| /* |
| * Copyright (c) 2016 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/delay_based_bwe.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <cstdio> |
| #include <string> |
| |
| #include "absl/memory/memory.h" |
| #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "modules/congestion_controller/goog_cc/trendline_estimator.h" |
| #include "modules/pacing/paced_sender.h" |
| #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace { |
| constexpr int kTimestampGroupLengthMs = 5; |
| constexpr int kAbsSendTimeFraction = 18; |
| constexpr int kAbsSendTimeInterArrivalUpshift = 8; |
| constexpr int kInterArrivalShift = |
| kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift; |
| constexpr double kTimestampToMs = |
| 1000.0 / static_cast<double>(1 << kInterArrivalShift); |
| // This ssrc is used to fulfill the current API but will be removed |
| // after the API has been changed. |
| constexpr uint32_t kFixedSsrc = 0; |
| |
| // Parameters for linear least squares fit of regression line to noisy data. |
| constexpr size_t kDefaultTrendlineWindowSize = 20; |
| constexpr double kDefaultTrendlineSmoothingCoeff = 0.9; |
| constexpr double kDefaultTrendlineThresholdGain = 4.0; |
| |
| constexpr int kMaxConsecutiveFailedLookups = 5; |
| |
| const char kBweWindowSizeInPacketsExperiment[] = |
| "WebRTC-BweWindowSizeInPackets"; |
| |
| size_t ReadTrendlineFilterWindowSize() { |
| std::string experiment_string = |
| webrtc::field_trial::FindFullName(kBweWindowSizeInPacketsExperiment); |
| size_t window_size; |
| int parsed_values = |
| sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size); |
| if (parsed_values == 1) { |
| if (window_size > 1) |
| return window_size; |
| RTC_LOG(WARNING) << "Window size must be greater than 1."; |
| } |
| RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweTrendlineFilter " |
| "experiment from field trial string. Using default."; |
| return kDefaultTrendlineWindowSize; |
| } |
| } // namespace |
| |
| namespace webrtc { |
| |
| DelayBasedBwe::Result::Result() |
| : updated(false), |
| probe(false), |
| target_bitrate_bps(0), |
| recovered_from_overuse(false) {} |
| |
| DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps) |
| : updated(true), |
| probe(probe), |
| target_bitrate_bps(target_bitrate_bps), |
| recovered_from_overuse(false) {} |
| |
| DelayBasedBwe::Result::~Result() {} |
| |
| DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log) |
| : event_log_(event_log), |
| inter_arrival_(), |
| delay_detector_(), |
| last_seen_packet_ms_(-1), |
| uma_recorded_(false), |
| probe_bitrate_estimator_(event_log), |
| trendline_window_size_( |
| webrtc::field_trial::IsEnabled(kBweWindowSizeInPacketsExperiment) |
| ? ReadTrendlineFilterWindowSize() |
| : kDefaultTrendlineWindowSize), |
| trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff), |
| trendline_threshold_gain_(kDefaultTrendlineThresholdGain), |
| consecutive_delayed_feedbacks_(0), |
| prev_bitrate_(0), |
| prev_state_(BandwidthUsage::kBwNormal) { |
| RTC_LOG(LS_INFO) |
| << "Using Trendline filter for delay change estimation with window size " |
| << trendline_window_size_; |
| delay_detector_.reset(new TrendlineEstimator(trendline_window_size_, |
| trendline_smoothing_coeff_, |
| trendline_threshold_gain_)); |
| } |
| |
| DelayBasedBwe::~DelayBasedBwe() {} |
| |
| DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector( |
| const std::vector<PacketFeedback>& packet_feedback_vector, |
| absl::optional<uint32_t> acked_bitrate_bps, |
| int64_t at_time_ms) { |
| RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(), |
| packet_feedback_vector.end(), |
| PacketFeedbackComparator())); |
| RTC_DCHECK_RUNS_SERIALIZED(&network_race_); |
| |
| // TOOD(holmer): An empty feedback vector here likely means that |
| // all acks were too late and that the send time history had |
| // timed out. We should reduce the rate when this occurs. |
| if (packet_feedback_vector.empty()) { |
| RTC_LOG(LS_WARNING) << "Very late feedback received."; |
| return DelayBasedBwe::Result(); |
| } |
| |
| if (!uma_recorded_) { |
| RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, |
| BweNames::kSendSideTransportSeqNum, |
| BweNames::kBweNamesMax); |
| uma_recorded_ = true; |
| } |
| bool delayed_feedback = true; |
| bool recovered_from_overuse = false; |
| BandwidthUsage prev_detector_state = delay_detector_->State(); |
| for (const auto& packet_feedback : packet_feedback_vector) { |
| if (packet_feedback.send_time_ms < 0) |
| continue; |
| delayed_feedback = false; |
| IncomingPacketFeedback(packet_feedback, at_time_ms); |
| if (prev_detector_state == BandwidthUsage::kBwUnderusing && |
| delay_detector_->State() == BandwidthUsage::kBwNormal) { |
| recovered_from_overuse = true; |
| } |
| prev_detector_state = delay_detector_->State(); |
| } |
| |
| if (delayed_feedback) { |
| ++consecutive_delayed_feedbacks_; |
| if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) { |
| consecutive_delayed_feedbacks_ = 0; |
| return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms); |
| } |
| } else { |
| consecutive_delayed_feedbacks_ = 0; |
| return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse, |
| at_time_ms); |
| } |
| return Result(); |
| } |
| |
| DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay( |
| int64_t arrival_time_ms) { |
| // Estimate should always be valid since a start bitrate always is set in the |
| // Call constructor. An alternative would be to return an empty Result here, |
| // or to estimate the throughput based on the feedback we received. |
| RTC_DCHECK(rate_control_.ValidEstimate()); |
| rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, |
| arrival_time_ms); |
| Result result; |
| result.updated = true; |
| result.probe = false; |
| result.target_bitrate_bps = rate_control_.LatestEstimate(); |
| RTC_LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to " |
| << result.target_bitrate_bps; |
| return result; |
| } |
| |
| void DelayBasedBwe::IncomingPacketFeedback( |
| const PacketFeedback& packet_feedback, |
| int64_t at_time_ms) { |
| int64_t now_ms = at_time_ms; |
| // Reset if the stream has timed out. |
| if (last_seen_packet_ms_ == -1 || |
| now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) { |
| inter_arrival_.reset( |
| new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000, |
| kTimestampToMs, true)); |
| delay_detector_.reset(new TrendlineEstimator(trendline_window_size_, |
| trendline_smoothing_coeff_, |
| trendline_threshold_gain_)); |
| } |
| last_seen_packet_ms_ = now_ms; |
| |
| uint32_t send_time_24bits = |
| static_cast<uint32_t>( |
| ((static_cast<uint64_t>(packet_feedback.send_time_ms) |
| << kAbsSendTimeFraction) + |
| 500) / |
| 1000) & |
| 0x00FFFFFF; |
| // Shift up send time to use the full 32 bits that inter_arrival works with, |
| // so wrapping works properly. |
| uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift; |
| |
| uint32_t ts_delta = 0; |
| int64_t t_delta = 0; |
| int size_delta = 0; |
| if (inter_arrival_->ComputeDeltas(timestamp, packet_feedback.arrival_time_ms, |
| now_ms, packet_feedback.payload_size, |
| &ts_delta, &t_delta, &size_delta)) { |
| double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift); |
| delay_detector_->Update(t_delta, ts_delta_ms, |
| packet_feedback.arrival_time_ms); |
| } |
| if (packet_feedback.pacing_info.probe_cluster_id != |
| PacedPacketInfo::kNotAProbe) { |
| probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback); |
| } |
| } |
| |
| DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate( |
| absl::optional<uint32_t> acked_bitrate_bps, |
| bool recovered_from_overuse, |
| int64_t at_time_ms) { |
| Result result; |
| int64_t now_ms = at_time_ms; |
| |
| absl::optional<int> probe_bitrate_bps = |
| probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps(); |
| // Currently overusing the bandwidth. |
| if (delay_detector_->State() == BandwidthUsage::kBwOverusing) { |
| if (acked_bitrate_bps && |
| rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) { |
| result.updated = |
| UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps); |
| } else if (!acked_bitrate_bps && rate_control_.ValidEstimate() && |
| rate_control_.InitialTimeToReduceFurther(now_ms)) { |
| // Overusing before we have a measured acknowledged bitrate. Reduce send |
| // rate by 50% every 200 ms. |
| // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator |
| // so that we (almost) always have a bitrate estimate. |
| rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms); |
| result.updated = true; |
| result.probe = false; |
| result.target_bitrate_bps = rate_control_.LatestEstimate(); |
| } |
| } else { |
| if (probe_bitrate_bps) { |
| result.probe = true; |
| result.updated = true; |
| result.target_bitrate_bps = *probe_bitrate_bps; |
| rate_control_.SetEstimate(*probe_bitrate_bps, now_ms); |
| } else { |
| result.updated = |
| UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps); |
| result.recovered_from_overuse = recovered_from_overuse; |
| } |
| } |
| BandwidthUsage detector_state = delay_detector_->State(); |
| if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) || |
| detector_state != prev_state_) { |
| uint32_t bitrate_bps = |
| result.updated ? result.target_bitrate_bps : prev_bitrate_; |
| |
| BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps); |
| |
| if (event_log_) { |
| event_log_->Log(absl::make_unique<RtcEventBweUpdateDelayBased>( |
| bitrate_bps, detector_state)); |
| } |
| |
| prev_bitrate_ = bitrate_bps; |
| prev_state_ = detector_state; |
| } |
| return result; |
| } |
| |
| bool DelayBasedBwe::UpdateEstimate(int64_t now_ms, |
| absl::optional<uint32_t> acked_bitrate_bps, |
| uint32_t* target_bitrate_bps) { |
| const RateControlInput input(delay_detector_->State(), acked_bitrate_bps); |
| *target_bitrate_bps = rate_control_.Update(&input, now_ms); |
| return rate_control_.ValidEstimate(); |
| } |
| |
| void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms) { |
| rate_control_.SetRtt(avg_rtt_ms); |
| } |
| |
| bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs, |
| uint32_t* bitrate_bps) const { |
| // Currently accessed from both the process thread (see |
| // ModuleRtpRtcpImpl::Process()) and the configuration thread (see |
| // Call::GetStats()). Should in the future only be accessed from a single |
| // thread. |
| RTC_DCHECK(ssrcs); |
| RTC_DCHECK(bitrate_bps); |
| if (!rate_control_.ValidEstimate()) |
| return false; |
| |
| *ssrcs = {kFixedSsrc}; |
| *bitrate_bps = rate_control_.LatestEstimate(); |
| return true; |
| } |
| |
| void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) { |
| RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: " << start_bitrate_bps; |
| rate_control_.SetStartBitrate(start_bitrate_bps); |
| } |
| |
| void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) { |
| // Called from both the configuration thread and the network thread. Shouldn't |
| // be called from the network thread in the future. |
| rate_control_.SetMinBitrate(min_bitrate_bps); |
| } |
| |
| int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const { |
| return rate_control_.GetExpectedBandwidthPeriodMs(); |
| } |
| } // namespace webrtc |
