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These field trials can be set with a string similar to: WebRTC-BweWindowSizeInPackets/Enabled-150/WebRTC-BweBackOffFactor/Enabled-0.95/ BweWindowSizeInPackets Number of packets which the delay-based BWE window is based on. A larger value means lower delay-sensitivity. Default in WebRTC: 20 Reasonable values for streaming: 50-150 BweBackOffFactor How far the BWE will back off when the delay increases. A value closer to 1.0 means smaller back-off. Range: > 0.0, < 1.0 Default in WebRTC: 0.85 Reasonable values for streaming: 0.85-0.95 Bug: webrtc:8212 Change-Id: I61f0883788b689847a43273b63cef663042f4d42 Reviewed-on: https://webrtc-review.googlesource.com/6764 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20172}
339 lines
13 KiB
C++
339 lines
13 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/congestion_controller/delay_based_bwe.h"
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#include <algorithm>
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#include <cmath>
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#include <string>
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#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/pacing/paced_sender.h"
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/thread_annotations.h"
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#include "system_wrappers/include/field_trial.h"
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#include "system_wrappers/include/metrics.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace {
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constexpr int kTimestampGroupLengthMs = 5;
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constexpr int kAbsSendTimeFraction = 18;
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constexpr int kAbsSendTimeInterArrivalUpshift = 8;
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constexpr int kInterArrivalShift =
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kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
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constexpr double kTimestampToMs =
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1000.0 / static_cast<double>(1 << kInterArrivalShift);
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// This ssrc is used to fulfill the current API but will be removed
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// after the API has been changed.
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constexpr uint32_t kFixedSsrc = 0;
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// Parameters for linear least squares fit of regression line to noisy data.
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constexpr size_t kDefaultTrendlineWindowSize = 20;
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constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
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constexpr double kDefaultTrendlineThresholdGain = 4.0;
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constexpr int kMaxConsecutiveFailedLookups = 5;
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const char kBweSparseUpdateExperiment[] = "WebRTC-BweSparseUpdateExperiment";
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const char kBweWindowSizeInPacketsExperiment[] =
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"WebRTC-BweWindowSizeInPackets";
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size_t ReadTrendlineFilterWindowSize() {
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std::string experiment_string =
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webrtc::field_trial::FindFullName(kBweWindowSizeInPacketsExperiment);
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size_t window_size;
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int parsed_values =
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sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size);
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if (parsed_values == 1) {
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if (window_size > 1)
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return window_size;
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LOG(WARNING) << "Window size must be greater than 1.";
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}
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LOG(LS_WARNING) << "Failed to parse parameters for BweTrendlineFilter "
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"experiment from field trial string. Using default.";
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return kDefaultTrendlineWindowSize;
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}
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} // namespace
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namespace webrtc {
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DelayBasedBwe::Result::Result()
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: updated(false),
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probe(false),
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target_bitrate_bps(0),
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recovered_from_overuse(false) {}
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DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps)
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: updated(true),
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probe(probe),
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target_bitrate_bps(target_bitrate_bps),
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recovered_from_overuse(false) {}
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DelayBasedBwe::Result::~Result() {}
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DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log, const Clock* clock)
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: event_log_(event_log),
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clock_(clock),
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inter_arrival_(),
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trendline_estimator_(),
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detector_(),
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last_seen_packet_ms_(-1),
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uma_recorded_(false),
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probe_bitrate_estimator_(event_log),
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trendline_window_size_(
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webrtc::field_trial::IsEnabled(kBweWindowSizeInPacketsExperiment)
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? ReadTrendlineFilterWindowSize()
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: kDefaultTrendlineWindowSize),
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trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
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trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
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consecutive_delayed_feedbacks_(0),
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prev_bitrate_(0),
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prev_state_(BandwidthUsage::kBwNormal),
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in_sparse_update_experiment_(
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webrtc::field_trial::IsEnabled(kBweSparseUpdateExperiment)) {
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LOG(LS_INFO)
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<< "Using Trendline filter for delay change estimation with window size "
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<< trendline_window_size_;
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}
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DelayBasedBwe::~DelayBasedBwe() {}
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DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
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const std::vector<PacketFeedback>& packet_feedback_vector,
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rtc::Optional<uint32_t> acked_bitrate_bps) {
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RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
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packet_feedback_vector.end(),
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PacketFeedbackComparator()));
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RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
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// TOOD(holmer): An empty feedback vector here likely means that
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// all acks were too late and that the send time history had
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// timed out. We should reduce the rate when this occurs.
