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/*
* 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 <cmath>
#include "webrtc/system_wrappers/interface/logging.h"
namespace webrtc {
namespace {
enum { kBweIncreaseIntervalMs = 1000 };
enum { kBweDecreaseIntervalMs = 300 };
enum { kLimitNumPackets = 20 };
enum { kAvgPacketSizeBytes = 1000 };
// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
// The formula in RFC 3448, Section 3.1, is used.
uint32_t CalcTfrcBps(uint16_t rtt, uint8_t loss) {
if (rtt == 0 || loss == 0) {
// Input variables out of range.
return 0;
}
double R = static_cast<double>(rtt) / 1000; // RTT in seconds.
int b = 1; // Number of packets acknowledged by a single TCP acknowledgement:
// recommended = 1.
double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds
// recommended = 4*R.
double p = static_cast<double>(loss) / 255; // Packet loss rate in [0, 1).
double s = static_cast<double>(kAvgPacketSizeBytes);
// Calculate send rate in bytes/second.
double X =
s / (R * std::sqrt(2 * b * p / 3) +
(t_RTO * (3 * std::sqrt(3 * b * p / 8) * p * (1 + 32 * p * p))));
// Convert to bits/second.
return (static_cast<uint32_t>(X * 8));
}
}
SendSideBandwidthEstimation::SendSideBandwidthEstimation()
: accumulate_lost_packets_Q8_(0),
accumulate_expected_packets_(0),
bitrate_(0),
min_bitrate_configured_(0),
max_bitrate_configured_(0),
time_last_receiver_block_ms_(0),
last_fraction_loss_(0),
last_round_trip_time_ms_(0),
bwe_incoming_(0),
time_last_decrease_ms_(0) {}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
void SendSideBandwidthEstimation::SetSendBitrate(uint32_t bitrate) {
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(uint32_t min_bitrate,
uint32_t max_bitrate) {
min_bitrate_configured_ = min_bitrate;
max_bitrate_configured_ = max_bitrate;
}
void SendSideBandwidthEstimation::SetMinBitrate(uint32_t min_bitrate) {
min_bitrate_configured_ = min_bitrate;
}
void SendSideBandwidthEstimation::CurrentEstimate(uint32_t* bitrate,
uint8_t* loss,
uint32_t* rtt) const {
*bitrate = bitrate_;
*loss = last_fraction_loss_;
*rtt = last_round_trip_time_ms_;
}
void SendSideBandwidthEstimation::UpdateReceiverEstimate(uint32_t bandwidth) {
bwe_incoming_ = bandwidth;
CapBitrateToThresholds();
}
void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
uint32_t rtt,
int number_of_packets,
uint32_t 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.
accumulate_lost_packets_Q8_ += num_lost_packets_Q8;
accumulate_expected_packets_ += number_of_packets;
// Report loss if the total report is based on sufficiently many packets.
if (accumulate_expected_packets_ >= kLimitNumPackets) {
last_fraction_loss_ =
accumulate_lost_packets_Q8_ / accumulate_expected_packets_;
// Reset accumulators.
accumulate_lost_packets_Q8_ = 0;
accumulate_expected_packets_ = 0;
} else {
// Early return without updating estimate.
return;
}
}
time_last_receiver_block_ms_ = now_ms;
UpdateEstimate(now_ms);
}
void SendSideBandwidthEstimation::UpdateEstimate(uint32_t now_ms) {
UpdateMinHistory(now_ms);
// Only start updating bitrate when receiving receiver blocks.
if (time_last_receiver_block_ms_ != 0) {
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 ((now_ms - time_last_decrease_ms_) >=
static_cast<uint32_t>(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);
// Calculate what rate TFRC would apply in this situation and to not
// reduce further than it.
bitrate_ = std::max(
bitrate_,
CalcTfrcBps(last_round_trip_time_ms_, last_fraction_loss_));
}
}
}
CapBitrateToThresholds();
}
void SendSideBandwidthEstimation::UpdateMinHistory(uint32_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_));
}
void SendSideBandwidthEstimation::CapBitrateToThresholds() {
if (bwe_incoming_ > 0 && bitrate_ > bwe_incoming_) {
bitrate_ = bwe_incoming_;
}
if (bitrate_ > max_bitrate_configured_) {
bitrate_ = max_bitrate_configured_;
}
if (bitrate_ < min_bitrate_configured_) {
LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate_ / 1000
<< " kbps is below configured min bitrate "
<< min_bitrate_configured_ / 1000 << " kbps.";
bitrate_ = min_bitrate_configured_;
}
}
} // namespace webrtc