blob: 86096a1b736520b0bf876a8098a6aa33948b402e [file] [log] [blame]
/*
* Copyright (c) 2012 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/bitrate_controller/send_side_bandwidth_estimation.h"
#include <algorithm>
#include <cmath>
#include <limits>
#include <string>
#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
const int64_t kBweIncreaseIntervalMs = 1000;
const int64_t kBweDecreaseIntervalMs = 300;
const int64_t kStartPhaseMs = 2000;
const int64_t kBweConverganceTimeMs = 20000;
const int kLimitNumPackets = 20;
const int kDefaultMaxBitrateBps = 1000000000;
const int64_t kLowBitrateLogPeriodMs = 10000;
const int64_t kRtcEventLogPeriodMs = 5000;
// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
const int64_t kFeedbackIntervalMs = 5000;
const int64_t kFeedbackTimeoutIntervals = 3;
const int64_t kTimeoutIntervalMs = 1000;
const float kDefaultLowLossThreshold = 0.02f;
const float kDefaultHighLossThreshold = 0.1f;
const int kDefaultBitrateThresholdKbps = 0;
struct UmaRampUpMetric {
const char* metric_name;
int bitrate_kbps;
};
const UmaRampUpMetric kUmaRampupMetrics[] = {
{"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
{"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
{"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
const size_t kNumUmaRampupMetrics =
sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
const char kBweLosExperiment[] = "WebRTC-BweLossExperiment";
bool BweLossExperimentIsEnabled() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweLosExperiment);
// The experiment is enabled iff the field trial string begins with "Enabled".
return experiment_string.find("Enabled") == 0;
}
bool ReadBweLossExperimentParameters(float* low_loss_threshold,
float* high_loss_threshold,
uint32_t* bitrate_threshold_kbps) {
RTC_DCHECK(low_loss_threshold);
RTC_DCHECK(high_loss_threshold);
RTC_DCHECK(bitrate_threshold_kbps);
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweLosExperiment);
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold,
high_loss_threshold, bitrate_threshold_kbps);
if (parsed_values == 3) {
RTC_CHECK_GT(*low_loss_threshold, 0.0f)
<< "Loss threshold must be greater than 0.";
RTC_CHECK_LE(*low_loss_threshold, 1.0f)
<< "Loss threshold must be less than or equal to 1.";
RTC_CHECK_GT(*high_loss_threshold, 0.0f)
<< "Loss threshold must be greater than 0.";
RTC_CHECK_LE(*high_loss_threshold, 1.0f)
<< "Loss threshold must be less than or equal to 1.";
RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold)
<< "The low loss threshold must be less than or equal to the high loss "
"threshold.";
RTC_CHECK_GE(*bitrate_threshold_kbps, 0)
<< "Bitrate threshold can't be negative.";
RTC_CHECK_LT(*bitrate_threshold_kbps,
std::numeric_limits<int>::max() / 1000)
<< "Bitrate must be smaller enough to avoid overflows.";
return true;
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment "
"experiment from field trial string. Using default.";
*low_loss_threshold = kDefaultLowLossThreshold;
*high_loss_threshold = kDefaultHighLossThreshold;
*bitrate_threshold_kbps = kDefaultBitrateThresholdKbps;
return false;
}
} // namespace
SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
: lost_packets_since_last_loss_update_Q8_(0),
expected_packets_since_last_loss_update_(0),
current_bitrate_bps_(0),
min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),
max_bitrate_configured_(kDefaultMaxBitrateBps),
last_low_bitrate_log_ms_(-1),
has_decreased_since_last_fraction_loss_(false),
last_feedback_ms_(-1),
last_packet_report_ms_(-1),
last_timeout_ms_(-1),
last_fraction_loss_(0),
last_logged_fraction_loss_(0),
last_round_trip_time_ms_(0),
bwe_incoming_(0),
delay_based_bitrate_bps_(0),
time_last_decrease_ms_(0),
first_report_time_ms_(-1),
initially_lost_packets_(0),
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate),
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
event_log_(event_log),
last_rtc_event_log_ms_(-1),
in_timeout_experiment_(
webrtc::field_trial::IsEnabled("WebRTC-FeedbackTimeout")),
low_loss_threshold_(kDefaultLowLossThreshold),
high_loss_threshold_(kDefaultHighLossThreshold),
bitrate_threshold_bps_(1000 * kDefaultBitrateThresholdKbps) {
RTC_DCHECK(event_log);
if (BweLossExperimentIsEnabled()) {
uint32_t bitrate_threshold_kbps;
if (ReadBweLossExperimentParameters(&low_loss_threshold_,
&high_loss_threshold_,
&bitrate_threshold_kbps)) {
RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "
<< low_loss_threshold_ << ", " << high_loss_threshold_
<< ", " << bitrate_threshold_kbps;
bitrate_threshold_bps_ = bitrate_threshold_kbps * 1000;
}
}
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
int min_bitrate,
int max_bitrate) {
SetMinMaxBitrate(min_bitrate, max_bitrate);
if (send_bitrate > 0)
SetSendBitrate(send_bitrate);
}
void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
RTC_DCHECK_GT(bitrate, 0);
delay_based_bitrate_bps_ = 0; // Reset to avoid being capped by the estimate.
