modules/remote_bitrate_estimator/aimd_rate_control.cc - src - Git at Google

 /*
  *  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 <algorithm>
 #include <cassert>
 #include <cmath>
 #include <cstdio>
 #include <string>

 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"

 #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/logging.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;

 const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";

 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;
 }

 AimdRateControl::AimdRateControl()
     : min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
       max_configured_bitrate_bps_(30000000),
       current_bitrate_bps_(max_configured_bitrate_bps_),
       latest_incoming_bitrate_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_first_incoming_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")) {
   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_incoming_bitrate_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 time_now,
                                           uint32_t incoming_bitrate_bps) const {
   const int64_t bitrate_reduction_interval =
       std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
   if (time_now - 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 incoming_bitrate_bps < threshold;
   }
   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_incoming_estimate_ < 0) {
       if (input->incoming_bitrate)
         time_first_incoming_estimate_ = now_ms;
     } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
                input->incoming_bitrate) {
       current_bitrate_bps_ = *input->incoming_bitrate;
       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;
   current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
   time_last_bitrate_change_ = 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 incoming_bitrate_bps =
       input.incoming_bitrate.value_or(latest_incoming_bitrate_bps_);
   if (input.incoming_bitrate)
     latest_incoming_bitrate_bps_ = *input.incoming_bitrate;

   // 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 incoming_bitrate_kbps = incoming_bitrate_bps / 1000.0f;
   // Calculate the max bit rate std dev given the normalized
   // variance and the current incoming bit rate.
   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 &&
           incoming_bitrate_kbps >
               avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
         ChangeRegion(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_ * incoming_bitrate_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_);
       }
       ChangeRegion(kRcNearMax);

       if (bitrate_is_initialized_ &&
           incoming_bitrate_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 (incoming_bitrate_kbps <
           avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
         avg_max_bitrate_kbps_ = -1.0f;
       }

       bitrate_is_initialized_ = true;
       UpdateMaxBitRateEstimate(incoming_bitrate_kbps);
       // Stay on hold until the pipes are cleared.
       rate_control_state_ = kRcHold;
       time_last_bitrate_change_ = now_ms;
       break;

     default:
       assert(false);
   }
   return ClampBitrate(new_bitrate_bps, incoming_bitrate_bps);
 }

 uint32_t AimdRateControl::ClampBitrate(uint32_t new_bitrate_bps,
                                        uint32_t incoming_bitrate_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 * incoming_bitrate_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::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
   const float alpha = 0.05f;
   if (avg_max_bitrate_kbps_ == -1.0f) {
     avg_max_bitrate_kbps_ = incoming_bitrate_kbps;
   } else {
     avg_max_bitrate_kbps_ = (1 - alpha) * avg_max_bitrate_kbps_ +
         alpha * incoming_bitrate_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_ - incoming_bitrate_kbps) *
           (avg_max_bitrate_kbps_ - incoming_bitrate_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);
   }
 }

 void AimdRateControl::ChangeRegion(RateControlRegion region) {
   rate_control_region_ = region;
 }

 }  // namespace webrtc
	/*
	* 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 <algorithm>
	#include <cassert>
	#include <cmath>
	#include <cstdio>
	#include <string>

	#include "rtc_base/checks.h"
	#include "rtc_base/numerics/safe_minmax.h"

	#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/logging.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;

	const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";

	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;
	}

	AimdRateControl::AimdRateControl()
	: min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
	max_configured_bitrate_bps_(30000000),
	current_bitrate_bps_(max_configured_bitrate_bps_),
	latest_incoming_bitrate_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_first_incoming_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")) {
	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_incoming_bitrate_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 time_now,
	uint32_t incoming_bitrate_bps) const {
	const int64_t bitrate_reduction_interval =
	std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
	if (time_now - 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 incoming_bitrate_bps < threshold;
	}
	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_incoming_estimate_ < 0) {
	if (input->incoming_bitrate)
	time_first_incoming_estimate_ = now_ms;
	} else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
	input->incoming_bitrate) {
	current_bitrate_bps_ = *input->incoming_bitrate;
	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;
	current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
	time_last_bitrate_change_ = 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 incoming_bitrate_bps =
	input.incoming_bitrate.value_or(latest_incoming_bitrate_bps_);
	if (input.incoming_bitrate)
	latest_incoming_bitrate_bps_ = *input.incoming_bitrate;

	// 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 incoming_bitrate_kbps = incoming_bitrate_bps / 1000.0f;
	// Calculate the max bit rate std dev given the normalized
	// variance and the current incoming bit rate.
	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 &&
	incoming_bitrate_kbps >
	avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
	ChangeRegion(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_ * incoming_bitrate_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_);
	}
	ChangeRegion(kRcNearMax);

	if (bitrate_is_initialized_ &&
	incoming_bitrate_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 (incoming_bitrate_kbps <
	avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
	avg_max_bitrate_kbps_ = -1.0f;
	}

	bitrate_is_initialized_ = true;
	UpdateMaxBitRateEstimate(incoming_bitrate_kbps);
	// Stay on hold until the pipes are cleared.
	rate_control_state_ = kRcHold;
	time_last_bitrate_change_ = now_ms;
	break;

	default:
	assert(false);
	}
	return ClampBitrate(new_bitrate_bps, incoming_bitrate_bps);
	}

	uint32_t AimdRateControl::ClampBitrate(uint32_t new_bitrate_bps,
	uint32_t incoming_bitrate_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 * incoming_bitrate_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::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
	const float alpha = 0.05f;
	if (avg_max_bitrate_kbps_ == -1.0f) {
	avg_max_bitrate_kbps_ = incoming_bitrate_kbps;
	} else {
	avg_max_bitrate_kbps_ = (1 - alpha) * avg_max_bitrate_kbps_ +
	alpha * incoming_bitrate_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_ - incoming_bitrate_kbps) *
	(avg_max_bitrate_kbps_ - incoming_bitrate_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);
	}
	}

	void AimdRateControl::ChangeRegion(RateControlRegion region) {
	rate_control_region_ = region;
	}

	} // namespace webrtc