call/simulated_network.cc - src/ - 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 "call/simulated_network.h"

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

 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/checks.h"

 namespace webrtc {
 namespace {

 // Calculate the time (in microseconds) that takes to send N `bits` on a
 // network with link capacity equal to `capacity_kbps` starting at time
 // `start_time_us`.
 int64_t CalculateArrivalTimeUs(int64_t start_time_us,
                                int64_t bits,
                                int capacity_kbps) {
   // If capacity is 0, the link capacity is assumed to be infinite.
   if (capacity_kbps == 0) {
     return start_time_us;
   }
   // 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_us + ((1000 * bits + capacity_kbps - 1) / capacity_kbps);
 }

 }  // namespace

 SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)
     : random_(random_seed),
       bursting_(false),
       last_enqueue_time_us_(0),
       last_capacity_link_exit_time_(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.
   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::UpdateConfig(
     std::function<void(BuiltInNetworkBehaviorConfig*)> config_modifier) {
   MutexLock lock(&config_lock_);
   config_modifier(&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;
   }

   // If the packet has been sent before the previous packet in the network left
   // the capacity queue, let's ensure the new packet will start its trip in the
   // network after the last bit of the previous packet has left it.
   int64_t packet_send_time_us = packet.send_time_us;
   if (!capacity_link_.empty()) {
     packet_send_time_us =
         std::max(packet_send_time_us, capacity_link_.back().arrival_time_us);
   }
   capacity_link_.push({.packet = packet,
                        .arrival_time_us = CalculateArrivalTimeUs(
                            packet_send_time_us, packet.size * 8,
                            state.config.link_capacity_kbps)});

   // Only update `next_process_time_us_` if not already set (if set, there is no
   // way that a new packet will make the `next_process_time_us_` change).
   if (!next_process_time_us_) {
     RTC_DCHECK_EQ(capacity_link_.size(), 1);
     next_process_time_us_ = capacity_link_.front().arrival_time_us;
   }

   last_enqueue_time_us_ = packet.send_time_us;
   return true;
 }

 absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
   RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
   return next_process_time_us_;
 }

 void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
                                            int64_t time_now_us) {
   // If there is at least one packet in the `capacity_link_`, let's update its
   // arrival time to take into account changes in the network configuration
   // since the last call to UpdateCapacityQueue.
   if (!capacity_link_.empty()) {
     capacity_link_.front().arrival_time_us = CalculateArrivalTimeUs(
         std::max(capacity_link_.front().packet.send_time_us,
                  last_capacity_link_exit_time_),
         capacity_link_.front().packet.size * 8,
         state.config.link_capacity_kbps);
   }

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

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

     // 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_us = PacketDeliveryInfo::kNotReceived;
     } else {
       // If packets are not dropped, apply extra delay as configured.
       bursting_ = false;
       int64_t arrival_time_jitter_us = 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.
       int64_t last_arrival_time_us =
           delay_link_.empty() ? -1 : delay_link_.back().arrival_time_us;
       if (!state.config.allow_reordering && !delay_link_.empty() &&
           packet.arrival_time_us + arrival_time_jitter_us <
               last_arrival_time_us) {
         arrival_time_jitter_us = last_arrival_time_us - packet.arrival_time_us;
       }
       packet.arrival_time_us += arrival_time_jitter_us;

       // Optimization: Schedule a reorder only when a packet will exit before
       // the one in front.
       if (last_arrival_time_us > packet.arrival_time_us) {
         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_us based on the latest config (which might
     // have been changed since it was enqueued).
     int64_t next_start = std::max(last_capacity_link_exit_time_,
                                   capacity_link_.front().packet.send_time_us);
     capacity_link_.front().arrival_time_us = CalculateArrivalTimeUs(
         next_start, capacity_link_.front().packet.size * 8,
         state.config.link_capacity_kbps);
     // And if the next packet in the queue needs to exit, let's dequeue it.
   } while (capacity_link_.front().arrival_time_us <= time_now_us);

   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_us 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_us < p2.arrival_time_us;
                      });
   }
 }

 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_);

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

   // Check the extra delay queue.
   while (!delay_link_.empty() &&
          receive_time_us >= delay_link_.front().arrival_time_us) {
     PacketInfo packet_info = delay_link_.front();
     packets_to_deliver.emplace_back(
         PacketDeliveryInfo(packet_info.packet, packet_info.arrival_time_us));
     delay_link_.pop_front();
   }

   if (!delay_link_.empty()) {
     next_process_time_us_ = delay_link_.front().arrival_time_us;
   } else if (!capacity_link_.empty()) {
     next_process_time_us_ = capacity_link_.front().arrival_time_us;
   } else {
     next_process_time_us_.reset();
   }
   return packets_to_deliver;
 }

 }  // 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 "call/simulated_network.h"

