diff options
Diffstat (limited to 'third_party/libwebrtc/api/test/pclf')
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/BUILD.gn | 114 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/DEPS | 13 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/media_configuration.cc | 314 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/media_configuration.h | 484 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/media_quality_test_params.h | 189 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/peer_configurer.cc | 245 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/test/pclf/peer_configurer.h | 200 |
7 files changed, 1559 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/test/pclf/BUILD.gn b/third_party/libwebrtc/api/test/pclf/BUILD.gn new file mode 100644 index 0000000000..0526478e8b --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/BUILD.gn @@ -0,0 +1,114 @@ +# 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. + +import("../../../webrtc.gni") + +rtc_source_set("media_configuration") { + visibility = [ "*" ] + testonly = true + sources = [ + "media_configuration.cc", + "media_configuration.h", + ] + + deps = [ + "../..:array_view", + "../..:audio_options_api", + "../..:audio_quality_analyzer_api", + "../..:callfactory_api", + "../..:fec_controller_api", + "../..:frame_generator_api", + "../..:function_view", + "../..:libjingle_peerconnection_api", + "../..:media_stream_interface", + "../..:packet_socket_factory", + "../..:peer_network_dependencies", + "../..:rtp_parameters", + "../..:simulated_network_api", + "../..:stats_observer_interface", + "../..:track_id_stream_info_map", + "../..:video_quality_analyzer_api", + "../../../modules/audio_processing:api", + "../../../rtc_base:checks", + "../../../rtc_base:network", + "../../../rtc_base:rtc_certificate_generator", + "../../../rtc_base:ssl", + "../../../rtc_base:stringutils", + "../../../rtc_base:threading", + "../../../test:fileutils", + "../../../test:video_test_support", + "../../../test/pc/e2e/analyzer/video:video_dumping", + "../../audio:audio_mixer_api", + "../../rtc_event_log", + "../../task_queue", + "../../transport:network_control", + "../../units:time_delta", + "../../video_codecs:video_codecs_api", + "../video:video_frame_writer", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("media_quality_test_params") { + visibility = [ "*" ] + testonly = true + sources = [ "media_quality_test_params.h" ] + + deps = [ + ":media_configuration", + "../../../api:callfactory_api", + "../../../api:fec_controller_api", + "../../../api:field_trials_view", + "../../../api:libjingle_peerconnection_api", + "../../../api:packet_socket_factory", + "../../../api/audio:audio_mixer_api", + "../../../api/rtc_event_log", + "../../../api/task_queue", + "../../../api/transport:network_control", + "../../../api/video_codecs:video_codecs_api", + "../../../modules/audio_processing:api", + "../../../p2p:rtc_p2p", + "../../../rtc_base:network", + "../../../rtc_base:rtc_certificate_generator", + "../../../rtc_base:ssl", + "../../../rtc_base:threading", + ] +} + +rtc_library("peer_configurer") { + visibility = [ "*" ] + testonly = true + sources = [ + "peer_configurer.cc", + "peer_configurer.h", + ] + deps = [ + ":media_configuration", + ":media_quality_test_params", + "../../../api:callfactory_api", + "../../../api:create_peer_connection_quality_test_frame_generator", + "../../../api:fec_controller_api", + "../../../api:packet_socket_factory", + "../../../api:peer_network_dependencies", + "../../../api/audio:audio_mixer_api", + "../../../api/rtc_event_log", + "../../../api/task_queue", + "../../../api/transport:network_control", + "../../../api/video_codecs:video_codecs_api", + "../../../modules/audio_processing:api", + "../../../rtc_base:network", + "../../../rtc_base:rtc_certificate_generator", + "../../../rtc_base:ssl", + "../../../rtc_base:threading", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] +} diff --git a/third_party/libwebrtc/api/test/pclf/DEPS b/third_party/libwebrtc/api/test/pclf/DEPS new file mode 100644 index 0000000000..60cc0aeeb3 --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/DEPS @@ -0,0 +1,13 @@ +specific_include_rules = { + ".