webrtc::RtpRtcpAudioTest Class Reference

Inheritance diagram for webrtc::RtpRtcpAudioTest:

Protected Member Functions
	RtpRtcpAudioTest ()
	~RtpRtcpAudioTest ()
void	SetUp () override
void	RegisterPayload (const CodecInst &codec)
Protected Member Functions inherited from testing::Test
	Test ()
virtual void	TearDown ()

Protected Attributes
VerifyingAudioReceiver	data_receiver1
VerifyingAudioReceiver	data_receiver2
RTPCallback	rtp_callback
std::unique_ptr< ReceiveStatistics >	receive_statistics1_
std::unique_ptr< ReceiveStatistics >	receive_statistics2_
std::unique_ptr< RTPPayloadRegistry >	rtp_payload_registry1_
std::unique_ptr< RTPPayloadRegistry >	rtp_payload_registry2_
std::unique_ptr< RtpReceiver >	rtp_receiver1_
std::unique_ptr< RtpReceiver >	rtp_receiver2_
std::unique_ptr< RtpRtcp >	module1
std::unique_ptr< RtpRtcp >	module2
LoopBackTransport	transport1
LoopBackTransport	transport2
uint32_t	test_ssrc
uint32_t	test_timestamp
uint16_t	test_sequence_number
uint32_t	test_CSRC [webrtc::kRtpCsrcSize]
SimulatedClock	fake_clock
RateLimiter	retransmission_rate_limiter_

Constructor & Destructor Documentation

RtpRtcpAudioTest()

webrtc::RtpRtcpAudioTest::RtpRtcpAudioTest ( )

inlineprotected

~RtpRtcpAudioTest()

webrtc::RtpRtcpAudioTest::~RtpRtcpAudioTest ( )

inlineprotected

Member Function Documentation

RegisterPayload()

void webrtc::RtpRtcpAudioTest::RegisterPayload ( const CodecInst &  codec )

inlineprotected

SetUp()

void webrtc::RtpRtcpAudioTest::SetUp ( )

inlineoverrideprotectedvirtual

Reimplemented from testing::Test.

Member Data Documentation

data_receiver1

VerifyingAudioReceiver webrtc::RtpRtcpAudioTest::data_receiver1

protected

data_receiver2

VerifyingAudioReceiver webrtc::RtpRtcpAudioTest::data_receiver2

protected

fake_clock

SimulatedClock webrtc::RtpRtcpAudioTest::fake_clock

protected

module1

std::unique_ptr<RtpRtcp> webrtc::RtpRtcpAudioTest::module1

protected

module2

std::unique_ptr<RtpRtcp> webrtc::RtpRtcpAudioTest::module2

protected

receive_statistics1_

std::unique_ptr<ReceiveStatistics> webrtc::RtpRtcpAudioTest::receive_statistics1_

protected

receive_statistics2_

std::unique_ptr<ReceiveStatistics> webrtc::RtpRtcpAudioTest::receive_statistics2_

protected

retransmission_rate_limiter_

RateLimiter webrtc::RtpRtcpAudioTest::retransmission_rate_limiter_

protected

rtp_callback

RTPCallback webrtc::RtpRtcpAudioTest::rtp_callback

protected

rtp_payload_registry1_

std::unique_ptr<RTPPayloadRegistry> webrtc::RtpRtcpAudioTest::rtp_payload_registry1_

protected

rtp_payload_registry2_

std::unique_ptr<RTPPayloadRegistry> webrtc::RtpRtcpAudioTest::rtp_payload_registry2_

protected

rtp_receiver1_

std::unique_ptr<RtpReceiver> webrtc::RtpRtcpAudioTest::rtp_receiver1_

protected

rtp_receiver2_

std::unique_ptr<RtpReceiver> webrtc::RtpRtcpAudioTest::rtp_receiver2_

protected

test_CSRC

uint32_t webrtc::RtpRtcpAudioTest::test_CSRC[webrtc::kRtpCsrcSize]

protected

test_sequence_number

uint16_t webrtc::RtpRtcpAudioTest::test_sequence_number

protected

test_ssrc

uint32_t webrtc::RtpRtcpAudioTest::test_ssrc

protected

test_timestamp

uint32_t webrtc::RtpRtcpAudioTest::test_timestamp

protected

transport1

LoopBackTransport webrtc::RtpRtcpAudioTest::transport1

protected

transport2

LoopBackTransport webrtc::RtpRtcpAudioTest::transport2

protected

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

• Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc