| /* |
| * 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 <cmath> |
| #include <limits> |
| #include <string> |
| |
| #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 "rtc_base/ptr_util.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace webrtc { |
| namespace { |
| const int64_t kBweIncreaseIntervalMs = 1000; |
| const int64_t kBweDecreaseIntervalMs = 300; |
| const int64_t kStartPhaseMs = 2000; |
| const int64_t kBweConverganceTimeMs = 20000; |
| const int kLimitNumPackets = 20; |
| const int kDefaultMaxBitrateBps = 1000000000; |
| const int64_t kLowBitrateLogPeriodMs = 10000; |
| const int64_t kRtcEventLogPeriodMs = 5000; |
| // Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals. |
| const int64_t kFeedbackIntervalMs = 5000; |
| const int64_t kFeedbackTimeoutIntervals = 3; |
| const int64_t kTimeoutIntervalMs = 1000; |
| |
| const float kDefaultLowLossThreshold = 0.02f; |
| const float kDefaultHighLossThreshold = 0.1f; |
| const int kDefaultBitrateThresholdKbps = 0; |
| |
| 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 = kDefaultBitrateThresholdKbps; |
| return false; |
| } |
| } // namespace |
| |
| SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log) |
| : lost_packets_since_last_loss_update_Q8_(0), |
| expected_packets_since_last_loss_update_(0), |
| current_bitrate_bps_(0), |
| min_bitrate_configured_(congestion_controller::GetMinBitrateBps()), |
| max_bitrate_configured_(kDefaultMaxBitrateBps), |
| last_low_bitrate_log_ms_(-1), |
| has_decreased_since_last_fraction_loss_(false), |
| last_feedback_ms_(-1), |
| last_packet_report_ms_(-1), |
| last_timeout_ms_(-1), |
| last_fraction_loss_(0), |
| last_logged_fraction_loss_(0), |
| last_round_trip_time_ms_(0), |
| bwe_incoming_(0), |
| delay_based_bitrate_bps_(0), |
| time_last_decrease_ms_(0), |
| first_report_time_ms_(-1), |
| initially_lost_packets_(0), |
| bitrate_at_2_seconds_kbps_(0), |
| uma_update_state_(kNoUpdate), |
| rampup_uma_stats_updated_(kNumUmaRampupMetrics, false), |
| event_log_(event_log), |
| last_rtc_event_log_ms_(-1), |
| in_timeout_experiment_( |
| webrtc::field_trial::IsEnabled("WebRTC-FeedbackTimeout")), |
| low_loss_threshold_(kDefaultLowLossThreshold), |
| high_loss_threshold_(kDefaultHighLossThreshold), |
| bitrate_threshold_bps_(1000 * kDefaultBitrateThresholdKbps) { |
| 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_bps_ = bitrate_threshold_kbps * 1000; |
| } |
| } |
| } |
| |
| SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {} |
| |
| void SendSideBandwidthEstimation::SetBitrates(int send_bitrate, |
| int min_bitrate, |
| int max_bitrate) { |
| SetMinMaxBitrate(min_bitrate, max_bitrate); |
| if (send_bitrate > 0) |
| SetSendBitrate(send_bitrate); |
| } |
| |
| void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) { |
| RTC_DCHECK_GT(bitrate, 0); |
| delay_based_bitrate_bps_ = 0; // Reset to avoid being capped by the estimate. |
| CapBitrateToThresholds(Clock::GetRealTimeClock()->TimeInMilliseconds(), |
| bitrate); |
| // Clear last sent bitrate history so the new value can be used directly |
| // and not capped. |
| min_bitrate_history_.clear(); |
| } |
| |
| void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate, |
| int max_bitrate) { |
| RTC_DCHECK_GE(min_bitrate, 0); |
| min_bitrate_configured_ = |
| std::max(min_bitrate, congestion_controller::GetMinBitrateBps()); |
| if (max_bitrate > 0) { |
| max_bitrate_configured_ = |
| std::max<uint32_t>(min_bitrate_configured_, max_bitrate); |
| } else { |
| max_bitrate_configured_ = kDefaultMaxBitrateBps; |
| } |
| } |
| |
| int SendSideBandwidthEstimation::GetMinBitrate() const { |
| return min_bitrate_configured_; |
| } |
| |
| void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate, |
| uint8_t* loss, |
| int64_t* rtt) const { |
| *bitrate = current_bitrate_bps_; |
