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836fee1e1a
Currently there's an implicit requirement that users of SimulatedNetwork should call it repeatedly, even if the return value of NextDeliveryTimeUs is unset. With this change, it will indicate that there might be a delivery in 5 ms at any time there are packets in queue. Which results in unchanged behavior compared to current usage but allows new users to expect robust behavior. Bug: webrtc:9510 Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069 Reviewed-on: https://webrtc-review.googlesource.com/c/120402 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26617}
96 lines
3.2 KiB
C++
96 lines
3.2 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_SIMULATED_NETWORK_H_
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#define CALL_SIMULATED_NETWORK_H_
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#include <stdint.h>
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#include <deque>
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#include <queue>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/test/simulated_network.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/race_checker.h"
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#include "rtc_base/random.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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// Class simulating a network link. This is a simple and naive solution just
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// faking capacity and adding an extra transport delay in addition to the
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// capacity introduced delay.
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class SimulatedNetwork : public NetworkBehaviorInterface {
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public:
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using Config = BuiltInNetworkBehaviorConfig;
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explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
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~SimulatedNetwork() override;
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// Sets a new configuration. This won't affect packets already in the pipe.
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void SetConfig(const Config& config);
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void PauseTransmissionUntil(int64_t until_us);
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// NetworkBehaviorInterface
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bool EnqueuePacket(PacketInFlightInfo packet) override;
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std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
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int64_t receive_time_us) override;
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absl::optional<int64_t> NextDeliveryTimeUs() const override;
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private:
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struct PacketInfo {
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PacketInFlightInfo packet;
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int64_t arrival_time_us;
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};
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// Contains current configuration state.
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struct ConfigState {
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// Static link configuration.
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Config config;
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// The probability to drop the packet if we are currently dropping a
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// burst of packet
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double prob_loss_bursting;
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// The probability to drop a burst of packets.
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double prob_start_bursting;
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// Used for temporary delay spikes.
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int64_t pause_transmission_until_us = 0;
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};
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// Moves packets from capacity- to delay link.
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void UpdateCapacityQueue(ConfigState state, int64_t time_now_us)
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RTC_RUN_ON(&process_checker_);
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ConfigState GetConfigState() const;
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rtc::CriticalSection config_lock_;
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// |process_checker_| guards the data structures involved in delay and loss
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// processes, such as the packet queues.
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rtc::RaceChecker process_checker_;
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std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_checker_);
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Random random_;
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std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_checker_);
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ConfigState config_state_ RTC_GUARDED_BY(config_lock_);
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// Are we currently dropping a burst of packets?
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bool bursting_;
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int64_t queue_size_bytes_ RTC_GUARDED_BY(process_checker_) = 0;
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int64_t pending_drain_bits_ RTC_GUARDED_BY(process_checker_) = 0;
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absl::optional<int64_t> last_capacity_link_visit_us_
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RTC_GUARDED_BY(process_checker_);
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absl::optional<int64_t> next_process_time_us_
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RTC_GUARDED_BY(process_checker_);
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};
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} // namespace webrtc
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#endif // CALL_SIMULATED_NETWORK_H_
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