webrtc::EndToEndTest Class Reference

Inheritance diagram for webrtc::EndToEndTest:

Classes
class	RequiredTransport
class	UnusedTransport

Public Member Functions
	EndToEndTest ()
virtual	~EndToEndTest ()
Public Member Functions inherited from webrtc::test::CallTest
	CallTest ()
virtual	~CallTest ()
	CallTest ()
virtual	~CallTest ()

Protected Member Functions
void	DecodesRetransmittedFrame (bool enable_rtx, bool enable_red)
void	ReceivesPliAndRecovers (int rtp_history_ms)
void	RespectsRtcpMode (RtcpMode rtcp_mode)
void	TestSendsSetSsrcs (size_t num_ssrcs, bool send_single_ssrc_first)
void	TestRtpStatePreservation (bool use_rtx, bool provoke_rtcpsr_before_rtp)
void	VerifyHistogramStats (bool use_rtx, bool use_red, bool screenshare)
void	VerifyNewVideoSendStreamsRespectNetworkState (MediaType network_to_bring_up, VideoEncoder *encoder, Transport *transport)
void	VerifyNewVideoReceiveStreamsRespectNetworkState (MediaType network_to_bring_up, Transport *transport)
Protected Member Functions inherited from webrtc::test::CallTest
void	RunBaseTest (BaseTest *test)
void	CreateCalls (const Call::Config &sender_config, const Call::Config &receiver_config)
void	CreateSenderCall (const Call::Config &config)
void	CreateReceiverCall (const Call::Config &config)
void	DestroyCalls ()
void	CreateSendConfig (size_t num_video_streams, size_t num_audio_streams, size_t num_flexfec_streams, Transport *send_transport)
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 ()
void	CreateVideoStreams ()
void	CreateAudioStreams ()
void	CreateFlexfecStreams ()
void	Start ()
void	Stop ()
void	DestroyStreams ()
void	SetFakeVideoCaptureRotation (VideoRotation rotation)
void	RunBaseTest (BaseTest *test)
void	CreateCalls (const Call::Config &sender_config, const Call::Config &receiver_config)
void	CreateSenderCall (const Call::Config &config)
void	CreateReceiverCall (const Call::Config &config)
void	DestroyCalls ()
void	CreateSendConfig (size_t num_video_streams, size_t num_audio_streams, size_t num_flexfec_streams, Transport *send_transport)
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 ()
void	CreateVideoStreams ()
void	CreateAudioStreams ()
void	CreateFlexfecStreams ()
void	Start ()
void	Stop ()
void	DestroyStreams ()
void	SetFakeVideoCaptureRotation (VideoRotation rotation)
Protected Member Functions inherited from testing::Test
	Test ()
virtual void	SetUp ()
virtual void	TearDown ()

Protected Attributes
test::ScopedFieldTrials	scoped_field_trial_
Protected Attributes inherited from webrtc::test::CallTest
Clock *const	clock_
webrtc::RtcEventLogNullImpl	event_log_
std::unique_ptr< Call >	sender_call_
std::unique_ptr< PacketTransport >	send_transport_
VideoSendStream::Config	video_send_config_
VideoEncoderConfig	video_encoder_config_
VideoSendStream *	video_send_stream_
AudioSendStream::Config	audio_send_config_
AudioSendStream *	audio_send_stream_
std::unique_ptr< Call >	receiver_call_
std::unique_ptr< PacketTransport >	receive_transport_
std::vector< VideoReceiveStream::Config >	video_receive_configs_
std::vector< VideoReceiveStream * >	video_receive_streams_
std::vector< AudioReceiveStream::Config >	audio_receive_configs_
std::vector< AudioReceiveStream * >	audio_receive_streams_
std::vector< FlexfecReceiveStream::Config >	flexfec_receive_configs_
std::vector< FlexfecReceiveStream * >	flexfec_receive_streams_
std::unique_ptr< test::FrameGeneratorCapturer >	frame_generator_capturer_
test::FakeEncoder	fake_encoder_
std::vector< std::unique_ptr< VideoDecoder > >	allocated_decoders_
size_t	num_video_streams_
size_t	num_audio_streams_
size_t	num_flexfec_streams_
rtc::scoped_refptr< AudioDecoderFactory >	decoder_factory_
test::FakeVideoRenderer	fake_renderer_

Additional Inherited Members
Static Public Attributes inherited from webrtc::test::CallTest
static const size_t	kNumSsrcs = 3
static const int	kDefaultWidth = 320
static const int	kDefaultHeight = 180
static const int	kDefaultFramerate = 30
static const int	kDefaultTimeoutMs = 30 * 1000
static const int	kLongTimeoutMs = 120 * 1000
static const uint8_t	kVideoSendPayloadType = 100
static const uint8_t	kSendRtxPayloadType = 98
static const uint8_t	kFakeVideoSendPayloadType = 125
static const uint8_t	kRedPayloadType = 118
static const uint8_t	kRtxRedPayloadType = 99
static const uint8_t	kUlpfecPayloadType = 119
static const uint8_t	kFlexfecPayloadType = 120
static const uint8_t	kAudioSendPayloadType = 103
static const uint32_t	kSendRtxSsrcs [kNumSsrcs]
static const uint32_t	kVideoSendSsrcs [kNumSsrcs]
static const uint32_t	kAudioSendSsrc = 0xDEADBEEF
static const uint32_t	kFlexfecSendSsrc = 0xBADBEEF
static const uint32_t	kReceiverLocalVideoSsrc = 0x123456
static const uint32_t	kReceiverLocalAudioSsrc = 0x1234567
static const int	kNackRtpHistoryMs = 1000

Constructor & Destructor Documentation

EndToEndTest()

webrtc::EndToEndTest::EndToEndTest ( )

inline

~EndToEndTest()

virtual webrtc::EndToEndTest::~EndToEndTest ( )

inlinevirtual

Member Function Documentation

DecodesRetransmittedFrame()

void webrtc::EndToEndTest::DecodesRetransmittedFrame ( bool  enable_rtx, bool  enable_red  )

protected

ReceivesPliAndRecovers()

void webrtc::EndToEndTest::ReceivesPliAndRecovers ( int  rtp_history_ms )

protected

RespectsRtcpMode()

void webrtc::EndToEndTest::RespectsRtcpMode ( RtcpMode  rtcp_mode )

protected

TestRtpStatePreservation()

void webrtc::EndToEndTest::TestRtpStatePreservation ( bool  use_rtx, bool  provoke_rtcpsr_before_rtp  )

protected

TestSendsSetSsrcs()

void webrtc::EndToEndTest::TestSendsSetSsrcs ( size_t  num_ssrcs, bool  send_single_ssrc_first  )

protected

VerifyHistogramStats()

void webrtc::EndToEndTest::VerifyHistogramStats ( bool  use_rtx, bool  use_red, bool  screenshare  )

protected

VerifyNewVideoReceiveStreamsRespectNetworkState()

void webrtc::EndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState ( MediaType  network_to_bring_up, Transport *  transport  )

protected

VerifyNewVideoSendStreamsRespectNetworkState()

void webrtc::EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState ( MediaType  network_to_bring_up, VideoEncoder *  encoder, Transport *  transport  )

protected

Member Data Documentation

scoped_field_trial_

test::ScopedFieldTrials webrtc::EndToEndTest::scoped_field_trial_

protected

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The documentation for this class was generated from the following file:

• Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests.cc