*": [ + "+modules/audio_processing/include/audio_processing.h", + "+rtc_base/checks.h", + "+rtc_base/network.h", + "+rtc_base/rtc_certificate_generator.h", + "+rtc_base/ssl_certificate.h", + "+rtc_base/thread.h", + ], + "media_quality_test_params\.h": [ + "+p2p/base/port_allocator.h", + ], +} diff --git a/third_party/libwebrtc/api/test/pclf/media_configuration.cc b/third_party/libwebrtc/api/test/pclf/media_configuration.cc new file mode 100644 index 0000000000..56b9e52e01 --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/media_configuration.cc @@ -0,0 +1,314 @@ +/* + * Copyright 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. + */ + +#include "api/test/pclf/media_configuration.h" + +#include <string> +#include <utility> + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "api/test/video/video_frame_writer.h" +#include "rtc_base/checks.h" +#include "rtc_base/strings/string_builder.h" +#include "test/pc/e2e/analyzer/video/video_dumping.h" +#include "test/testsupport/file_utils.h" +#include "test/testsupport/video_frame_writer.h" + +namespace webrtc { +namespace webrtc_pc_e2e { +namespace { + +std::string SpecToString(VideoResolution::Spec spec) { + switch (spec) { + case VideoResolution::Spec::kNone: + return "None"; + case VideoResolution::Spec::kMaxFromSender: + return "MaxFromSender"; + } +} + +void AppendResolution(const VideoResolution& resolution, + rtc::StringBuilder& builder) { + builder << "_" << resolution.width() << "x" << resolution.height() << "_" + << resolution.fps(); +} + +} // namespace + +ScreenShareConfig::ScreenShareConfig(TimeDelta slide_change_interval) + : slide_change_interval(slide_change_interval) { + RTC_CHECK_GT(slide_change_interval.ms(), 0); +} +VideoSimulcastConfig::VideoSimulcastConfig(int simulcast_streams_count) + : simulcast_streams_count(simulcast_streams_count) { + RTC_CHECK_GT(simulcast_streams_count, 1); +} +EmulatedSFUConfig::EmulatedSFUConfig(int target_layer_index) + : target_layer_index(target_layer_index) { + RTC_CHECK_GE(target_layer_index, 0); +} + +EmulatedSFUConfig::EmulatedSFUConfig(absl::optional<int> target_layer_index, + absl::optional<int> target_temporal_index) + : target_layer_index(target_layer_index), + target_temporal_index(target_temporal_index) { + RTC_CHECK_GE(target_temporal_index.value_or(0), 0); + if (target_temporal_index) + RTC_CHECK_GE(*target_temporal_index, 0); +} + +VideoResolution::VideoResolution(size_t width, size_t height, int32_t fps) + : width_(width), height_(height), fps_(fps), spec_(Spec::kNone) {} +VideoResolution::VideoResolution(Spec spec) + : width_(0), height_(0), fps_(0), spec_(spec) {} + +bool VideoResolution::operator==(const VideoResolution& other) const { + if (spec_ != Spec::kNone && spec_ == other.spec_) { + // If there is some particular spec set, then it doesn't matter what + // values we have in other fields. + return true; + } + return width_ == other.width_ && height_ == other.height_ && + fps_ == other.fps_ && spec_ == other.spec_; +} +bool VideoResolution::operator!=(const VideoResolution& other) const { + return !(*this == other); +} + +bool VideoResolution::IsRegular() const { + return spec_ == Spec::kNone; +} +std::string VideoResolution::ToString() const { + rtc::StringBuilder out; + out << "{ width=" << width_ << ", height=" << height_ << ", fps=" << fps_ + << ", spec=" << SpecToString(spec_) << " }"; + return out.Release(); +} + +VideoDumpOptions::VideoDumpOptions( + absl::string_view output_directory, + int sampling_modulo, + bool export_frame_ids, + std::function<std::unique_ptr<test::VideoFrameWriter>( + absl::string_view file_name_prefix, + const VideoResolution& resolution)> video_frame_writer_factory) + : output_directory_(output_directory), + sampling_modulo_(sampling_modulo), + export_frame_ids_(export_frame_ids), + video_frame_writer_factory_(video_frame_writer_factory) { + RTC_CHECK_GT(sampling_modulo, 0); +} + +VideoDumpOptions::VideoDumpOptions(absl::string_view output_directory, + bool