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if (packet_feedback_vector.empty()) {
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LOG(LS_WARNING) << "Very late feedback received.";
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return DelayBasedBwe::Result();
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}
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if (!uma_recorded_) {
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RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
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BweNames::kSendSideTransportSeqNum,
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BweNames::kBweNamesMax);
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uma_recorded_ = true;
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}
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bool overusing = false;
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bool delayed_feedback = true;
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bool recovered_from_overuse = false;
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BandwidthUsage prev_detector_state = detector_.State();
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for (const auto& packet_feedback : packet_feedback_vector) {
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if (packet_feedback.send_time_ms < 0)
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continue;
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delayed_feedback = false;
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IncomingPacketFeedback(packet_feedback);
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if (!in_sparse_update_experiment_)
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overusing |= (detector_.State() == BandwidthUsage::kBwOverusing);
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if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
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detector_.State() == BandwidthUsage::kBwNormal) {
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recovered_from_overuse = true;
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}
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prev_detector_state = detector_.State();
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}
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if (in_sparse_update_experiment_)
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overusing = (detector_.State() == BandwidthUsage::kBwOverusing);
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if (delayed_feedback) {
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++consecutive_delayed_feedbacks_;
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if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
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consecutive_delayed_feedbacks_ = 0;
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return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms);
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}
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} else {
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consecutive_delayed_feedbacks_ = 0;
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return MaybeUpdateEstimate(overusing, acked_bitrate_bps,
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recovered_from_overuse);
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}
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return Result();
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}
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DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
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int64_t arrival_time_ms) {
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// Estimate should always be valid since a start bitrate always is set in the
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// Call constructor. An alternative would be to return an empty Result here,
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// or to estimate the throughput based on the feedback we received.
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RTC_DCHECK(rate_control_.ValidEstimate());
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rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2,
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arrival_time_ms);
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Result result;
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result.updated = true;
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result.probe = false;
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result.target_bitrate_bps = rate_control_.LatestEstimate();
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LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to "
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<< result.target_bitrate_bps;
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return result;
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}
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void DelayBasedBwe::IncomingPacketFeedback(
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const PacketFeedback& packet_feedback) {
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int64_t now_ms = clock_->TimeInMilliseconds();
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// Reset if the stream has timed out.
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if (last_seen_packet_ms_ == -1 ||
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now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
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inter_arrival_.reset(
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new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
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kTimestampToMs, true));
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trendline_estimator_.reset(new TrendlineEstimator(
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trendline_window_size_, trendline_smoothing_coeff_,
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trendline_threshold_gain_));
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}
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last_seen_packet_ms_ = now_ms;
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uint32_t send_time_24bits =
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static_cast<uint32_t>(
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((static_cast<uint64_t>(packet_feedback.send_time_ms)
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<< kAbsSendTimeFraction) +
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500) /
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1000) &
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0x00FFFFFF;
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// Shift up send time to use the full 32 bits that inter_arrival works with,
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// so wrapping works properly.
