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This makes it safer to reason about the common case where send time information is available. We don't have to either assume that it's available, or check it everywhere the PacketResult struct is used. To achieve this, a new field is added to TransportPacketsFeedback and a new interface is introduced to clearly separate which field is used. A possible followup would be to introduce a separate struct. That would complicate the signature of ProcessTransportFeedback. Bug: webrtc:9934 Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15 Reviewed-on: https://webrtc-review.googlesource.com/c/107080 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25465}
330 lines
12 KiB
C++
330 lines
12 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/goog_cc/delay_based_bwe.h"
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#include <algorithm>
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#include <cstdint>
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#include <cstdio>
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#include <string>
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#include "absl/memory/memory.h"
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#include "api/transport/network_types.h" // For PacedPacketInfo
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#include "logging/rtc_event_log/events/rtc_event.h"
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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/congestion_controller/goog_cc/trendline_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/logging.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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namespace {
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static const int64_t kStreamTimeOutMs = 2000;
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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 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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RTC_LOG(WARNING) << "Window size must be greater than 1.";
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}
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RTC_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)
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: event_log_(event_log),
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inter_arrival_(),
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delay_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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RTC_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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delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
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trendline_smoothing_coeff_,
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trendline_threshold_gain_));
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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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absl::optional<uint32_t> acked_bitrate_bps,
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int64_t at_time_ms) {
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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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RTC_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 delayed_feedback = true;
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bool recovered_from_overuse = false;
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BandwidthUsage prev_detector_state = delay_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, at_time_ms);
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if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
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delay_detector_->State() == BandwidthUsage::kBwNormal) {
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recovered_from_overuse = true;
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}
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prev_detector_state = delay_detector_->State();
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}
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if (delayed_feedback) {
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return OnDelayedFeedback(packet_feedback_vector.back().arrival_time_ms);
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} else {
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consecutive_delayed_feedbacks_ = 0;
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return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse,
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at_time_ms);
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}
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return Result();
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}
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DelayBasedBwe::Result DelayBasedBwe::OnDelayedFeedback(
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int64_t receive_time_ms) {
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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(receive_time_ms);
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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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RTC_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 at_time_ms) {
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int64_t now_ms = at_time_ms;
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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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delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
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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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delay_detector_->Update(t_delta, ts_delta_ms,
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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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absl::optional<uint32_t> acked_bitrate_bps,
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bool recovered_from_overuse,
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int64_t at_time_ms) {
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Result result;
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int64_t now_ms = at_time_ms;
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absl::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 (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
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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 =
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UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
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} else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
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rate_control_.InitialTimeToReduceFurther(now_ms)) {
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// Overusing before we have a measured acknowledged bitrate. Reduce send
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// rate by 50% every 200 ms.
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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 =
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UpdateEstimate(now_ms, acked_bitrate_bps, &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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BandwidthUsage detector_state = delay_detector_->State();
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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(absl::make_unique<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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absl::optional<uint32_t> acked_bitrate_bps,
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uint32_t* target_bitrate_bps) {
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const RateControlInput input(delay_detector_->State(), acked_bitrate_bps);
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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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}
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void DelayBasedBwe::OnRttUpdate(int64_t avg_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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RTC_LOG(LS_INFO) << "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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