_ = DataRate::Zero(); |
| CapBitrateToThresholds(at_time, bitrate); |
| // 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, congestion_controller::GetMinBitrate()); |
| if (max_bitrate > DataRate::Zero() && max_bitrate.IsFinite()) { |
| max_bitrate_configured_ = std::max(min_bitrate_configured_, max_bitrate); |
| } else { |
| max_bitrate_configured_ = kDefaultMaxBitrate; |
| } |
| } |
| |
| int SendSideBandwidthEstimation::GetMinBitrate() const { |
| return min_bitrate_configured_.bps<int>(); |
| } |
| |
| void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate, |
| uint8_t* loss, |
| int64_t* rtt) const { |
| *bitrate = current_bitrate_.bps<int>(); |
| *loss = last_fraction_loss_; |
| *rtt = last_round_trip_time_.ms<int64_t>(); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time, |
| DataRate bandwidth) { |
| bwe_incoming_ = bandwidth; |
| CapBitrateToThresholds(at_time, current_bitrate_); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time, |
| DataRate bitrate) { |
| delay_based_bitrate_ = bitrate; |
| CapBitrateToThresholds(at_time, current_bitrate_); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss, |
| TimeDelta rtt, |
| int number_of_packets, |
| Timestamp at_time) { |
| const int kRoundingConstant = 128; |
| int packets_lost = (static_cast<int>(fraction_loss) * number_of_packets + |
| kRoundingConstant) >> |
| 8; |
| UpdatePacketsLost(packets_lost, number_of_packets, at_time); |
| UpdateRtt(rtt, at_time); |
| } |
| |
| void SendSideBandwidthEstimation::UpdatePacketsLost(int packets_lost, |
| int 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) { |
| // Accumulate reports. |
| lost_packets_since_last_loss_update_ += packets_lost; |
| 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_packets_since_last_loss_update_ < kLimitNumPackets) |
| return; |
| |
| has_decreased_since_last_fraction_loss_ = false; |
| int64_t lost_q8 = lost_packets_since_last_loss_update_ << 8; |
| int64_t expected = expected_packets_since_last_loss_update_; |
| 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::kbps((current_bitrate_.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) { |
| DataRate new_bitrate = current_bitrate_; |
| TimeDelta time_since_rtt = at_time - last_propagation_rtt_update_; |
| if (time_since_rtt + last_propagation_rtt_ > rtt_backoff_config_.rtt_limit) { |
| if (at_time - time_last_decrease_ >= rtt_backoff_config_.drop_interval) { |
| time_last_decrease_ = at_time; |
| new_bitrate = current_bitrate_ * rtt_backoff_config_.drop_fraction; |
| } |
| CapBitrateToThresholds(at_time, new_bitrate); |
| 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)) { |
| new_bitrate = std::max(bwe_incoming_, new_bitrate); |
| new_bitrate = std::max(delay_based_bitrate_, new_bitrate); |
| |
| if (new_bitrate != current_bitrate_) { |
| min_bitrate_history_.clear(); |
| min_bitrate_history_.push_back(std::make_pair(at_time, current_bitrate_)); |
| CapBitrateToThresholds(at_time, new_bitrate); |
| return; |
| } |
| } |
| UpdateMinHistory(at_time); |
| if (last_loss_packet_report_.IsInfinite()) { |
| // No feedback received. |
| CapBitrateToThresholds(at_time, current_bitrate_); |
| return; |
| } |
| TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_; |
| TimeDelta time_since_loss_feedback = at_time - last_loss_feedback_; |
| 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_bitrate_ < 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 immediatly 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. |
| new_bitrate = |
| DataRate::bps(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::bps(1000); |
| } else if (current_bitrate_ > 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; |
| new_bitrate = |
| DataRate::bps((current_bitrate_.bps() * |
| static_cast<double>(512 - last_fraction_loss_)) / |
| 512.0); |
| has_decreased_since_last_fraction_loss_ = true; |
| } |
| } |
| } |
| } else if (time_since_loss_feedback > |
| kFeedbackTimeoutIntervals * kMaxRtcpFeedbackInterval && |
| (last_timeout_.IsInfinite() || |
| at_time - last_timeout_ > kTimeoutInterval)) { |
| if (in_timeout_experiment_) { |
| RTC_LOG(LS_WARNING) << "Feedback timed out (" |
| << ToString(time_since_loss_feedback) |
| << "), reducing bitrate."; |
| new_bitrate = new_bitrate * 0.8; |
| // Reset accumulators since we've already acted on missing feedback and |
| // shouldn't to act again on these old lost packets. |
| lost_packets_since_last_loss_update_ = 0; |
| expected_packets_since_last_loss_update_ = 0; |
| last_timeout_ = at_time; |
| } |
| } |
| |
| CapBitrateToThresholds(at_time, new_bitrate); |
| } |
| |
| void SendSideBandwidthEstimation::UpdatePropagationRtt( |
| Timestamp at_time, |
| TimeDelta propagation_rtt) { |
| last_propagation_rtt_update_ = at_time; |
| last_propagation_rtt_ = propagation_rtt; |
| } |
| |
| 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::ms(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_bitrate_ <= min_bitrate_history_.back().second) { |
| min_bitrate_history_.pop_back(); |
| } |
| |
| min_bitrate_history_.push_back(std::make_pair(at_time, current_bitrate_)); |
| } |
| |
| void SendSideBandwidthEstimation::CapBitrateToThresholds(Timestamp at_time, |
| DataRate bitrate) { |
| if (bwe_incoming_ > DataRate::Zero() && bitrate > bwe_incoming_) { |
| bitrate = bwe_incoming_; |
| } |
| if (delay_based_bitrate_ > DataRate::Zero() && |
| bitrate > delay_based_bitrate_) { |
| bitrate = delay_based_bitrate_; |
| } |
| if (bitrate > max_bitrate_configured_) { |
| bitrate = max_bitrate_configured_; |
| } |
| if (bitrate < min_bitrate_configured_) { |
| if (last_low_bitrate_log_.IsInfinite() || |
| 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; |
| } |
| bitrate = min_bitrate_configured_; |
| } |
| |
| if (bitrate != current_bitrate_ || |
| last_fraction_loss_ != last_logged_fraction_loss_ || |
| at_time - last_rtc_event_log_ > kRtcEventLogPeriod) { |
| event_log_->Log(absl::make_unique<RtcEventBweUpdateLossBased>( |
| bitrate.bps(), last_fraction_loss_, |
| expected_packets_since_last_loss_update_)); |
| last_logged_fraction_loss_ = last_fraction_loss_; |
| last_rtc_event_log_ = at_time; |
| } |
| current_bitrate_ = bitrate; |
| } |
| } // namespace webrtc |
