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/*
* Copyright (c) 2014 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/remote_bitrate_estimator/aimd_rate_control.h"
#include <inttypes.h>
#include <algorithm>
#include <cassert>
#include <cmath>
#include <cstdio>
#include <string>
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
static const int64_t kDefaultRttMs = 200;
static const int64_t kMaxFeedbackIntervalMs = 1000;
static const float kDefaultBackoffFactor = 0.85f;
static const int64_t kDefaultInitialBackOffIntervalMs = 200;
const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
const char kBweInitialBackOffIntervalExperiment[] =
"WebRTC-BweInitialBackOffInterval";
float ReadBackoffFactor() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweBackOffFactorExperiment);
float backoff_factor;
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%f", &backoff_factor);
if (parsed_values == 1) {
if (backoff_factor >= 1.0f) {
RTC_LOG(WARNING) << "Back-off factor must be less than 1.";
} else if (backoff_factor <= 0.0f) {
RTC_LOG(WARNING) << "Back-off factor must be greater than 0.";
} else {
return backoff_factor;
}
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for AimdRateControl "
"experiment from field trial string. Using default.";
return kDefaultBackoffFactor;
}
int64_t ReadInitialBackoffIntervalMs() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweInitialBackOffIntervalExperiment);
int64_t backoff_interval;
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%" SCNd64, &backoff_interval);
if (parsed_values == 1) {
if (10 <= backoff_interval && backoff_interval <= 200) {
return backoff_interval;
}
RTC_LOG(WARNING)
<< "Initial back-off interval must be between 10 and 200 ms.";
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for "
<< kBweInitialBackOffIntervalExperiment
<< " experiment. Using default.";
return kDefaultInitialBackOffIntervalMs;
}
AimdRateControl::AimdRateControl()
: min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
max_configured_bitrate_bps_(30000000),
current_bitrate_bps_(max_configured_bitrate_bps_),
latest_estimated_throughput_bps_(current_bitrate_bps_),
avg_max_bitrate_kbps_(-1.0f),
var_max_bitrate_kbps_(0.4f),
rate_control_state_(kRcHold),
rate_control_region_(kRcMaxUnknown),
time_last_bitrate_change_(-1),
time_last_bitrate_decrease_(-1),
time_first_throughput_estimate_(-1),
bitrate_is_initialized_(false),
beta_(webrtc::field_trial::IsEnabled(kBweBackOffFactorExperiment)
? ReadBackoffFactor()
: kDefaultBackoffFactor),
rtt_(kDefaultRttMs),
in_experiment_(!AdaptiveThresholdExperimentIsDisabled()),
smoothing_experiment_(
webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")),
in_initial_backoff_interval_experiment_(
webrtc::field_trial::IsEnabled(kBweInitialBackOffIntervalExperiment)),
initial_backoff_interval_ms_(kDefaultInitialBackOffIntervalMs) {
if (in_initial_backoff_interval_experiment_) {
initial_backoff_interval_ms_ = ReadInitialBackoffIntervalMs();
RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
<< " " << initial_backoff_interval_ms_ << " ms.";
}
RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
}
AimdRateControl::~AimdRateControl() {}
void AimdRateControl::SetStartBitrate(int start_bitrate_bps) {
current_bitrate_bps_ = start_bitrate_bps;
latest_estimated_throughput_bps_ = current_bitrate_bps_;
bitrate_is_initialized_ = true;
}
void AimdRateControl::SetMinBitrate(int min_bitrate_bps) {
min_configured_bitrate_bps_ = min_bitrate_bps;
current_bitrate_bps_ = std::max<int>(min_bitrate_bps, current_bitrate_bps_);
}
bool AimdRateControl::ValidEstimate() const {
return bitrate_is_initialized_;
}
int64_t AimdRateControl::GetFeedbackInterval() const {
// Estimate how often we can send RTCP if we allocate up to 5% of bandwidth
// to feedback.
static const int kRtcpSize = 80;
const int64_t interval = static_cast<int64_t>(
kRtcpSize * 8.0 * 1000.0 / (0.05 * current_bitrate_bps_) + 0.5);
const int64_t kMinFeedbackIntervalMs = 200;
return rtc::SafeClamp(interval, kMinFeedbackIntervalMs,
kMaxFeedbackIntervalMs);
}
bool AimdRateControl::TimeToReduceFurther(
int64_t now_ms,
uint32_t estimated_throughput_bps) const {
const int64_t bitrate_reduction_interval =
std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
if (now_ms - time_last_bitrate_change_ >= bitrate_reduction_interval) {
return true;
}
if (ValidEstimate()) {
// TODO(terelius/holmer): Investigate consequences of increasing
// the threshold to 0.95 * LatestEstimate().
const uint32_t threshold = static_cast<uint32_t>(0.5 * LatestEstimate());
return estimated_throughput_bps < threshold;
}
return false;
}
bool AimdRateControl::InitialTimeToReduceFurther(int64_t now_ms) const {
if (!in_initial_backoff_interval_experiment_) {
return ValidEstimate() &&
TimeToReduceFurther(now_ms, LatestEstimate() / 2 - 1);
}
// TODO(terelius): We could use the RTT (clamped to suitable limits) instead
// of a fixed bitrate_reduction_interval.
