blob: f28758190c54b550ec8335e92592a6939ccd3cd5 [file] [log] [blame]
/*
* 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 <string>
#include "absl/memory/memory.h"
#include "api/transport/network_types.h" // For PacedPacketInfo
#include "logging/rtc_event_log/events/rtc_event.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/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;
// 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;
const char kBweWindowSizeInPacketsExperiment[] =
"WebRTC-BweWindowSizeInPackets";
size_t ReadTrendlineFilterWindowSize(
const WebRtcKeyValueConfig* key_value_config) {
std::string experiment_string =
key_value_config->Lookup(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 BweWindowSizeInPackets"
" experiment from field trial string. Using default.";
return kDefaultTrendlineWindowSize;
}
} // namespace
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)
: event_log_(event_log),
inter_arrival_(),
delay_detector_(),
last_seen_packet_(Timestamp::MinusInfinity()),
uma_recorded_(false),
trendline_window_size_(
key_value_config->Lookup(kBweWindowSizeInPacketsExperiment)
.find("Enabled") == 0
? ReadTrendlineFilterWindowSize(key_value_config)
: kDefaultTrendlineWindowSize),
trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
prev_bitrate_(DataRate::Zero()),
prev_state_(BandwidthUsage::kBwNormal),
alr_limited_backoff_enabled_(
key_value_config->Lookup("WebRTC-Bwe-AlrLimitedBackoff")
.find("Enabled") == 0) {
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<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
bool in_alr,
Timestamp at_time) {
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);
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();
}
return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
recovered_from_overuse, in_alr, at_time);
}
void DelayBasedBwe::IncomingPacketFeedback(
const PacketFeedback& 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(trendline_window_size_,
trendline_smoothing_coeff_,
trendline_threshold_gain_));
}
last_seen_packet_ = at_time;
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,
at_time.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);
}
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
bool recovered_from_overuse,
bool in_alr,
Timestamp at_time) {
Result result;
// Currently overusing the bandwidth.
if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
if (in_alr && alr_limited_backoff_enabled_ &&
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();
}
} 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(absl::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