| /* |
| * 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 "absl/memory/memory.h" |
| #include "api/async_resolver_factory.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/fec_controller.h" |
| #include "api/function_view.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/units/time_delta.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 { |
| namespace webrtc_pc_e2e { |
| |
| // TODO(titovartem) move to API when it will be stabilized. |
| class PeerConnectionE2EQualityTestFixture { |
| public: |
| // Contains screen share video stream properties. |
| struct ScreenShareConfig { |
| // If true, slides will be generated programmatically. |
| bool generate_slides; |
| // Shows how long one slide should be presented on the screen during |
| // slide generation. |
| TimeDelta slide_change_interval; |
| // If equal to 0, no scrolling will be applied. |
| TimeDelta scroll_duration; |
| // If empty, default set of slides will be used. |
| std::vector<std::string> slides_yuv_file_names; |
| }; |
| |
| enum VideoGeneratorType { kDefault, kI420A, kI010 }; |
| |
| // Contains properties of single video stream. |
| struct VideoConfig { |
| VideoConfig(size_t width, size_t height, int32_t fps) |
| : width(width), height(height), fps(fps) {} |
| |
| const size_t width; |
| const size_t height; |
| const int32_t fps; |
| // Have to be unique among all specified configs for all peers in the call. |
| // Will be auto generated if omitted. |
| absl::optional<std::string> stream_label; |
| // Only 1 from |generator|, |input_file_name| and |screen_share_config| can |
| // be specified. If none of them are specified, then |generator| will be set |
| // to VideoGeneratorType::kDefault. |
| // If specified generator of this type will be used to produce input video. |
| absl::optional<VideoGeneratorType> generator; |
| // If specified this file will be used as input. Input video will be played |
| // in a circle. |
| absl::optional<std::string> input_file_name; |
| // If specified screen share video stream will be created as input. |
| absl::optional<ScreenShareConfig> screen_share_config; |
| // Specifies spatial index of the video stream to analyze. |
| // There are 3 cases: |
| // 1. |target_spatial_index| omitted: in such case it will be assumed that |
| // video stream has not spatial layers and simulcast streams. |
| // 2. |target_spatial_index| presented and simulcast encoder is used: |
| // in such case |target_spatial_index| will specify the index of |
| // simulcast stream, that should be analyzed. Other streams will be |
| // dropped. |
| // 3. |target_spatial_index| presented and SVP encoder is used: |
| // in such case |target_spatial_index| will specify the top interesting |
| // spatial layer and all layers bellow, including target one will be |
| // processed. All layers above target one will be dropped. |
| absl::optional<int> target_spatial_index; |
| // If specified the input stream will be also copied to specified file. |
| // It is actually one of the test's output file, which contains copy of what |
| // was captured during the test for this video stream on sender side. |
| // It is useful when generator is used as input. |
| 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. The produced files contains |
| // what was rendered for this video stream on receiver side. |
| absl::optional<std::string> output_dump_file_name; |
| }; |
| |
| // Contains properties for audio in the call. |
| struct AudioConfig { |
| enum Mode { |
| kGenerated, |
| kFile, |
| }; |
| // Have to be unique among all specified configs for all peers in the call. |
| // Will be auto generated if omitted. |
| absl::optional<std::string> stream_label; |
| Mode mode = kGenerated; |
| // 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_dump_file_name; |
| // Audio options to use. |
| cricket::AudioOptions audio_options; |
| }; |
| |
| // This class is used to fully configure one peer inside the call. |
| class PeerConfigurer { |
| public: |
| virtual ~PeerConfigurer() = default; |
| |
| // The parameters of the following 7 methods will be passed to the |
| // PeerConnectionFactoryInterface implementation that will be created for |
| // this peer. |
| virtual PeerConfigurer* SetCallFactory( |
| std::unique_ptr<CallFactoryInterface> call_factory) = 0; |
| virtual PeerConfigurer* SetEventLogFactory( |
| std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) = 0; |
| virtual PeerConfigurer* SetFecControllerFactory( |
| std::unique_ptr<FecControllerFactoryInterface> |
| fec_controller_factory) = 0; |
| virtual PeerConfigurer* SetNetworkControllerFactory( |
| std::unique_ptr<NetworkControllerFactoryInterface> |
| network_controller_factory) = 0; |
| virtual PeerConfigurer* SetMediaTransportFactory( |
| std::unique_ptr<MediaTransportFactory> media_transport_factory) = 0; |
| virtual PeerConfigurer* SetVideoEncoderFactory( |
| std::unique_ptr<VideoEncoderFactory> video_encoder_factory) = 0; |
| virtual PeerConfigurer* SetVideoDecoderFactory( |
| std::unique_ptr<VideoDecoderFactory> video_decoder_factory) = 0; |
| |
| // The parameters of the following 3 methods will be passed to the |
| // PeerConnectionInterface implementation that will be created for this |
| // peer. |
| virtual PeerConfigurer* SetAsyncResolverFactory( |
| std::unique_ptr<webrtc::AsyncResolverFactory> |
| async_resolver_factory) = 0; |
| virtual PeerConfigurer* SetRTCCertificateGenerator( |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> |
| cert_generator) = 0; |
| virtual PeerConfigurer* SetSSLCertificateVerifier( |
| std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier) = 0; |
| |
| // Add new video stream to the call that will be sent from this peer. |
| virtual PeerConfigurer* AddVideoConfig(VideoConfig config) = 0; |
| // Set the audio stream for the call from this peer. If this method won't |
| // be invoked, this peer will send no audio. |
| virtual PeerConfigurer* SetAudioConfig(AudioConfig config) = 0; |
| // If is set, an RTCEventLog will be saved in that location and it will be |
| // available for further analysis. |
| virtual PeerConfigurer* SetRtcEventLogPath(std::string path) = 0; |
| virtual PeerConfigurer* SetRTCConfiguration( |
| PeerConnectionInterface::RTCConfiguration configuration) = 0; |
| }; |
| |
| // Contains parameters, that describe how long framework should run quality |
| // test. |
| struct RunParams { |
| // 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; |
| }; |
| |
| virtual ~PeerConnectionE2EQualityTestFixture() = default; |
| |
| // Add activity that will be executed on the best effort at least after |
| // |target_time_since_start| after call will be set up (after offer/answer |
| // exchange, ICE gathering will be done and ICE candidates will passed to |
| // remote side). |func| param is amount of time spent from the call set up. |
| virtual void ExecuteAt(TimeDelta target_time_since_start, |
| std::function<void(TimeDelta)> func) = 0; |
| // Add activity that will be executed every |interval| with first execution |
| // on the best effort at least after |initial_delay_since_start| after call |
| // will be set up (after all participants will be connected). |func| param is |
| // amount of time spent from the call set up. |
| virtual void ExecuteEvery(TimeDelta initial_delay_since_start, |
| TimeDelta interval, |
| std::function<void(TimeDelta)> func) = 0; |
| |
| // Add a new peer to the call and return an object through which caller |
| // can configure peer's behavior. |
| // |network_thread| will be used as network thread for peer's peer connection |
| // |network_manager| will be used to provide network interfaces for peer's |
| // peer connection. |
| // |configurer| function will be used to configure peer in the call. |
| virtual void AddPeer(rtc::Thread* network_thread, |
| rtc::NetworkManager* network_manager, |
| rtc::FunctionView<void(PeerConfigurer*)> configurer) = 0; |
| virtual void Run(RunParams run_params) = 0; |
| }; |
| |
| } // namespace webrtc_pc_e2e |
| } // namespace webrtc |
| |
| #endif // TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_ |