Also introduce interface for video quality analyze and mock interface, that then will be extended for audio quality analyze. Bug: webrtc:10138 Change-Id: I0e3957fb2af1b12e796f154765580ddf562c7814 Reviewed-on: https://webrtc-review.googlesource.com/c/116500 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26157}
139 lines
5.2 KiB
C++
139 lines
5.2 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_
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#define TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/asyncresolverfactory.h"
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#include "api/call/callfactoryinterface.h"
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#include "api/fec_controller.h"
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#include "api/media_transport_interface.h"
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#include "api/peerconnectioninterface.h"
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#include "api/test/simulated_network.h"
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#include "api/transport/network_control.h"
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#include "api/video_codecs/video_decoder_factory.h"
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#include "api/video_codecs/video_encoder.h"
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#include "api/video_codecs/video_encoder_factory.h"
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#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
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#include "rtc_base/network.h"
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#include "rtc_base/rtccertificategenerator.h"
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#include "rtc_base/sslcertificate.h"
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#include "rtc_base/thread.h"
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#include "test/pc/e2e/api/audio_quality_analyzer_interface.h"
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#include "test/pc/e2e/api/video_quality_analyzer_interface.h"
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namespace webrtc {
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// TODO(titovartem) move to API when it will be stabilized.
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class PeerConnectionE2EQualityTestFixture {
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public:
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struct PeerConnectionFactoryComponents {
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std::unique_ptr<CallFactoryInterface> call_factory;
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std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
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std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
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std::unique_ptr<NetworkControllerFactoryInterface>
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network_controller_factory;
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std::unique_ptr<MediaTransportFactory> media_transport_factory;
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// Will be passed to MediaEngineInterface, that will be used in
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// PeerConnectionFactory.
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std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
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std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
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};
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struct PeerConnectionComponents {
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std::unique_ptr<rtc::NetworkManager> network_manager;
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std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
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std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
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};
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struct InjectableComponents {
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explicit InjectableComponents(rtc::Thread* network_thread)
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: network_thread(network_thread) {}
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rtc::Thread* network_thread;
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std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies;
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std::unique_ptr<PeerConnectionComponents> pc_dependencies;
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};
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struct ScreenShareConfig {
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// If true, slides will be generated programmatically.
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bool generate_slides;
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int32_t slide_change_interval;
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// If equal to 0, no scrolling will be applied.
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int32_t scroll_duration;
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// If empty, default set of slides will be used.
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std::vector<std::string> slides_yuv_file_names;
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};
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struct VideoConfig {
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size_t width;
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size_t height;
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int32_t fps;
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// Have to be unique among all specified configs for all peers in the call.
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absl::optional<std::string> stream_label;
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// Only single from 3 next fields can be specified.
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// If specified generator with this name will be used as input.
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absl::optional<std::string> generator_name;
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// If specified this file will be used as input.
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absl::optional<std::string> input_file_name;
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// If specified screen share video stream will be created as input.
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absl::optional<ScreenShareConfig> screen_share_config;
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// If specified the input stream will be also copied to specified file.
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absl::optional<std::string> input_dump_file_name;
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// If specified this file will be used as output on the receiver side for
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// this stream. If multiple streams will be produced by input stream,
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// output files will be appended with indexes.
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absl::optional<std::string> output_file_name;
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};
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struct AudioConfig {
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enum Mode {
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kGenerated,
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kFile,
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};
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Mode mode;
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// Have to be specified only if mode = kFile
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absl::optional<std::string> input_file_name;
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// If specified the input stream will be also copied to specified file.
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absl::optional<std::string> input_dump_file_name;
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// If specified the output stream will be copied to specified file.
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absl::optional<std::string> output_file_name;
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// Audio options to use.
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cricket::AudioOptions audio_options;
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};
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struct Params {
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// If |video_configs| is empty - no video should be added to the test call.
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std::vector<VideoConfig> video_configs;
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// If |audio_config| is presented audio stream will be configured
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absl::optional<AudioConfig> audio_config;
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PeerConnectionInterface::RTCConfiguration rtc_configuration;
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};
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struct Analyzers {
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std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer;
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std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer;
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};
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virtual void Run() = 0;
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virtual ~PeerConnectionE2EQualityTestFixture() = default;
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};
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} // namespace webrtc
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#endif // TEST_PC_E2E_API_PEERCONNECTION_QUALITY_TEST_FIXTURE_H_
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