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The Mode is currently redundant with the optional input_file_name. Change-Id: Ib4f0a363e86d925107d61867a7f743d6663e7071 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298743 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#39754}
478 lines
20 KiB
C++
478 lines
20 KiB
C++
/*
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* Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_TEST_PCLF_MEDIA_CONFIGURATION_H_
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#define API_TEST_PCLF_MEDIA_CONFIGURATION_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <functional>
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#include <map>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "absl/strings/string_view.h"
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/async_resolver_factory.h"
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#include "api/audio/audio_mixer.h"
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#include "api/audio_options.h"
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#include "api/call/call_factory_interface.h"
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#include "api/fec_controller.h"
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#include "api/function_view.h"
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#include "api/media_stream_interface.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
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#include "api/rtp_parameters.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "api/test/audio_quality_analyzer_interface.h"
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#include "api/test/frame_generator_interface.h"
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#include "api/test/peer_network_dependencies.h"
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#include "api/test/simulated_network.h"
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#include "api/test/stats_observer_interface.h"
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#include "api/test/track_id_stream_info_map.h"
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#include "api/test/video/video_frame_writer.h"
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#include "api/test/video_quality_analyzer_interface.h"
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#include "api/transport/network_control.h"
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#include "api/units/time_delta.h"
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#include "api/video_codecs/video_decoder_factory.h"
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#include "api/video_codecs/video_encoder.h"
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#include "api/video_codecs/video_encoder_factory.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/network.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "rtc_base/ssl_certificate.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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namespace webrtc_pc_e2e {
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constexpr size_t kDefaultSlidesWidth = 1850;
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constexpr size_t kDefaultSlidesHeight = 1110;
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// The index of required capturing device in OS provided list of video
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// devices. On Linux and Windows the list will be obtained via
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// webrtc::VideoCaptureModule::DeviceInfo, on Mac OS via
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// [RTCCameraVideoCapturer captureDevices].
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enum class CapturingDeviceIndex : size_t {};
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// Contains parameters for screen share scrolling.
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//
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// If scrolling is enabled, then it will be done by putting sliding window
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// on source video and moving this window from top left corner to the
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// bottom right corner of the picture.
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//
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// In such case source dimensions must be greater or equal to the sliding
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// window dimensions. So `source_width` and `source_height` are the dimensions
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// of the source frame, while `VideoConfig::width` and `VideoConfig::height`
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// are the dimensions of the sliding window.
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//
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// Because `source_width` and `source_height` are dimensions of the source
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// frame, they have to be width and height of videos from
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// `ScreenShareConfig::slides_yuv_file_names`.
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//
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// Because scrolling have to be done on single slide it also requires, that
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// `duration` must be less or equal to
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// `ScreenShareConfig::slide_change_interval`.
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struct ScrollingParams {
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// Duration of scrolling.
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TimeDelta duration;
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// Width of source slides video.
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size_t source_width = kDefaultSlidesWidth;
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// Height of source slides video.
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size_t source_height = kDefaultSlidesHeight;
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};
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// Contains screen share video stream properties.
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struct ScreenShareConfig {
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explicit ScreenShareConfig(TimeDelta slide_change_interval);
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// Shows how long one slide should be presented on the screen during
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// slide generation.
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TimeDelta slide_change_interval;
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// If true, slides will be generated programmatically. No scrolling params
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// will be applied in such case.
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bool generate_slides = false;
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// If present scrolling will be applied. Please read extra requirement on
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// `slides_yuv_file_names` for scrolling.
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absl::optional<ScrollingParams> scrolling_params;
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// Contains list of yuv files with slides.
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//
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// If empty, default set of slides will be used. In such case
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// `VideoConfig::width` must be equal to `kDefaultSlidesWidth` and
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// `VideoConfig::height` must be equal to `kDefaultSlidesHeight` or if
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// `scrolling_params` are specified, then `ScrollingParams::source_width`
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// must be equal to `kDefaultSlidesWidth` and
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// `ScrollingParams::source_height` must be equal to `kDefaultSlidesHeight`.
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std::vector<std::string> slides_yuv_file_names;
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};
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// Config for Vp8 simulcast or non-standard Vp9 SVC testing.
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//
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// To configure standard SVC setting, use `scalability_mode` in the
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// `encoding_params` array.
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// This configures Vp9 SVC by requesting simulcast layers, the request is
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// internally converted to a request for SVC layers.
