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/*
* 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 <cstdint>
#include <memory>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/bandwidth_usage.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/race_checker.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds(2);
constexpr TimeDelta kSendTimeGroupLength = TimeDelta::Millis(5);
// This ssrc is used to fulfill the current API but will be removed
// after the API has been changed.
constexpr uint32_t kFixedSsrc = 0;
} // namespace
constexpr char BweSeparateAudioPacketsSettings::kKey[];
BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
const FieldTrialsView* key_value_config) {
Parser()->Parse(
key_value_config->Lookup(BweSeparateAudioPacketsSettings::kKey));
}
std::unique_ptr<StructParametersParser>
BweSeparateAudioPacketsSettings::Parser() {
return StructParametersParser::Create( //
"enabled", &enabled, //
"packet_threshold", &packet_threshold, //
"time_threshold", &time_threshold);
}
DelayBasedBwe::Result::Result()
: updated(false),
probe(false),
target_bitrate(DataRate::Zero()),
recovered_from_overuse(false),
delay_detector_state(BandwidthUsage::kBwNormal) {}
DelayBasedBwe::DelayBasedBwe(const FieldTrialsView* key_value_config,
RtcEventLog* event_log,
NetworkStatePredictor* network_state_predictor)
: event_log_(event_log),
key_value_config_(key_value_config),
separate_audio_(key_value_config),
audio_packets_since_last_video_(0),
last_video_packet_recv_time_(Timestamp::MinusInfinity()),
network_state_predictor_(network_state_predictor),
video_delay_detector_(
new TrendlineEstimator(key_value_config_, network_state_predictor_)),
audio_delay_detector_(
new TrendlineEstimator(key_value_config_, network_state_predictor_)),
active_delay_detector_(video_delay_detector_.get()),
last_seen_packet_(Timestamp::MinusInfinity()),
uma_recorded_(false),
rate_control_(*key_value_config, /*send_side=*/true),
prev_bitrate_(DataRate::Zero()),
prev_state_(BandwidthUsage::kBwNormal) {
RTC_LOG(LS_INFO)
<< "Initialized DelayBasedBwe with separate audio overuse detection"
<< separate_audio_.Parser()->Encode();
}
DelayBasedBwe::~DelayBasedBwe() {}
DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
const TransportPacketsFeedback& msg,
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::optional<NetworkStateEstimate> network_estimate,
bool in_alr) {
RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
auto packet_feedback_vector = msg.SortedByReceiveTime();
// TODO(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 = active_delay_detector_->State();
for (const auto& packet_feedback : packet_feedback_vector) {
delayed_feedback = false;
IncomingPacketFeedback(packet_feedback, msg.feedback_time);
if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
active_delay_detector_->State() == BandwidthUsage::kBwNormal) {
recovered_from_overuse = true;
}
prev_detector_state = active_delay_detector_->State();
}
if (delayed_feedback) {
// TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
// against building very large network queues.
return Result();
}
rate_control_.SetInApplicationLimitedRegion(in_alr);
rate_control_.SetNetworkStateEstimate(network_estimate);
return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
std::move(network_estimate),
recovered_from_overuse, in_alr, msg.feedback_time);
}
void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
Timestamp at_time) {
// Reset if the stream has timed out.
if (last_seen_packet_.IsInfinite() ||
at_time - last_seen_packet_ > kStreamTimeOut) {
video_inter_arrival_delta_ =
std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
audio_inter_arrival_delta_ =
std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
video_delay_detector_.reset(
new TrendlineEstimator(key_value_config_, network_state_predictor_));
audio_delay_detector_.reset(
new TrendlineEstimator(key_value_config_, network_state_predictor_));
active_delay_detector_ = video_delay_detector_.get();
}
last_seen_packet_ = at_time;
// As an alternative to ignoring small packets, we can separate audio and
// video packets for overuse detection.
DelayIncreaseDetectorInterface* delay_detector_for_packet =
video_delay_detector_.get();
if (separate_audio_.enabled) {
if (packet_feedback.sent_packet.audio) {
delay_detector_for_packet = audio_delay_detector_.get();
audio_packets_since_last_video_++;
if (audio_packets_since_last_video_ > separate_audio_.packet_threshold &&
packet_feedback.receive_time - last_video_packet_recv_time_ >
separate_audio_.time_threshold) {
active_delay_detector_ = audio_delay_detector_.get();
}
} else {
audio_packets_since_last_video_ = 0;
last_video_packet_recv_time_ =
std::max(last_video_packet_recv_time_, packet_feedback.receive_time);
active_delay_detector_ = video_delay_detector_.get();
}
}
DataSize packet_size = packet_feedback.sent_packet.size;
TimeDelta send_delta = TimeDelta::Zero();
TimeDelta recv_delta = TimeDelta::Zero();
int size_delta = 0;
InterArrivalDelta* inter_arrival_for_packet =
(separate_audio_.enabled && packet_feedback.sent_packet.audio)
? audio_inter_arrival_delta_.get()
: video_inter_arrival_delta_.get();
bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
packet_feedback.sent_packet.send_time, packet_feedback.receive_time,
at_time, packet_size.bytes(), &send_delta, &recv_delta, &size_delta);
delay_detector_for_packet->Update(recv_delta.ms<double>(),
send_delta.ms<double>(),
packet_feedback.sent_packet.send_time.ms(),
packet_feedback.receive_time.ms(),
packet_size.bytes(), calculated_deltas);
}
DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
absl::optional<DataRate> link_capacity) {
RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
return rate_control_.Update(input, at_time);
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::optional<NetworkStateEstimate> state_estimate,
bool recovered_from_overuse,
bool in_alr,
Timestamp at_time) {
Result result;
// Currently overusing the bandwidth.
if (active_delay_detector_->State() == BandwidthUsage::kBwOverusing) {
if (acked_bitrate &&
rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
result.updated =
UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
} else if (!acked_bitrate && rate_control_.ValidEstimate() &&
rate_control_.InitialTimeToReduceFurther(at_time)) {
// 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, at_time);
result.updated = true;
result.probe = false;
result.target_bitrate = rate_control_.LatestEstimate();
}
} else {
if (probe_bitrate) {
result.probe = true;
result.updated = true;
rate_control_.SetEstimate(*probe_bitrate, at_time);
result.target_bitrate = rate_control_.LatestEstimate();
} else {
result.updated =
UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
result.recovered_from_overuse = recovered_from_overuse;
}
}
BandwidthUsage detector_state = active_delay_detector_->State();
if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
detector_state != prev_state_) {
DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
bitrate.bps(), detector_state));
}
prev_bitrate_ = bitrate;
prev_state_ = detector_state;
}
result.delay_detector_state = detector_state;
return result;
}
bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
absl::optional<DataRate> acked_bitrate,
DataRate* target_rate) {
const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
*target_rate = rate_control_.Update(input, at_time);
return rate_control_.ValidEstimate();
}
void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
rate_control_.SetRtt(avg_rtt);
}
bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
DataRate* bitrate) 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);
if (!rate_control_.ValidEstimate())
return false;
*ssrcs = {kFixedSsrc};
*bitrate = rate_control_.LatestEstimate();
return true;
}
void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
<< ToString(start_bitrate);
rate_control_.SetStartBitrate(start_bitrate);
}
void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
// 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);
}
TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
return rate_control_.GetExpectedBandwidthPeriod();
}
} // namespace webrtc