| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <functional> |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/base/macros.h" |
| #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/pclf/media_configuration.h" |
| #include "api/test/pclf/media_quality_test_params.h" |
| #include "api/test/pclf/peer_configurer.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/video_frame_writer.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/checks.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 { |
| |
| // API is in development. Can be changed/removed without notice. |
| class PeerConnectionE2EQualityTestFixture { |
| public: |
| // 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(std::unique_ptr<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_ |