| /* |
| * 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 <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/async_dns_resolver.h" |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/fec_controller.h" |
| #include "api/field_trials_view.h" |
| #include "api/rtc_event_log/rtc_event_log_factory_interface.h" |
| #include "api/test/pclf/media_configuration.h" |
| #include "api/transport/network_control.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/network.h" |
| #include "rtc_base/rtc_certificate_generator.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<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; |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory; |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory; |
| |
| std::unique_ptr<FieldTrialsView> trials; |
| |
| rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing; |
| rtc::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. Also only network manager can be overridden |
| // inside port allocator. |
| struct PeerConnectionComponents { |
| PeerConnectionComponents(rtc::NetworkManager* network_manager, |
| rtc::PacketSocketFactory* packet_socket_factory) |
| : network_manager(network_manager), |
| packet_socket_factory(packet_socket_factory) { |
| RTC_CHECK(network_manager); |
| } |
| |
| rtc::NetworkManager* const network_manager; |
| rtc::PacketSocketFactory* const packet_socket_factory; |
| std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface> |
| async_dns_resolver_factory; |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator; |
| std::unique_ptr<rtc::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(rtc::Thread* network_thread, |
| rtc::NetworkManager* network_manager, |
| rtc::PacketSocketFactory* packet_socket_factory) |
| : network_thread(network_thread), |
| worker_thread(nullptr), |
| pcf_dependencies(std::make_unique<PeerConnectionFactoryComponents>()), |
| pc_dependencies( |
| std::make_unique<PeerConnectionComponents>(network_manager, |
| packet_socket_factory)) { |
| RTC_CHECK(network_thread); |
| } |
| |
| rtc::Thread* const network_thread; |
| rtc::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. |
| absl::optional<std::string> name; |
| // If `audio_config` is set audio stream will be configured |
| absl::optional<AudioConfig> audio_config; |
| // Flags to set on `cricket::PortAllocator`. These flags will be added |
| // to the default ones that are presented on the port allocator. |
| uint32_t port_allocator_extra_flags = cricket::kDefaultPortAllocatorFlags; |
| // If `rtc_event_log_path` is set, an RTCEventLog will be saved in that |
| // location and it will be available for further analysis. |
| absl::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. |
| absl::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 set to true peers will be able to use Flex FEC, otherwise they won't |
| // be able to negotiate it even if it's enabled on per peer level. |
| bool enable_flex_fec_support = false; |
| // 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. |
| absl::optional<EchoEmulationConfig> echo_emulation_config; |
| }; |
| |
| } // namespace webrtc_pc_e2e |
| } // namespace webrtc |
| |
| #endif // API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_ |