| *loss = last_fraction_loss_; |
| *rtt = last_round_trip_time_ms_; |
| } |
| |
| void SendSideBandwidthEstimation::UpdateReceiverEstimate( |
| int64_t now_ms, uint32_t bandwidth) { |
| bwe_incoming_ = bandwidth; |
| CapBitrateToThresholds(now_ms, current_bitrate_bps_); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateDelayBasedEstimate( |
| int64_t now_ms, |
| uint32_t bitrate_bps) { |
| delay_based_bitrate_bps_ = bitrate_bps; |
| CapBitrateToThresholds(now_ms, current_bitrate_bps_); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss, |
| int64_t rtt, |
| int number_of_packets, |
| int64_t now_ms) { |
| last_feedback_ms_ = now_ms; |
| if (first_report_time_ms_ == -1) |
| first_report_time_ms_ = now_ms; |
| |
| // 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 > 0) |
| last_round_trip_time_ms_ = rtt; |
| |
| // Check sequence number diff and weight loss report |
| if (number_of_packets > 0) { |
| // Calculate number of lost packets. |
| const int num_lost_packets_Q8 = fraction_loss * number_of_packets; |
| // Accumulate reports. |
| lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8; |
| 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; |
| last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ / |
| expected_packets_since_last_loss_update_; |
| |
| // Reset accumulators. |
| lost_packets_since_last_loss_update_Q8_ = 0; |
| expected_packets_since_last_loss_update_ = 0; |
| last_packet_report_ms_ = now_ms; |
| UpdateEstimate(now_ms); |
| } |
| UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8); |
| } |
| |
| void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms, |
| int64_t rtt, |
| int lost_packets) { |
| int bitrate_kbps = static_cast<int>((current_bitrate_bps_ + 500) / 1000); |
| for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) { |
| if (!rampup_uma_stats_updated_[i] && |
| bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) { |
| RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name, |
| now_ms - first_report_time_ms_); |
| rampup_uma_stats_updated_[i] = true; |
| } |
| } |
| if (IsInStartPhase(now_ms)) { |
| initially_lost_packets_ += lost_packets; |
| } else if (uma_update_state_ == kNoUpdate) { |
| uma_update_state_ = kFirstDone; |
| bitrate_at_2_seconds_kbps_ = bitrate_kbps; |
| RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets", |
| initially_lost_packets_, 0, 100, 50); |
| RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0, |
| 2000, 50); |
| RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate", |
| bitrate_at_2_seconds_kbps_, 0, 2000, 50); |
| } else if (uma_update_state_ == kFirstDone && |
| now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) { |
| uma_update_state_ = kDone; |
| int bitrate_diff_kbps = |
| std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0); |
| RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, |
| 0, 2000, 50); |
| } |
| } |
| |
| void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) { |
| uint32_t new_bitrate = current_bitrate_bps_; |
| // 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(now_ms)) { |
| new_bitrate = std::max(bwe_incoming_, new_bitrate); |
| new_bitrate = std::max(delay_based_bitrate_bps_, new_bitrate); |
| |
| if (new_bitrate != current_bitrate_bps_) { |
| min_bitrate_history_.clear(); |
| min_bitrate_history_.push_back( |
| std::make_pair(now_ms, current_bitrate_bps_)); |
| CapBitrateToThresholds(now_ms, new_bitrate); |
| return; |
| } |
| } |
| UpdateMinHistory(now_ms); |
| if (last_packet_report_ms_ == -1) { |
| // No feedback received. |
| CapBitrateToThresholds(now_ms, current_bitrate_bps_); |
| return; |
| } |
| int64_t time_since_packet_report_ms = now_ms - last_packet_report_ms_; |
| int64_t time_since_feedback_ms = now_ms - last_feedback_ms_; |
| if (time_since_packet_report_ms < 1.2 * kFeedbackIntervalMs) { |