export_frame_ids) + : VideoDumpOptions(output_directory, + kDefaultSamplingModulo, + export_frame_ids) {} + +std::unique_ptr<test::VideoFrameWriter> +VideoDumpOptions::CreateInputDumpVideoFrameWriter( + absl::string_view stream_label, + const VideoResolution& resolution) const { + std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_( + GetInputDumpFileName(stream_label, resolution), resolution); + absl::optional<std::string> frame_ids_file = + GetInputFrameIdsDumpFileName(stream_label, resolution); + if (frame_ids_file.has_value()) { + writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file); + } + return writer; +} + +std::unique_ptr<test::VideoFrameWriter> +VideoDumpOptions::CreateOutputDumpVideoFrameWriter( + absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const { + std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_( + GetOutputDumpFileName(stream_label, receiver, resolution), resolution); + absl::optional<std::string> frame_ids_file = + GetOutputFrameIdsDumpFileName(stream_label, receiver, resolution); + if (frame_ids_file.has_value()) { + writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file); + } + return writer; +} + +std::unique_ptr<test::VideoFrameWriter> +VideoDumpOptions::Y4mVideoFrameWriterFactory( + absl::string_view file_name_prefix, + const VideoResolution& resolution) { + return std::make_unique<test::Y4mVideoFrameWriterImpl>( + std::string(file_name_prefix) + ".y4m", resolution.width(), + resolution.height(), resolution.fps()); +} + +std::string VideoDumpOptions::GetInputDumpFileName( + absl::string_view stream_label, + const VideoResolution& resolution) const { + rtc::StringBuilder file_name; + file_name << stream_label; + AppendResolution(resolution, file_name); + return test::JoinFilename(output_directory_, file_name.Release()); +} + +absl::optional<std::string> VideoDumpOptions::GetInputFrameIdsDumpFileName( + absl::string_view stream_label, + const VideoResolution& resolution) const { + if (!export_frame_ids_) { + return absl::nullopt; + } + return GetInputDumpFileName(stream_label, resolution) + ".frame_ids.txt"; +} + +std::string VideoDumpOptions::GetOutputDumpFileName( + absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const { + rtc::StringBuilder file_name; + file_name << stream_label << "_" << receiver; + AppendResolution(resolution, file_name); + return test::JoinFilename(output_directory_, file_name.Release()); +} + +absl::optional<std::string> VideoDumpOptions::GetOutputFrameIdsDumpFileName( + absl::string_view stream_label, + absl::string_view receiver, + const VideoResolution& resolution) const { + if (!export_frame_ids_) { + return absl::nullopt; + } + return GetOutputDumpFileName(stream_label, receiver, resolution) + + ".frame_ids.txt"; +} + +std::string VideoDumpOptions::ToString() const { + rtc::StringBuilder out; + out << "{ output_directory_=" << output_directory_ + << ", sampling_modulo_=" << sampling_modulo_ + << ", export_frame_ids_=" << export_frame_ids_ << " }"; + return out.Release(); +} + +VideoConfig::VideoConfig(const VideoResolution& resolution) + : width(resolution.width()), + height(resolution.height()), + fps(resolution.fps()) { + RTC_CHECK(resolution.IsRegular()); +} +VideoConfig::VideoConfig(size_t width, size_t height, int32_t fps) + : width(width), height(height), fps(fps) {} +VideoConfig::VideoConfig(std::string stream_label, + size_t width, + size_t height, + int32_t fps) + : width(width), + height(height), + fps(fps), + stream_label(std::move(stream_label)) {} + +AudioConfig::AudioConfig(std::string stream_label) + : stream_label(std::move(stream_label)) {} + +VideoCodecConfig::VideoCodecConfig(std::string name) + : name(std::move(name)), required_params() {} +VideoCodecConfig::VideoCodecConfig( + std::string name, + std::map<std::string, std::string> required_params) + : name(std::move(name)), required_params(std::move(required_params)) {} + +absl::optional<VideoResolution> VideoSubscription::GetMaxResolution( + rtc::ArrayView<const VideoConfig> video_configs) { + std::vector<VideoResolution> resolutions; + for (const auto& video_config : video_configs) { + resolutions.push_back(video_config.GetResolution()); + } + return GetMaxResolution(resolutions); +} + +absl::optional<VideoResolution> VideoSubscription::GetMaxResolution( + rtc::ArrayView<const VideoResolution> resolutions) { + if (resolutions.empty()) { + return absl::nullopt; + } + + VideoResolution max_resolution; + for (const VideoResolution& resolution : resolutions) { + if (max_resolution.width() < resolution.width()) { + max_resolution.set_width(resolution.width()); + } + if (max_resolution.height() < resolution.height()) { + max_resolution.set_height(resolution.height()); + } + if (max_resolution.fps() < resolution.fps()) { + max_resolution.set_fps(resolution.fps()); + } + } + return max_resolution; +} + +bool VideoSubscription::operator==(const VideoSubscription& other) const { + return default_resolution_ == other.default_resolution_ && + peers_resolution_ == other.peers_resolution_; +} +bool VideoSubscription::operator!=(const VideoSubscription& other) const { + return !(*this == other); +} + +VideoSubscription& VideoSubscription::SubscribeToPeer( + absl::string_view peer_name, + VideoResolution resolution) { + peers_resolution_[std::string(peer_name)] = resolution; + return *this; +} + +VideoSubscription& VideoSubscription::SubscribeToAllPeers( + VideoResolution resolution) { + default_resolution_ = resolution; + return *this; +} + +absl::optional<VideoResolution> VideoSubscription::GetResolutionForPeer( + absl::string_view peer_name) const { + auto it = peers_resolution_.find(std::string(peer_name)); + if (it == peers_resolution_.end()) { + return default_resolution_; + } + return it->second; +} + +std::vector<std::string> VideoSubscription::GetSubscribedPeers() const { + std::vector<std::string> subscribed_streams; + subscribed_streams.reserve(peers_resolution_.size()); + for (const auto& entry : peers_resolution_) { + subscribed_streams.push_back(entry.first); + } + return subscribed_streams; +} + +std::string VideoSubscription::ToString() const { + rtc::StringBuilder out; + out << "{ default_resolution_=["; + if (default_resolution_.has_value()) { + out << default_resolution_->ToString(); + } else { + out << "undefined"; + } + out << "], {"; + for (const auto& [peer_name, resolution] : peers_resolution_) { + out << "[" << peer_name << ": " << resolution.ToString() << "], "; + } + out << "} }"; + return out.Release(); +} +} // namespace webrtc_pc_e2e +} // namespace webrtc 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_ diff --git a/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h b/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h new file mode 100644 index 0000000000..3377d31f68 --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h @@ -0,0 +1,189 @@ +/* + * 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_resolver_factory.h" +#include "api/audio/audio_mixer.h" +#include "api/call/call_factory_interface.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/task_queue/task_queue_factory.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 "modules/audio_processing/include/audio_processing.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<TaskQueueFactory> task_queue_factory; + 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<NetEqFactory> neteq_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; + + 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::AsyncResolverFactory> async_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; + + 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_ diff --git a/third_party/libwebrtc/api/test/pclf/peer_configurer.cc b/third_party/libwebrtc/api/test/pclf/peer_configurer.cc new file mode 100644 index 0000000000..2203a4c4f0 --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/peer_configurer.cc @@ -0,0 +1,245 @@ +/* + * 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. + */ + +#include "api/test/pclf/peer_configurer.h" + +#include <set> + +#include "absl/strings/string_view.h" +#include "api/test/pclf/media_configuration.h" +#include "api/test/pclf/media_quality_test_params.h" +#include "api/test/peer_network_dependencies.h" + +namespace