| /* |
| * Copyright (c) 2014 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 TEST_CALL_TEST_H_ |
| #define TEST_CALL_TEST_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/task_queue/task_queue_factory.h" |
| #include "api/test/simulated_network.h" |
| #include "api/test/video/function_video_decoder_factory.h" |
| #include "api/test/video/function_video_encoder_factory.h" |
| #include "api/units/time_delta.h" |
| #include "api/video/video_bitrate_allocator_factory.h" |
| #include "call/call.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_device/include/test_audio_device.h" |
| #include "test/encoder_settings.h" |
| #include "test/fake_decoder.h" |
| #include "test/fake_videorenderer.h" |
| #include "test/fake_vp8_encoder.h" |
| #include "test/frame_generator_capturer.h" |
| #include "test/rtp_rtcp_observer.h" |
| #include "test/run_loop.h" |
| #include "test/scoped_key_value_config.h" |
| #include "test/test_video_capturer.h" |
| #include "test/video_test_constants.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class BaseTest; |
| |
| class CallTest : public ::testing::Test, public RtpPacketSinkInterface { |
| public: |
| CallTest(); |
| virtual ~CallTest(); |
| |
| static const std::map<uint8_t, MediaType> payload_type_map_; |
| |
| protected: |
| void RegisterRtpExtension(const RtpExtension& extension); |
| // Returns header extensions that can be parsed by the transport. |
| rtc::ArrayView<const RtpExtension> GetRegisteredExtensions() { |
| return rtp_extensions_; |
| } |
| |
| // RunBaseTest overwrites the audio_state of the send and receive Call configs |
| // to simplify test code. |
| void RunBaseTest(BaseTest* test); |
| |
| void CreateCalls(); |
| void CreateCalls(const CallConfig& sender_config, |
| const CallConfig& receiver_config); |
| void CreateSenderCall(); |
| void CreateSenderCall(const CallConfig& config); |
| void CreateReceiverCall(const CallConfig& config); |
| void DestroyCalls(); |
| |
| void CreateVideoSendConfig(VideoSendStream::Config* video_config, |
| size_t num_video_streams, |
| size_t num_used_ssrcs, |
| Transport* send_transport); |
| void CreateAudioAndFecSendConfigs(size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| Transport* send_transport); |
| void SetAudioConfig(const AudioSendStream::Config& config); |
| |
| void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs); |
| void SetSendUlpFecConfig(VideoSendStream::Config* send_config); |
| void SetReceiveUlpFecConfig( |
| VideoReceiveStreamInterface::Config* receive_config); |
| |
| void CreateSendConfig(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams) { |
| CreateSendConfig(num_video_streams, num_audio_streams, num_flexfec_streams, |
| send_transport_.get()); |
| } |
| void CreateSendConfig(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| Transport* send_transport); |
| |
| void CreateMatchingVideoReceiveConfigs( |
| const VideoSendStream::Config& video_send_config) { |
| CreateMatchingVideoReceiveConfigs(video_send_config, |
| receive_transport_.get()); |
| } |
| void CreateMatchingVideoReceiveConfigs( |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport); |
| void CreateMatchingVideoReceiveConfigs( |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport, |
| VideoDecoderFactory* decoder_factory, |
| absl::optional<size_t> decode_sub_stream, |
| bool receiver_reference_time_report, |
| int rtp_history_ms); |
| void AddMatchingVideoReceiveConfigs( |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport, |
| VideoDecoderFactory* decoder_factory, |
| absl::optional<size_t> decode_sub_stream, |
| bool receiver_reference_time_report, |
| int rtp_history_ms); |
| |
| void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport); |
| void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group); |
| static AudioReceiveStreamInterface::Config CreateMatchingAudioConfig( |
| const AudioSendStream::Config& send_config, |
| rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory, |
| Transport* transport, |
| std::string sync_group); |
| void CreateMatchingFecConfig( |
| Transport* transport, |
| const VideoSendStream::Config& video_send_config); |
| void CreateMatchingReceiveConfigs() { |
| CreateMatchingReceiveConfigs(receive_transport_.get()); |
| } |
| void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); |
| |
| void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, |
| float speed, |
| int framerate, |
| int width, |
| int height); |
| void CreateFrameGeneratorCapturer(int framerate, int width, int height); |
| void CreateFakeAudioDevices( |
| std::unique_ptr<TestAudioDeviceModule::Capturer> capturer, |
| std::unique_ptr<TestAudioDeviceModule::Renderer> renderer); |
| |
| void CreateVideoStreams(); |
| void CreateVideoSendStreams(); |
| void CreateVideoSendStream(const VideoEncoderConfig& encoder_config); |
| void CreateAudioStreams(); |
| void CreateFlexfecStreams(); |
| |
| // Receiver call must be created before calling CreateSendTransport in order |
| // to set a receiver. |
| // Rtp header extensions must be registered (RegisterRtpExtension(..)) before |
| // the transport is created in order for the receiving call object receive RTP |
| // packets with extensions. |
| void CreateSendTransport(const BuiltInNetworkBehaviorConfig& config, |
| RtpRtcpObserver* observer); |
| void CreateReceiveTransport(const BuiltInNetworkBehaviorConfig& config, |
| RtpRtcpObserver* observer); |
| |
| void ConnectVideoSourcesToStreams(); |
| |
| void Start(); |
| void StartVideoSources(); |
| void StartVideoStreams(); |
| void Stop(); |
| void StopVideoStreams(); |
| void DestroyStreams(); |
| void DestroyVideoSendStreams(); |
| void SetFakeVideoCaptureRotation(VideoRotation rotation); |
| |
| void SetVideoDegradation(DegradationPreference preference); |
| |
| VideoSendStream::Config* GetVideoSendConfig(); |
| void SetVideoSendConfig(const VideoSendStream::Config& config); |
| VideoEncoderConfig* GetVideoEncoderConfig(); |
| void SetVideoEncoderConfig(const VideoEncoderConfig& config); |
| VideoSendStream* GetVideoSendStream(); |
| FlexfecReceiveStream::Config* GetFlexFecConfig(); |
| TaskQueueBase* task_queue() { return task_queue_.get(); } |
| |
| // RtpPacketSinkInterface implementation. |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| test::RunLoop loop_; |
| |
| Clock* const clock_; |
| test::ScopedKeyValueConfig field_trials_; |
| |
| std::unique_ptr<TaskQueueFactory> task_queue_factory_; |
| std::unique_ptr<webrtc::RtcEventLog> send_event_log_; |
| std::unique_ptr<webrtc::RtcEventLog> recv_event_log_; |
| std::unique_ptr<Call> sender_call_; |
| std::unique_ptr<PacketTransport> send_transport_; |
| SimulatedNetworkInterface* send_simulated_network_ = nullptr; |
| std::vector<VideoSendStream::Config> video_send_configs_; |
| std::vector<VideoEncoderConfig> video_encoder_configs_; |
| std::vector<VideoSendStream*> video_send_streams_; |
| AudioSendStream::Config audio_send_config_; |
| AudioSendStream* audio_send_stream_; |
| |
| std::unique_ptr<Call> receiver_call_; |
| std::unique_ptr<PacketTransport> receive_transport_; |
| SimulatedNetworkInterface* receive_simulated_network_ = nullptr; |
| std::vector<VideoReceiveStreamInterface::Config> video_receive_configs_; |
| std::vector<VideoReceiveStreamInterface*> video_receive_streams_; |
| std::vector<AudioReceiveStreamInterface::Config> audio_receive_configs_; |
| std::vector<AudioReceiveStreamInterface*> audio_receive_streams_; |
| std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_; |
| std::vector<FlexfecReceiveStream*> flexfec_receive_streams_; |
| |
| test::FrameGeneratorCapturer* frame_generator_capturer_; |
| std::vector<std::unique_ptr<TestVideoCapturer>> video_sources_; |
| DegradationPreference degradation_preference_ = |
| DegradationPreference::MAINTAIN_FRAMERATE; |
| |
| std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_; |
