
BUG=3153 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11069005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5851 4adac7df-926f-26a2-2b94-8c16560cd09d
214 lines
7.5 KiB
C++
214 lines
7.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
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#include <cmath>
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#include "webrtc/system_wrappers/interface/logging.h"
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namespace webrtc {
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namespace {
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enum { kBweIncreaseIntervalMs = 1000 };
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enum { kBweDecreaseIntervalMs = 300 };
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enum { kLimitNumPackets = 20 };
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enum { kAvgPacketSizeBytes = 1000 };
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// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
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// The formula in RFC 3448, Section 3.1, is used.
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uint32_t CalcTfrcBps(uint16_t rtt, uint8_t loss) {
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if (rtt == 0 || loss == 0) {
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// Input variables out of range.
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return 0;
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}
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double R = static_cast<double>(rtt) / 1000; // RTT in seconds.
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int b = 1; // Number of packets acknowledged by a single TCP acknowledgement:
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// recommended = 1.
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double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds
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// recommended = 4*R.
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double p = static_cast<double>(loss) / 255; // Packet loss rate in [0, 1).
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double s = static_cast<double>(kAvgPacketSizeBytes);
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// Calculate send rate in bytes/second.
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double X =
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s / (R * std::sqrt(2 * b * p / 3) +
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(t_RTO * (3 * std::sqrt(3 * b * p / 8) * p * (1 + 32 * p * p))));
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// Convert to bits/second.
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return (static_cast<uint32_t>(X * 8));
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}
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}
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SendSideBandwidthEstimation::SendSideBandwidthEstimation()
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: accumulate_lost_packets_Q8_(0),
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accumulate_expected_packets_(0),
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bitrate_(0),
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min_bitrate_configured_(0),
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max_bitrate_configured_(0),
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time_last_receiver_block_ms_(0),
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last_fraction_loss_(0),
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last_round_trip_time_ms_(0),
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bwe_incoming_(0),
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time_last_decrease_ms_(0) {}
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SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
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void SendSideBandwidthEstimation::SetSendBitrate(uint32_t bitrate) {
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bitrate_ = bitrate;
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// Clear last sent bitrate history so the new value can be used directly
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// and not capped.
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min_bitrate_history_.clear();
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}
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void SendSideBandwidthEstimation::SetMinMaxBitrate(uint32_t min_bitrate,
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uint32_t max_bitrate) {
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min_bitrate_configured_ = min_bitrate;
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max_bitrate_configured_ = max_bitrate;
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}
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void SendSideBandwidthEstimation::SetMinBitrate(uint32_t min_bitrate) {
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min_bitrate_configured_ = min_bitrate;
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}
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void SendSideBandwidthEstimation::CurrentEstimate(uint32_t* bitrate,
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uint8_t* loss,
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uint32_t* rtt) const {
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*bitrate = bitrate_;
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*loss = last_fraction_loss_;
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*rtt = last_round_trip_time_ms_;
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}
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void SendSideBandwidthEstimation::UpdateReceiverEstimate(uint32_t bandwidth) {
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bwe_incoming_ = bandwidth;
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CapBitrateToThresholds();
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}
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void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
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uint32_t rtt,
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int number_of_packets,
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uint32_t now_ms) {
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// Update RTT.
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last_round_trip_time_ms_ = rtt;
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// Check sequence number diff and weight loss report
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if (number_of_packets > 0) {
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// Calculate number of lost packets.
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const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
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// Accumulate reports.
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accumulate_lost_packets_Q8_ += num_lost_packets_Q8;
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accumulate_expected_packets_ += number_of_packets;
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// Report loss if the total report is based on sufficiently many packets.
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if (accumulate_expected_packets_ >= kLimitNumPackets) {
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last_fraction_loss_ =
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accumulate_lost_packets_Q8_ / accumulate_expected_packets_;
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// Reset accumulators.
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accumulate_lost_packets_Q8_ = 0;
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accumulate_expected_packets_ = 0;
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} else {
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// Early return without updating estimate.
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return;
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}
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}
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time_last_receiver_block_ms_ = now_ms;
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UpdateEstimate(now_ms);
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}
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void SendSideBandwidthEstimation::UpdateEstimate(uint32_t now_ms) {
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UpdateMinHistory(now_ms);
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// Only start updating bitrate when receiving receiver blocks.
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if (time_last_receiver_block_ms_ != 0) {
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if (last_fraction_loss_ <= 5) {
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// Loss < 2%: Increase rate by 8% of the min bitrate in the last
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// kBweIncreaseIntervalMs.
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// Note that by remembering the bitrate over the last second one can
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// rampup up one second faster than if only allowed to start ramping
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// at 8% per second rate now. E.g.:
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// If sending a constant 100kbps it can rampup immediatly to 108kbps
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// whenever a receiver report is received with lower packet loss.
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// If instead one would do: bitrate_ *= 1.08^(delta time), it would
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// take over one second since the lower packet loss to achieve 108kbps.
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bitrate_ = static_cast<uint32_t>(
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min_bitrate_history_.front().second * 1.08 + 0.5);
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// Add 1 kbps extra, just to make sure that we do not get stuck
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// (gives a little extra increase at low rates, negligible at higher
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// rates).
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bitrate_ += 1000;
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} else if (last_fraction_loss_ <= 26) {
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// Loss between 2% - 10%: Do nothing.
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} else {
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// Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs +
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// rtt.
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if ((now_ms - time_last_decrease_ms_) >=
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static_cast<uint32_t>(kBweDecreaseIntervalMs +
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last_round_trip_time_ms_)) {
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time_last_decrease_ms_ = now_ms;
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// Reduce rate:
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// newRate = rate * (1 - 0.5*lossRate);
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// where packetLoss = 256*lossRate;
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bitrate_ = static_cast<uint32_t>(
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(bitrate_ * static_cast<double>(512 - last_fraction_loss_)) /
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512.0);
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// Calculate what rate TFRC would apply in this situation and to not
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// reduce further than it.
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bitrate_ = std::max(
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bitrate_,
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CalcTfrcBps(last_round_trip_time_ms_, last_fraction_loss_));
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}
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}
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}
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CapBitrateToThresholds();
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}
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void SendSideBandwidthEstimation::UpdateMinHistory(uint32_t now_ms) {
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// Remove old data points from history.
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// Since history precision is in ms, add one so it is able to increase
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// bitrate if it is off by as little as 0.5ms.
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while (!min_bitrate_history_.empty() &&
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now_ms - min_bitrate_history_.front().first + 1 >
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kBweIncreaseIntervalMs) {
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min_bitrate_history_.pop_front();
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}
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// Typical minimum sliding-window algorithm: Pop values higher than current
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// bitrate before pushing it.
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while (!min_bitrate_history_.empty() &&
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bitrate_ <= min_bitrate_history_.back().second) {
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min_bitrate_history_.pop_back();
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}
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min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
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}
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void SendSideBandwidthEstimation::CapBitrateToThresholds() {
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if (bwe_incoming_ > 0 && bitrate_ > bwe_incoming_) {
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bitrate_ = bwe_incoming_;
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}
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if (bitrate_ > max_bitrate_configured_) {
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bitrate_ = max_bitrate_configured_;
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}
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if (bitrate_ < min_bitrate_configured_) {
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LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate_ / 1000
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<< " kbps is below configured min bitrate "
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<< min_bitrate_configured_ / 1000 << " kbps.";
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bitrate_ = min_bitrate_configured_;
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}
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}
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} // namespace webrtc
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