summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/test/scenario/scenario_config.h
diff options
context:
space:
mode:
authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/scenario/scenario_config.h
parentInitial commit. (diff)
downloadfirefox-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.h231
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_