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webrtc / src / e39115a0ca3fffc87cdc602a60caa26fd3b3949f / . / api / test / peerconnection_quality_test_fixture.h
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/*
* Copyright (c) 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 API_TEST_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_
#define API_TEST_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_
#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/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_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 "media/base/media_constants.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 {
constexpr size_t kDefaultSlidesWidth = 1850;
constexpr size_t kDefaultSlidesHeight = 1110;
// API is in development. Can be changed/removed without notice.
class PeerConnectionE2EQualityTestFixture {
public:
// 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 {
ScrollingParams(TimeDelta duration,
size_t source_width,
size_t source_height)
: duration(duration),
source_width(source_width),
source_height(source_height) {
RTC_CHECK_GT(duration.ms(), 0);
}
// Duration of scrolling.
TimeDelta duration;
// Width of source slides video.
size_t source_width;
// Height of source slides video.
size_t source_height;
};
// Contains screen share video stream properties.
struct ScreenShareConfig {
explicit ScreenShareConfig(TimeDelta slide_change_interval)
: slide_change_interval(slide_change_interval) {
RTC_CHECK_GT(slide_change_interval.ms(), 0);
}
// 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)
: simulcast_streams_count(simulcast_streams_count) {
RTC_CHECK_GT(simulcast_streams_count, 1);
}
// 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() {}
explicit EmulatedSFUConfig(int target_layer_index)
: target_layer_index(target_layer_index) {
RTC_CHECK_GE(target_layer_index, 0);
}
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);
}
// 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 {
return !(*this == other);
}
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 { return spec_ == Spec::kNone; }
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>.<extension> for output dumps and
// <stream_name>.<extension> for input dumps. By default <extension> is
// "y4m".
// 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)
explicit VideoDumpOptions(absl::string_view output_directory,
int sampling_modulo = kDefaultSamplingModulo,
bool export_frame_ids = false);
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::string GetInputDumpFileName(absl::string_view stream_label) 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;
std::string GetOutputDumpFileName(absl::string_view stream_label,
absl::string_view receiver) 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;
std::string ToString() const;
private:
std::string output_directory_;
int sampling_modulo_ = 1;
bool export_frame_ids_ = false;
};
// Contains properties of single video stream.
struct VideoConfig {
explicit VideoConfig(const VideoResolution& resolution);
VideoConfig(size_t width, size_t height, int32_t fps)
: width(width), height(height), fps(fps) {}
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)) {}
// 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. `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)
: stream_label(std::move(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)
: name(std::move(name)), required_params() {}
VideoCodecConfig(std::string name,
std::map<std::string, std::string> required_params)
: name(std::move(name)), required_params(std::move(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 = cricket::kVp8CodecName;
// 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 {
return !(*this == other);
}
// 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)) {
peers_resolution_[std::string(peer_name)] = resolution;
return *this;
}
// 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)) {
default_resolution_ = resolution;
return *this;
}
// 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 {
auto it = peers_resolution_.find(std::string(peer_name));
if (it == peers_resolution_.end()) {
return default_resolution_;
}
return it->second;
}
// Returns a maybe empty list of senders for which peer explicitly
// subscribed to with specific resolution.
std::vector<std::string> 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 ToString() const;
private:
absl::optional<VideoResolution> default_resolution_ = absl::nullopt;
std::map<std::string, VideoResolution> peers_resolution_;
};
// This class is used to fully configure one peer inside the call.
class PeerConfigurer {
public:
virtual ~PeerConfigurer() = default;
// 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.
virtual PeerConfigurer* SetName(absl::string_view name) = 0;
// The parameters of the following 9 methods will be passed to the
// PeerConnectionFactoryInterface implementation that will be created for
// this peer.
virtual PeerConfigurer* SetTaskQueueFactory(
std::unique_ptr<TaskQueueFactory> task_queue_factory) = 0;
virtual PeerConfigurer* SetCallFactory(
std::unique_ptr<CallFactoryInterface> call_factory) = 0;
virtual PeerConfigurer* SetEventLogFactory(
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) = 0;
virtual PeerConfigurer* SetFecControllerFactory(
std::unique_ptr<FecControllerFactoryInterface>
fec_controller_factory) = 0;
virtual PeerConfigurer* SetNetworkControllerFactory(
std::unique_ptr<NetworkControllerFactoryInterface>
network_controller_factory) = 0;
virtual PeerConfigurer* SetVideoEncoderFactory(
std::unique_ptr<VideoEncoderFactory> video_encoder_factory) = 0;
virtual PeerConfigurer* SetVideoDecoderFactory(
std::unique_ptr<VideoDecoderFactory> video_decoder_factory) = 0;
// Set a custom NetEqFactory to be used in the call.
virtual PeerConfigurer* SetNetEqFactory(
std::unique_ptr<NetEqFactory> neteq_factory) = 0;
virtual PeerConfigurer* SetAudioProcessing(
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) = 0;
virtual PeerConfigurer* SetAudioMixer(
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) = 0;
// The parameters of the following 4 methods will be passed to the
// PeerConnectionInterface implementation that will be created for this
// peer.
virtual PeerConfigurer* SetAsyncResolverFactory(
std::unique_ptr<webrtc::AsyncResolverFactory>
async_resolver_factory) = 0;
virtual PeerConfigurer* SetRTCCertificateGenerator(
std::unique_ptr<rtc::RTCCertificateGeneratorInterface>
cert_generator) = 0;
virtual PeerConfigurer* SetSSLCertificateVerifier(
std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier) = 0;
virtual PeerConfigurer* SetIceTransportFactory(
std::unique_ptr<IceTransportFactory> factory) = 0;
// 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.
virtual PeerConfigurer* SetPortAllocatorExtraFlags(
uint32_t extra_flags) = 0;
// Add new video stream to the call that will be sent from this peer.
