blob: 3955da5040386df34e15f2579494c326682c8961 [file] [log] [blame]
/*
* 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 "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_transport_interface.h"
#include "api/peer_connection_interface.h"
#include "api/test/audio_quality_analyzer_interface.h"
#include "api/test/simulated_network.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 "logging/rtc_event_log/rtc_event_log_factory_interface.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:
// 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;
};
enum VideoGeneratorType { kDefault, kI420A, kI010 };
// 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;
// Only 1 from |generator|, |input_file_name| and |screen_share_config| can
// be specified. If none of them are specified, then |generator| will be set
// to VideoGeneratorType::kDefault.
// If specified generator of this type will be used to produce input video.
absl::optional<VideoGeneratorType> generator;
// If specified this file will be used as input. Input video will be played
// in a circle.
absl::optional<std::string> input_file_name;
// If specified screen share video stream will be created as input.
absl::optional<ScreenShareConfig> screen_share_config;
// Specifies spatial index of the video stream to analyze.
// There are 3 cases:
// 1. |target_spatial_index| omitted: in such case it will be assumed that
// video stream has not spatial layers and simulcast streams.
// 2. |target_spatial_index| presented and 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.
// 3. |target_spatial_index| presented and SVP encoder is used:
// in such case |target_spatial_index| will specify the top interesting
// spatial layer and all layers bellow, including target one will be
// processed. All layers above target one will be dropped.
absl::optional<int> target_spatial_index;
// 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;
};
// 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;
};
// This class is used to fully configure one peer inside the call.
class PeerConfigurer {
public:
virtual ~PeerConfigurer() = default;
// The parameters of the following 7 methods will be passed to the
// PeerConnectionFactoryInterface implementation that will be created for
// this peer.
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* SetMediaTransportFactory(
std::unique_ptr<MediaTransportFactory> media_transport_factory) = 0;
virtual PeerConfigurer* SetVideoEncoderFactory(
std::unique_ptr<VideoEncoderFactory> video_encoder_factory) = 0;
virtual PeerConfigurer* SetVideoDecoderFactory(
std::unique_ptr<VideoDecoderFactory> video_decoder_factory) = 0;
// The parameters of the following 3 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;
// Add new video stream to the call that will be sent from this peer.
virtual PeerConfigurer* AddVideoConfig(VideoConfig config) = 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;
};
// 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;
// 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 video_codec_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 |video_codec_required_params| there
// will be a parameter with name equal to this key and parameter value will
// be equal to the value from |video_codec_required_params| for this key.
// If empty then only name will be used to match the codec.
std::map<std::string, std::string> video_codec_required_params;
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;
};
// Represent an entity that will report quality metrics after test.
class QualityMetricsReporter {
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.
virtual void Start(absl::string_view test_case_name) = 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;
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_