webrtc { +namespace webrtc_pc_e2e { + +PeerConfigurer::PeerConfigurer( + const PeerNetworkDependencies& network_dependencies) + : components_(std::make_unique<InjectableComponents>( + network_dependencies.network_thread, + network_dependencies.network_manager, + network_dependencies.packet_socket_factory)), + params_(std::make_unique<Params>()), + configurable_params_(std::make_unique<ConfigurableParams>()) {} + +PeerConfigurer* PeerConfigurer::SetName(absl::string_view name) { + params_->name = std::string(name); + return this; +} + +PeerConfigurer* PeerConfigurer::SetTaskQueueFactory( + std::unique_ptr<TaskQueueFactory> task_queue_factory) { + components_->pcf_dependencies->task_queue_factory = + std::move(task_queue_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetCallFactory( + std::unique_ptr<CallFactoryInterface> call_factory) { + components_->pcf_dependencies->call_factory = std::move(call_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetEventLogFactory( + std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) { + components_->pcf_dependencies->event_log_factory = + std::move(event_log_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetFecControllerFactory( + std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) { + components_->pcf_dependencies->fec_controller_factory = + std::move(fec_controller_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetNetworkControllerFactory( + std::unique_ptr<NetworkControllerFactoryInterface> + network_controller_factory) { + components_->pcf_dependencies->network_controller_factory = + std::move(network_controller_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetVideoEncoderFactory( + std::unique_ptr<VideoEncoderFactory> video_encoder_factory) { + components_->pcf_dependencies->video_encoder_factory = + std::move(video_encoder_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetVideoDecoderFactory( + std::unique_ptr<VideoDecoderFactory> video_decoder_factory) { + components_->pcf_dependencies->video_decoder_factory = + std::move(video_decoder_factory); + return this; +} + +PeerConfigurer* PeerConfigurer::SetAsyncResolverFactory( + std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory) { + components_->pc_dependencies->async_resolver_factory = + std::move(async_resolver_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetRTCCertificateGenerator( + std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) { + components_->pc_dependencies->cert_generator = std::move(cert_generator); + return this; +} +PeerConfigurer* PeerConfigurer::SetSSLCertificateVerifier( + std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier) { + components_->pc_dependencies->tls_cert_verifier = + std::move(tls_cert_verifier); + return this; +} + +PeerConfigurer* PeerConfigurer::AddVideoConfig(VideoConfig config) { + video_sources_.push_back( + CreateSquareFrameGenerator(config, /*type=*/absl::nullopt)); + configurable_params_->video_configs.push_back(std::move(config)); + return this; +} +PeerConfigurer* PeerConfigurer::AddVideoConfig( + VideoConfig config, + std::unique_ptr<test::FrameGeneratorInterface> generator) { + configurable_params_->video_configs.push_back(std::move(config)); + video_sources_.push_back(std::move(generator)); + return this; +} +PeerConfigurer* PeerConfigurer::AddVideoConfig(VideoConfig config, + CapturingDeviceIndex index) { + configurable_params_->video_configs.push_back(std::move(config)); + video_sources_.push_back(index); + return this; +} +PeerConfigurer* PeerConfigurer::SetVideoSubscription( + VideoSubscription subscription) { + configurable_params_->video_subscription = std::move(subscription); + return this; +} +PeerConfigurer* PeerConfigurer::SetVideoCodecs( + std::vector<VideoCodecConfig> video_codecs) { + params_->video_codecs = std::move(video_codecs); + return this; +} +PeerConfigurer* PeerConfigurer::SetExtraVideoRtpHeaderExtensions( + std::vector<std::string> extensions) { + params_->extra_video_rtp_header_extensions = std::move(extensions); + return this; +} +PeerConfigurer* PeerConfigurer::SetAudioConfig(AudioConfig