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//
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// SVC support is limited:
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// During SVC testing there is no SFU, so framework will try to emulate SFU
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// behavior in regular p2p call. Because of it there are such limitations:
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// * if `target_spatial_index` is not equal to the highest spatial layer
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// then no packet/frame drops are allowed.
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//
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// If there will be any drops, that will affect requested layer, then
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// WebRTC SVC implementation will continue decoding only the highest
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// available layer and won't restore lower layers, so analyzer won't
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// receive required data which will cause wrong results or test failures.
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struct VideoSimulcastConfig {
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explicit VideoSimulcastConfig(int simulcast_streams_count);
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// Specified amount of simulcast streams/SVC layers, depending on which
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// encoder is used.
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int simulcast_streams_count;
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};
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// Configuration for the emulated Selective Forward Unit (SFU)
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//
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// The framework can optionally filter out frames that are decoded
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// using an emulated SFU.
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// When using simulcast or SVC, it's not always desirable to receive
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// all frames. In a real world call, a SFU will only forward a subset
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// of the frames.
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// The emulated SFU is not able to change its configuration dynamically,
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// if adaptation happens during the call, layers may be dropped and the
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// analyzer won't receive the required data which will cause wrong results or
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// test failures.
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struct EmulatedSFUConfig {
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EmulatedSFUConfig() = default;
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explicit EmulatedSFUConfig(int target_layer_index);
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EmulatedSFUConfig(absl::optional<int> target_layer_index,
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absl::optional<int> target_temporal_index);
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// Specifies simulcast or spatial index of the video stream to analyze.
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// There are 2 cases:
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// 1. simulcast encoding is used:
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// in such case `target_layer_index` will specify the index of
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// simulcast stream, that should be analyzed. Other streams will be
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// dropped.
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// 2. SVC encoding is used:
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// in such case `target_layer_index` will specify the top interesting
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// spatial layer and all layers below, including target one will be
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// processed. All layers above target one will be dropped.
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// If not specified then all streams will be received and analyzed.
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// When set, it instructs the framework to create an emulated Selective
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// Forwarding Unit (SFU) that will propagate only the requested layers.
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absl::optional<int> target_layer_index;
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// Specifies the index of the maximum temporal unit to keep.
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// If not specified then all temporal layers will be received and analyzed.
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// When set, it instructs the framework to create an emulated Selective
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// Forwarding Unit (SFU) that will propagate only up to the requested layer.
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absl::optional<int> target_temporal_index;
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};
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class VideoResolution {
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public:
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// Determines special resolutions, which can't be expressed in terms of
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// width, height and fps.
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enum class Spec {
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// No extra spec set. It describes a regular resolution described by
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// width, height and fps.
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kNone,
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// Describes resolution which contains max value among all sender's
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// video streams in each dimension (width, height, fps).
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kMaxFromSender
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};
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VideoResolution(size_t width, size_t height, int32_t fps);
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explicit VideoResolution(Spec spec = Spec::kNone);
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bool operator==(const VideoResolution& other) const;
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bool operator!=(const VideoResolution& other) const;
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size_t width() const { return width_; }
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void set_width(size_t width) { width_ = width; }
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size_t height() const { return height_; }
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void set_height(size_t height) { height_ = height; }
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int32_t fps() const { return fps_; }
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void set_fps(int32_t fps) { fps_ = fps; }
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// Returns if it is a regular resolution or not. The resolution is regular
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// if it's spec is `Spec::kNone`.
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bool IsRegular() const;
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std::string ToString() const;
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private:
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size_t width_ = 0;
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size_t height_ = 0;
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int32_t fps_ = 0;
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Spec spec_ = Spec::kNone;
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};
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class VideoDumpOptions {
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public:
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static constexpr int kDefaultSamplingModulo = 1;
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// output_directory - the output directory where stream will be dumped. The
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// output files' names will be constructed as
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// <stream_name>_<receiver_name>_<resolution>.<extension> for output dumps
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// and <stream_name>_<resolution>.<extension> for input dumps.
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// By default <extension> is "y4m". Resolution is in the format
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// <width>x<height>_<fps>.
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// sampling_modulo - the module for the video frames to be dumped. Modulo
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// equals X means every Xth frame will be written to the dump file. The
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// value must be greater than 0. (Default: 1)
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// export_frame_ids - specifies if frame ids should be exported together
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// with content of the stream. If true, an output file with the same name as
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// video dump and suffix ".frame_ids.txt" will be created. It will contain
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// the frame ids in the same order as original frames in the output
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// file with stream content. File will contain one frame id per line.
