test/network/simulated_network.cc - src.git - Git at Google

 /*
  *  Copyright 2018 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 "test/network/simulated_network.h"

 #include <algorithm>
 #include <cmath>
 #include <cstdint>
 #include <utility>

 #include "absl/types/optional.h"
 #include "api/test/simulated_network.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/checks.h"

 namespace webrtc {
 namespace {

 // Calculate the time that it takes to send N `bits` on a
 // network with link capacity equal to `capacity_kbps` starting at time
 // `start_time`.
 Timestamp CalculateArrivalTime(Timestamp start_time,
                                int64_t bits,
                                DataRate capacity) {
   if (capacity.IsInfinite()) {
     return start_time;
   }
   if (capacity.IsZero()) {
     return Timestamp::PlusInfinity();
   }

   // Adding `capacity - 1` to the numerator rounds the extra delay caused by
   // capacity constraints up to an integral microsecond. Sending 0 bits takes 0
   // extra time, while sending 1 bit gets rounded up to 1 (the multiplication by
   // 1000 is because capacity is in kbps).
   // The factor 1000 comes from 10^6 / 10^3, where 10^6 is due to the time unit
   // being us and 10^3 is due to the rate unit being kbps.
   return start_time + TimeDelta::Micros((1000 * bits + capacity.kbps() - 1) /
                                         capacity.kbps());
 }

 void UpdateLegacyConfiguration(SimulatedNetwork::Config& config) {
   if (config.link_capacity_kbps != 0) {
     RTC_DCHECK(config.link_capacity ==
                    DataRate::KilobitsPerSec(config.link_capacity_kbps) ||
                config.link_capacity == DataRate::Infinity());
     config.link_capacity = DataRate::KilobitsPerSec(config.link_capacity_kbps);
   }
 }

 }  // namespace

 SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)
     : random_(random_seed), bursting_(false), last_enqueue_time_us_(0) {
   SetConfig(config);
 }

 SimulatedNetwork::~SimulatedNetwork() = default;

 void SimulatedNetwork::SetConfig(const Config& config) {
   MutexLock lock(&config_lock_);
   config_state_.config = config;  // Shallow copy of the struct.
   UpdateLegacyConfiguration(config_state_.config);

   double prob_loss = config.loss_percent / 100.0;
   if (config_state_.config.avg_burst_loss_length == -1) {
     // Uniform loss
     config_state_.prob_loss_bursting = prob_loss;
     config_state_.prob_start_bursting = prob_loss;
   } else {
     // Lose packets according to a gilbert-elliot model.
     int avg_burst_loss_length = config.avg_burst_loss_length;
     int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));

     RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
         << "For a total packet loss of " << config.loss_percent
         << "%% then"
            " avg_burst_loss_length must be "
         << min_avg_burst_loss_length + 1 << " or higher.";

     config_state_.prob_loss_bursting = (1.0 - 1.0 / avg_burst_loss_length);
     config_state_.prob_start_bursting =
         prob_loss / (1 - prob_loss) / avg_burst_loss_length;
   }
 }

 void SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& new_config,
                                  Timestamp config_update_time) {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);

   if (!capacity_link_.empty()) {
     // Calculate and update how large portion of the packet first in the
     // capacity link is left to to send at time `config_update_time`.
     const BuiltInNetworkBehaviorConfig& current_config =
         GetConfigState().config;
     TimeDelta duration_with_current_config =
         config_update_time - capacity_link_.front().last_update_time;
     RTC_DCHECK_GE(duration_with_current_config, TimeDelta::Zero());
     capacity_link_.front().bits_left_to_send -= std::min(
         duration_with_current_config.ms() * current_config.link_capacity.kbps(),
         capacity_link_.front().bits_left_to_send);
     capacity_link_.front().last_update_time = config_update_time;
   }
   SetConfig(new_config);
   UpdateCapacityQueue(GetConfigState(), config_update_time);
   if (UpdateNextProcessTime() && next_process_time_changed_callback_) {
     next_process_time_changed_callback_();
   }
 }

 void SimulatedNetwork::UpdateConfig(
     std::function<void(BuiltInNetworkBehaviorConfig*)> config_modifier) {
   MutexLock lock(&config_lock_);
   config_modifier(&config_state_.config);
   UpdateLegacyConfiguration(config_state_.config);
 }

 void SimulatedNetwork::PauseTransmissionUntil(int64_t until_us) {
   MutexLock lock(&config_lock_);
   config_state_.pause_transmission_until_us = until_us;
 }

 bool SimulatedNetwork::EnqueuePacket(PacketInFlightInfo packet) {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);

   // Check that old packets don't get enqueued, the SimulatedNetwork expect that
   // the packets' send time is monotonically increasing. The tolerance for
   // non-monotonic enqueue events is 0.5 ms because on multi core systems
   // clock_gettime(CLOCK_MONOTONIC) can show non-monotonic behaviour between
   // theads running on different cores.
   // TODO(bugs.webrtc.org/14525): Open a bug on this with the goal to re-enable
   // the DCHECK.
   // At the moment, we see more than 130ms between non-monotonic events, which
   // is more than expected.
   // RTC_DCHECK_GE(packet.send_time_us - last_enqueue_time_us_, -2000);

   ConfigState state = GetConfigState();

   // If the network config requires packet overhead, let's apply it as early as
   // possible.
   packet.size += state.config.packet_overhead;

   // If `queue_length_packets` is 0, the queue size is infinite.
   if (state.config.queue_length_packets > 0 &&
       capacity_link_.size() >= state.config.queue_length_packets) {
     // Too many packet on the link, drop this one.
     return false;
   }

   // Note that arrival time will be updated when previous packets are dequeued
   // from the capacity link.
   // A packet can not enter the narrow section before the last packet has exit.
   Timestamp enqueue_time = Timestamp::Micros(packet.send_time_us);
   Timestamp arrival_time =
       capacity_link_.empty()
           ? CalculateArrivalTime(
                 std::max(enqueue_time, last_capacity_link_exit_time_),
                 packet.size * 8, state.config.link_capacity)
           : Timestamp::PlusInfinity();
   capacity_link_.push(
       {.packet = packet,
        .last_update_time = enqueue_time,
        .bits_left_to_send = 8 * static_cast<int64_t>(packet.size),
        .arrival_time = arrival_time});

   // Only update `next_process_time_` if not already set. Otherwise,
   // next_process_time_ is calculated when a packet is dequeued. Note that this
   // means that the newly enqueued packet risk having an arrival time before
   // `next_process_time_` if packet reordering is allowed and
   // config.delay_standard_deviation_ms is set.
   // TODO(bugs.webrtc.org/14525): Consider preventing this.
   if (next_process_time_.IsInfinite() && arrival_time.IsFinite()) {
     RTC_DCHECK_EQ(capacity_link_.size(), 1);
     next_process_time_ = arrival_time;
   }

   last_enqueue_time_us_ = packet.send_time_us;
   return true;
 }

 absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
   if (next_process_time_.IsFinite()) {
     return next_process_time_.us();
   }
   return absl::nullopt;
 }

 void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
                                            Timestamp time_now) {
   // Only the first packet in capacity_link_ have a calculated arrival time
   // (when packet leave the narrow section), and time when it entered the narrow
   // section. Also, the configuration may have changed. Thus we need to
   // calculate the arrival time again before maybe moving the packet to the
   // delay link.
   if (!capacity_link_.empty()) {
     capacity_link_.front().last_update_time = std::max(
         capacity_link_.front().last_update_time, last_capacity_link_exit_time_);
     capacity_link_.front().arrival_time = CalculateArrivalTime(
         capacity_link_.front().last_update_time,
         capacity_link_.front().bits_left_to_send, state.config.link_capacity);
   }

   // The capacity link is empty or the first packet is not expected to exit yet.
   if (capacity_link_.empty() ||
       time_now < capacity_link_.front().arrival_time) {
     return;
   }
   bool reorder_packets = false;

   do {
     // Time to get this packet (the original or just updated arrival_time is
     // smaller or equal to time_now_us).
     PacketInfo packet = capacity_link_.front();
     RTC_DCHECK(packet.arrival_time.IsFinite());
     capacity_link_.pop();

     // If the network is paused, the pause will be implemented as an extra delay
     // to be spent in the `delay_link_` queue.
     if (state.pause_transmission_until_us > packet.arrival_time.us()) {
       packet.arrival_time =
           Timestamp::Micros(state.pause_transmission_until_us);
     }

     // Store the original arrival time, before applying packet loss or extra
     // delay. This is needed to know when it is the first available time the
     // next packet in the `capacity_link_` queue can start transmitting.
     last_capacity_link_exit_time_ = packet.arrival_time;

