blob: 776fccbbf626d883afe572def9142c97599bd24f [file] [log] [blame]
/*
* 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 "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