webrtc::test::CallTest Class Reference

#include <call_test.h>

Inheritance diagram for webrtc::test::CallTest:

Public Member Functions
	CallTest ()
virtual	~CallTest ()
	CallTest ()
virtual	~CallTest ()

Static Public Attributes
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

Protected Member Functions
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
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_

Constructor & Destructor Documentation

CallTest() [1/2]

webrtc::test::CallTest::CallTest

(

)

~CallTest() [1/2]

webrtc::test::CallTest::~CallTest ( )

virtual

CallTest() [2/2]

webrtc::test::CallTest::CallTest

(

)

~CallTest() [2/2]

virtual webrtc::test::CallTest::~CallTest ( )

virtual

Member Function Documentation

CreateAudioStreams() [1/2]

void webrtc::test::CallTest::CreateAudioStreams ( )

protected

CreateAudioStreams() [2/2]

void webrtc::test::CallTest::CreateAudioStreams ( )

protected

CreateCalls() [1/2]

void webrtc::test::CallTest::CreateCalls ( const Call::Config &  sender_config, const Call::Config &  receiver_config  )

protected

CreateCalls() [2/2]

void webrtc::test::CallTest::CreateCalls ( const Call::Config &  sender_config, const Call::Config &  receiver_config  )

protected

CreateFakeAudioDevices() [1/2]

void webrtc::test::CallTest::CreateFakeAudioDevices ( )

protected

CreateFakeAudioDevices() [2/2]

void webrtc::test::CallTest::CreateFakeAudioDevices ( )

protected

CreateFlexfecStreams() [1/2]

void webrtc::test::CallTest::CreateFlexfecStreams ( )

protected

CreateFlexfecStreams() [2/2]

void webrtc::test::CallTest::CreateFlexfecStreams ( )

protected

CreateFrameGeneratorCapturer() [1/2]

void webrtc::test::CallTest::CreateFrameGeneratorCapturer ( int  framerate, int  width, int  height  )

protected

CreateFrameGeneratorCapturer() [2/2]

void webrtc::test::CallTest::CreateFrameGeneratorCapturer ( int  framerate, int  width, int  height  )

protected

CreateFrameGeneratorCapturerWithDrift() [1/2]

void webrtc::test::CallTest::CreateFrameGeneratorCapturerWithDrift ( Clock *  drift_clock, float  speed, int  framerate, int  width, int  height  )

protected

CreateFrameGeneratorCapturerWithDrift() [2/2]

void webrtc::test::CallTest::CreateFrameGeneratorCapturerWithDrift ( Clock *  drift_clock, float  speed, int  framerate, int  width, int  height  )

protected

CreateMatchingReceiveConfigs() [1/2]

void webrtc::test::CallTest::CreateMatchingReceiveConfigs ( Transport *  rtcp_send_transport )

protected

CreateMatchingReceiveConfigs() [2/2]

void webrtc::test::CallTest::CreateMatchingReceiveConfigs ( Transport *  rtcp_send_transport )

protected

CreateReceiverCall() [1/2]

void webrtc::test::CallTest::CreateReceiverCall ( const Call::Config &  config )

protected

CreateReceiverCall() [2/2]

void webrtc::test::CallTest::CreateReceiverCall ( const Call::Config &  config )

protected

CreateSendConfig() [1/2]

void webrtc::test::CallTest::CreateSendConfig ( size_t  num_video_streams, size_t  num_audio_streams, size_t  num_flexfec_streams, Transport *  send_transport  )

protected

CreateSendConfig() [2/2]

void webrtc::test::CallTest::CreateSendConfig ( size_t  num_video_streams, size_t  num_audio_streams, size_t  num_flexfec_streams, Transport *  send_transport  )

protected

CreateSenderCall() [1/2]

void webrtc::test::CallTest::CreateSenderCall ( const Call::Config &  config )

protected

CreateSenderCall() [2/2]

void webrtc::test::CallTest::CreateSenderCall ( const Call::Config &  config )

protected

CreateVideoStreams() [1/2]

void webrtc::test::CallTest::CreateVideoStreams ( )

protected

CreateVideoStreams() [2/2]

void webrtc::test::CallTest::CreateVideoStreams ( )

protected

DestroyCalls() [1/2]

void webrtc::test::CallTest::DestroyCalls ( )

protected

DestroyCalls() [2/2]

void webrtc::test::CallTest::DestroyCalls ( )

protected

DestroyStreams() [1/2]

void webrtc::test::CallTest::DestroyStreams ( )

protected

DestroyStreams() [2/2]

void webrtc::test::CallTest::DestroyStreams ( )

protected

RunBaseTest() [1/2]

void webrtc::test::CallTest::RunBaseTest ( BaseTest *  test )

protected

RunBaseTest() [2/2]

void webrtc::test::CallTest::RunBaseTest ( BaseTest *  test )

protected

SetFakeVideoCaptureRotation() [1/2]

void webrtc::test::CallTest::SetFakeVideoCaptureRotation ( VideoRotation  rotation )

protected

SetFakeVideoCaptureRotation() [2/2]

void webrtc::test::CallTest::SetFakeVideoCaptureRotation ( VideoRotation  rotation )

protected

Start() [1/2]

void webrtc::test::CallTest::Start ( )

protected

Start() [2/2]

void webrtc::test::CallTest::Start ( )

protected

Stop() [1/2]

void webrtc::test::CallTest::Stop ( )

protected

Stop() [2/2]

void webrtc::test::CallTest::Stop ( )