     // Drop packets at an average rate of `state.config.loss_percent` with
     // and average loss burst length of `state.config.avg_burst_loss_length`.
     if ((bursting_ && random_.Rand<double>() < state.prob_loss_bursting) ||
         (!bursting_ && random_.Rand<double>() < state.prob_start_bursting)) {
       bursting_ = true;
       packet.arrival_time = Timestamp::MinusInfinity();
     } else {
       // If packets are not dropped, apply extra delay as configured.
       bursting_ = false;
       TimeDelta arrival_time_jitter = TimeDelta::Micros(std::max(
           random_.Gaussian(state.config.queue_delay_ms * 1000,
                            state.config.delay_standard_deviation_ms * 1000),
           0.0));

       // If reordering is not allowed then adjust arrival_time_jitter
       // to make sure all packets are sent in order.
       Timestamp last_arrival_time = delay_link_.empty()
                                         ? Timestamp::MinusInfinity()
                                         : delay_link_.back().arrival_time;
       if (!state.config.allow_reordering && !delay_link_.empty() &&
           packet.arrival_time + arrival_time_jitter < last_arrival_time) {
         arrival_time_jitter = last_arrival_time - packet.arrival_time;
       }
       packet.arrival_time += arrival_time_jitter;

       // Optimization: Schedule a reorder only when a packet will exit before
       // the one in front.
       if (last_arrival_time > packet.arrival_time) {
         reorder_packets = true;
       }
     }
     delay_link_.emplace_back(packet);

     // If there are no packets in the queue, there is nothing else to do.
     if (capacity_link_.empty()) {
       break;
     }
     // If instead there is another packet in the `capacity_link_` queue, let's
     // calculate its arrival_time based on the latest config (which might
     // have been changed since it was enqueued).
     Timestamp next_start = std::max(last_capacity_link_exit_time_,
                                     capacity_link_.front().last_update_time);
     capacity_link_.front().arrival_time =
         CalculateArrivalTime(next_start, capacity_link_.front().packet.size * 8,
                              state.config.link_capacity);
     // And if the next packet in the queue needs to exit, let's dequeue it.
   } while (capacity_link_.front().arrival_time <= time_now);

   if (state.config.allow_reordering && reorder_packets) {
     // Packets arrived out of order and since the network config allows
     // reordering, let's sort them per arrival_time to make so they will also
     // be delivered out of order.
     std::stable_sort(delay_link_.begin(), delay_link_.end(),
                      [](const PacketInfo& p1, const PacketInfo& p2) {
                        return p1.arrival_time < p2.arrival_time;
                      });
   }
 }

 SimulatedNetwork::ConfigState SimulatedNetwork::GetConfigState() const {
   MutexLock lock(&config_lock_);
   return config_state_;
 }

 std::vector<PacketDeliveryInfo> SimulatedNetwork::DequeueDeliverablePackets(
     int64_t receive_time_us) {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
   Timestamp receive_time = Timestamp::Micros(receive_time_us);

   UpdateCapacityQueue(GetConfigState(), receive_time);
   std::vector<PacketDeliveryInfo> packets_to_deliver;

   // Check the extra delay queue.
   while (!delay_link_.empty() &&
          receive_time >= delay_link_.front().arrival_time) {
     PacketInfo packet_info = delay_link_.front();
     packets_to_deliver.emplace_back(PacketDeliveryInfo(
         packet_info.packet, packet_info.arrival_time.IsFinite()
                                 ? packet_info.arrival_time.us()
                                 : PacketDeliveryInfo::kNotReceived));
     delay_link_.pop_front();
   }
   // There is no need to invoke `next_process_time_changed_callback_` here since
   // it is expected that the user of NetworkBehaviorInterface calls
   // NextDeliveryTimeUs after DequeueDeliverablePackets. See
   // NetworkBehaviorInterface.
   UpdateNextProcessTime();
   return packets_to_deliver;
 }

 bool SimulatedNetwork::UpdateNextProcessTime() {
   Timestamp next_process_time = next_process_time_;

   next_process_time_ = Timestamp::PlusInfinity();
   for (const PacketInfo& packet : delay_link_) {
     if (packet.arrival_time.IsFinite()) {
       next_process_time_ = packet.arrival_time;
       break;
     }
   }
   if (next_process_time_.IsInfinite() && !capacity_link_.empty()) {
     next_process_time_ = capacity_link_.front().arrival_time;
   }
   return next_process_time != next_process_time_;
 }

 void SimulatedNetwork::RegisterDeliveryTimeChangedCallback(
     absl::AnyInvocable<void()> callback) {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
   next_process_time_changed_callback_ = std::move(callback);
 }

 }  // namespace webrtc
	/*
	* Copyright 2018 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 "test/network/simulated_network.h"

