blob: 7e19eb1629ff1ed96c2afecc5a2e234c84c88373 [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.
#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/audio/audio_mixer.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 {
// Represent an entity that will report quality metrics after test.
class QualityMetricsReporter : public StatsObserverInterface {
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 {
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