protected

Member Data Documentation

allocated_decoders_

std::vector< std::unique_ptr< VideoDecoder > > webrtc::test::CallTest::allocated_decoders_

protected

audio_receive_configs_

std::vector< AudioReceiveStream::Config > webrtc::test::CallTest::audio_receive_configs_

protected

audio_receive_streams_

std::vector< AudioReceiveStream * > webrtc::test::CallTest::audio_receive_streams_

protected

audio_send_config_

AudioSendStream::Config webrtc::test::CallTest::audio_send_config_

protected

audio_send_stream_

AudioSendStream * webrtc::test::CallTest::audio_send_stream_

protected

clock_

Clock *const webrtc::test::CallTest::clock_

protected

decoder_factory_

rtc::scoped_refptr< AudioDecoderFactory > webrtc::test::CallTest::decoder_factory_

protected

event_log_

webrtc::RtcEventLogNullImpl webrtc::test::CallTest::event_log_

protected

fake_encoder_

test::FakeEncoder webrtc::test::CallTest::fake_encoder_

protected

fake_renderer_

test::FakeVideoRenderer webrtc::test::CallTest::fake_renderer_

protected

flexfec_receive_configs_

std::vector< FlexfecReceiveStream::Config > webrtc::test::CallTest::flexfec_receive_configs_

protected

flexfec_receive_streams_

std::vector< FlexfecReceiveStream * > webrtc::test::CallTest::flexfec_receive_streams_

protected

frame_generator_capturer_

std::unique_ptr< test::FrameGeneratorCapturer > webrtc::test::CallTest::frame_generator_capturer_

protected

kAudioSendPayloadType

static const uint8_t webrtc::test::CallTest::kAudioSendPayloadType = 103

static

kAudioSendSsrc

static const uint32_t webrtc::test::CallTest::kAudioSendSsrc = 0xDEADBEEF

static

kDefaultFramerate

static const int webrtc::test::CallTest::kDefaultFramerate = 30

static

kDefaultHeight

static const int webrtc::test::CallTest::kDefaultHeight = 180

static

kDefaultTimeoutMs

static const int webrtc::test::CallTest::kDefaultTimeoutMs = 30 * 1000

static

kDefaultWidth

static const int webrtc::test::CallTest::kDefaultWidth = 320

static

kFakeVideoSendPayloadType

static const uint8_t webrtc::test::CallTest::kFakeVideoSendPayloadType = 125

static

kFlexfecPayloadType

static const uint8_t webrtc::test::CallTest::kFlexfecPayloadType = 120

static

kFlexfecSendSsrc

static const uint32_t webrtc::test::CallTest::kFlexfecSendSsrc = 0xBADBEEF

static

kLongTimeoutMs

static const int webrtc::test::CallTest::kLongTimeoutMs = 120 * 1000

static

kNackRtpHistoryMs

static const int webrtc::test::CallTest::kNackRtpHistoryMs = 1000

static

kNumSsrcs

static const size_t webrtc::test::CallTest::kNumSsrcs = 3

static

kReceiverLocalAudioSsrc

static const uint32_t webrtc::test::CallTest::kReceiverLocalAudioSsrc = 0x1234567

static

kReceiverLocalVideoSsrc

static const uint32_t webrtc::test::CallTest::kReceiverLocalVideoSsrc = 0x123456

static

kRedPayloadType

static const uint8_t webrtc::test::CallTest::kRedPayloadType = 118

static

kRtxRedPayloadType

static const uint8_t webrtc::test::CallTest::kRtxRedPayloadType = 99

static

kSendRtxPayloadType

static const uint8_t webrtc::test::CallTest::kSendRtxPayloadType = 98

static

kSendRtxSsrcs

static const uint32_t webrtc::test::CallTest::kSendRtxSsrcs

static

Initial value:

= {0xBADCAFD, 0xBADCAFE,
                                                     0xBADCAFF}

kUlpfecPayloadType

static const uint8_t webrtc::test::CallTest::kUlpfecPayloadType = 119

static

kVideoSendPayloadType

static const uint8_t webrtc::test::CallTest::kVideoSendPayloadType = 100

static

kVideoSendSsrcs

static const uint32_t webrtc::test::CallTest::kVideoSendSsrcs

static

Initial value:

= {0xC0FFED, 0xC0FFEE,
                                                       0xC0FFEF}

num_audio_streams_

size_t webrtc::test::CallTest::num_audio_streams_

protected

num_flexfec_streams_

size_t webrtc::test::CallTest::num_flexfec_streams_

protected

num_video_streams_

size_t webrtc::test::CallTest::num_video_streams_

protected

receive_transport_

std::unique_ptr< PacketTransport > webrtc::test::CallTest::receive_transport_

protected

receiver_call_

std::unique_ptr< Call > webrtc::test::CallTest::receiver_call_

protected

send_transport_

std::unique_ptr< PacketTransport > webrtc::test::CallTest::send_transport_

protected

sender_call_

std::unique_ptr< Call > webrtc::test::CallTest::sender_call_

protected

video_encoder_config_

VideoEncoderConfig webrtc::test::CallTest::video_encoder_config_

protected

video_receive_configs_

std::vector< VideoReceiveStream::Config > webrtc::test::CallTest::video_receive_configs_

protected

video_receive_streams_

std::vector< VideoReceiveStream * > webrtc::test::CallTest::video_receive_streams_

protected

video_send_config_

VideoSendStream::Config webrtc::test::CallTest::video_send_config_

protected

video_send_stream_

VideoSendStream * webrtc::test::CallTest::video_send_stream_

protected

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/test/call_test.h
• Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc