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baf5c9fabd
This reverts commit c1d5fda22c.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
101 lines
3.4 KiB
C++
101 lines
3.4 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/sequence_checker.h"
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#include "api/test/simulated_network.h"
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#include "api/units/data_size.h"
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#include "api/units/timestamp.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/synchronization/mutex.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 SimulatedNetworkInterface {
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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) override;
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void UpdateConfig(std::function<void(BuiltInNetworkBehaviorConfig*)>
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config_modifier) override;
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void PauseTransmissionUntil(int64_t until_us) override;
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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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mutable Mutex 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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