// Default implementation of video frames generator will be used.
virtual PeerConfigurer* AddVideoConfig(VideoConfig config) = 0;
// Add new video stream to the call that will be sent from this peer with
// provided own implementation of video frames generator.
virtual PeerConfigurer* AddVideoConfig(
VideoConfig config,
std::unique_ptr<test::FrameGeneratorInterface> generator) = 0;
// 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.
virtual PeerConfigurer* AddVideoConfig(
VideoConfig config,
CapturingDeviceIndex capturing_device_index) = 0;
// Sets video subscription for the peer. By default subscription will
// include all streams with `VideoSubscription::kSameAsSendStream`
// resolution. To override this behavior use this method.
virtual PeerConfigurer* SetVideoSubscription(
VideoSubscription subscription) = 0;
// Set 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.
virtual PeerConfigurer* SetVideoCodecs(
std::vector<VideoCodecConfig> video_codecs) = 0;
// Set the audio stream for the call from this peer. If this method won't
// be invoked, this peer will send no audio.
virtual PeerConfigurer* SetAudioConfig(AudioConfig config) = 0;
// Set if ULP FEC should be used or not. False by default.
virtual PeerConfigurer* SetUseUlpFEC(bool value) = 0;
// 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.
virtual PeerConfigurer* SetUseFlexFEC(bool value) = 0;
// 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.
virtual PeerConfigurer* SetVideoEncoderBitrateMultiplier(
double multiplier) = 0;
// If is set, an RTCEventLog will be saved in that location and it will be
// available for further analysis.
virtual PeerConfigurer* SetRtcEventLogPath(std::string path) = 0;
// If is set, an AEC dump will be saved in that location and it will be
// available for further analysis.
virtual PeerConfigurer* SetAecDumpPath(std::string path) = 0;
virtual PeerConfigurer* SetRTCConfiguration(
PeerConnectionInterface::RTCConfiguration configuration) = 0;
virtual PeerConfigurer* SetRTCOfferAnswerOptions(
PeerConnectionInterface::RTCOfferAnswerOptions options) = 0;
// Set bitrate parameters on PeerConnection. This constraints will be
// applied to all summed RTP streams for this peer.
virtual PeerConfigurer* SetBitrateSettings(
BitrateSettings bitrate_settings) = 0;
};
// 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);
};
// 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;
};
// Represent an entity that will report quality metrics after test.
class QualityMetricsReporter : public StatsObserverInterface {
public:
virtual ~QualityMetricsReporter() = default;
// Invoked by framework after peer connection factory and peer connection
// itself will be created but before offer/answer exchange will be started.
// `test_case_name` is name of test case, that should be used to report all
// metrics.
// `reporter_helper` is a pointer to a class that will allow track_id to
// stream_id matching. The caller is responsible for ensuring the
// TrackIdStreamInfoMap will be valid from Start() to
// StopAndReportResults().
virtual void Start(absl::string_view test_case_name,
const TrackIdStreamInfoMap* reporter_helper) = 0;
// Invoked by framework after call is ended and peer connection factory and
// peer connection are destroyed.
virtual void StopAndReportResults() = 0;
};
// Represents single participant in call and can be used to perform different
// in-call actions. Might be extended in future.
class PeerHandle {
public:
virtual ~PeerHandle() = default;
};
virtual ~PeerConnectionE2EQualityTestFixture() = default;
// Add activity that will be executed on the best effort at least after
// `target_time_since_start` after call will be set up (after offer/answer
// exchange, ICE gathering will be done and ICE candidates will passed to
// remote side). `func` param is amount of time spent from the call set up.
virtual void ExecuteAt(TimeDelta target_time_since_start,
std::function<void(TimeDelta)> func) = 0;
// Add activity that will be executed every `interval` with first execution
// on the best effort at least after `initial_delay_since_start` after call
// will be set up (after all participants will be connected). `func` param is
// amount of time spent from the call set up.
virtual void ExecuteEvery(TimeDelta initial_delay_since_start,
TimeDelta interval,
std::function<void(TimeDelta)> func) = 0;
// Add stats reporter entity to observe the test.
virtual void AddQualityMetricsReporter(
std::unique_ptr<QualityMetricsReporter> quality_metrics_reporter) = 0;
// Add a new peer to the call and return an object through which caller
// can configure peer's behavior.
// `network_dependencies` are used to provide networking for peer's peer
// connection. Members must be non-null.
// `configurer` function will be used to configure peer in the call.
virtual PeerHandle* AddPeer(
const PeerNetworkDependencies& network_dependencies,
rtc::FunctionView<void(PeerConfigurer*)> configurer) = 0;
// Runs the media quality test, which includes setting up the call with
// configured participants, running it according to provided `run_params` and
// terminating it properly at the end. During call duration media quality
// metrics are gathered, which are then reported to stdout and (if configured)
// to the json/protobuf output file through the WebRTC perf test results
// reporting system.
virtual void Run(RunParams run_params) = 0;
// Returns real test duration - the time of test execution measured during
// test. Client must call this method only after test is finished (after
// Run(...) method returned). Test execution time is time from end of call
// setup (offer/answer, ICE candidates exchange done and ICE connected) to
// start of call tear down (PeerConnection closed).
virtual TimeDelta GetRealTestDuration() const = 0;
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // API_TEST_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_