| /* |
| * 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. |
| */ |
| #ifndef CALL_SIMULATED_NETWORK_H_ |
| #define CALL_SIMULATED_NETWORK_H_ |
| |
| #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 { |
| public: |
| 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; |
| |
| private: |
| 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) |
| RTC_RUN_ON(&process_checker_); |
| 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_ |
| RTC_GUARDED_BY(process_checker_); |
| absl::optional<int64_t> next_process_time_us_ |
| RTC_GUARDED_BY(process_checker_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_SIMULATED_NETWORK_H_ |