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/*
* 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 Call::Config& sender_config,
const Call::Config& receiver_config);
void CreateSenderCall();
void CreateSenderCall(const Call::Config& config);
void CreateReceiverCall(const Call::Config& 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_