e9721f2f08
Make SimulatedNetwrokConfig configuration of default implementation of NetworkSimulationInterface, that will be used by WebRTC in case of network simulation. Bug: webrtc:9630 Change-Id: Ib7c3d0c69fc09627f3d8694e61ac8409101e8392 Reviewed-on: https://webrtc-review.googlesource.com/94154 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24311}
88 lines
3.0 KiB
C++
88 lines
3.0 KiB
C++
/*
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* Copyright (c) 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 API_TEST_SIMULATED_NETWORK_H_
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#define API_TEST_SIMULATED_NETWORK_H_
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#include <stddef.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 "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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struct PacketInFlightInfo {
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PacketInFlightInfo(size_t size, int64_t send_time_us, uint64_t packet_id)
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: size(size), send_time_us(send_time_us), packet_id(packet_id) {}
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size_t size;
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int64_t send_time_us;
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// Unique identifier for the packet in relation to other packets in flight.
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uint64_t packet_id;
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};
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struct PacketDeliveryInfo {
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static constexpr int kNotReceived = -1;
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PacketDeliveryInfo(PacketInFlightInfo source, int64_t receive_time_us)
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: receive_time_us(receive_time_us), packet_id(source.packet_id) {}
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int64_t receive_time_us;
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uint64_t packet_id;
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};
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// DefaultNetworkSimulationConfig is a default network simulation configuration
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// for default network simulation that will be used by WebRTC if no custom
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// NetworkSimulationInterface is provided.
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struct DefaultNetworkSimulationConfig {
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DefaultNetworkSimulationConfig() {}
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// Queue length in number of packets.
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size_t queue_length_packets = 0;
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// Delay in addition to capacity induced delay.
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int queue_delay_ms = 0;
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// Standard deviation of the extra delay.
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int delay_standard_deviation_ms = 0;
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// Link capacity in kbps.
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int link_capacity_kbps = 0;
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// Random packet loss.
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int loss_percent = 0;
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// If packets are allowed to be reordered.
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bool allow_reordering = false;
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// The average length of a burst of lost packets.
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int avg_burst_loss_length = -1;
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};
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class NetworkSimulationInterface {
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public:
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// DO NOT USE. Use DefaultNetworkSimulationConfig directly. This reference
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// should be removed when all users will be switched on direct usage.
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using SimulatedNetworkConfig = DefaultNetworkSimulationConfig;
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// DO NOT USE. Method added temporary for further refactoring and will be
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// removed soon.
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// Sets a new configuration. This won't affect packets already in the pipe.
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virtual void SetConfig(const SimulatedNetworkConfig& config) = 0;
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virtual bool EnqueuePacket(PacketInFlightInfo packet_info) = 0;
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// Retrieves all packets that should be delivered by the given receive time.
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virtual std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
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int64_t receive_time_us) = 0;
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virtual absl::optional<int64_t> NextDeliveryTimeUs() const = 0;
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virtual ~NetworkSimulationInterface() = default;
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};
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} // namespace webrtc
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#endif // API_TEST_SIMULATED_NETWORK_H_
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