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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/async_resolver_factory.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/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 "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 Vp9 SVC testing.
//
// 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);
}
VideoSimulcastConfig(int simulcast_streams_count, int target_spatial_index)
: simulcast_streams_count(simulcast_streams_count),
target_spatial_index(target_spatial_index) {
RTC_CHECK_GT(simulcast_streams_count, 1);
RTC_CHECK_GE(target_spatial_index, 0);
RTC_CHECK_LT(target_spatial_index, simulcast_streams_count);
}
// Specified amount of simulcast streams/SVC layers, depending on which
// encoder is used.
int simulcast_streams_count;
// Specifies spatial index of the video stream to analyze.
// There are 2 cases:
// 1. simulcast encoder is used:
// in such case |target_spatial_index| will specify the index of
// simulcast stream, that should be analyzed. Other streams will be
// dropped.
// 2. SVC encoder is used:
// in such case |target_spatial_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 than whatever stream will be received will be analyzed.
// It requires Selective Forwarding Unit (SFU) to be configured in the
// network.
absl::optional<int> target_spatial_index;
// Encoding parameters per simulcast layer. If not empty, |encoding_params|
// size have to be equal to |simulcast_streams_count|. Will be used to set
// transceiver send encoding params for simulcast layers. Applicable only
// for codecs that support simulcast (ex. Vp8) and will be ignored
// otherwise. RtpEncodingParameters::rid may be changed by fixture
// implementation to ensure signaling correctness.
std::vector<RtpEncodingParameters> encoding_params;
};
// Contains properties of single video stream.
struct VideoConfig {
VideoConfig(size_t width, size_t height, int32_t fps)
: width(width), height(height), fps(fps) {}
// Video stream width.
const size_t width;
// Video stream height.
const size_t height;
const int32_t 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;
// 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;
// Sets the maximum encode bitrate in bps. If this value is not set, the
// encoder will be capped at an internal maximum value around 2 Mbps
// depending on the resolution. This means that it will never be able to
// utilize a high bandwidth link.
absl::optional<int> max_encode_bitrate_bps;
// Sets the minimum encode bitrate in bps. If this value is not set, the
// encoder will use an internal minimum value. Please note that if this
// value is set higher than the bandwidth of the link, the encoder will
// generate more data than the link can handle regardless of the bandwidth
// estimation.
absl::optional<int> min_encode_bitrate_bps;
// If specified the input stream will be also copied to specified file.
// 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<std::string> input_dump_file_name;
// If specified this file will be used as output on the receiver side for
// this stream. If multiple streams will be produced by input stream,
// output files will be appended with indexes. The produced files contains
// what was rendered for this video stream on receiver side.
absl::optional<std::string> output_dump_file_name;
// 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;
};
// Contains properties for audio in the call.
struct AudioConfig {
enum Mode {
kGenerated,
kFile,
};
// 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;
};
// 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;
// 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;
// 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;
// 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;
// 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;
// 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);
};
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;
};
// 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;
// List of video codecs to use 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.
// If list is empty Vp8 with no required_params will be used.
std::vector<VideoCodecConfig> video_codecs;
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 on this multiplier and then
// provided into VideoEncoder::SetRates(...).
double video_encoder_bitrate_multiplier = 1.0;
// 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;
};
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_thread| will be used as network thread for peer's peer connection
// |network_manager| will be used to provide network interfaces for peer's
// peer connection.
// |configurer| function will be used to configure peer in the call.
virtual void AddPeer(rtc::Thread* network_thread,
rtc::NetworkManager* network_manager,
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_
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