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Diffstat (limited to 'third_party/libwebrtc/api/test/pclf/media_configuration.h')
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diff --git a/third_party/libwebrtc/api/test/pclf/media_configuration.h b/third_party/libwebrtc/api/test/pclf/media_configuration.h new file mode 100644 index 0000000000..8e841a265b --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/media_configuration.h @@ -0,0 +1,484 @@ +/* + * 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_CONFIGURATION_H_ +#define API_TEST_PCLF_MEDIA_CONFIGURATION_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <functional> +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/memory/memory.h" +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "api/async_resolver_factory.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_options.h" +#include "api/call/call_factory_interface.h" +#include "api/fec_controller.h" +#include "api/function_view.h" +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_event_log/rtc_event_log_factory_interface.h" +#include "api/rtp_parameters.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/test/audio_quality_analyzer_interface.h" +#include "api/test/frame_generator_interface.h" +#include "api/test/peer_network_dependencies.h" +#include "api/test/simulated_network.h" +#include "api/test/stats_observer_interface.h" +#include "api/test/track_id_stream_info_map.h" +#include "api/test/video/video_frame_writer.h" +#include "api/test/video_quality_analyzer_interface.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 "modules/audio_processing/include/audio_processing.h" +#include "rtc_base/checks.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 { + +constexpr size_t kDefaultSlidesWidth = 1850; +constexpr size_t kDefaultSlidesHeight = 1110; + +// The index of required capturing device in OS provided list of video +// devices. On Linux and Windows the list will be obtained via +// webrtc::VideoCaptureModule::DeviceInfo, on Mac OS via +// [RTCCameraVideoCapturer captureDevices]. +enum class CapturingDeviceIndex : size_t {}; + +// Contains parameters for screen share scrolling. +// +// If scrolling is enabled, then it will be done by putting sliding window +// on source video and moving this window from top left corner to the +// bottom right corner of the picture. +// +// In such case source dimensions must be greater or equal to the sliding +// window dimensions. So `source_width` and `source_height` are the dimensions +// of the source frame, while `VideoConfig::width` and `VideoConfig::height` +// are the dimensions of the sliding window. +// +// Because `source_width` and `source_height` are dimensions of the source +// frame, they have to be width and height of videos from +// `ScreenShareConfig::slides_yuv_file_names`. +// +// Because scrolling have to be done on single slide it also requires, that +// `duration` must be less or equal to +// `ScreenShareConfig::slide_change_interval`. +struct ScrollingParams { + // Duration of scrolling. + TimeDelta duration; + // Width of source slides video. + size_t source_width = kDefaultSlidesWidth; + // Height of source slides video. + size_t source_height = kDefaultSlidesHeight; +}; + +// Contains screen share video stream properties. +struct ScreenShareConfig { + explicit ScreenShareConfig(TimeDelta slide_change_interval); + + // Shows how long one slide should be presented on the screen during + // slide generation. + TimeDelta slide_change_interval; + // If true, slides will be generated programmatically. No scrolling params + // will be applied in such case. + bool generate_slides = false; + // If present scrolling will be applied. Please read extra requirement on + // `slides_yuv_file_names` for scrolling. + absl::optional<ScrollingParams> scrolling_params; + // Contains list of yuv files with slides. + // + // If empty, default set of slides will be used. In such case + // `VideoConfig::width` must be equal to `kDefaultSlidesWidth` and + // `VideoConfig::height` must be equal to `kDefaultSlidesHeight` or if + // `scrolling_params` are specified, then `ScrollingParams::source_width` + // must be equal to `kDefaultSlidesWidth` and + // `ScrollingParams::source_height` must be equal to `kDefaultSlidesHeight`. + std::vector<std::string> slides_yuv_file_names; +}; + +// Config for Vp8 simulcast or non-standard Vp9 SVC testing. +// +// To configure standard SVC setting, use `scalability_mode` in the +// `encoding_params` array. +// This configures Vp9 SVC by requesting simulcast layers, the request is +// internally converted to a request for SVC layers. +// +// SVC support is limited: +// During SVC testing there is no SFU, so framework will try to emulate SFU +// behavior in regular p2p call. Because of it there are such limitations: +// * if `target_spatial_index` is not equal to the highest spatial layer +// then no packet/frame drops are allowed. +// +// If there will be any drops, that will affect requested layer, then +// WebRTC SVC implementation will continue decoding only the highest +// available layer and won't restore lower layers, so analyzer won't +// receive required data which will cause wrong results or test failures. +struct VideoSimulcastConfig { + explicit VideoSimulcastConfig(int simulcast_streams_count); + + // Specified amount of simulcast streams/SVC layers, depending on which + // encoder is used. + int simulcast_streams_count; +}; + +// Configuration for the emulated Selective Forward Unit (SFU) +// +// The framework can optionally filter out frames that are decoded +// using an emulated SFU. +// When using simulcast or SVC, it's not always desirable to receive +// all frames. In a real world call, a SFU will only forward a subset +// of the frames. +// The emulated SFU is not able to change its configuration dynamically, +// if adaptation happens during the call, layers may be dropped and the +// analyzer won't receive the required data which will cause wrong results or +// test failures. +struct EmulatedSFUConfig { + EmulatedSFUConfig() = default; + explicit EmulatedSFUConfig(int target_layer_index); + EmulatedSFUConfig(absl::optional<int> target_layer_index, + absl::optional<int> target_temporal_index); + + // Specifies simulcast or spatial index of the video stream to analyze. + // There are 2 cases: + // 1. simulcast encoding is used: + // in such case `target_layer_index` will specify the index of + // simulcast stream, that should be analyzed. Other streams will be + // dropped. + // 2. SVC encoding is used: + // in such case `target_layer_index` will specify the top interesting + // spatial layer and all layers below, including target one will be + // processed. All layers above target one will be dropped. + // If not specified then all streams will be received and analyzed. + // When set, it instructs the framework to create an emulated Selective + // Forwarding Unit (SFU) that will propagate only the requested layers. + absl::optional<int> target_layer_index; + // Specifies the index of the maximum temporal unit to keep. + // If not specified then all temporal layers will be received and analyzed. + // When set, it instructs the framework to create an emulated Selective + // Forwarding Unit (SFU) that will propagate only up to the requested layer. + absl::optional<int> target_temporal_index; +}; + +class VideoResolution { + public: + // Determines special resolutions, which can't be expressed in terms of + // width, height and fps. + enum class Spec { + // No extra spec set. It describes a regular resolution described by + // width, height and fps. + kNone, + // Describes resolution which contains max value among all sender's + // video streams in each dimension (width, height, fps). + kMaxFromSender + }; + + VideoResolution(size_t width, size_t height, int32_t fps); + explicit VideoResolution(Spec spec = Spec::kNone); + + bool operator==(const VideoResolution& other) const; + bool operator!=(const VideoResolution& other) const; + + size_t width() const { return width_; } + void set_width(size_t width) { width_ = width; } + size_t height() const { return height_; } + void set_height(size_t height) { height_ = height; } + int32_t fps() const { return fps_; } + void set_fps(int32_t fps) { fps_ = fps; } + + // Returns if it is a regular resolution or not. The resolution is regular + // if it's spec is `Spec::kNone`. + bool IsRegular() const; + + std::string ToString() const; + + private: + size_t width_ = 0; + size_t height_ = 0; + int32_t fps_ = 0; + Spec spec_ = Spec::kNone; +}; + +class VideoDumpOptions { + public: + static constexpr int kDefaultSamplingModulo = 1; + + // output_directory - the output directory where stream will be dumped. The + // output files' names will be constructed as + // <stream_name>_<receiver_name>_<resolution>.<extension> for output dumps + // and <stream_name>_<resolution>.<extension> for input dumps. + // By default <extension> is "y4m". Resolution is in the format + // <width>x<height>_<fps>. + // sampling_modulo - the module for the video frames to be dumped. Modulo + // equals X means every Xth frame will be written to the dump file. The + // value must be greater than 0. (Default: 1) + // export_frame_ids - specifies if frame ids should be exported together + // with content of the stream. If true, an output file with the same name as + // video dump and suffix ".frame_ids.txt" will be created. It will contain + // the frame ids in the same order as original frames in the output + // file with stream content. File will contain one frame id per line. + // (Default: false) + // `video_frame_writer_factory` - factory