| // 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_bps_ < bitrate_threshold_bps_ || |
| loss <= low_loss_threshold_) { |
| // Loss < 2%: Increase rate by 8% of the min bitrate in the last |
| // kBweIncreaseIntervalMs. |
| // 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_bps_ *= 1.08^(delta time), |
| // it would take over one second since the lower packet loss to achieve |
| // 108kbps. |
| new_bitrate = static_cast<uint32_t>( |
| min_bitrate_history_.front().second * 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 += 1000; |
| } else if (current_bitrate_bps_ > bitrate_threshold_bps_) { |
| if (loss <= high_loss_threshold_) { |
| // Loss between 2% - 10%: Do nothing. |
| } else { |
| // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs |
| // + rtt. |
| if (!has_decreased_since_last_fraction_loss_ && |
| (now_ms - time_last_decrease_ms_) >= |
| (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) { |
| time_last_decrease_ms_ = now_ms; |
| |
| // Reduce rate: |
| // newRate = rate * (1 - 0.5*lossRate); |
| // where packetLoss = 256*lossRate; |
| new_bitrate = static_cast<uint32_t>( |
| (current_bitrate_bps_ * |
| static_cast<double>(512 - last_fraction_loss_)) / |
| 512.0); |
| has_decreased_since_last_fraction_loss_ = true; |
| } |
| } |
| } |
| } else if (time_since_feedback_ms > |
| kFeedbackTimeoutIntervals * kFeedbackIntervalMs && |
| (last_timeout_ms_ == -1 || |
| now_ms - last_timeout_ms_ > kTimeoutIntervalMs)) { |
| if (in_timeout_experiment_) { |
| RTC_LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms |
| << " ms), reducing 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_Q8_ = 0; |
| expected_packets_since_last_loss_update_ = 0; |
| last_timeout_ms_ = now_ms; |
| } |
| } |
| |
| CapBitrateToThresholds(now_ms, new_bitrate); |
| } |
| |
| bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const { |
| return first_report_time_ms_ == -1 || |
| now_ms - first_report_time_ms_ < kStartPhaseMs; |
| } |
| |
| void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) { |
| // 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() && |
| now_ms - min_bitrate_history_.front().first + 1 > |
| kBweIncreaseIntervalMs) { |
| 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_bps_ <= min_bitrate_history_.back().second) { |
| min_bitrate_history_.pop_back(); |
| } |
| |
| min_bitrate_history_.push_back(std::make_pair(now_ms, current_bitrate_bps_)); |
| } |
| |
| void SendSideBandwidthEstimation::CapBitrateToThresholds(int64_t now_ms, |
| uint32_t bitrate_bps) { |
| if (bwe_incoming_ > 0 && bitrate_bps > bwe_incoming_) { |
| bitrate_bps = bwe_incoming_; |
| } |
| if (delay_based_bitrate_bps_ > 0 && bitrate_bps > delay_based_bitrate_bps_) { |
| bitrate_bps = delay_based_bitrate_bps_; |
| } |
| if (bitrate_bps > max_bitrate_configured_) { |
| bitrate_bps = max_bitrate_configured_; |
| } |
| if (bitrate_bps < min_bitrate_configured_) { |
| if (last_low_bitrate_log_ms_ == -1 || |
| now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) { |
| RTC_LOG(LS_WARNING) << "Estimated available bandwidth " |
| << bitrate_bps / 1000 |
| << " kbps is below configured min bitrate " |
| << min_bitrate_configured_ / 1000 << " kbps."; |
| last_low_bitrate_log_ms_ = now_ms; |
| } |
| bitrate_bps = min_bitrate_configured_; |
| } |
| |
| if (bitrate_bps != current_bitrate_bps_ || |
| last_fraction_loss_ != last_logged_fraction_loss_ || |
| now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) { |
| event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>( |
| bitrate_bps, last_fraction_loss_, |
| expected_packets_since_last_loss_update_)); |
| last_logged_fraction_loss_ = last_fraction_loss_; |
| last_rtc_event_log_ms_ = now_ms; |
| } |
| current_bitrate_bps_ = bitrate_bps; |
| } |
| } // namespace webrtc |