CapBitrateToThresholds(Clock::GetRealTimeClock()->TimeInMilliseconds(),
bitrate);
// Clear last sent bitrate history so the new value can be used directly
// and not capped.
min_bitrate_history_.clear();
}
void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate,
int max_bitrate) {
RTC_DCHECK_GE(min_bitrate, 0);
min_bitrate_configured_ =
std::max(min_bitrate, congestion_controller::GetMinBitrateBps());
if (max_bitrate > 0) {
max_bitrate_configured_ =
std::max<uint32_t>(min_bitrate_configured_, max_bitrate);
} else {
max_bitrate_configured_ = kDefaultMaxBitrateBps;
}
}
int SendSideBandwidthEstimation::GetMinBitrate() const {
return min_bitrate_configured_;
}
void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate,
uint8_t* loss,
int64_t* rtt) const {
*bitrate = current_bitrate_bps_;
*loss = last_fraction_loss_;
*rtt = last_round_trip_time_ms_;
}
void SendSideBandwidthEstimation::UpdateReceiverEstimate(
int64_t now_ms, uint32_t bandwidth) {
bwe_incoming_ = bandwidth;
CapBitrateToThresholds(now_ms, current_bitrate_bps_);
}
void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
int64_t now_ms,
uint32_t bitrate_bps) {
delay_based_bitrate_bps_ = bitrate_bps;
CapBitrateToThresholds(now_ms, current_bitrate_bps_);
}
void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
int64_t rtt,
int number_of_packets,
int64_t now_ms) {
last_feedback_ms_ = now_ms;
if (first_report_time_ms_ == -1)
first_report_time_ms_ = now_ms;
// Update RTT if we were able to compute an RTT based on this RTCP.
// FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
if (rtt > 0)
last_round_trip_time_ms_ = rtt;
// Check sequence number diff and weight loss report
if (number_of_packets > 0) {
// Calculate number of lost packets.
const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
// Accumulate reports.
lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
expected_packets_since_last_loss_update_ += number_of_packets;
// Don't generate a loss rate until it can be based on enough packets.
if (expected_packets_since_last_loss_update_ < kLimitNumPackets)
return;
has_decreased_since_last_fraction_loss_ = false;
last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
expected_packets_since_last_loss_update_;
// Reset accumulators.
lost_packets_since_last_loss_update_Q8_ = 0;
expected_packets_since_last_loss_update_ = 0;
last_packet_report_ms_ = now_ms;
UpdateEstimate(now_ms);
}
UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
}
void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
int64_t rtt,
int lost_packets) {
int bitrate_kbps = static_cast<int>((current_bitrate_bps_ + 500) / 1000);
for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
if (!rampup_uma_stats_updated_[i] &&
bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
now_ms - first_report_time_ms_);
rampup_uma_stats_updated_[i] = true;
}
}
if (IsInStartPhase(now_ms)) {
initially_lost_packets_ += lost_packets;
} else if (uma_update_state_ == kNoUpdate) {
uma_update_state_ = kFirstDone;
bitrate_at_2_seconds_kbps_ = bitrate_kbps;
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
2000, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
bitrate_at_2_seconds_kbps_, 0, 2000, 50);
} else if (uma_update_state_ == kFirstDone &&
now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
uma_update_state_ = kDone;
int bitrate_diff_kbps =
std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
0, 2000, 50);
}
}
void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
uint32_t new_bitrate = current_bitrate_bps_;
// We trust the REMB and/or delay-based estimate during the first 2 seconds if
// we haven't had any packet loss reported, to allow startup bitrate probing.
if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms)) {
new_bitrate = std::max(bwe_incoming_, new_bitrate);
new_bitrate = std::max(delay_based_bitrate_bps_, new_bitrate);
if (new_bitrate != current_bitrate_bps_) {
min_bitrate_history_.clear();
min_bitrate_history_.push_back(
std::make_pair(now_ms, current_bitrate_bps_));
CapBitrateToThresholds(now_ms, new_bitrate);
return;
}
}
UpdateMinHistory(now_ms);
if (last_packet_report_ms_ == -1) {
// No feedback received.