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// (Default: false)
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// `video_frame_writer_factory` - factory function to create a video frame
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// writer for input and output video files. (Default: Y4M video writer
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// factory).
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explicit VideoDumpOptions(
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absl::string_view output_directory,
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int sampling_modulo = kDefaultSamplingModulo,
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bool export_frame_ids = false,
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std::function<std::unique_ptr<test::VideoFrameWriter>(
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absl::string_view file_name_prefix,
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const VideoResolution& resolution)> video_frame_writer_factory =
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Y4mVideoFrameWriterFactory);
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VideoDumpOptions(absl::string_view output_directory, bool export_frame_ids);
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VideoDumpOptions(const VideoDumpOptions&) = default;
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VideoDumpOptions& operator=(const VideoDumpOptions&) = default;
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VideoDumpOptions(VideoDumpOptions&&) = default;
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VideoDumpOptions& operator=(VideoDumpOptions&&) = default;
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std::string output_directory() const { return output_directory_; }
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int sampling_modulo() const { return sampling_modulo_; }
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bool export_frame_ids() const { return export_frame_ids_; }
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std::unique_ptr<test::VideoFrameWriter> CreateInputDumpVideoFrameWriter(
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absl::string_view stream_label,
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const VideoResolution& resolution) const;
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std::unique_ptr<test::VideoFrameWriter> CreateOutputDumpVideoFrameWriter(
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absl::string_view stream_label,
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absl::string_view receiver,
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const VideoResolution& resolution) const;
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std::string ToString() const;
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private:
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static std::unique_ptr<test::VideoFrameWriter> Y4mVideoFrameWriterFactory(
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absl::string_view file_name_prefix,
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const VideoResolution& resolution);
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std::string GetInputDumpFileName(absl::string_view stream_label,
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const VideoResolution& resolution) const;
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// Returns file name for input frame ids dump if `export_frame_ids()` is
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// true, absl::nullopt otherwise.
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absl::optional<std::string> GetInputFrameIdsDumpFileName(
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absl::string_view stream_label,
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const VideoResolution& resolution) const;
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std::string GetOutputDumpFileName(absl::string_view stream_label,
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absl::string_view receiver,
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const VideoResolution& resolution) const;
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// Returns file name for output frame ids dump if `export_frame_ids()` is
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// true, absl::nullopt otherwise.
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absl::optional<std::string> GetOutputFrameIdsDumpFileName(
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absl::string_view stream_label,
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absl::string_view receiver,
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const VideoResolution& resolution) const;
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std::string output_directory_;
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int sampling_modulo_ = 1;
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bool export_frame_ids_ = false;
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std::function<std::unique_ptr<test::VideoFrameWriter>(
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absl::string_view file_name_prefix,
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const VideoResolution& resolution)>
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video_frame_writer_factory_;
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};
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// Contains properties of single video stream.
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struct VideoConfig {
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explicit VideoConfig(const VideoResolution& resolution);
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VideoConfig(size_t width, size_t height, int32_t fps);
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VideoConfig(absl::string_view stream_label,
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size_t width,
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size_t height,
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int32_t fps);
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// Video stream width.
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size_t width;
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// Video stream height.
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size_t height;
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int32_t fps;
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VideoResolution GetResolution() const {
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return VideoResolution(width, height, fps);
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}
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// Have to be unique among all specified configs for all peers in the call.
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// Will be auto generated if omitted.
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absl::optional<std::string> stream_label;
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// Will be set for current video track. If equals to kText or kDetailed -
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// screencast in on.
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absl::optional<VideoTrackInterface::ContentHint> content_hint;
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// If presented video will be transfered in simulcast/SVC mode depending on
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// which encoder is used.
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//
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// Simulcast is supported only from 1st added peer. For VP8 simulcast only
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// without RTX is supported so it will be automatically disabled for all
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// simulcast tracks. For VP9 simulcast enables VP9 SVC mode and support RTX,
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// but only on non-lossy networks. See more in documentation to
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// VideoSimulcastConfig.
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absl::optional<VideoSimulcastConfig> simulcast_config;
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// Configuration for the emulated Selective Forward Unit (SFU).
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absl::optional<EmulatedSFUConfig> emulated_sfu_config;
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// Encoding parameters for both singlecast and per simulcast layer.
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// If singlecast is used, if not empty, a single value can be provided.
