| /* |
| * 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/test/simulated_network.h" |
| #include "rtc_base/criticalsection.h" |
| #include "rtc_base/random.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 NetworkBehaviorInterface { |
| 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); |
| void PauseTransmissionUntil(int64_t until_us); |
| |
| // 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; |
| }; |
| rtc::CriticalSection config_lock_; |
| bool reset_capacity_delay_error_ RTC_GUARDED_BY(config_lock_) = false; |
| |
| // |process_lock| guards the data structures involved in delay and loss |
| // processes, such as the packet queues. |
| rtc::CriticalSection process_lock_; |
| std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_lock_); |
| Random random_; |
| |
| std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_lock_); |
| |
| // Link configuration. |
| Config config_ RTC_GUARDED_BY(config_lock_); |
| absl::optional<int64_t> pause_transmission_until_us_ |
| RTC_GUARDED_BY(config_lock_); |
| |
| // Are we currently dropping a burst of packets? |
| bool bursting_; |
| |
| // The probability to drop the packet if we are currently dropping a |
| // burst of packet |
| double prob_loss_bursting_ RTC_GUARDED_BY(config_lock_); |
| |
| // The probability to drop a burst of packets. |
| double prob_start_bursting_ RTC_GUARDED_BY(config_lock_); |
| int64_t capacity_delay_error_bytes_ = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_SIMULATED_NETWORK_H_ |