b6b29e0718
The purpose is to make the fixture reusable in downstream projects. The CL adds the following things to API: - api/test/video_quality_test_fixture.h - api/test/create_video_quality_test_fixture.h The following things are moved to API: - call/bitrate_constraints.h (api/bitrate_constraints.h) - call/simulated_network.h (api/test/simulated_network.h) - call/media_type.h (api/mediatypes.h) These are required by the params struct passed to the fixture. I didn't attempt to split the params struct into an internal-only and public version in this CL, and as a result we need to pull in the above things. They are quite harmless though, so I think it's worth it in order to avoid splitting up the test config struct. This CL doesn't solve all the problems we need to implement downstream tests; we probably need to upstream tracing variants of FakeNetworkPipe for instance, but that will come later. This puts in place the basic structure for now. Bug: None Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911 Reviewed-on: https://webrtc-review.googlesource.com/69601 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23714}
78 lines
2.5 KiB
C++
78 lines
2.5 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 "api/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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class NetworkSimulationInterface {
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public:
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// TODO(phoglund): this one shouldn't really be here; make fake network pipes
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// injectable instead in the video quality test fixture.
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struct SimulatedNetworkConfig {
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SimulatedNetworkConfig() {}
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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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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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