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Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
187 lines
8 KiB
C++
187 lines
8 KiB
C++
/*
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* Copyright (c) 2022 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_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
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#define API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
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#include <cstddef>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/async_dns_resolver.h"
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#include "api/audio/audio_mixer.h"
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#include "api/audio/audio_processing.h"
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#include "api/fec_controller.h"
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#include "api/field_trials_view.h"
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#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
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#include "api/test/pclf/media_configuration.h"
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#include "api/transport/network_control.h"
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#include "api/video_codecs/video_decoder_factory.h"
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#include "api/video_codecs/video_encoder_factory.h"
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#include "p2p/base/port_allocator.h"
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#include "rtc_base/network.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "rtc_base/ssl_certificate.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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namespace webrtc_pc_e2e {
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// Contains most part from PeerConnectionFactoryDependencies. Also all fields
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// are optional and defaults will be provided by fixture implementation if
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// any will be omitted.
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//
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// Separate class was introduced to clarify which components can be
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// overridden. For example worker and signaling threads will be provided by
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// fixture implementation. The same is applicable to the media engine. So user
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// can override only some parts of media engine like video encoder/decoder
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// factories.
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struct PeerConnectionFactoryComponents {
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std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
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std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
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std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
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std::unique_ptr<NetEqFactory> neteq_factory;
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std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
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std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory;
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory;
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std::unique_ptr<FieldTrialsView> trials;
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rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
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rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer;
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};
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// Contains most parts from PeerConnectionDependencies. Also all fields are
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// optional and defaults will be provided by fixture implementation if any
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// will be omitted.
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//
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// Separate class was introduced to clarify which components can be
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// overridden. For example observer, which is required to
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// PeerConnectionDependencies, will be provided by fixture implementation,
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// so client can't inject its own. Also only network manager can be overridden
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// inside port allocator.
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struct PeerConnectionComponents {
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PeerConnectionComponents(rtc::NetworkManager* network_manager,
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rtc::PacketSocketFactory* packet_socket_factory)
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: network_manager(network_manager),
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packet_socket_factory(packet_socket_factory) {
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RTC_CHECK(network_manager);
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}
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rtc::NetworkManager* const network_manager;
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rtc::PacketSocketFactory* const packet_socket_factory;
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std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
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async_dns_resolver_factory;
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
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std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
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std::unique_ptr<IceTransportFactory> ice_transport_factory;
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};
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// Contains all components, that can be overridden in peer connection. Also
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// has a network thread, that will be used to communicate with another peers.
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struct InjectableComponents {
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InjectableComponents(rtc::Thread* network_thread,
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rtc::NetworkManager* network_manager,
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rtc::PacketSocketFactory* packet_socket_factory)
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: network_thread(network_thread),
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worker_thread(nullptr),
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pcf_dependencies(std::make_unique<PeerConnectionFactoryComponents>()),
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pc_dependencies(
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std::make_unique<PeerConnectionComponents>(network_manager,
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packet_socket_factory)) {
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RTC_CHECK(network_thread);
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}
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rtc::Thread* const network_thread;
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rtc::Thread* worker_thread;
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std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies;
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std::unique_ptr<PeerConnectionComponents> pc_dependencies;
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};
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// Contains information about call media streams (up to 1 audio stream and
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// unlimited amount of video streams) and rtc configuration, that will be used
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// to set up peer connection.
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struct Params {
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// Peer name. If empty - default one will be set by the fixture.
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absl::optional<std::string> name;
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// If `audio_config` is set audio stream will be configured
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absl::optional<AudioConfig> audio_config;
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// Flags to set on `cricket::PortAllocator`. These flags will be added
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// to the default ones that are presented on the port allocator.
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uint32_t port_allocator_extra_flags = cricket::kDefaultPortAllocatorFlags;
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// If `rtc_event_log_path` is set, an RTCEventLog will be saved in that
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// location and it will be available for further analysis.
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absl::optional<std::string> rtc_event_log_path;
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// If `aec_dump_path` is set, an AEC dump will be saved in that location and
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// it will be available for further analysis.
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absl::optional<std::string> aec_dump_path;
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bool use_ulp_fec = false;
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bool use_flex_fec = false;
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// Specifies how much video encoder target bitrate should be different than
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// target bitrate, provided by WebRTC stack. Must be greater then 0. Can be
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// used to emulate overshooting of video encoders. This multiplier will
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// be applied for all video encoder on both sides for all layers. Bitrate
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// estimated by WebRTC stack will be multiplied by this multiplier and then
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// provided into VideoEncoder::SetRates(...).
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double video_encoder_bitrate_multiplier = 1.0;
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PeerConnectionFactoryInterface::Options peer_connection_factory_options;
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PeerConnectionInterface::RTCConfiguration rtc_configuration;
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PeerConnectionInterface::RTCOfferAnswerOptions rtc_offer_answer_options;
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BitrateSettings bitrate_settings;
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std::vector<VideoCodecConfig> video_codecs;
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// A list of RTP header extensions which will be enforced on all video streams
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// added to this peer.
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std::vector<std::string> extra_video_rtp_header_extensions;
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// A list of RTP header extensions which will be enforced on all audio streams
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// added to this peer.
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std::vector<std::string> extra_audio_rtp_header_extensions;
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};
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// Contains parameters that maybe changed by test writer during the test call.
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struct ConfigurableParams {
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// If `video_configs` is empty - no video should be added to the test call.
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std::vector<VideoConfig> video_configs;
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VideoSubscription video_subscription =
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VideoSubscription().SubscribeToAllPeers();
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};
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// Contains parameters, that describe how long framework should run quality
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// test.
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struct RunParams {
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explicit RunParams(TimeDelta run_duration) : run_duration(run_duration) {}
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// Specifies how long the test should be run. This time shows how long
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// the media should flow after connection was established and before
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// it will be shut downed.
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TimeDelta run_duration;
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// If set to true peers will be able to use Flex FEC, otherwise they won't
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// be able to negotiate it even if it's enabled on per peer level.
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bool enable_flex_fec_support = false;
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// If true will set conference mode in SDP media section for all video
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// tracks for all peers.
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bool use_conference_mode = false;
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// If specified echo emulation will be done, by mixing the render audio into
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// the capture signal. In such case input signal will be reduced by half to
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// avoid saturation or compression in the echo path simulation.
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absl::optional<EchoEmulationConfig> echo_emulation_config;
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};
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} // namespace webrtc_pc_e2e
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} // namespace webrtc
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#endif // API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
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