webrtc::Channel Class Reference

#include <Channel.h>

Inheritance diagram for webrtc::Channel:

Public Member Functions
	Channel (int16_t chID=-1)
	~Channel () override
int32_t	SendData (FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override
void	RegisterReceiverACM (AudioCodingModule *acm)
void	ResetStats ()
int16_t	Stats (CodecInst &codecInst, ACMTestPayloadStats &payloadStats)
void	Stats (uint32_t *numPackets)
void	Stats (uint8_t *payloadType, uint32_t *payloadLenByte)
void	PrintStats (CodecInst &codecInst)
void	SetIsStereo (bool isStereo)
uint32_t	LastInTimestamp ()
void	SetFECTestWithPacketLoss (bool usePacketLoss)
double	BitRate ()
void	set_send_timestamp (uint32_t new_send_ts)
void	set_sequence_number (uint16_t new_sequence_number)
void	set_num_packets_to_drop (int new_num_packets_to_drop)
	Channel (int16_t chID=-1)
	~Channel () override
int32_t	SendData (FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override
void	RegisterReceiverACM (AudioCodingModule *acm)
void	ResetStats ()
int16_t	Stats (CodecInst &codecInst, ACMTestPayloadStats &payloadStats)
void	Stats (uint32_t *numPackets)
void	Stats (uint8_t *payloadType, uint32_t *payloadLenByte)
void	PrintStats (CodecInst &codecInst)
void	SetIsStereo (bool isStereo)
uint32_t	LastInTimestamp ()
void	SetFECTestWithPacketLoss (bool usePacketLoss)
double	BitRate ()
void	set_send_timestamp (uint32_t new_send_ts)
void	set_sequence_number (uint16_t new_sequence_number)
void	set_num_packets_to_drop (int new_num_packets_to_drop)
Public Member Functions inherited from webrtc::AudioPacketizationCallback
virtual	~AudioPacketizationCallback ()
virtual	~AudioPacketizationCallback ()

Constructor & Destructor Documentation

Channel() [1/2]

webrtc::Channel::Channel

(

int16_t

chID = -1

)

~Channel() [1/2]

webrtc::Channel::~Channel ( )

override

Channel() [2/2]

webrtc::Channel::Channel

(

int16_t

chID = -1

)

~Channel() [2/2]

webrtc::Channel::~Channel ( )

override

Member Function Documentation

BitRate() [1/2]

double webrtc::Channel::BitRate

(

)

BitRate() [2/2]

double webrtc::Channel::BitRate

(

)

LastInTimestamp() [1/2]

uint32_t webrtc::Channel::LastInTimestamp

(

)

LastInTimestamp() [2/2]

uint32_t webrtc::Channel::LastInTimestamp

(

)

PrintStats() [1/2]

void webrtc::Channel::PrintStats

(

CodecInst &

codecInst

)

PrintStats() [2/2]

void webrtc::Channel::PrintStats

(

CodecInst &

codecInst

)

RegisterReceiverACM() [1/2]

void webrtc::Channel::RegisterReceiverACM

(

AudioCodingModule *

acm

)

RegisterReceiverACM() [2/2]

void webrtc::Channel::RegisterReceiverACM

(

AudioCodingModule *

acm

)

ResetStats() [1/2]

void webrtc::Channel::ResetStats

(

)

ResetStats() [2/2]

void webrtc::Channel::ResetStats

(

)

SendData() [1/2]

int32_t webrtc::Channel::SendData ( FrameType  frameType, uint8_t  payloadType, uint32_t  timeStamp, const uint8_t *  payloadData, size_t  payloadSize, const RTPFragmentationHeader *  fragmentation  )

overridevirtual

Implements webrtc::AudioPacketizationCallback.

SendData() [2/2]

int32_t webrtc::Channel::SendData ( FrameType  frameType, uint8_t  payloadType, uint32_t  timeStamp, const uint8_t *  payloadData, size_t  payloadSize, const RTPFragmentationHeader *  fragmentation  )

overridevirtual

Implements webrtc::AudioPacketizationCallback.

set_num_packets_to_drop() [1/2]

void webrtc::Channel::set_num_packets_to_drop ( int  new_num_packets_to_drop )

inline

set_num_packets_to_drop() [2/2]

void webrtc::Channel::set_num_packets_to_drop ( int  new_num_packets_to_drop )

inline

set_send_timestamp() [1/2]

void webrtc::Channel::set_send_timestamp ( uint32_t  new_send_ts )

inline

set_send_timestamp() [2/2]

void webrtc::Channel::set_send_timestamp ( uint32_t  new_send_ts )

inline

set_sequence_number() [1/2]

void webrtc::Channel::set_sequence_number ( uint16_t  new_sequence_number )

inline

set_sequence_number() [2/2]

void webrtc::Channel::set_sequence_number ( uint16_t  new_sequence_number )

inline

SetFECTestWithPacketLoss() [1/2]

void webrtc::Channel::SetFECTestWithPacketLoss ( bool  usePacketLoss )

inline

SetFECTestWithPacketLoss() [2/2]

void webrtc::Channel::SetFECTestWithPacketLoss ( bool  usePacketLoss )

inline

SetIsStereo() [1/2]

void webrtc::Channel::SetIsStereo ( bool  isStereo )

inline

SetIsStereo() [2/2]

void webrtc::Channel::SetIsStereo ( bool  isStereo )

inline

Stats() [1/6]

int16_t webrtc::Channel::Stats	(	CodecInst &	codecInst,
		ACMTestPayloadStats &	payloadStats
	)

Stats() [2/6]

int16_t webrtc::Channel::Stats	(	CodecInst &	codecInst,
		ACMTestPayloadStats &	payloadStats
	)

Stats() [3/6]

void webrtc::Channel::Stats

(

uint32_t *

numPackets

)

Stats() [4/6]

void webrtc::Channel::Stats

(

uint32_t *

numPackets

)

Stats() [5/6]

void webrtc::Channel::Stats	(	uint8_t *	payloadType,
		uint32_t *	payloadLenByte
	)

Stats() [6/6]

void webrtc::Channel::Stats	(	uint8_t *	payloadType,
		uint32_t *	payloadLenByte
	)

---

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

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