blob: 2dee92d8cd0cfd3292d494bfb731841633707a48 [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 "call/simulated_network.h"
#include <algorithm>
#include <cmath>
#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 {
SimulatedNetwork::SimulatedNetwork(SimulatedNetwork::Config config,
uint64_t random_seed)
: random_(random_seed), bursting_(false) {
SetConfig(config);
}
SimulatedNetwork::~SimulatedNetwork() = default;
void SimulatedNetwork::SetConfig(const SimulatedNetwork::Config& config) {
rtc::CritScope crit(&config_lock_);
if (config_.link_capacity_kbps != config.link_capacity_kbps) {
reset_capacity_delay_error_ = true;
}
config_ = config; // Shallow copy of the struct.
double prob_loss = config.loss_percent / 100.0;
if (config_.avg_burst_loss_length == -1) {
// Uniform loss
prob_loss_bursting_ = prob_loss;
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.";
prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
}
}
void SimulatedNetwork::PauseTransmissionUntil(int64_t until_us) {
rtc::CritScope crit(&config_lock_);
pause_transmission_until_us_ = until_us;
}
bool SimulatedNetwork::EnqueuePacket(PacketInFlightInfo packet) {
Config config;
{
rtc::CritScope crit(&config_lock_);
config = config_;
}
rtc::CritScope crit(&process_lock_);
if (config.queue_length_packets > 0 &&
capacity_link_.size() >= config.queue_length_packets) {
// Too many packet on the link, drop this one.
return false;
}
int64_t network_start_time_us = packet.send_time_us;
{
rtc::CritScope crit(&config_lock_);
if (reset_capacity_delay_error_) {
capacity_delay_error_bytes_ = 0;
reset_capacity_delay_error_ = false;
}
if (pause_transmission_until_us_) {
network_start_time_us =
std::max(network_start_time_us, *pause_transmission_until_us_);
pause_transmission_until_us_.reset();
}
}
// Delay introduced by the link capacity.
TimeDelta capacity_delay = TimeDelta::Zero();
if (config.link_capacity_kbps > 0) {
const DataRate link_capacity = DataRate::kbps(config.link_capacity_kbps);
int64_t compensated_size =
static_cast<int64_t>(packet.size) + capacity_delay_error_bytes_;
capacity_delay = DataSize::bytes(compensated_size) / link_capacity;
capacity_delay_error_bytes_ +=
packet.size - (capacity_delay * link_capacity).bytes();
}
// Check if there already are packets on the link and change network start
// time forward if there is.
if (!capacity_link_.empty() &&
network_start_time_us < capacity_link_.back().arrival_time_us)
network_start_time_us = capacity_link_.back().arrival_time_us;
int64_t arrival_time_us = network_start_time_us + capacity_delay.us();
capacity_link_.push({packet, arrival_time_us});
return true;
}
absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
rtc::CritScope crit(&process_lock_);
if (!delay_link_.empty())
return delay_link_.begin()->arrival_time_us;
return absl::nullopt;
}
std::vector<PacketDeliveryInfo> SimulatedNetwork::DequeueDeliverablePackets(
int64_t receive_time_us) {
int64_t time_now_us = receive_time_us;
Config config;
double prob_loss_bursting;
double prob_start_bursting;
{
rtc::CritScope crit(&config_lock_);
config = config_;
prob_loss_bursting = prob_loss_bursting_;
prob_start_bursting = prob_start_bursting_;
}
{
rtc::CritScope crit(&process_lock_);
// Check the capacity link first.
if (!capacity_link_.empty()) {
int64_t last_arrival_time_us =
delay_link_.empty() ? -1 : delay_link_.back().arrival_time_us;
bool needs_sort = false;
while (!capacity_link_.empty() &&
time_now_us >= capacity_link_.front().arrival_time_us) {
// Time to get this packet.
PacketInfo packet = std::move(capacity_link_.front());
capacity_link_.pop();
// Drop packets at an average rate of |config_.loss_percent| with
// and average loss burst length of |config_.avg_burst_loss_length|.
if ((bursting_ && random_.Rand<double>() < prob_loss_bursting) ||
(!bursting_ && random_.Rand<double>() < prob_start_bursting)) {
bursting_ = true;
continue;
} else {
bursting_ = false;
}
int64_t arrival_time_jitter_us = std::max(
random_.Gaussian(config.queue_delay_ms * 1000,
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.
if (!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;
if (packet.arrival_time_us >= last_arrival_time_us) {
last_arrival_time_us = packet.arrival_time_us;
} else {
needs_sort = true;
}
delay_link_.emplace_back(std::move(packet));
}
if (needs_sort) {
// Packet(s) arrived out of order, make sure list is sorted.
std::sort(delay_link_.begin(), delay_link_.end(),
[](const PacketInfo& p1, const PacketInfo& p2) {
return p1.arrival_time_us < p2.arrival_time_us;
});
}
}
std::vector<PacketDeliveryInfo> packets_to_deliver;
// Check the extra delay queue.
while (!delay_link_.empty() &&
time_now_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();
}
return packets_to_deliver;
}
}
} // namespace webrtc