| /* | |
| * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_ | |
| #define API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_ | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <memory> | |
| #include <optional> | |
| #include <string> | |
| #include <utility> | |
| #include <vector> | |
| #include "api/async_dns_resolver.h" | |
| #include "api/audio/audio_mixer.h" | |
| #include "api/audio/audio_processing.h" | |
| #include "api/audio_codecs/audio_decoder_factory.h" | |
| #include "api/audio_codecs/audio_encoder_factory.h" | |
| #include "api/fec_controller.h" | |
| #include "api/field_trials.h" | |
| #include "api/ice_transport_interface.h" | |
| #include "api/neteq/neteq_factory.h" | |
| #include "api/peer_connection_interface.h" | |
| #include "api/rtc_event_log/rtc_event_log_factory_interface.h" | |
| #include "api/scoped_refptr.h" | |
| #include "api/test/pclf/media_configuration.h" | |
| #include "api/transport/bitrate_settings.h" | |
| #include "api/transport/network_control.h" | |
| #include "api/units/time_delta.h" | |
| #include "api/video_codecs/video_decoder_factory.h" | |
| #include "api/video_codecs/video_encoder_factory.h" | |
| #include "p2p/base/port_allocator.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/network.h" | |
| #include "rtc_base/rtc_certificate_generator.h" | |
| #include "rtc_base/socket_factory.h" | |
| #include "rtc_base/ssl_certificate.h" | |
| #include "rtc_base/thread.h" | |
| namespace webrtc { | |
| namespace webrtc_pc_e2e { | |
| // Contains most part from PeerConnectionFactoryDependencies. Also all fields | |
| // are optional and defaults will be provided by fixture implementation if | |
| // any will be omitted. | |
| // | |
| // Separate class was introduced to clarify which components can be | |
| // overridden. For example worker and signaling threads will be provided by | |
| // fixture implementation. The same is applicable to the media engine. So user | |
| // can override only some parts of media engine like video encoder/decoder | |
| // factories. | |
| struct PeerConnectionFactoryComponents { | |
| std::unique_ptr<NetworkManager> network_manager; | |
| SocketFactory* socket_factory = nullptr; | |
| std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory; | |
| std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory; | |
| std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory; | |
| std::unique_ptr<NetEqFactory> neteq_factory; | |
| std::unique_ptr<VideoEncoderFactory> video_encoder_factory; | |
| std::unique_ptr<VideoDecoderFactory> video_decoder_factory; | |
| scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory; | |
| scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory; | |
| std::unique_ptr<FieldTrials> field_trials; | |
| std::unique_ptr<AudioProcessingBuilderInterface> audio_processing; | |
| scoped_refptr<webrtc::AudioMixer> audio_mixer; | |
| }; | |
| // Contains most parts from PeerConnectionDependencies. Also all fields are | |
| // optional and defaults will be provided by fixture implementation if any | |
| // will be omitted. | |
| // | |
| // Separate class was introduced to clarify which components can be | |
| // overridden. For example observer, which is required to | |
| // PeerConnectionDependencies, will be provided by fixture implementation, | |
| // so client can't inject its own. | |
| struct PeerConnectionComponents { | |
| std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface> | |
| async_dns_resolver_factory; | |
| std::unique_ptr<RTCCertificateGeneratorInterface> cert_generator; | |
| std::unique_ptr<SSLCertificateVerifier> tls_cert_verifier; | |
| std::unique_ptr<IceTransportFactory> ice_transport_factory; | |
| }; | |
| // Contains all components, that can be overridden in peer connection. Also | |
| // has a network thread, that will be used to communicate with another peers. | |
| struct InjectableComponents { | |
| InjectableComponents(Thread* network_thread, | |
| std::unique_ptr<NetworkManager> network_manager, | |
| SocketFactory* socket_factory) | |
| : network_thread(network_thread), | |
| worker_thread(nullptr), | |
| pcf_dependencies(std::make_unique<PeerConnectionFactoryComponents>()), | |
| pc_dependencies(std::make_unique<PeerConnectionComponents>()) { | |
| RTC_CHECK(network_thread); | |
| pcf_dependencies->network_manager = std::move(network_manager); | |
| pcf_dependencies->socket_factory = socket_factory; | |
| } | |
| Thread* const network_thread; | |
| Thread* worker_thread; | |
| std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies; | |
| std::unique_ptr<PeerConnectionComponents> pc_dependencies; | |
| }; | |
| // Contains information about call media streams (up to 1 audio stream and | |
| // unlimited amount of video streams) and rtc configuration, that will be used | |
| // to set up peer connection. | |
| struct Params { | |
| // Peer name. If empty - default one will be set by the fixture. | |
| std::optional<std::string> name; | |
| // If `audio_config` is set audio stream will be configured | |
| std::optional<AudioConfig> audio_config; | |
| // Flags to override `rtc_configuration.port_allocator_config.flags` | |
| // | |
| // IMPORTANT: if you use WebRTC Network Emulation | |
| // (api/test/network_emulation_manager.h) and set this field, remember to set | |
| // webrtc::PORTALLOCATOR_DISABLE_TCP. | |
| uint32_t port_allocator_flags = PORTALLOCATOR_DISABLE_TCP; | |
| // If `rtc_event_log_path` is set, an RTCEventLog will be saved in that | |
| // location and it will be available for further analysis. | |
| std::optional<std::string> rtc_event_log_path; | |
| // If `aec_dump_path` is set, an AEC dump will be saved in that location and | |
| // it will be available for further analysis. | |
| std::optional<std::string> aec_dump_path; | |
| bool use_ulp_fec = false; | |
| bool use_flex_fec = false; | |
| // Specifies how much video encoder target bitrate should be different than | |
| // target bitrate, provided by WebRTC stack. Must be greater then 0. Can be | |
| // used to emulate overshooting of video encoders. This multiplier will | |
| // be applied for all video encoder on both sides for all layers. Bitrate | |
| // estimated by WebRTC stack will be multiplied by this multiplier and then | |
| // provided into VideoEncoder::SetRates(...). | |
| double video_encoder_bitrate_multiplier = 1.0; | |
| PeerConnectionFactoryInterface::Options peer_connection_factory_options; | |
| PeerConnectionInterface::RTCConfiguration rtc_configuration; | |
| PeerConnectionInterface::RTCOfferAnswerOptions rtc_offer_answer_options; | |
| BitrateSettings bitrate_settings; | |
| std::vector<VideoCodecConfig> video_codecs; | |
| // A list of RTP header extensions which will be enforced on all video streams | |
| // added to this peer. | |
| std::vector<std::string> extra_video_rtp_header_extensions; | |
| // A list of RTP header extensions which will be enforced on all audio streams | |
| // added to this peer. | |
| std::vector<std::string> extra_audio_rtp_header_extensions; | |
| }; | |
| // Contains parameters that maybe changed by test writer during the test call. | |
| struct ConfigurableParams { | |
| // If `video_configs` is empty - no video should be added to the test call. | |
| std::vector<VideoConfig> video_configs; | |
| VideoSubscription video_subscription = | |
| VideoSubscription().SubscribeToAllPeers(); | |
| }; | |
| // Contains parameters, that describe how long framework should run quality | |
| // test. | |
| struct RunParams { | |
| explicit RunParams(TimeDelta run_duration) : run_duration(run_duration) {} | |
| // Specifies how long the test should be run. This time shows how long | |
| // the media should flow after connection was established and before | |
| // it will be shut downed. | |
| TimeDelta run_duration; | |
| // If true will set conference mode in SDP media section for all video | |
| // tracks for all peers. | |
| bool use_conference_mode = false; | |
| // If specified echo emulation will be done, by mixing the render audio into | |
| // the capture signal. In such case input signal will be reduced by half to | |
| // avoid saturation or compression in the echo path simulation. | |
| std::optional<EchoEmulationConfig> echo_emulation_config; | |
| }; | |
| } // namespace webrtc_pc_e2e | |
| } // namespace webrtc | |
| #endif // API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_ |