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Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934}
332 lines
13 KiB
C++
332 lines
13 KiB
C++
/*
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* Copyright (c) 2014 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_CALL_TEST_H_
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#define TEST_CALL_TEST_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/task_queue/task_queue_base.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "api/test/simulated_network.h"
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#include "api/test/video/function_video_decoder_factory.h"
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#include "api/test/video/function_video_encoder_factory.h"
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#include "api/units/time_delta.h"
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#include "api/video/video_bitrate_allocator_factory.h"
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#include "call/call.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_device/include/test_audio_device.h"
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#include "test/encoder_settings.h"
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#include "test/fake_decoder.h"
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#include "test/fake_videorenderer.h"
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#include "test/fake_vp8_encoder.h"
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#include "test/frame_generator_capturer.h"
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#include "test/rtp_rtcp_observer.h"
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#include "test/run_loop.h"
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#include "test/scoped_key_value_config.h"
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#include "test/test_video_capturer.h"
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#include "test/video_test_constants.h"
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namespace webrtc {
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namespace test {
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class BaseTest;
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class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
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public:
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CallTest();
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virtual ~CallTest();
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static const std::map<uint8_t, MediaType> payload_type_map_;
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protected:
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void RegisterRtpExtension(const RtpExtension& extension);
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// Returns header extensions that can be parsed by the transport.
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rtc::ArrayView<const RtpExtension> GetRegisteredExtensions() {
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return rtp_extensions_;
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}
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// RunBaseTest overwrites the audio_state of the send and receive Call configs
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// to simplify test code.
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void RunBaseTest(BaseTest* test);
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void CreateCalls();
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void CreateCalls(const CallConfig& sender_config,
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const CallConfig& receiver_config);
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void CreateSenderCall();
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void CreateSenderCall(const CallConfig& config);
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void CreateReceiverCall(const CallConfig& config);
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void DestroyCalls();
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void CreateVideoSendConfig(VideoSendStream::Config* video_config,
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size_t num_video_streams,
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size_t num_used_ssrcs,
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Transport* send_transport);
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void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport);
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void SetAudioConfig(const AudioSendStream::Config& config);
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void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
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void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
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void SetReceiveUlpFecConfig(
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VideoReceiveStreamInterface::Config* receive_config);
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void CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams) {
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CreateSendConfig(num_video_streams, num_audio_streams, num_flexfec_streams,
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send_transport_.get());
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}
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void CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport);
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void CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config) {
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CreateMatchingVideoReceiveConfigs(video_send_config,
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receive_transport_.get());
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}
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void CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport);
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void CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport,
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VideoDecoderFactory* decoder_factory,
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absl::optional<size_t> decode_sub_stream,
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bool receiver_reference_time_report,
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int rtp_history_ms);
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void AddMatchingVideoReceiveConfigs(
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std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport,
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VideoDecoderFactory* decoder_factory,
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absl::optional<size_t> decode_sub_stream,
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bool receiver_reference_time_report,
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int rtp_history_ms);
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void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
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void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
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static AudioReceiveStreamInterface::Config CreateMatchingAudioConfig(
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const AudioSendStream::Config& send_config,
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rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
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Transport* transport,
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std::string sync_group);
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void CreateMatchingFecConfig(
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Transport* transport,
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const VideoSendStream::Config& video_send_config);
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void CreateMatchingReceiveConfigs() {
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CreateMatchingReceiveConfigs(receive_transport_.get());
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}
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void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
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void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
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float speed,
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int framerate,
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int width,
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int height);
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void CreateFrameGeneratorCapturer(int framerate, int width, int height);
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void CreateFakeAudioDevices(
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std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
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std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
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void CreateVideoStreams();
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void CreateVideoSendStreams();
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void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
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void CreateAudioStreams();
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void CreateFlexfecStreams();
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// Receiver call must be created before calling CreateSendTransport in order
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// to set a receiver.
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// Rtp header extensions must be registered (RegisterRtpExtension(..)) before
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// the transport is created in order for the receiving call object receive RTP
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// packets with extensions.
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void CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
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RtpRtcpObserver* observer);
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void CreateReceiveTransport(const BuiltInNetworkBehaviorConfig& config,
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RtpRtcpObserver* observer);
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void ConnectVideoSourcesToStreams();
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void Start();
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void StartVideoSources();
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void StartVideoStreams();
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void Stop();
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void StopVideoStreams();
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void DestroyStreams();
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void DestroyVideoSendStreams();
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void SetFakeVideoCaptureRotation(VideoRotation rotation);
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void SetVideoDegradation(DegradationPreference preference);
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VideoSendStream::Config* GetVideoSendConfig();
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void SetVideoSendConfig(const VideoSendStream::Config& config);
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VideoEncoderConfig* GetVideoEncoderConfig();
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void SetVideoEncoderConfig(const VideoEncoderConfig& config);
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VideoSendStream* GetVideoSendStream();
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FlexfecReceiveStream::Config* GetFlexFecConfig();
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TaskQueueBase* task_queue() { return task_queue_.get(); }
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// RtpPacketSinkInterface implementation.
