blob: d3092aefba0a2d68711686d1a81db91ba5c6d28a [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 <stdint.h>
#include <deque>
#include <queue>
#include <vector>
#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/test/simulated_network.h"
#include "api/units/data_size.h"
#include "api/units/timestamp.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// Class simulating a network link. This is a simple and naive solution just
// faking capacity and adding an extra transport delay in addition to the
// capacity introduced delay.
class SimulatedNetwork : public SimulatedNetworkInterface {
using Config = BuiltInNetworkBehaviorConfig;
explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
~SimulatedNetwork() override;
// Sets a new configuration. This won't affect packets already in the pipe.
void SetConfig(const Config& config) override;
void UpdateConfig(std::function<void(BuiltInNetworkBehaviorConfig*)>
config_modifier) override;
void PauseTransmissionUntil(int64_t until_us) override;
// NetworkBehaviorInterface
bool EnqueuePacket(PacketInFlightInfo packet) override;
std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
int64_t receive_time_us) override;
absl::optional<int64_t> NextDeliveryTimeUs() const override;
struct PacketInfo {
PacketInFlightInfo packet;
int64_t arrival_time_us;
// Contains current configuration state.
struct ConfigState {
// Static link configuration.
Config config;
// The probability to drop the packet if we are currently dropping a
// burst of packet
double prob_loss_bursting;
// The probability to drop a burst of packets.
double prob_start_bursting;
// Used for temporary delay spikes.
int64_t pause_transmission_until_us = 0;
// Moves packets from capacity- to delay link.
void UpdateCapacityQueue(ConfigState state, int64_t time_now_us)
ConfigState GetConfigState() const;
mutable Mutex config_lock_;
// `process_checker_` guards the data structures involved in delay and loss
// processes, such as the packet queues.
rtc::RaceChecker process_checker_;
std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_checker_);
Random random_;
std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_checker_);
ConfigState config_state_ RTC_GUARDED_BY(config_lock_);
// Are we currently dropping a burst of packets?
bool bursting_;
int64_t queue_size_bytes_ RTC_GUARDED_BY(process_checker_) = 0;
int64_t pending_drain_bits_ RTC_GUARDED_BY(process_checker_) = 0;
absl::optional<int64_t> last_capacity_link_visit_us_
absl::optional<int64_t> next_process_time_us_
} // namespace webrtc