	#include <algorithm>
	#include <cmath>
	#include <cstdint>
	#include <utility>

	#include "absl/types/optional.h"
	#include "api/test/simulated_network.h"
	#include "api/units/data_rate.h"
	#include "api/units/data_size.h"
	#include "api/units/time_delta.h"
	#include "api/units/timestamp.h"
	#include "rtc_base/checks.h"

	namespace webrtc {
	namespace {

	// Calculate the time that it takes to send N `bits` on a
	// network with link capacity equal to `capacity_kbps` starting at time
	// `start_time`.
	Timestamp CalculateArrivalTime(Timestamp start_time,
	int64_t bits,
	DataRate capacity) {
	if (capacity.IsInfinite()) {
	return start_time;
	}
	if (capacity.IsZero()) {
	return Timestamp::PlusInfinity();
	}

	// Adding `capacity - 1` to the numerator rounds the extra delay caused by
	// capacity constraints up to an integral microsecond. Sending 0 bits takes 0
	// extra time, while sending 1 bit gets rounded up to 1 (the multiplication by
	// 1000 is because capacity is in kbps).
	// The factor 1000 comes from 10^6 / 10^3, where 10^6 is due to the time unit
	// being us and 10^3 is due to the rate unit being kbps.
	return start_time + TimeDelta::Micros((1000 * bits + capacity.kbps() - 1) /
	capacity.kbps());
	}

	void UpdateLegacyConfiguration(SimulatedNetwork::Config& config) {
	if (config.link_capacity_kbps != 0) {
	RTC_DCHECK(config.link_capacity ==
	DataRate::KilobitsPerSec(config.link_capacity_kbps) \|\|
	config.link_capacity == DataRate::Infinity());
	config.link_capacity = DataRate::KilobitsPerSec(config.link_capacity_kbps);
	}
	}

	} // namespace

	SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)
	: random_(random_seed), bursting_(false), last_enqueue_time_us_(0) {
	SetConfig(config);
	}

	SimulatedNetwork::~SimulatedNetwork() = default;

	void SimulatedNetwork::SetConfig(const Config& config) {
	MutexLock lock(&config_lock_);
	config_state_.config = config; // Shallow copy of the struct.
	UpdateLegacyConfiguration(config_state_.config);

	double prob_loss = config.loss_percent / 100.0;
	if (config_state_.config.avg_burst_loss_length == -1) {
	// Uniform loss
	config_state_.prob_loss_bursting = prob_loss;
	config_state_.prob_start_bursting = prob_loss;
	} else {
	// Lose packets according to a gilbert-elliot model.
	int avg_burst_loss_length = config.avg_burst_loss_length;
	int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));

	RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
	<< "For a total packet loss of " << config.loss_percent
	<< "%% then"
	" avg_burst_loss_length must be "
	<< min_avg_burst_loss_length + 1 << " or higher.";

	config_state_.prob_loss_bursting = (1.0 - 1.0 / avg_burst_loss_length);
	config_state_.prob_start_bursting =
	prob_loss / (1 - prob_loss) / avg_burst_loss_length;
	}
	}

	void SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& new_config,
	Timestamp config_update_time) {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);

	if (!capacity_link_.empty()) {
	// Calculate and update how large portion of the packet first in the
	// capacity link is left to to send at time `config_update_time`.
	const BuiltInNetworkBehaviorConfig& current_config =
	GetConfigState().config;
	TimeDelta duration_with_current_config =
	config_update_time - capacity_link_.front().last_update_time;
	RTC_DCHECK_GE(duration_with_current_config, TimeDelta::Zero());
	capacity_link_.front().bits_left_to_send -= std::min(
	duration_with_current_config.ms() * current_config.link_capacity.kbps(),
	capacity_link_.front().bits_left_to_send);
	capacity_link_.front().last_update_time = config_update_time;
	}
	SetConfig(new_config);
	UpdateCapacityQueue(GetConfigState(), config_update_time);
	if (UpdateNextProcessTime() && next_process_time_changed_callback_) {
	next_process_time_changed_callback_();
	}
	}

	void SimulatedNetwork::UpdateConfig(
	std::function<void(BuiltInNetworkBehaviorConfig*)> config_modifier) {
	MutexLock lock(&config_lock_);
	config_modifier(&config_state_.config);
	UpdateLegacyConfiguration(config_state_.config);
	}

	void SimulatedNetwork::PauseTransmissionUntil(int64_t until_us) {
	MutexLock lock(&config_lock_);
	config_state_.pause_transmission_until_us = until_us;
	}

	bool SimulatedNetwork::EnqueuePacket(PacketInFlightInfo packet) {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);