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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test::RunLoop loop_;
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Clock* const clock_;
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test::ScopedKeyValueConfig field_trials_;
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std::unique_ptr<TaskQueueFactory> task_queue_factory_;
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std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
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std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
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std::unique_ptr<Call> sender_call_;
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std::unique_ptr<PacketTransport> send_transport_;
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SimulatedNetworkInterface* send_simulated_network_ = nullptr;
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std::vector<VideoSendStream::Config> video_send_configs_;
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std::vector<VideoEncoderConfig> video_encoder_configs_;
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std::vector<VideoSendStream*> video_send_streams_;
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AudioSendStream::Config audio_send_config_;
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AudioSendStream* audio_send_stream_;
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std::unique_ptr<Call> receiver_call_;
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std::unique_ptr<PacketTransport> receive_transport_;
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SimulatedNetworkInterface* receive_simulated_network_ = nullptr;
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std::vector<VideoReceiveStreamInterface::Config> video_receive_configs_;
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std::vector<VideoReceiveStreamInterface*> video_receive_streams_;
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std::vector<AudioReceiveStreamInterface::Config> audio_receive_configs_;
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std::vector<AudioReceiveStreamInterface*> audio_receive_streams_;
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std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
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std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
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test::FrameGeneratorCapturer* frame_generator_capturer_;
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std::vector<std::unique_ptr<TestVideoCapturer>> video_sources_;
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DegradationPreference degradation_preference_ =
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DegradationPreference::MAINTAIN_FRAMERATE;
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std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
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std::unique_ptr<NetworkStatePredictorFactoryInterface>
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network_state_predictor_factory_;
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std::unique_ptr<NetworkControllerFactoryInterface>
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network_controller_factory_;
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test::FunctionVideoEncoderFactory fake_encoder_factory_;
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int fake_encoder_max_bitrate_ = -1;
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test::FunctionVideoDecoderFactory fake_decoder_factory_;
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std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
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// Number of simulcast substreams.
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size_t num_video_streams_;
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size_t num_audio_streams_;
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size_t num_flexfec_streams_;
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rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
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rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
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test::FakeVideoRenderer fake_renderer_;
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private:
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absl::optional<RtpExtension> GetRtpExtensionByUri(
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const std::string& uri) const;
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void AddRtpExtensionByUri(const std::string& uri,
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std::vector<RtpExtension>* extensions) const;
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std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
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std::vector<RtpExtension> rtp_extensions_;
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rtc::scoped_refptr<AudioProcessing> apm_send_;
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rtc::scoped_refptr<AudioProcessing> apm_recv_;
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rtc::scoped_refptr<AudioDeviceModule> fake_send_audio_device_;
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rtc::scoped_refptr<AudioDeviceModule> fake_recv_audio_device_;
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};
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class BaseTest : public RtpRtcpObserver {
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public:
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BaseTest();
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explicit BaseTest(TimeDelta timeout);
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virtual ~BaseTest();
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virtual void PerformTest() = 0;
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virtual bool ShouldCreateReceivers() const = 0;
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virtual size_t GetNumVideoStreams() const;
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virtual size_t GetNumAudioStreams() const;
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virtual size_t GetNumFlexfecStreams() const;
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virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
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virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
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virtual void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
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AudioDeviceModule* recv_audio_device);
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virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
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virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
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virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
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virtual void OnTransportCreated(PacketTransport* to_receiver,
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SimulatedNetworkInterface* sender_network,
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PacketTransport* to_sender,
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SimulatedNetworkInterface* receiver_network);
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virtual BuiltInNetworkBehaviorConfig GetSendTransportConfig() const;
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virtual BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const;
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virtual void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
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VideoEncoderConfig* encoder_config);
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virtual void ModifyVideoCaptureStartResolution(int* width,
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int* heigt,
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int* frame_rate);
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virtual void ModifyVideoDegradationPreference(
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DegradationPreference* degradation_preference);
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virtual void OnVideoStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStreamInterface*>& receive_streams);
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virtual void ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStreamInterface::Config>* receive_configs);
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virtual void OnAudioStreamsCreated(
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AudioSendStream* send_stream,
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const std::vector<AudioReceiveStreamInterface*>& receive_streams);
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virtual void ModifyFlexfecConfigs(
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std::vector<FlexfecReceiveStream::Config>* receive_configs);
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virtual void OnFlexfecStreamsCreated(
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const std::vector<FlexfecReceiveStream*>& receive_streams);
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virtual void OnFrameGeneratorCapturerCreated(
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FrameGeneratorCapturer* frame_generator_capturer);
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virtual void OnStreamsStopped();
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};
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class SendTest : public BaseTest {
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public:
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explicit SendTest(TimeDelta timeout);
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bool ShouldCreateReceivers() const override;
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};
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class EndToEndTest : public BaseTest {
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public:
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EndToEndTest();
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explicit EndToEndTest(TimeDelta timeout);
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bool ShouldCreateReceivers() const override;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_CALL_TEST_H_
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