| /* |
| * 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 { |
| namespace { |
| constexpr TimeDelta kDefaultProcessDelay = TimeDelta::Millis(5); |
| } // namespace |
| |
| SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed) |
| : random_(random_seed), bursting_(false) { |
| 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_); |
| ConfigState state = GetConfigState(); |
| |
| UpdateCapacityQueue(state, packet.send_time_us); |
| |
| packet.size += state.config.packet_overhead; |
| |
| 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; |
| } |
| |
| // Set arrival time = send time for now; actual arrival time will be |
| // calculated in UpdateCapacityQueue. |
| queue_size_bytes_ += packet.size; |
| capacity_link_.push({packet, packet.send_time_us}); |
| if (!next_process_time_us_) { |
| next_process_time_us_ = packet.send_time_us + kDefaultProcessDelay.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) { |
| bool needs_sort = false; |
| |
| // Catch for thread races. |
| if (time_now_us < last_capacity_link_visit_us_.value_or(time_now_us)) |
| return; |
| |
| int64_t time_us = last_capacity_link_visit_us_.value_or(time_now_us); |
| // Check the capacity link first. |
| while (!capacity_link_.empty()) { |
| int64_t time_until_front_exits_us = 0; |
| if (state.config.link_capacity_kbps > 0) { |
| int64_t remaining_bits = |
| capacity_link_.front().packet.size * 8 - pending_drain_bits_; |
| RTC_DCHECK(remaining_bits > 0); |
| // Division rounded up - packet not delivered until its last bit is. |
| time_until_front_exits_us = |
| (1000 * remaining_bits + state.config.link_capacity_kbps - 1) / |
| state.config.link_capacity_kbps; |
| } |
| |
| if (time_us + time_until_front_exits_us > time_now_us) { |
| // Packet at front will not exit yet. Will not enter here on infinite |
| // capacity(=0) so no special handling needed. |
| pending_drain_bits_ += |
| ((time_now_us - time_us) * state.config.link_capacity_kbps) / 1000; |
| break; |
| } |
| if (state.config.link_capacity_kbps > 0) { |
| pending_drain_bits_ += |
| (time_until_front_exits_us * state.config.link_capacity_kbps) / 1000; |
| } else { |
| // Enough to drain the whole queue. |
| pending_drain_bits_ = queue_size_bytes_ * 8; |
| } |
| |
| // Time to get this packet. |
| PacketInfo packet = capacity_link_.front(); |
| capacity_link_.pop(); |
| |
| time_us += time_until_front_exits_us; |
| RTC_DCHECK(time_us >= packet.packet.send_time_us); |
| packet.arrival_time_us = |
| std::max(state.pause_transmission_until_us, time_us); |
| queue_size_bytes_ -= packet.packet.size; |
| pending_drain_bits_ -= packet.packet.size * 8; |
| RTC_DCHECK(pending_drain_bits_ >= 0); |
| |
| // 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 { |
| 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; |
| 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(packet); |
| } |
| last_capacity_link_visit_us_ = time_now_us; |
| // Cannot save unused capacity for later. |
| pending_drain_bits_ = std::min(pending_drain_bits_, queue_size_bytes_ * 8); |
| |
| 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; |
| }); |
| } |
| } |
| |
| 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_ = receive_time_us + kDefaultProcessDelay.us(); |
| } else { |
| next_process_time_us_.reset(); |
| } |
| return packets_to_deliver; |
| } |
| |
| } // namespace webrtc |