config) { + params_->audio_config = std::move(config); + return this; +} +PeerConfigurer* PeerConfigurer::SetExtraAudioRtpHeaderExtensions( + std::vector<std::string> extensions) { + params_->extra_audio_rtp_header_extensions = std::move(extensions); + return this; +} +PeerConfigurer* PeerConfigurer::SetUseUlpFEC(bool value) { + params_->use_ulp_fec = value; + return this; +} +PeerConfigurer* PeerConfigurer::SetUseFlexFEC(bool value) { + params_->use_flex_fec = value; + return this; +} +PeerConfigurer* PeerConfigurer::SetVideoEncoderBitrateMultiplier( + double multiplier) { + params_->video_encoder_bitrate_multiplier = multiplier; + return this; +} +PeerConfigurer* PeerConfigurer::SetNetEqFactory( + std::unique_ptr<NetEqFactory> neteq_factory) { + components_->pcf_dependencies->neteq_factory = std::move(neteq_factory); + return this; +} +PeerConfigurer* PeerConfigurer::SetAudioProcessing( + rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) { + components_->pcf_dependencies->audio_processing = audio_processing; + return this; +} +PeerConfigurer* PeerConfigurer::SetAudioMixer( + rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) { + components_->pcf_dependencies->audio_mixer = audio_mixer; + return this; +} + +PeerConfigurer* PeerConfigurer::SetUseNetworkThreadAsWorkerThread() { + components_->worker_thread = components_->network_thread; + return this; +} + +PeerConfigurer* PeerConfigurer::SetRtcEventLogPath(std::string path) { + params_->rtc_event_log_path = std::move(path); + return this; +} +PeerConfigurer* PeerConfigurer::SetAecDumpPath(std::string path) { + params_->aec_dump_path = std::move(path); + return this; +} +PeerConfigurer* PeerConfigurer::SetRTCConfiguration( + PeerConnectionInterface::RTCConfiguration configuration) { + params_->rtc_configuration = std::move(configuration); + return this; +} +PeerConfigurer* PeerConfigurer::SetRTCOfferAnswerOptions( + PeerConnectionInterface::RTCOfferAnswerOptions options) { + params_->rtc_offer_answer_options = std::move(options); + return this; +} +PeerConfigurer* PeerConfigurer::SetBitrateSettings( + BitrateSettings bitrate_settings) { + params_->bitrate_settings = bitrate_settings; + return this; +} + +PeerConfigurer* PeerConfigurer::SetIceTransportFactory( + std::unique_ptr<IceTransportFactory> factory) { + components_->pc_dependencies->ice_transport_factory = std::move(factory); + return this; +} + +PeerConfigurer* PeerConfigurer::SetPortAllocatorExtraFlags( + uint32_t extra_flags) { + params_->port_allocator_extra_flags = extra_flags; + return this; +} +std::unique_ptr<InjectableComponents> PeerConfigurer::ReleaseComponents() { + RTC_CHECK(components_); + auto components = std::move(components_); + components_ = nullptr; + return components; +} + +// Returns Params and transfer ownership to the caller. +// Can be called once. +std::unique_ptr<Params> PeerConfigurer::ReleaseParams() { + RTC_CHECK(params_); + auto params = std::move(params_); + params_ = nullptr; + return params; +} + +// Returns ConfigurableParams and transfer ownership to the caller. +// Can be called once. +std::unique_ptr<ConfigurableParams> +PeerConfigurer::ReleaseConfigurableParams() { + RTC_CHECK(configurable_params_); + auto configurable_params = std::move(configurable_params_); + configurable_params_ = nullptr; + return configurable_params; +} + +// Returns video sources and transfer frame generators ownership to the +// caller. Can be called once. +std::vector<PeerConfigurer::VideoSource> PeerConfigurer::ReleaseVideoSources() { + auto video_sources = std::move(video_sources_); + video_sources_.clear(); + return video_sources; +} + +} // namespace webrtc_pc_e2e +} // namespace webrtc diff --git a/third_party/libwebrtc/api/test/pclf/peer_configurer.h b/third_party/libwebrtc/api/test/pclf/peer_configurer.h new file mode 100644 index 0000000000..996dc5cd89 --- /dev/null +++ b/third_party/libwebrtc/api/test/pclf/peer_configurer.h @@ -0,0 +1,200 @@ +/* + * 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_PEER_CONFIGURER_H_ +#define API_TEST_PCLF_PEER_CONFIGURER_H_ + +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/strings/string_view.h" +#include "api/async_resolver_factory.h" +#include "api/audio/audio_mixer.h" +#include "api/call/call_factory_interface.h" +#include "api/fec_controller.h" +#include "api/rtc_event_log/rtc_event_log_factory_interface.