function to create a video frame + // writer for input and output video files. (Default: Y4M video writer + // factory). + explicit VideoDumpOptions( + absl::string_view output_directory, + int sampling_modulo = kDefaultSamplingModulo, + bool export_frame_ids = false, + std::function<std::unique_ptr<test::VideoFrameWriter>( + absl::string_view file_name_prefix, + const VideoResolution& resolution)> video_frame_writer_factory = + Y4mVideoFrameWriterFactory); + VideoDumpOptions(absl::string_view output_directory, bool export_frame_ids); + + VideoDumpOptions(const VideoDumpOptions&) = default; + VideoDumpOptions& operator=(const VideoDumpOptions&) = default; + VideoDumpOptions(VideoDumpOptions&&) = default; + VideoDumpOptions& operator=(VideoDumpOptions&&) = default; + + std::string output_directory() const { return output_directory_; } + int sampling_modulo() const { return sampling_modulo_; } + bool export_frame_ids() const { return export_frame_ids_; } + + std::unique_ptr<test::VideoFrameWriter> CreateInputDumpVideoFrameWriter( + absl::string_view stream_label, + const VideoResolution& resolution) const; + + std::unique_ptr<test::VideoFrameWriter> CreateOutputDumpVideoFrameWriter( + absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const; + + std::string ToString() const; + + private: + static std::unique_ptr<test::VideoFrameWriter> Y4mVideoFrameWriterFactory( + absl::string_view file_name_prefix, + const VideoResolution& resolution); + std::string GetInputDumpFileName(absl::string_view stream_label, + const VideoResolution& resolution) const; + // Returns file name for input frame ids dump if `export_frame_ids()` is + // true, absl::nullopt otherwise. + absl::optional<std::string> GetInputFrameIdsDumpFileName( + absl::string_view stream_label, + const VideoResolution& resolution) const; + std::string GetOutputDumpFileName(absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const; + // Returns file name for output frame ids dump if `export_frame_ids()` is + // true, absl::nullopt otherwise. + absl::optional<std::string> GetOutputFrameIdsDumpFileName( + absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const; + + std::string output_directory_; + int sampling_modulo_ = 1; + bool export_frame_ids_ = false; + std::function<std::unique_ptr<test::VideoFrameWriter>( + absl::string_view file_name_prefix, + const VideoResolution& resolution)> + video_frame_writer_factory_; +}; + +// Contains properties of single video stream. +struct VideoConfig { + explicit VideoConfig(const VideoResolution& resolution); + VideoConfig(size_t width, size_t height, int32_t fps); + VideoConfig(std::string stream_label, + size_t width, + size_t height, + int32_t fps); + + // Video stream width. + size_t width; + // Video stream height. + size_t height; + int32_t fps; + VideoResolution GetResolution() const { + return VideoResolution(width, height, 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; + // Will be set for current video track. If equals to kText or kDetailed - + // screencast in on. + absl::optional<VideoTrackInterface::ContentHint> content_hint; + // If presented video will be transfered in simulcast/SVC mode depending on + // which encoder is used. + // + // Simulcast is supported only from 1st added peer. For VP8 simulcast only + // without RTX is supported so it will be automatically disabled for all + // simulcast tracks. For VP9 simulcast enables VP9 SVC mode and support RTX, + // but only on non-lossy networks. See more in documentation to + // VideoSimulcastConfig. + absl::optional<VideoSimulcastConfig> simulcast_config; + // Configuration for the emulated Selective Forward Unit (SFU). + absl::optional<EmulatedSFUConfig> emulated_sfu_config; + // Encoding parameters for both singlecast and per simulcast layer. + // If singlecast is used, if not empty, a single value can be provided. + // If simulcast is used, if not empty, `encoding_params` size have to be + // equal to `simulcast_config.simulcast_streams_count`. Will be used to set + // transceiver send encoding params for each layer. + // RtpEncodingParameters::rid may be changed by fixture implementation to + // ensure signaling correctness. + std::vector<RtpEncodingParameters> encoding_params; + // Count of temporal layers for video stream. This value will be set into + // each RtpEncodingParameters of RtpParameters of corresponding + // RtpSenderInterface for this video stream. + absl::optional<int> temporal_layers_count; + // If specified defines how input should be dumped. 