webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
Björn Terelius fd0e32a87a Fix filtering of small packets in delay-based BWE
crodbro@ found that the previous field trial, which filtered the deltas
in the trendline estimator, can increase the noise caused by varying
packet sizes. Moving the filtering to the DelayBasedBwe class fixes the
issue.

To avoid confusion, we've updated the field trial name, so e.g.
WebRTC-BweIgnoreSmallPacketsFix/small:200bytes,large:200bytes,
                                fraction_large:0.25,smoothing:0.1/
should be used to enable the feature.

Bug: webrtc:10932
Change-Id: If77e83043c37fff909038405f634e541ce41abb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159711
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29804}
2019-11-15 14:53:59 +00:00

323 lines
12 KiB
C++

/*
* 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 <cstdio>
#include <memory>
#include <string>
#include <utility>
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds<2>();
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;
} // namespace
constexpr char BweIgnoreSmallPacketsSettings::kKey[];
BweIgnoreSmallPacketsSettings::BweIgnoreSmallPacketsSettings(
const WebRtcKeyValueConfig* key_value_config) {
Parser()->Parse(
key_value_config->Lookup(BweIgnoreSmallPacketsSettings::kKey));
}
std::unique_ptr<StructParametersParser>
BweIgnoreSmallPacketsSettings::Parser() {
return StructParametersParser::Create("smoothing", &smoothing_factor, //
"fraction_large", &fraction_large, //
"large", &large_threshold, //
"small", &small_threshold);
}
DelayBasedBwe::Result::Result()
: updated(false),
probe(false),
target_bitrate(DataRate::Zero()),
recovered_from_overuse(false),
backoff_in_alr(false) {}
DelayBasedBwe::Result::Result(bool probe, DataRate target_bitrate)
: updated(true),
probe(probe),
target_bitrate(target_bitrate),
recovered_from_overuse(false),
backoff_in_alr(false) {}
DelayBasedBwe::Result::~Result() {}
DelayBasedBwe::DelayBasedBwe(const WebRtcKeyValueConfig* key_value_config,
RtcEventLog* event_log,
NetworkStatePredictor* network_state_predictor)
: event_log_(event_log),
key_value_config_(key_value_config),
ignore_small_(key_value_config),
fraction_large_packets_(0.5),
network_state_predictor_(network_state_predictor),
inter_arrival_(),
delay_detector_(
new TrendlineEstimator(key_value_config_, network_state_predictor_)),
last_seen_packet_(Timestamp::MinusInfinity()),
uma_recorded_(false),
rate_control_(key_value_config, /*send_side=*/true),
prev_bitrate_(DataRate::Zero()),
has_once_detected_overuse_(false),
prev_state_(BandwidthUsage::kBwNormal),
alr_limited_backoff_enabled_(
key_value_config->Lookup("WebRTC-Bwe-AlrLimitedBackoff")
.find("Enabled") == 0) {
RTC_LOG(LS_INFO) << "Initialized DelayBasedBwe with field trial "
<< ignore_small_.Parser()->Encode()
<< " and alr limited backoff "
<< (alr_limited_backoff_enabled_ ? "enabled" : "disabled");
}
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 = 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 &&
delay_detector_->State() == BandwidthUsage::kBwNormal) {
recovered_from_overuse = true;
}
prev_detector_state = 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) {
inter_arrival_.reset(
new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
kTimestampToMs, true));
delay_detector_.reset(
new TrendlineEstimator(key_value_config_, network_state_predictor_));
}
last_seen_packet_ = at_time;
uint32_t send_time_24bits =
static_cast<uint32_t>(
((static_cast<uint64_t>(packet_feedback.sent_packet.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;
// Ignore "small" packets if many/most packets in the call are "large". The
// packet size may have a significant effect on the propagation delay,
// especially at low bandwidths. Variations in packet size will then show up
// as noise in the delay measurement. By default, we include all packets.
DataSize packet_size = packet_feedback.sent_packet.size;
if (!ignore_small_.small_threshold.IsZero()) {
double is_large =
static_cast<double>(packet_size >= ignore_small_.large_threshold);
fraction_large_packets_ +=
ignore_small_.smoothing_factor * (is_large - fraction_large_packets_);
if (packet_size <= ignore_small_.small_threshold &&
fraction_large_packets_ >= ignore_small_.fraction_large) {
return;
}
}
uint32_t ts_delta = 0;
int64_t t_delta = 0;
int size_delta = 0;
bool calculated_deltas = inter_arrival_->ComputeDeltas(
timestamp, packet_feedback.receive_time.ms(), at_time.ms(),
packet_size.bytes(), &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.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 (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
if (has_once_detected_overuse_ && in_alr && alr_limited_backoff_enabled_) {
if (rate_control_.TimeToReduceFurther(at_time, prev_bitrate_)) {
result.updated =
UpdateEstimate(at_time, prev_bitrate_, &result.target_bitrate);
result.backoff_in_alr = true;
}
} else 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();
}
has_once_detected_overuse_ = true;
} else {
if (probe_bitrate) {
result.probe = true;
result.updated = true;
result.target_bitrate = *probe_bitrate;
rate_control_.SetEstimate(*probe_bitrate, at_time);
} else {
result.updated =
UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
result.recovered_from_overuse = recovered_from_overuse;
}
}
BandwidthUsage detector_state = delay_detector_->State();
if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
detector_state != prev_state_) {
DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
bitrate.bps(), detector_state));
}
prev_bitrate_ = bitrate;
prev_state_ = detector_state;
}
return result;
}
bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
absl::optional<DataRate> acked_bitrate,
DataRate* target_rate) {
const RateControlInput input(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();
}
void DelayBasedBwe::SetAlrLimitedBackoffExperiment(bool enabled) {
alr_limited_backoff_enabled_ = enabled;
}
} // namespace webrtc