Sebastian Jansson | f96b1ca | 2018-08-07 18:58:05 +0200 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2018 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | #ifndef CALL_SIMULATED_NETWORK_H_ |
| 11 | #define CALL_SIMULATED_NETWORK_H_ |
| 12 | |
| 13 | #include <deque> |
| 14 | #include <queue> |
| 15 | #include <vector> |
| 16 | |
| 17 | #include "absl/memory/memory.h" |
| 18 | #include "absl/types/optional.h" |
| 19 | #include "api/test/simulated_network.h" |
| 20 | #include "rtc_base/criticalsection.h" |
| 21 | #include "rtc_base/random.h" |
| 22 | #include "rtc_base/thread_annotations.h" |
| 23 | |
| 24 | namespace webrtc { |
| 25 | |
| 26 | // Class simulating a network link. This is a simple and naive solution just |
| 27 | // faking capacity and adding an extra transport delay in addition to the |
| 28 | // capacity introduced delay. |
| 29 | class SimulatedNetwork : public NetworkSimulationInterface { |
| 30 | public: |
| 31 | using Config = NetworkSimulationInterface::SimulatedNetworkConfig; |
| 32 | explicit SimulatedNetwork(Config config, uint64_t random_seed = 1); |
| 33 | ~SimulatedNetwork() override; |
| 34 | |
| 35 | // Sets a new configuration. This won't affect packets already in the pipe. |
| 36 | void SetConfig(const Config& config); |
| 37 | void PauseTransmissionUntil(int64_t until_us); |
| 38 | |
| 39 | // NetworkSimulationInterface |
| 40 | bool EnqueuePacket(PacketInFlightInfo packet) override; |
| 41 | std::vector<PacketDeliveryInfo> DequeueDeliverablePackets( |
| 42 | int64_t receive_time_us) override; |
| 43 | |
| 44 | absl::optional<int64_t> NextDeliveryTimeUs() const override; |
| 45 | |
| 46 | private: |
| 47 | struct PacketInfo { |
| 48 | PacketInFlightInfo packet; |
| 49 | int64_t arrival_time_us; |
| 50 | }; |
| 51 | rtc::CriticalSection config_lock_; |
| 52 | |
| 53 | // |process_lock| guards the data structures involved in delay and loss |
| 54 | // processes, such as the packet queues. |
| 55 | rtc::CriticalSection process_lock_; |
| 56 | std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_lock_); |
| 57 | Random random_; |
| 58 | |
| 59 | std::deque<PacketInfo> delay_link_; |
| 60 | |
| 61 | // Link configuration. |
| 62 | Config config_ RTC_GUARDED_BY(config_lock_); |
| 63 | absl::optional<int64_t> pause_transmission_until_us_ |
| 64 | RTC_GUARDED_BY(config_lock_); |
| 65 | |
| 66 | // Are we currently dropping a burst of packets? |
| 67 | bool bursting_; |
| 68 | |
| 69 | // The probability to drop the packet if we are currently dropping a |
| 70 | // burst of packet |
| 71 | double prob_loss_bursting_ RTC_GUARDED_BY(config_lock_); |
| 72 | |
| 73 | // The probability to drop a burst of packets. |
| 74 | double prob_start_bursting_ RTC_GUARDED_BY(config_lock_); |
| 75 | int64_t capacity_delay_error_bytes_ = 0; |
| 76 | }; |
| 77 | |
| 78 | } // namespace webrtc |
| 79 | |
| 80 | #endif // CALL_SIMULATED_NETWORK_H_ |