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/test/create_peer_connection_quality_test_frame_generator.h" +#include "api/test/pclf/media_configuration.h" +#include "api/test/pclf/media_quality_test_params.h" +#include "api/test/peer_network_dependencies.h" +#include "api/transport/network_control.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "modules/audio_processing/include/audio_processing.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 { + +// This class is used to fully configure one peer inside a call. +class PeerConfigurer { + public: + using VideoSource = + absl::variant<std::unique_ptr<test::FrameGeneratorInterface>, + CapturingDeviceIndex>; + + explicit PeerConfigurer(const PeerNetworkDependencies& network_dependencies); + + // Sets peer name that will be used to report metrics related to this peer. + // If not set, some default name will be assigned. All names have to be + // unique. + PeerConfigurer* SetName(absl::string_view name); + + // The parameters of the following 9 methods will be passed to the + // PeerConnectionFactoryInterface implementation that will be created for + // this peer. + PeerConfigurer* SetTaskQueueFactory( + std::unique_ptr<TaskQueueFactory> task_queue_factory); + PeerConfigurer* SetCallFactory( + std::unique_ptr<CallFactoryInterface> call_factory); + PeerConfigurer* SetEventLogFactory( + std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory); + PeerConfigurer* SetFecControllerFactory( + std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory); + PeerConfigurer* SetNetworkControllerFactory( + std::unique_ptr<NetworkControllerFactoryInterface> + network_controller_factory); + PeerConfigurer* SetVideoEncoderFactory( + std::unique_ptr<VideoEncoderFactory> video_encoder_factory); + PeerConfigurer* SetVideoDecoderFactory( + std::unique_ptr<VideoDecoderFactory> video_decoder_factory); + // Set a custom NetEqFactory to be used in the call. + PeerConfigurer* SetNetEqFactory(std::unique_ptr<NetEqFactory> neteq_factory); + PeerConfigurer* SetAudioProcessing( + rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing); + PeerConfigurer* SetAudioMixer( + rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer); + + // Forces the Peerconnection to use the network thread as the worker thread. + // Ie, worker thread and the network thread is the same thread. + PeerConfigurer* SetUseNetworkThreadAsWorkerThread(); + + // The parameters of the following 4 methods will be passed to the + // PeerConnectionInterface implementation that will be created for this + // peer. + PeerConfigurer* SetAsyncResolverFactory( + std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory); + PeerConfigurer* SetRTCCertificateGenerator( + std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator); + PeerConfigurer* SetSSLCertificateVerifier( + std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier); + PeerConfigurer* SetIceTransportFactory( + std::unique_ptr<IceTransportFactory> factory); + // Flags to set on `cricket::PortAllocator`. These flags will be added + // to the default ones that are presented on the port allocator. + // For possible values check p2p/base/port_allocator.h. + PeerConfigurer* SetPortAllocatorExtraFlags(uint32_t extra_flags); + + // Add new video stream to the call that will be sent from this peer. + // Default implementation of video frames generator will be used. + PeerConfigurer* AddVideoConfig(VideoConfig config); + // Add new video stream to the call that will be sent from this peer with + // provided own implementation of video frames generator. + PeerConfigurer* AddVideoConfig( + VideoConfig config, + std::unique_ptr<test::FrameGeneratorInterface> generator); + // Add new video stream to the call that will be sent from this peer. + // Capturing device with specified index will be used to get input