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

	#include "api/units/data_rate.h"
	#include "api/units/data_size.h"
	#include "api/units/time_delta.h"
	#include "rtc_base/checks.h"

	namespace webrtc {
	namespace {

	// Calculate the time (in microseconds) that takes to send N `bits` on a
	// network with link capacity equal to `capacity_kbps` starting at time
	// `start_time_us`.
	int64_t CalculateArrivalTimeUs(int64_t start_time_us,
	int64_t bits,
	int capacity_kbps) {
	// If capacity is 0, the link capacity is assumed to be infinite.
	if (capacity_kbps == 0) {
	return start_time_us;
	}
	// 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_us + ((1000 * bits + capacity_kbps - 1) / capacity_kbps);
	}

	} // namespace

	SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)
	: random_(random_seed),
	bursting_(false),
	last_enqueue_time_us_(0),
	last_capacity_link_exit_time_(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.
	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::UpdateConfig(
	std::function<void(BuiltInNetworkBehaviorConfig*)> config_modifier) {
	MutexLock lock(&config_lock_);
	config_modifier(&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;
	}

	// If the packet has been sent before the previous packet in the network left
	// the capacity queue, let's ensure the new packet will start its trip in the
	// network after the last bit of the previous packet has left it.
	int64_t packet_send_time_us = packet.send_time_us;
	if (!capacity_link_.empty()) {
	packet_send_time_us =
	std::max(packet_send_time_us, capacity_link_.back().arrival_time_us);
	}
	capacity_link_.push({.packet = packet,
	.arrival_time_us = CalculateArrivalTimeUs(
	packet_send_time_us, packet.size * 8,
	state.config.link_capacity_kbps)});

	// Only update `next_process_time_us_` if not already set (if set, there is no
	// way that a new packet will make the `next_process_time_us_` change).
	if (!next_process_time_us_) {
	RTC_DCHECK_EQ(capacity_link_.size(), 1);
	next_process_time_us_ = capacity_link_.front().arrival_time_us;
	}

	last_enqueue_time_us_ = packet.send_time_us;
	return true;
	}

	absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
	RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
	return next_process_time_us_;
	}

	void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
	int64_t time_now_us) {
	// If there is at least one packet in the `capacity_link_`, let's update its
	// arrival time to take into account changes in the network configuration
	// since the last call to UpdateCapacityQueue.
	if (!capacity_link_.empty()) {
	capacity_link_.front().arrival_time_us = CalculateArrivalTimeUs(
	std::max(capacity_link_.front().packet.send_time_us,
	last_capacity_link_exit_time_),
	capacity_link_.front().packet.size * 8,
	state.config.link_capacity_kbps);
	}

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

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

	// 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_us = PacketDeliveryInfo::kNotReceived;
	} else {
	// If packets are not dropped, apply extra delay as configured.
	bursting_ = false;
	int64_t arrival_time_jitter_us = 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.
	int64_t last_arrival_time_us =
	delay_link_.empty() ? -1 : delay_link_.back().arrival_time_us;
	if (!state.config.allow_reordering && !delay_link_.empty() &&
	packet.arrival_time_us + arrival_time_jitter_us <
	last_arrival_time_us) {
	arrival_time_jitter_us = last_arrival_time_us - packet.arrival_time_us;
	}
	packet.arrival_time_us += arrival_time_jitter_us;

	// Optimization: Schedule a reorder only when a packet will exit before
	// the one in front.
	if (last_arrival_time_us > packet.arrival_time_us) {
	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_us based on the latest config (which might
	// have been changed since it was enqueued).
	int64_t next_start = std::max(last_capacity_link_exit_time_,
	capacity_link_.front().packet.send_time_us);
	capacity_link_.front().arrival_time_us = CalculateArrivalTimeUs(
	next_start, capacity_link_.front().packet.size * 8,
	state.config.link_capacity_kbps);
	// And if the next packet in the queue needs to exit, let's dequeue it.
	} while (capacity_link_.front().arrival_time_us <= time_now_us);

	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_us 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_us < p2.arrival_time_us;
	});
	}
	}

	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_);

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

	// Check the extra delay queue.
	while (!delay_link_.empty() &&
	receive_time_us >= delay_link_.front().arrival_time_us) {
	PacketInfo packet_info = delay_link_.front();
	packets_to_deliver.emplace_back(
	PacketDeliveryInfo(packet_info.packet, packet_info.arrival_time_us));
	delay_link_.pop_front();
	}

	if (!delay_link_.empty()) {
	next_process_time_us_ = delay_link_.front().arrival_time_us;
	} else if (!capacity_link_.empty()) {
	next_process_time_us_ = capacity_link_.front().arrival_time_us;
	} else {
	next_process_time_us_.reset();
	}
	return packets_to_deliver;
	}

	} // namespace webrtc