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// If simulcast is used, if not empty, `encoding_params` size have to be
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// equal to `simulcast_config.simulcast_streams_count`. Will be used to set
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// transceiver send encoding params for each layer.
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// RtpEncodingParameters::rid may be changed by fixture implementation to
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// ensure signaling correctness.
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std::vector<RtpEncodingParameters> encoding_params;
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// Count of temporal layers for video stream. This value will be set into
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// each RtpEncodingParameters of RtpParameters of corresponding
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// RtpSenderInterface for this video stream.
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absl::optional<int> temporal_layers_count;
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// If specified defines how input should be dumped. It is actually one of
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// the test's output file, which contains copy of what was captured during
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// the test for this video stream on sender side. It is useful when
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// generator is used as input.
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absl::optional<VideoDumpOptions> input_dump_options;
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// If specified defines how output should be dumped on the receiver side for
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// this stream. The produced files contain what was rendered for this video
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// stream on receiver side per each receiver.
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absl::optional<VideoDumpOptions> output_dump_options;
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// If set to true uses fixed frame rate while dumping output video to the
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// file. Requested `VideoSubscription::fps()` will be used as frame rate.
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bool output_dump_use_fixed_framerate = false;
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// If true will display input and output video on the user's screen.
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bool show_on_screen = false;
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// If specified, determines a sync group to which this video stream belongs.
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// According to bugs.webrtc.org/4762 WebRTC supports synchronization only
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// for pair of single audio and single video stream.
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absl::optional<std::string> sync_group;
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// If specified, it will be set into RtpParameters of corresponding
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// RtpSenderInterface for this video stream.
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// Note that this setting takes precedence over `content_hint`.
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absl::optional<DegradationPreference> degradation_preference;
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};
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// Contains properties for audio in the call.
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struct AudioConfig {
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AudioConfig() = default;
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explicit AudioConfig(absl::string_view stream_label);
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// Have to be unique among all specified configs for all peers in the call.
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// Will be auto generated if omitted.
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absl::optional<std::string> stream_label;
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// If no file is specified an audio will be generated.
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absl::optional<std::string> input_file_name;
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// If specified the input stream will be also copied to specified file.
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absl::optional<std::string> input_dump_file_name;
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// If specified the output stream will be copied to specified file.
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absl::optional<std::string> output_dump_file_name;
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// Audio options to use.
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cricket::AudioOptions audio_options;
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// Sampling frequency of input audio data (from file or generated).
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int sampling_frequency_in_hz = 48000;
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// If specified, determines a sync group to which this audio stream belongs.
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// According to bugs.webrtc.org/4762 WebRTC supports synchronization only
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// for pair of single audio and single video stream.
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absl::optional<std::string> sync_group;
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};
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struct VideoCodecConfig {
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explicit VideoCodecConfig(absl::string_view name);
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VideoCodecConfig(absl::string_view name,
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std::map<std::string, std::string> required_params);
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// Next two fields are used to specify concrete video codec, that should be
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// used in the test. Video code will be negotiated in SDP during offer/
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// answer exchange.
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// Video codec name. You can find valid names in
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// media/base/media_constants.h
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std::string name;
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// Map of parameters, that have to be specified on SDP codec. Each parameter
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// is described by key and value. Codec parameters will match the specified
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// map if and only if for each key from `required_params` there will be
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// a parameter with name equal to this key and parameter value will be equal
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// to the value from `required_params` for this key.
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// If empty then only name will be used to match the codec.
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std::map<std::string, std::string> required_params;
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};
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// Subscription to the remote video streams. It declares which remote stream
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// peer should receive and in which resolution (width x height x fps).
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class VideoSubscription {
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public:
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// Returns the resolution constructed as maximum from all resolution
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// dimensions: width, height and fps.
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static absl::optional<VideoResolution> GetMaxResolution(
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rtc::ArrayView<const VideoConfig> video_configs);
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static absl::optional<VideoResolution> GetMaxResolution(
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rtc::ArrayView<const VideoResolution> resolutions);
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|
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bool operator==(const VideoSubscription& other) const;
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|
bool operator!=(const VideoSubscription& other) const;
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|
|
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// 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(
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|
absl::string_view peer_name,
|
|
VideoResolution resolution =
|
|
VideoResolution(VideoResolution::Spec::kMaxFromSender));
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|
|
|
// 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.
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|
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:
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|
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
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|
} // namespace webrtc
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|
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#endif // API_TEST_PCLF_MEDIA_CONFIGURATION_H_
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