CapBitrateToThresholds(now_ms, current_bitrate_bps_);
return;
}
int64_t time_since_packet_report_ms = now_ms - last_packet_report_ms_;
int64_t time_since_feedback_ms = now_ms - last_feedback_ms_;
if (time_since_packet_report_ms < 1.2 * kFeedbackIntervalMs) {
// We only care about loss above a given bitrate threshold.
float loss = last_fraction_loss_ / 256.0f;
// We only make decisions based on loss when the bitrate is above a
// threshold. This is a crude way of handling loss which is uncorrelated
// to congestion.
if (current_bitrate_bps_ < bitrate_threshold_bps_ ||
loss <= low_loss_threshold_) {
// Loss < 2%: Increase rate by 8% of the min bitrate in the last
// kBweIncreaseIntervalMs.
// Note that by remembering the bitrate over the last second one can
// rampup up one second faster than if only allowed to start ramping
// at 8% per second rate now. E.g.:
// If sending a constant 100kbps it can rampup immediatly to 108kbps
// whenever a receiver report is received with lower packet loss.
// If instead one would do: current_bitrate_bps_ *= 1.08^(delta time),
// it would take over one second since the lower packet loss to achieve
// 108kbps.
new_bitrate = static_cast<uint32_t>(
min_bitrate_history_.front().second * 1.08 + 0.5);
// Add 1 kbps extra, just to make sure that we do not get stuck
// (gives a little extra increase at low rates, negligible at higher
// rates).
new_bitrate += 1000;
} else if (current_bitrate_bps_ > bitrate_threshold_bps_) {
if (loss <= high_loss_threshold_) {
// Loss between 2% - 10%: Do nothing.
} else {
// Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs
// + rtt.
if (!has_decreased_since_last_fraction_loss_ &&
(now_ms - time_last_decrease_ms_) >=
(kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
time_last_decrease_ms_ = now_ms;
// Reduce rate:
// newRate = rate * (1 - 0.5*lossRate);
// where packetLoss = 256*lossRate;
new_bitrate = static_cast<uint32_t>(
(current_bitrate_bps_ *
static_cast<double>(512 - last_fraction_loss_)) /
512.0);
has_decreased_since_last_fraction_loss_ = true;
}
}
}
} else if (time_since_feedback_ms >
kFeedbackTimeoutIntervals * kFeedbackIntervalMs &&
(last_timeout_ms_ == -1 ||
now_ms - last_timeout_ms_ > kTimeoutIntervalMs)) {
if (in_timeout_experiment_) {
RTC_LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms
<< " ms), reducing bitrate.";
new_bitrate *= 0.8;
// Reset accumulators since we've already acted on missing feedback and
// shouldn't to act again on these old lost packets.
lost_packets_since_last_loss_update_Q8_ = 0;
expected_packets_since_last_loss_update_ = 0;
last_timeout_ms_ = now_ms;
}
}
CapBitrateToThresholds(now_ms, new_bitrate);
}
bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const {
return first_report_time_ms_ == -1 ||
now_ms - first_report_time_ms_ < kStartPhaseMs;
}
void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) {
// Remove old data points from history.
// Since history precision is in ms, add one so it is able to increase
// bitrate if it is off by as little as 0.5ms.
while (!min_bitrate_history_.empty() &&
now_ms - min_bitrate_history_.front().first + 1 >
kBweIncreaseIntervalMs) {
min_bitrate_history_.pop_front();
}
// Typical minimum sliding-window algorithm: Pop values higher than current
// bitrate before pushing it.
while (!min_bitrate_history_.empty() &&
current_bitrate_bps_ <= min_bitrate_history_.back().second) {
min_bitrate_history_.pop_back();
}
min_bitrate_history_.push_back(std::make_pair(now_ms, current_bitrate_bps_));
}
void SendSideBandwidthEstimation::CapBitrateToThresholds(int64_t now_ms,
uint32_t bitrate_bps) {
if (bwe_incoming_ > 0 && bitrate_bps > bwe_incoming_) {
bitrate_bps = bwe_incoming_;
}
if (delay_based_bitrate_bps_ > 0 && bitrate_bps > delay_based_bitrate_bps_) {
bitrate_bps = delay_based_bitrate_bps_;
}
if (bitrate_bps > max_bitrate_configured_) {
bitrate_bps = max_bitrate_configured_;
}
if (bitrate_bps < min_bitrate_configured_) {
if (last_low_bitrate_log_ms_ == -1 ||
now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
RTC_LOG(LS_WARNING) << "Estimated available bandwidth "
<< bitrate_bps / 1000
<< " kbps is below configured min bitrate "
<< min_bitrate_configured_ / 1000 << " kbps.";
last_low_bitrate_log_ms_ = now_ms;
}
bitrate_bps = min_bitrate_configured_;
}
if (bitrate_bps != current_bitrate_bps_ ||
last_fraction_loss_ != last_logged_fraction_loss_ ||
now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(
bitrate_bps, last_fraction_loss_,
expected_packets_since_last_loss_update_));
last_logged_fraction_loss_ = last_fraction_loss_;
last_rtc_event_log_ms_ = now_ms;
}
current_bitrate_bps_ = bitrate_bps;
}
} // namespace webrtc