| std::unique_ptr<NetworkStatePredictorFactoryInterface> |
| network_state_predictor_factory_; |
| std::unique_ptr<NetworkControllerFactoryInterface> |
| network_controller_factory_; |
| |
| test::FunctionVideoEncoderFactory fake_encoder_factory_; |
| int fake_encoder_max_bitrate_ = -1; |
| test::FunctionVideoDecoderFactory fake_decoder_factory_; |
| std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_; |
| // Number of simulcast substreams. |
| size_t num_video_streams_; |
| size_t num_audio_streams_; |
| size_t num_flexfec_streams_; |
| rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_; |
| rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_; |
| test::FakeVideoRenderer fake_renderer_; |
| |
| private: |
| absl::optional<RtpExtension> GetRtpExtensionByUri( |
| const std::string& uri) const; |
| |
| void AddRtpExtensionByUri(const std::string& uri, |
| std::vector<RtpExtension>* extensions) const; |
| |
| std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_; |
| std::vector<RtpExtension> rtp_extensions_; |
| rtc::scoped_refptr<AudioProcessing> apm_send_; |
| rtc::scoped_refptr<AudioProcessing> apm_recv_; |
| rtc::scoped_refptr<AudioDeviceModule> fake_send_audio_device_; |
| rtc::scoped_refptr<AudioDeviceModule> fake_recv_audio_device_; |
| }; |
| |
| class BaseTest : public RtpRtcpObserver { |
| public: |
| BaseTest(); |
| explicit BaseTest(TimeDelta timeout); |
| virtual ~BaseTest(); |
| |
| virtual void PerformTest() = 0; |
| virtual bool ShouldCreateReceivers() const = 0; |
| |
| virtual size_t GetNumVideoStreams() const; |
| virtual size_t GetNumAudioStreams() const; |
| virtual size_t GetNumFlexfecStreams() const; |
| |
| virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer(); |
| virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer(); |
| virtual void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device, |
| AudioDeviceModule* recv_audio_device); |
| |
| virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config); |
| virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config); |
| |
| virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); |
| virtual void OnTransportCreated(PacketTransport* to_receiver, |
| SimulatedNetworkInterface* sender_network, |
| PacketTransport* to_sender, |
| SimulatedNetworkInterface* receiver_network); |
| |
| virtual BuiltInNetworkBehaviorConfig GetSendTransportConfig() const; |
| virtual BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const; |
| |
| virtual void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config); |
| virtual void ModifyVideoCaptureStartResolution(int* width, |
| int* heigt, |
| int* frame_rate); |
| virtual void ModifyVideoDegradationPreference( |
| DegradationPreference* degradation_preference); |
| |
| virtual void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStreamInterface*>& receive_streams); |
| |
| virtual void ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStreamInterface::Config>* receive_configs); |
| virtual void OnAudioStreamsCreated( |
| AudioSendStream* send_stream, |
| const std::vector<AudioReceiveStreamInterface*>& receive_streams); |
| |
| virtual void ModifyFlexfecConfigs( |
| std::vector<FlexfecReceiveStream::Config>* receive_configs); |
| virtual void OnFlexfecStreamsCreated( |
| const std::vector<FlexfecReceiveStream*>& receive_streams); |
| |
| virtual void OnFrameGeneratorCapturerCreated( |
| FrameGeneratorCapturer* frame_generator_capturer); |
| |
| virtual void OnStreamsStopped(); |
| }; |
| |
| class SendTest : public BaseTest { |
| public: |
| explicit SendTest(TimeDelta timeout); |
| |
| bool ShouldCreateReceivers() const override; |
| }; |
| |
| class EndToEndTest : public BaseTest { |
| public: |
| EndToEndTest(); |
| explicit EndToEndTest(TimeDelta timeout); |
| |
| bool ShouldCreateReceivers() const override; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // TEST_CALL_TEST_H_ |