	// Check that old packets don't get enqueued, the SimulatedNetwork expect that
	// the packets' send time is monotonically increasing. The tolerance for
	// non-monotonic enqueue events is 0.5 ms because on multi core systems
	// clock_gettime(CLOCK_MONOTONIC) can show non-monotonic behaviour between
	// theads running on different cores.
	// TODO(bugs.webrtc.org/14525): Open a bug on this with the goal to re-enable
	// the DCHECK.
	// At the moment, we see more than 130ms between non-monotonic events, which
	// is more than expected.
	// RTC_DCHECK_GE(packet.send_time_us - last_enqueue_time_us_, -2000);

	ConfigState state = GetConfigState();

	// If the network config requires packet overhead, let's apply it as early as
	// possible.
	packet.size += state.config.packet_overhead;

	// If `queue_length_packets` is 0, the queue size is infinite.
	if (state.config.queue_length_packets > 0 &&
	capacity_link_.size() >= state.config.queue_length_packets) {
	// Too many packet on the link, drop this one.
	return false;
	}

	// Note that arrival time will be updated when previous packets are dequeued
	// from the capacity link.
	// A packet can not enter the narrow section before the last packet has exit.
	Timestamp enqueue_time = Timestamp::Micros(packet.send_time_us);
	Timestamp arrival_time =
	capacity_link_.empty()
	? CalculateArrivalTime(
	std::max(enqueue_time, last_capacity_link_exit_time_),
	packet.size * 8, state.config.link_capacity)
	: Timestamp::PlusInfinity();
	capacity_link_.push(
	{.packet = packet,
	.last_update_time = enqueue_time,
	.bits_left_to_send = 8 * static_cast<int64_t>(packet.size),
	.arrival_time = arrival_time});

	// Only update `next_process_time_` if not already set. Otherwise,
	// next_process_time_ is calculated when a packet is dequeued. Note that this
	// means that the newly enqueued packet risk having an arrival time before
	// `next_process_time_` if packet reordering is allowed and
	// config.delay_standard_deviation_ms is set.
	// TODO(bugs.webrtc.org/14525): Consider preventing this.
	if (next_process_time_.IsInfinite() && arrival_time.IsFinite()) {
	RTC_DCHECK_EQ(capacity_link_.size(), 1);
	next_process_time_ = arrival_time;
	}

	last_enqueue_time_us_ = packet.send_time_us;
	return true;
	}

	absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
	if (next_process_time_.IsFinite()) {
	return next_process_time_.us();
	}
	return absl::nullopt;
	}

	void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
	Timestamp time_now) {
	// Only the first packet in capacity_link_ have a calculated arrival time
	// (when packet leave the narrow section), and time when it entered the narrow
	// section. Also, the configuration may have changed. Thus we need to
	// calculate the arrival time again before maybe moving the packet to the
	// delay link.
	if (!capacity_link_.empty()) {
	capacity_link_.front().last_update_time = std::max(
	capacity_link_.front().last_update_time, last_capacity_link_exit_time_);
	capacity_link_.front().arrival_time = CalculateArrivalTime(
	capacity_link_.front().last_update_time,
	capacity_link_.front().bits_left_to_send, state.config.link_capacity);
	}

	// The capacity link is empty or the first packet is not expected to exit yet.
	if (capacity_link_.empty() \|\|
	time_now < capacity_link_.front().arrival_time) {
	return;
	}
	bool reorder_packets = false;

	do {
	// Time to get this packet (the original or just updated arrival_time is
	// smaller or equal to time_now_us).
	PacketInfo packet = capacity_link_.front();
	RTC_DCHECK(packet.arrival_time.IsFinite());
	capacity_link_.pop();

	// If the network is paused, the pause will be implemented as an extra delay
	// to be spent in the `delay_link_` queue.
	if (state.pause_transmission_until_us > packet.arrival_time.us()) {
	packet.arrival_time =
	Timestamp::Micros(state.pause_transmission_until_us);
	}

	// Store the original arrival time, before applying packet loss or extra
	// delay. This is needed to know when it is the first available time the
	// next packet in the `capacity_link_` queue can start transmitting.
	last_capacity_link_exit_time_ = packet.arrival_time;

