4ff63cc9a1
Remove temporary SetConfig method from NetworkSimulatioInterface and makes minor cleanup. Bug: webrtc:9630 Change-Id: If472da7c21ffc9c83fe8b80e6665c3d5fb94382b Reviewed-on: https://webrtc-review.googlesource.com/95644 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24400}
81 lines
2.6 KiB
C++
81 lines
2.6 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 <deque>
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#include <queue>
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#include <vector>
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#include "absl/memory/memory.h"
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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/criticalsection.h"
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#include "rtc_base/random.h"
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#include "rtc_base/thread_annotations.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 NetworkSimulationInterface {
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public:
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using Config = DefaultNetworkSimulationConfig;
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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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// NetworkSimulationInterface
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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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rtc::CriticalSection config_lock_;
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// |process_lock| guards the data structures involved in delay and loss
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// processes, such as the packet queues.
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rtc::CriticalSection process_lock_;
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std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_lock_);
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Random random_;
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std::deque<PacketInfo> delay_link_;
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// Link configuration.
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Config config_ RTC_GUARDED_BY(config_lock_);
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absl::optional<int64_t> pause_transmission_until_us_
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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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// 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_ RTC_GUARDED_BY(config_lock_);
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// The probability to drop a burst of packets.
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double prob_start_bursting_ RTC_GUARDED_BY(config_lock_);
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int64_t capacity_delay_error_bytes_ = 0;
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};
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} // namespace webrtc
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#endif // CALL_SIMULATED_NETWORK_H_
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