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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/scenario/scenario_config.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/scenario/scenario_config.h')
-rw-r--r-- | third_party/libwebrtc/test/scenario/scenario_config.h | 231 |
1 files changed, 231 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/scenario_config.h b/third_party/libwebrtc/test/scenario/scenario_config.h new file mode 100644 index 0000000000..9ce99401d7 --- /dev/null +++ b/third_party/libwebrtc/test/scenario/scenario_config.h @@ -0,0 +1,231 @@ +/* + * Copyright 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef TEST_SCENARIO_SCENARIO_CONFIG_H_ +#define TEST_SCENARIO_SCENARIO_CONFIG_H_ + +#include <stddef.h> + +#include <string> + +#include "absl/types/optional.h" +#include "api/fec_controller.h" +#include "api/rtp_parameters.h" +#include "api/test/frame_generator_interface.h" +#include "api/transport/network_control.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/scalability_mode.h" +#include "test/scenario/performance_stats.h" + +namespace webrtc { +namespace test { +struct PacketOverhead { + static constexpr size_t kSrtp = 10; + static constexpr size_t kStun = 4; + // TURN messages can be sent either with or without an establieshed channel. + // In the latter case, a TURN Send/Data Indication is sent which has + // significantly more overhead. + static constexpr size_t kTurnChannelMessage = 4; + static constexpr size_t kTurnIndicationMessage = 36; + static constexpr size_t kDefault = kSrtp; +}; +struct TransportControllerConfig { + struct Rates { + Rates(); + Rates(const Rates&); + ~Rates(); + DataRate min_rate = DataRate::KilobitsPerSec(30); + DataRate max_rate = DataRate::KilobitsPerSec(3000); + DataRate start_rate = DataRate::KilobitsPerSec(300); + } rates; + NetworkControllerFactoryInterface* cc_factory = nullptr; + TimeDelta state_log_interval = TimeDelta::Millis(100); +}; + +struct CallClientConfig { + TransportControllerConfig transport; + // Allows the pacer to send out multiple packets in a burst. + // The number of bites that can be sent in one burst is pacer_burst_interval * + // current bwe. 40ms is the default Chrome setting. + TimeDelta pacer_burst_interval = TimeDelta::Millis(40); + const FieldTrialsView* field_trials = nullptr; +}; + +struct PacketStreamConfig { + PacketStreamConfig(); + PacketStreamConfig(const PacketStreamConfig&); + ~PacketStreamConfig(); + int frame_rate = 30; + DataRate max_data_rate = DataRate::Infinity(); + DataSize max_packet_size = DataSize::Bytes(1400); + DataSize min_frame_size = DataSize::Bytes(100); + double keyframe_multiplier = 1; + DataSize packet_overhead = DataSize::Bytes(PacketOverhead::kDefault); +}; + +struct VideoStreamConfig { + bool autostart = true; + struct Source { + enum Capture { + kGenerator, + kVideoFile, + kGenerateSlides, + kImageSlides, + // Support for explicit frame triggers should be added here if needed. + } capture = Capture::kGenerator; + struct Slides { + TimeDelta change_interval = TimeDelta::Seconds(10); + struct Generator { + int width = 1600; + int height = 1200; + } generator; + struct Images { + struct Crop { + TimeDelta scroll_duration = TimeDelta::Seconds(0); + absl::optional<int> width; + absl::optional<int> height; + } crop; + int width = 1850; + int height = 1110; + std::vector<std::string> paths = { + "web_screenshot_1850_1110", + "presentation_1850_1110", + "photo_1850_1110", + "difficult_photo_1850_1110", + }; + } images; + } slides; + struct Generator { + using PixelFormat = FrameGeneratorInterface::OutputType; + PixelFormat pixel_format = PixelFormat::kI420; + int width = 320; + int height = 180; + } generator; + struct VideoFile { + std::string name; + // Must be set to width and height of the source video file. + int width = 0; + int height = 0; + } video_file; + int framerate = 30; + } source; + struct Encoder { + Encoder(); + Encoder(const Encoder&); + ~Encoder(); + enum class ContentType { + kVideo, + kScreen, + } content_type = ContentType::kVideo; + enum Implementation { kFake, kSoftware, kHardware } implementation = kFake; + struct Fake { + DataRate max_rate = DataRate::Infinity(); + } fake; + + using Codec = VideoCodecType; + Codec codec = Codec::kVideoCodecGeneric; + absl::optional<DataRate> max_data_rate; + absl::optional<DataRate> min_data_rate; + absl::optional<int> max_framerate; + // Counted in frame count. + absl::optional<int> key_frame_interval = 3000; + bool frame_dropping = true; + struct SingleLayer { + bool denoising = true; + bool automatic_scaling = true; + } single; + std::vector<webrtc::ScalabilityMode> simulcast_streams = { + webrtc::ScalabilityMode::kL1T1}; + + DegradationPreference degradation_preference = + DegradationPreference::MAINTAIN_FRAMERATE; + bool suspend_below_min_bitrate = false; + } encoder; + struct Stream { + Stream(); + Stream(const Stream&); + ~Stream(); + bool abs_send_time = false; + bool packet_feedback = true; + bool use_rtx = true; + DataRate pad_to_rate = DataRate::Zero(); + TimeDelta nack_history_time = TimeDelta::Millis(1000); + bool use_flexfec = false; + bool use_ulpfec = false; + FecControllerFactoryInterface* fec_controller_factory = nullptr; + } stream; + struct Rendering { + enum Type { kFake } type = kFake; + std::string sync_group; + } render; + struct Hooks { + std::vector<std::function<void(const VideoFramePair&)>> frame_pair_handlers; + } hooks; +}; + +struct AudioStreamConfig { + AudioStreamConfig(); + AudioStreamConfig(const AudioStreamConfig&); + ~AudioStreamConfig(); + bool autostart = true; + struct Source { + int channels = 1; + } source; + bool network_adaptation = false; + struct NetworkAdaptation { + struct FrameLength { + double min_packet_loss_for_decrease = 0; + double max_packet_loss_for_increase = 1; + DataRate min_rate_for_20_ms = DataRate::Zero(); + DataRate max_rate_for_60_ms = DataRate::Infinity(); + DataRate min_rate_for_60_ms = DataRate::Zero(); + DataRate max_rate_for_120_ms = DataRate::Infinity(); + } frame; + std::string binary_proto; + } adapt; + struct Encoder { + Encoder(); + Encoder(const Encoder&); + ~Encoder(); + bool allocate_bitrate = false; + bool enable_dtx = false; + DataRate fixed_rate = DataRate::KilobitsPerSec(32); + // Overrides fixed rate. + absl::optional<DataRate> min_rate; + absl::optional<DataRate> max_rate; + TimeDelta initial_frame_length = TimeDelta::Millis(20); + } encoder; + struct Stream { + Stream(); + Stream(const Stream&); + ~Stream(); + bool abs_send_time = true; + bool in_bandwidth_estimation = true; + } stream; + struct Rendering { + std::string sync_group; + } render; +}; + +// TODO(srte): Merge this with BuiltInNetworkBehaviorConfig. +struct NetworkSimulationConfig { + DataRate bandwidth = DataRate::Infinity(); + TimeDelta delay = TimeDelta::Zero(); + TimeDelta delay_std_dev = TimeDelta::Zero(); + double loss_rate = 0; + absl::optional<int> packet_queue_length_limit; + DataSize packet_overhead = DataSize::Zero(); +}; +} // namespace test +} // namespace webrtc + +#endif // TEST_SCENARIO_SCENARIO_CONFIG_H_ |