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<VideoDumpOptions> input_dump_options; + // If specified defines how output should be dumped on the receiver side for + // this stream. The produced files contain what was rendered for this video + // stream on receiver side per each receiver. + absl::optional<VideoDumpOptions> output_dump_options; + // If set to true uses fixed frame rate while dumping output video to the + // file. Requested `VideoSubscription::fps()` will be used as frame rate. + bool output_dump_use_fixed_framerate = false; + // If true will display input and output video on the user's screen. + bool show_on_screen = false; + // If specified, determines a sync group to which this video stream belongs. + // According to bugs.webrtc.org/4762 WebRTC supports synchronization only + // for pair of single audio and single video stream. + absl::optional<std::string> sync_group; + // If specified, it will be set into RtpParameters of corresponding + // RtpSenderInterface for this video stream. + // Note that this setting takes precedence over `content_hint`. + absl::optional<DegradationPreference> degradation_preference; +}; + +// Contains properties for audio in the call. +struct AudioConfig { + enum Mode { + kGenerated, + kFile, + }; + + AudioConfig() = default; + explicit AudioConfig(std::string stream_label); + + // 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; + // Sampling frequency of input audio data (from file or generated). + int sampling_frequency_in_hz = 48000; + // If specified, determines a sync group to which this audio stream belongs. + // According to bugs.webrtc.org/4762 WebRTC supports synchronization only + // for pair of single audio and single video stream. + absl::optional<std::string> sync_group; +}; + +struct VideoCodecConfig { + explicit VideoCodecConfig(std::string name); + VideoCodecConfig(std::string name, + std::map<std::string, std::string> required_params); + // Next two fields are used to specify concrete video codec, that should be + // used in the test. Video code will be negotiated in SDP during offer/ + // answer exchange. + // Video codec name. You can find valid names in + // media/base/media_constants.h + std::string name; + // Map of parameters, that have to be specified on SDP codec. Each parameter + // is described by key and value. Codec parameters will match the specified + // map if and only if for each key from `required_params` there will be + // a parameter with name equal to this key and parameter value will be equal + // to the value from `required_params` for this key. + // If empty then only name will be used to match the codec. + std::map<std::string, std::string> required_params; +}; + +// Subscription to the remote video streams. It declares which remote stream +// peer should receive and in which resolution (width x height x fps). +class VideoSubscription { + public: + // Returns the resolution constructed as maximum from all resolution + // dimensions: width, height and fps. + static absl::optional<VideoResolution> GetMaxResolution( + rtc::ArrayView<const VideoConfig> video_configs); + static absl::optional<VideoResolution> GetMaxResolution( + rtc::ArrayView<const VideoResolution> resolutions); + + bool operator==(const VideoSubscription& other) const; + bool operator!=(const VideoSubscription& other) const; + + // Subscribes receiver to all streams sent by the specified peer with + // specified resolution. It will override any resolution that was used in + // `SubscribeToAll` independently from methods call order. + VideoSubscription& SubscribeToPeer( + absl::string_view peer_name, + VideoResolution resolution = + VideoResolution(VideoResolution::Spec::kMaxFromSender)); + + // Subscribes receiver to the all sent streams with specified resolution. + // If any stream was subscribed to with `SubscribeTo` method that will + // override resolution passed to this function independently from methods + // call order. + VideoSubscription& SubscribeToAllPeers( + VideoResolution resolution = + VideoResolution(VideoResolution::Spec::kMaxFromSender)); + + // Returns resolution for specific sender. If no specific resolution was + // set for this sender, then will return resolution used for all streams. + // If subscription doesn't subscribe to all streams, `absl::nullopt` will be + // returned. + absl::optional<VideoResolution> GetResolutionForPeer( + absl::string_view peer_name) const; + + // Returns a maybe empty list of senders for which peer explicitly + // subscribed to with specific resolution. + std::vector<std::string> GetSubscribedPeers() const; + + std::string ToString() const; + + private: + absl::optional<VideoResolution> default_resolution_ = absl::nullopt; + std::map<std::string, VideoResolution> peers_resolution_; +}; + +// Contains configuration for echo emulator. +struct EchoEmulationConfig { + // Delay which represents the echo path delay, i.e. how soon rendered signal + // should reach capturer. + TimeDelta echo_delay = TimeDelta::Millis(50); +}; + +} // namespace webrtc_pc_e2e +} // namespace webrtc + +#endif // API_TEST_PCLF_MEDIA_CONFIGURATION_H_ |