if (time_last_bitrate_decrease_ == -1 ||
now_ms - time_last_bitrate_decrease_ >= initial_backoff_interval_ms_) {
return true;
}
return false;
}
uint32_t AimdRateControl::LatestEstimate() const {
return current_bitrate_bps_;
}
void AimdRateControl::SetRtt(int64_t rtt) {
rtt_ = rtt;
}
uint32_t AimdRateControl::Update(const RateControlInput* input,
int64_t now_ms) {
RTC_CHECK(input);
// Set the initial bit rate value to what we're receiving the first half
// second.
// TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
if (!bitrate_is_initialized_) {
const int64_t kInitializationTimeMs = 5000;
RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
if (time_first_throughput_estimate_ < 0) {
if (input->estimated_throughput_bps)
time_first_throughput_estimate_ = now_ms;
} else if (now_ms - time_first_throughput_estimate_ >
kInitializationTimeMs &&
input->estimated_throughput_bps) {
current_bitrate_bps_ = *input->estimated_throughput_bps;
bitrate_is_initialized_ = true;
}
}
current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_, *input, now_ms);
return current_bitrate_bps_;
}
void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
bitrate_is_initialized_ = true;
uint32_t prev_bitrate_bps = current_bitrate_bps_;
current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
time_last_bitrate_change_ = now_ms;
if (current_bitrate_bps_ < prev_bitrate_bps) {
time_last_bitrate_decrease_ = now_ms;
}
}
int AimdRateControl::GetNearMaxIncreaseRateBps() const {
RTC_DCHECK_GT(current_bitrate_bps_, 0);
double bits_per_frame = static_cast<double>(current_bitrate_bps_) / 30.0;
double packets_per_frame = std::ceil(bits_per_frame / (8.0 * 1200.0));
double avg_packet_size_bits = bits_per_frame / packets_per_frame;
// Approximate the over-use estimator delay to 100 ms.
const int64_t response_time = in_experiment_ ? (rtt_ + 100) * 2 : rtt_ + 100;
constexpr double kMinIncreaseRateBps = 4000;
return static_cast<int>(std::max(
kMinIncreaseRateBps, (avg_packet_size_bits * 1000) / response_time));
}
int AimdRateControl::GetExpectedBandwidthPeriodMs() const {
const int kMinPeriodMs = smoothing_experiment_ ? 500 : 2000;
constexpr int kDefaultPeriodMs = 3000;
constexpr int kMaxPeriodMs = 50000;
int increase_rate = GetNearMaxIncreaseRateBps();
if (!last_decrease_)
return smoothing_experiment_ ? kMinPeriodMs : kDefaultPeriodMs;
return std::min(kMaxPeriodMs,
std::max<int>(1000 * static_cast<int64_t>(*last_decrease_) /
increase_rate,
kMinPeriodMs));
}
uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
const RateControlInput& input,
int64_t now_ms) {
uint32_t estimated_throughput_bps =
input.estimated_throughput_bps.value_or(latest_estimated_throughput_bps_);
if (input.estimated_throughput_bps)
latest_estimated_throughput_bps_ = *input.estimated_throughput_bps;
// An over-use should always trigger us to reduce the bitrate, even though
// we have not yet established our first estimate. By acting on the over-use,
// we will end up with a valid estimate.
if (!bitrate_is_initialized_ &&
input.bw_state != BandwidthUsage::kBwOverusing)
return current_bitrate_bps_;
ChangeState(input, now_ms);
// Calculated here because it's used in multiple places.
const float estimated_throughput_kbps = estimated_throughput_bps / 1000.0f;
// Calculate the max bit rate std dev given the normalized
// variance and the current throughput bitrate.
const float std_max_bit_rate =
sqrt(var_max_bitrate_kbps_ * avg_max_bitrate_kbps_);
switch (rate_control_state_) {
case kRcHold:
break;
case kRcIncrease:
if (avg_max_bitrate_kbps_ >= 0 &&
estimated_throughput_kbps >
avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
rate_control_region_ = kRcMaxUnknown;
avg_max_bitrate_kbps_ = -1.0;
}
if (rate_control_region_ == kRcNearMax) {
uint32_t additive_increase_bps =
AdditiveRateIncrease(now_ms, time_last_bitrate_change_);
new_bitrate_bps += additive_increase_bps;
} else {
uint32_t multiplicative_increase_bps = MultiplicativeRateIncrease(
now_ms, time_last_bitrate_change_, new_bitrate_bps);
new_bitrate_bps += multiplicative_increase_bps;
}
time_last_bitrate_change_ = now_ms;
break;
case kRcDecrease:
// Set bit rate to something slightly lower than max
// to get rid of any self-induced delay.
new_bitrate_bps =
static_cast<uint32_t>(beta_ * estimated_throughput_bps + 0.5);
if (new_bitrate_bps > current_bitrate_bps_) {
// Avoid increasing the rate when over-using.