	// Drop packets at an average rate of `state.config.loss_percent` with
	// and average loss burst length of `state.config.avg_burst_loss_length`.
	if ((bursting_ && random_.Rand<double>() < state.prob_loss_bursting) \|\|
	(!bursting_ && random_.Rand<double>() < state.prob_start_bursting)) {
	bursting_ = true;
	packet.arrival_time = Timestamp::MinusInfinity();
	} else {
	// If packets are not dropped, apply extra delay as configured.
	bursting_ = false;
	TimeDelta arrival_time_jitter = TimeDelta::Micros(std::max(
	random_.Gaussian(state.config.queue_delay_ms * 1000,
	state.config.delay_standard_deviation_ms * 1000),
	0.0));

	// If reordering is not allowed then adjust arrival_time_jitter
	// to make sure all packets are sent in order.
	Timestamp last_arrival_time = delay_link_.empty()
	? Timestamp::MinusInfinity()
	: delay_link_.back().arrival_time;
	if (!state.config.allow_reordering && !delay_link_.empty() &&
	packet.arrival_time + arrival_time_jitter < last_arrival_time) {
	arrival_time_jitter = last_arrival_time - packet.arrival_time;
	}
	packet.arrival_time += arrival_time_jitter;

	// Optimization: Schedule a reorder only when a packet will exit before
	// the one in front.
	if (last_arrival_time > packet.arrival_time) {
	reorder_packets = true;
	}
	}
	delay_link_.emplace_back(packet);

	// If there are no packets in the queue, there is nothing else to do.
	if (capacity_link_.empty()) {
	break;
	}
	// If instead there is another packet in the `capacity_link_` queue, let's
	// calculate its arrival_time based on the latest config (which might
	// have been changed since it was enqueued).
	Timestamp next_start = std::max(last_capacity_link_exit_time_,
	capacity_link_.front().last_update_time);
	capacity_link_.front().arrival_time =
	CalculateArrivalTime(next_start, capacity_link_.front().packet.size * 8,
	state.config.link_capacity);
	// And if the next packet in the queue needs to exit, let's dequeue it.
	} while (capacity_link_.front().arrival_time <= time_now);

	if (state.config.allow_reordering && reorder_packets) {
	// Packets arrived out of order and since the network config allows
	// reordering, let's sort them per arrival_time to make so they will also
	// be delivered out of order.
	std::stable_sort(delay_link_.begin(), delay_link_.end(),
	[](const PacketInfo& p1, const PacketInfo& p2) {
	return p1.arrival_time < p2.arrival_time;
	});
	}
	}

	SimulatedNetwork::ConfigState SimulatedNetwork::GetConfigState() const {
	MutexLock lock(&config_lock_);
	return config_state_;
	}

	std::vector<PacketDeliveryInfo> SimulatedNetwork::DequeueDeliverablePackets(
	int64_t receive_time_us) {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
	Timestamp receive_time = Timestamp::Micros(receive_time_us);

	UpdateCapacityQueue(GetConfigState(), receive_time);
	std::vector<PacketDeliveryInfo> packets_to_deliver;

	// Check the extra delay queue.
	while (!delay_link_.empty() &&
	receive_time >= delay_link_.front().arrival_time) {
	PacketInfo packet_info = delay_link_.front();
	packets_to_deliver.emplace_back(PacketDeliveryInfo(
	packet_info.packet, packet_info.arrival_time.IsFinite()
	? packet_info.arrival_time.us()
	: PacketDeliveryInfo::kNotReceived));
	delay_link_.pop_front();
	}
	// There is no need to invoke `next_process_time_changed_callback_` here since
	// it is expected that the user of NetworkBehaviorInterface calls
	// NextDeliveryTimeUs after DequeueDeliverablePackets. See
	// NetworkBehaviorInterface.
	UpdateNextProcessTime();
	return packets_to_deliver;
	}

	bool SimulatedNetwork::UpdateNextProcessTime() {
	Timestamp next_process_time = next_process_time_;

	next_process_time_ = Timestamp::PlusInfinity();
	for (const PacketInfo& packet : delay_link_) {
	if (packet.arrival_time.IsFinite()) {
	next_process_time_ = packet.arrival_time;
	break;
	}
	}
	if (next_process_time_.IsInfinite() && !capacity_link_.empty()) {
	next_process_time_ = capacity_link_.front().arrival_time;
	}
	return next_process_time != next_process_time_;
	}

	void SimulatedNetwork::RegisterDeliveryTimeChangedCallback(
	absl::AnyInvocable<void()> callback) {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
	next_process_time_changed_callback_ = std::move(callback);
	}

	} // namespace webrtc