video. + PeerConfigurer* AddVideoConfig(VideoConfig config, + CapturingDeviceIndex capturing_device_index); + // Sets video subscription for the peer. By default subscription will + // include all streams with `VideoSubscription::kSameAsSendStream` + // resolution. To this behavior use this method. + PeerConfigurer* SetVideoSubscription(VideoSubscription subscription); + // Sets the list of video codecs used by the peer during the test. These + // codecs will be negotiated in SDP during offer/answer exchange. The order + // of these codecs during negotiation will be the same as in `video_codecs`. + // Codecs have to be available in codecs list provided by peer connection to + // be negotiated. If some of specified codecs won't be found, the test will + // crash. + PeerConfigurer* SetVideoCodecs(std::vector<VideoCodecConfig> video_codecs); + // Sets a list of RTP header extensions which will be enforced on all video + // streams added to this peer. + PeerConfigurer* SetExtraVideoRtpHeaderExtensions( + std::vector<std::string> extensions); + // Sets the audio stream for the call from this peer. If this method won't + // be invoked, this peer will send no audio. + PeerConfigurer* SetAudioConfig(AudioConfig config); + // Sets a list of RTP header extensions which will be enforced on all audio + // streams added to this peer. + PeerConfigurer* SetExtraAudioRtpHeaderExtensions( + std::vector<std::string> extensions); + + // Set if ULP FEC should be used or not. False by default. + PeerConfigurer* SetUseUlpFEC(bool value); + // Set if Flex FEC should be used or not. False by default. + // Client also must enable `enable_flex_fec_support` in the `RunParams` to + // be able to use this feature. + PeerConfigurer* SetUseFlexFEC(bool value); + // Specifies how much video encoder target bitrate should be different than + // target bitrate, provided by WebRTC stack. Must be greater than 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(...). 1.0 by default. + PeerConfigurer* SetVideoEncoderBitrateMultiplier(double multiplier); + + // If is set, an RTCEventLog will be saved in that location and it will be + // available for further analysis. + PeerConfigurer* SetRtcEventLogPath(std::string path); + // If is set, an AEC dump will be saved in that location and it will be + // available for further analysis. + PeerConfigurer* SetAecDumpPath(std::string path); + PeerConfigurer* SetRTCConfiguration( + PeerConnectionInterface::RTCConfiguration configuration); + PeerConfigurer* SetRTCOfferAnswerOptions( + PeerConnectionInterface::RTCOfferAnswerOptions options); + // Set bitrate parameters on PeerConnection. This constraints will be + // applied to all summed RTP streams for this peer. + PeerConfigurer* SetBitrateSettings(BitrateSettings bitrate_settings); + + // Returns InjectableComponents and transfer ownership to the caller. + // Can be called once. + std::unique_ptr<InjectableComponents> ReleaseComponents(); + + // Returns Params and transfer ownership to the caller. + // Can be called once. + std::unique_ptr<Params> ReleaseParams(); + + // Returns ConfigurableParams and transfer ownership to the caller. + // Can be called once. + std::unique_ptr<ConfigurableParams> ReleaseConfigurableParams(); + + // Returns video sources and transfer frame generators ownership to the + // caller. Can be called once. + std::vector<VideoSource> ReleaseVideoSources(); + + InjectableComponents* components() { return components_.get(); } + Params* params() { return params_.get(); } + ConfigurableParams* configurable_params() { + return configurable_params_.get(); + } + const Params& params() const { return *params_; } + const ConfigurableParams& configurable_params() const { + return *configurable_params_; + } + std::vector<VideoSource>* video_sources() { return &video_sources_; } + + private: + std::unique_ptr<InjectableComponents> components_; + std::unique_ptr<Params> params_; + std::unique_ptr<ConfigurableParams> configurable_params_; + std::vector<VideoSource> video_sources_; +}; + +} // namespace webrtc_pc_e2e +} // namespace webrtc + +#endif // API_TEST_PCLF_PEER_CONFIGURER_H_ |