if (rate_control_region_ != kRcMaxUnknown) {
new_bitrate_bps = static_cast<uint32_t>(
beta_ * avg_max_bitrate_kbps_ * 1000 + 0.5f);
}
new_bitrate_bps = std::min(new_bitrate_bps, current_bitrate_bps_);
}
rate_control_region_ = kRcNearMax;
if (bitrate_is_initialized_ &&
estimated_throughput_bps < current_bitrate_bps_) {
constexpr float kDegradationFactor = 0.9f;
if (smoothing_experiment_ &&
new_bitrate_bps <
kDegradationFactor * beta_ * current_bitrate_bps_) {
// If bitrate decreases more than a normal back off after overuse, it
// indicates a real network degradation. We do not let such a decrease
// to determine the bandwidth estimation period.
last_decrease_ = absl::nullopt;
} else {
last_decrease_ = current_bitrate_bps_ - new_bitrate_bps;
}
}
if (estimated_throughput_kbps <
avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
avg_max_bitrate_kbps_ = -1.0f;
}
bitrate_is_initialized_ = true;
UpdateMaxThroughputEstimate(estimated_throughput_kbps);
// Stay on hold until the pipes are cleared.
rate_control_state_ = kRcHold;
time_last_bitrate_change_ = now_ms;
time_last_bitrate_decrease_ = now_ms;
break;
default:
assert(false);
}
return ClampBitrate(new_bitrate_bps, estimated_throughput_bps);
}
uint32_t AimdRateControl::ClampBitrate(
uint32_t new_bitrate_bps,
uint32_t estimated_throughput_bps) const {
// Don't change the bit rate if the send side is too far off.
// We allow a bit more lag at very low rates to not too easily get stuck if
// the encoder produces uneven outputs.
const uint32_t max_bitrate_bps =
static_cast<uint32_t>(1.5f * estimated_throughput_bps) + 10000;
if (new_bitrate_bps > current_bitrate_bps_ &&
new_bitrate_bps > max_bitrate_bps) {
new_bitrate_bps = std::max(current_bitrate_bps_, max_bitrate_bps);
}
new_bitrate_bps = std::max(new_bitrate_bps, min_configured_bitrate_bps_);
return new_bitrate_bps;
}
uint32_t AimdRateControl::MultiplicativeRateIncrease(
int64_t now_ms,
int64_t last_ms,
uint32_t current_bitrate_bps) const {
double alpha = 1.08;
if (last_ms > -1) {
auto time_since_last_update_ms =
rtc::SafeMin<int64_t>(now_ms - last_ms, 1000);
alpha = pow(alpha, time_since_last_update_ms / 1000.0);
}
uint32_t multiplicative_increase_bps =
std::max(current_bitrate_bps * (alpha - 1.0), 1000.0);
return multiplicative_increase_bps;
}
uint32_t AimdRateControl::AdditiveRateIncrease(int64_t now_ms,
int64_t last_ms) const {
return static_cast<uint32_t>((now_ms - last_ms) *
GetNearMaxIncreaseRateBps() / 1000);
}
void AimdRateControl::UpdateMaxThroughputEstimate(
float estimated_throughput_kbps) {
const float alpha = 0.05f;
if (avg_max_bitrate_kbps_ == -1.0f) {
avg_max_bitrate_kbps_ = estimated_throughput_kbps;
} else {
avg_max_bitrate_kbps_ =
(1 - alpha) * avg_max_bitrate_kbps_ + alpha * estimated_throughput_kbps;
}
// Estimate the max bit rate variance and normalize the variance
// with the average max bit rate.
const float norm = std::max(avg_max_bitrate_kbps_, 1.0f);
var_max_bitrate_kbps_ =
(1 - alpha) * var_max_bitrate_kbps_ +
alpha * (avg_max_bitrate_kbps_ - estimated_throughput_kbps) *
(avg_max_bitrate_kbps_ - estimated_throughput_kbps) / norm;
// 0.4 ~= 14 kbit/s at 500 kbit/s
if (var_max_bitrate_kbps_ < 0.4f) {
var_max_bitrate_kbps_ = 0.4f;
}
// 2.5f ~= 35 kbit/s at 500 kbit/s
if (var_max_bitrate_kbps_ > 2.5f) {
var_max_bitrate_kbps_ = 2.5f;
}
}
void AimdRateControl::ChangeState(const RateControlInput& input,
int64_t now_ms) {
switch (input.bw_state) {
case BandwidthUsage::kBwNormal:
if (rate_control_state_ == kRcHold) {
time_last_bitrate_change_ = now_ms;
rate_control_state_ = kRcIncrease;
}
break;
case BandwidthUsage::kBwOverusing:
if (rate_control_state_ != kRcDecrease) {
rate_control_state_ = kRcDecrease;
}
break;
case BandwidthUsage::kBwUnderusing:
rate_control_state_ = kRcHold;
break;
default:
assert(false);
}
}
} // namespace webrtc