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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}
93 lines
3.2 KiB
C++
93 lines
3.2 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/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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// BuiltInNetworkBehaviorConfig is a built-in network behavior configuration
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// for built-in network behavior that will be used by WebRTC if no custom
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// NetworkBehaviorInterface is provided.
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struct BuiltInNetworkBehaviorConfig {
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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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// Additional bytes to add to packet size.
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int packet_overhead = 0;
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};
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class NetworkBehaviorInterface {
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public:
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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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// Returns time in microseconds when caller should call
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// DequeueDeliverablePackets to get next set of packets to deliver.
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virtual absl::optional<int64_t> NextDeliveryTimeUs() const = 0;
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virtual ~NetworkBehaviorInterface() = default;
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};
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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 SimulatedNetworkInterface : public NetworkBehaviorInterface {
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public:
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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 BuiltInNetworkBehaviorConfig& config) = 0;
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virtual void UpdateConfig(
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std::function<void(BuiltInNetworkBehaviorConfig*)> config_modifier) = 0;
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virtual void PauseTransmissionUntil(int64_t until_us) = 0;
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};
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} // namespace webrtc
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#endif // API_TEST_SIMULATED_NETWORK_H_
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