webrtc::ReceiverWithPacketLoss Class Reference

#include <PacketLossTest.h>

Inheritance diagram for webrtc::ReceiverWithPacketLoss:

Public Member Functions
	ReceiverWithPacketLoss ()
void	Setup (AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length)
bool	IncomingPacket () override
	ReceiverWithPacketLoss ()
void	Setup (AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length)
bool	IncomingPacket () override
Public Member Functions inherited from webrtc::Receiver
	Receiver ()
virtual	~Receiver ()
void	Setup (AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, size_t channels)
void	Teardown ()
void	Run ()
bool	PlayoutData ()
	Receiver ()
virtual	~Receiver ()
void	Setup (AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, size_t channels)
void	Teardown ()
void	Run ()
bool	PlayoutData ()

Protected Member Functions
bool	PacketLost ()
bool	PacketLost ()

Protected Attributes
int	loss_rate_
int	burst_length_
int	packet_counter_
int	lost_packet_counter_
int	burst_lost_counter_
Protected Attributes inherited from webrtc::Receiver
AudioCodingModule *	_acm
uint8_t	_incomingPayload [MAX_INCOMING_PAYLOAD]
RTPStream *	_rtpStream
WebRtcRTPHeader	_rtpInfo
size_t	_realPayloadSizeBytes
size_t	_payloadSizeBytes
uint32_t	_nextTime

Additional Inherited Members
Public Attributes inherited from webrtc::Receiver
uint8_t	codeId
uint8_t	testMode

Constructor & Destructor Documentation

ReceiverWithPacketLoss() [1/2]

webrtc::ReceiverWithPacketLoss::ReceiverWithPacketLoss

(

)

ReceiverWithPacketLoss() [2/2]

webrtc::ReceiverWithPacketLoss::ReceiverWithPacketLoss

(

)

Member Function Documentation

IncomingPacket() [1/2]

bool webrtc::ReceiverWithPacketLoss::IncomingPacket ( )

overridevirtual

Reimplemented from webrtc::Receiver.

IncomingPacket() [2/2]

bool webrtc::ReceiverWithPacketLoss::IncomingPacket ( )

overridevirtual

Reimplemented from webrtc::Receiver.

PacketLost() [1/2]

bool webrtc::ReceiverWithPacketLoss::PacketLost ( )

protected

PacketLost() [2/2]

bool webrtc::ReceiverWithPacketLoss::PacketLost ( )

protected

Setup() [1/2]

void webrtc::ReceiverWithPacketLoss::Setup	(	AudioCodingModule *	acm,
		RTPStream *	rtpStream,
		std::string	out_file_name,
		int	channels,
		int	loss_rate,
		int	burst_length
	)

Setup() [2/2]

void webrtc::ReceiverWithPacketLoss::Setup	(	AudioCodingModule *	acm,
		RTPStream *	rtpStream,
		std::string	out_file_name,
		int	channels,
		int	loss_rate,
		int	burst_length
	)

Member Data Documentation

burst_length_

int webrtc::ReceiverWithPacketLoss::burst_length_

protected

burst_lost_counter_

int webrtc::ReceiverWithPacketLoss::burst_lost_counter_

protected

loss_rate_

int webrtc::ReceiverWithPacketLoss::loss_rate_

protected

lost_packet_counter_

int webrtc::ReceiverWithPacketLoss::lost_packet_counter_

protected

packet_counter_

int webrtc::ReceiverWithPacketLoss::packet_counter_

protected

---

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

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