blob: f306fb255dc060c386569e343b19199eada30ab8 [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 "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
#include <algorithm>
#include <cmath>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/call/rtc_event_log.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 kDefaultMinBitrateBps = 10000;
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 = 1500;
const int64_t kFeedbackTimeoutIntervals = 3;
const int64_t kTimeoutIntervalMs = 1000;
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]);
} // namespace
SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
: lost_packets_since_last_loss_update_Q8_(0),
expected_packets_since_last_loss_update_(0),
bitrate_(0),
min_bitrate_configured_(kDefaultMinBitrateBps),
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::FindFullName(
"WebRTC-SendSideBwe") == "Enabled") {
RTC_DCHECK(event_log);
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
int min_bitrate,
int max_bitrate) {
if (send_bitrate > 0)
SetSendBitrate(send_bitrate);
SetMinMaxBitrate(min_bitrate, max_bitrate);
}
void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
RTC_DCHECK_GT(bitrate, 0);
bitrate_ = 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, kDefaultMinBitrateBps);
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 = bitrate_;
*loss = last_fraction_loss_;
*rtt = last_round_trip_time_ms_;
}
void SendSideBandwidthEstimation::UpdateReceiverEstimate(
int64_t now_ms, uint32_t bandwidth) {
bwe_incoming_ = bandwidth;
bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
}
void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
int64_t now_ms,
uint32_t bitrate_bps) {
delay_based_bitrate_bps_ = bitrate_bps;
bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
}
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.
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>((bitrate_ + 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) {
// 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)) {
uint32_t prev_bitrate = bitrate_;
if (bwe_incoming_ > bitrate_)
bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
if (delay_based_bitrate_bps_ > bitrate_) {
bitrate_ = CapBitrateToThresholds(now_ms, delay_based_bitrate_bps_);
}
if (bitrate_ != prev_bitrate) {
min_bitrate_history_.clear();
min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
return;
}
}
UpdateMinHistory(now_ms);
if (last_packet_report_ms_ == -1) {
// No feedback received.
bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
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) {
if (last_fraction_loss_ <= 5) {
// 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: bitrate_ *= 1.08^(delta time), it would
// take over one second since the lower packet loss to achieve
// 108kbps.
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).
bitrate_ += 1000;
} else if (last_fraction_loss_ <= 26) {
// 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;
bitrate_ = static_cast<uint32_t>(
(bitrate_ * 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_) {
LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms
<< " ms), reducing bitrate.";
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;
}
}
uint32_t capped_bitrate = CapBitrateToThresholds(now_ms, bitrate_);
if (capped_bitrate != bitrate_ ||
last_fraction_loss_ != last_logged_fraction_loss_ ||
last_rtc_event_log_ms_ == -1 ||
now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
event_log_->LogBwePacketLossEvent(capped_bitrate, last_fraction_loss_,
expected_packets_since_last_loss_update_);
last_logged_fraction_loss_ = last_fraction_loss_;
last_rtc_event_log_ms_ = now_ms;
}
bitrate_ = capped_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() &&
bitrate_ <= min_bitrate_history_.back().second) {
min_bitrate_history_.pop_back();
}
min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
}
uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(
int64_t now_ms, uint32_t bitrate) {
if (bwe_incoming_ > 0 && bitrate > bwe_incoming_) {
bitrate = bwe_incoming_;
}
if (delay_based_bitrate_bps_ > 0 && bitrate > delay_based_bitrate_bps_) {
bitrate = delay_based_bitrate_bps_;
}
if (bitrate > max_bitrate_configured_) {
bitrate = max_bitrate_configured_;
}
if (bitrate < min_bitrate_configured_) {
if (last_low_bitrate_log_ms_ == -1 ||
now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
<< " kbps is below configured min bitrate "
<< min_bitrate_configured_ / 1000 << " kbps.";
last_low_bitrate_log_ms_ = now_ms;
}
bitrate = min_bitrate_configured_;
}
return bitrate;
}
} // namespace webrtc