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uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift;
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uint32_t ts_delta = 0;
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int64_t t_delta = 0;
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int size_delta = 0;
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if (inter_arrival_->ComputeDeltas(timestamp, packet_feedback.arrival_time_ms,
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now_ms, packet_feedback.payload_size,
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&ts_delta, &t_delta, &size_delta)) {
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double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
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trendline_estimator_->Update(t_delta, ts_delta_ms,
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packet_feedback.arrival_time_ms);
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detector_.Detect(trendline_estimator_->trendline_slope(), ts_delta_ms,
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trendline_estimator_->num_of_deltas(),
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packet_feedback.arrival_time_ms);
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}
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if (packet_feedback.pacing_info.probe_cluster_id !=
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PacedPacketInfo::kNotAProbe) {
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probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
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}
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}
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DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
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bool overusing,
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rtc::Optional<uint32_t> acked_bitrate_bps,
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bool recovered_from_overuse) {
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Result result;
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int64_t now_ms = clock_->TimeInMilliseconds();
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rtc::Optional<int> probe_bitrate_bps =
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probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
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// Currently overusing the bandwidth.
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if (overusing) {
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if (acked_bitrate_bps &&
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rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
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result.updated = UpdateEstimate(now_ms, acked_bitrate_bps, overusing,
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&result.target_bitrate_bps);
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} else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
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rate_control_.TimeToReduceFurther(
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now_ms, rate_control_.LatestEstimate() / 2 - 1)) {
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// Overusing before we have a measured acknowledged bitrate. We check
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// TimeToReduceFurther (with a fake acknowledged bitrate) to avoid
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// reducing too often.
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// TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
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// so that we (almost) always have a bitrate estimate.
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rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms);
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result.updated = true;
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result.probe = false;
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result.target_bitrate_bps = rate_control_.LatestEstimate();
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}
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} else {
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if (probe_bitrate_bps) {
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result.probe = true;
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result.updated = true;
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result.target_bitrate_bps = *probe_bitrate_bps;
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rate_control_.SetEstimate(*probe_bitrate_bps, now_ms);
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} else {
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result.updated = UpdateEstimate(now_ms, acked_bitrate_bps, overusing,
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&result.target_bitrate_bps);
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result.recovered_from_overuse = recovered_from_overuse;
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}
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}
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if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) ||
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detector_.State() != prev_state_) {
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uint32_t bitrate_bps =
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result.updated ? result.target_bitrate_bps : prev_bitrate_;
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BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps);
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if (event_log_) {
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event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
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bitrate_bps, detector_.State()));
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}
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prev_bitrate_ = bitrate_bps;
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prev_state_ = detector_.State();
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}
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return result;
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}
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bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
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rtc::Optional<uint32_t> acked_bitrate_bps,
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bool overusing,
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uint32_t* target_bitrate_bps) {
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// TODO(terelius): RateControlInput::noise_var is deprecated and will be
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// removed. In the meantime, we set it to zero.
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const RateControlInput input(
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overusing ? BandwidthUsage::kBwOverusing : detector_.State(),
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acked_bitrate_bps, 0);
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uint32_t prev_target_bitrate_bps = rate_control_.LatestEstimate();
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*target_bitrate_bps = rate_control_.Update(&input, now_ms);
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return rate_control_.ValidEstimate() &&
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prev_target_bitrate_bps != *target_bitrate_bps;
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}
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void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
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rate_control_.SetRtt(avg_rtt_ms);
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}
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bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
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uint32_t* bitrate_bps) const {
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// Currently accessed from both the process thread (see
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// ModuleRtpRtcpImpl::Process()) and the configuration thread (see
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// Call::GetStats()). Should in the future only be accessed from a single
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// thread.
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RTC_DCHECK(ssrcs);
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RTC_DCHECK(bitrate_bps);
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if (!rate_control_.ValidEstimate())
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return false;
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*ssrcs = {kFixedSsrc};
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*bitrate_bps = rate_control_.LatestEstimate();
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return true;
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}
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void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) {
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LOG(LS_WARNING) << "BWE Setting start bitrate to: " << start_bitrate_bps;
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rate_control_.SetStartBitrate(start_bitrate_bps);
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}
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void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) {
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// Called from both the configuration thread and the network thread. Shouldn't
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// be called from the network thread in the future.
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rate_control_.SetMinBitrate(min_bitrate_bps);
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}
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int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const {
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return rate_control_.GetExpectedBandwidthPeriodMs();
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}
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} // namespace webrtc
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