| /* |
| * Copyright (c) 2018 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 TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_ |
| #define TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/async_resolver_factory.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/fec_controller.h" |
| #include "api/media_transport_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/test/simulated_network.h" |
| #include "api/transport/network_control.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "logging/rtc_event_log/rtc_event_log_factory_interface.h" |
| #include "rtc_base/network.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/ssl_certificate.h" |
| #include "rtc_base/thread.h" |
| #include "test/pc/e2e/api/audio_quality_analyzer_interface.h" |
| #include "test/pc/e2e/api/video_quality_analyzer_interface.h" |
| |
| namespace webrtc { |
| |
| // TODO(titovartem) move to API when it will be stabilized. |
| class PeerConnectionE2EQualityTestFixture { |
| public: |
| struct PeerConnectionFactoryComponents { |
| std::unique_ptr<CallFactoryInterface> call_factory; |
| std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory; |
| std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory; |
| std::unique_ptr<NetworkControllerFactoryInterface> |
| network_controller_factory; |
| std::unique_ptr<MediaTransportFactory> media_transport_factory; |
| |
| // Will be passed to MediaEngineInterface, that will be used in |
| // PeerConnectionFactory. |
| std::unique_ptr<VideoEncoderFactory> video_encoder_factory; |
| std::unique_ptr<VideoDecoderFactory> video_decoder_factory; |
| }; |
| |
| struct PeerConnectionComponents { |
| std::unique_ptr<rtc::NetworkManager> network_manager; |
| std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory; |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator; |
| std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier; |
| }; |
| |
| struct InjectableComponents { |
| explicit InjectableComponents(rtc::Thread* network_thread) |
| : network_thread(network_thread) {} |
| |
| rtc::Thread* network_thread; |
| |
| std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies; |
| std::unique_ptr<PeerConnectionComponents> pc_dependencies; |
| }; |
| |
| struct ScreenShareConfig { |
| // If true, slides will be generated programmatically. |
| bool generate_slides; |
| int32_t slide_change_interval; |
| // If equal to 0, no scrolling will be applied. |
| int32_t scroll_duration; |
| // If empty, default set of slides will be used. |
| std::vector<std::string> slides_yuv_file_names; |
| }; |
| |
| struct VideoConfig { |
| size_t width; |
| size_t height; |
| int32_t fps; |
| // Have to be unique among all specified configs for all peers in the call. |
| absl::optional<std::string> stream_label; |
| // Only single from 3 next fields can be specified. |
| // If specified generator with this name will be used as input. |
| absl::optional<std::string> generator_name; |
| // If specified this file will be used as input. |
| absl::optional<std::string> input_file_name; |
| // If specified screen share video stream will be created as input. |
| absl::optional<ScreenShareConfig> screen_share_config; |
| // If specified the input stream will be also copied to specified file. |
| absl::optional<std::string> input_dump_file_name; |
| // If specified this file will be used as output on the receiver side for |
| // this stream. If multiple streams will be produced by input stream, |
| // output files will be appended with indexes. |
| absl::optional<std::string> output_file_name; |
| }; |
| |
| struct AudioConfig { |
| enum Mode { |
| kGenerated, |
| kFile, |
| }; |
| Mode mode; |
| // Have to be specified only if mode = kFile |
| absl::optional<std::string> input_file_name; |
| // If specified the input stream will be also copied to specified file. |
| absl::optional<std::string> input_dump_file_name; |
| // If specified the output stream will be copied to specified file. |
| absl::optional<std::string> output_file_name; |
| // Audio options to use. |
| cricket::AudioOptions audio_options; |
| }; |
| |
| struct Params { |
| // If |video_configs| is empty - no video should be added to the test call. |
| std::vector<VideoConfig> video_configs; |
| // If |audio_config| is presented audio stream will be configured |
| absl::optional<AudioConfig> audio_config; |
| |
| PeerConnectionInterface::RTCConfiguration rtc_configuration; |
| }; |
| |
| struct Analyzers { |
| std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer; |
| std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer; |
| }; |
| |
| virtual void Run() = 0; |
| virtual ~PeerConnectionE2EQualityTestFixture() = default; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_ |