webrtc::RTPSender Class Reference

#include <rtp_sender.h>

Public Member Functions
	RTPSender (bool audio, Clock *clock, Transport *transport, RtpPacketSender *paced_sender, FlexfecSender *flexfec_sender, TransportSequenceNumberAllocator *sequence_number_allocator, TransportFeedbackObserver *transport_feedback_callback, BitrateStatisticsObserver *bitrate_callback, FrameCountObserver *frame_count_observer, SendSideDelayObserver *send_side_delay_observer, RtcEventLog *event_log, SendPacketObserver *send_packet_observer, RateLimiter *nack_rate_limiter, OverheadObserver *overhead_observer)
	~RTPSender ()
void	ProcessBitrate ()
uint16_t	ActualSendBitrateKbit () const
uint32_t	VideoBitrateSent () const
uint32_t	FecOverheadRate () const
uint32_t	NackOverheadRate () const
size_t	MaxPayloadSize () const
int32_t	RegisterPayload (const char *payload_name, const int8_t payload_type, const uint32_t frequency, const size_t channels, const uint32_t rate)
int32_t	DeRegisterSendPayload (const int8_t payload_type)
void	SetSendPayloadType (int8_t payload_type)
int8_t	SendPayloadType () const
void	SetSendingMediaStatus (bool enabled)
bool	SendingMedia () const
void	GetDataCounters (StreamDataCounters *rtp_stats, StreamDataCounters *rtx_stats) const
uint32_t	TimestampOffset () const
void	SetTimestampOffset (uint32_t timestamp)
void	SetSSRC (uint32_t ssrc)
uint16_t	SequenceNumber () const
void	SetSequenceNumber (uint16_t seq)
void	SetCsrcs (const std::vector< uint32_t > &csrcs)
void	SetMaxRtpPacketSize (size_t max_packet_size)
bool	SendOutgoingData (FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_header, uint32_t *transport_frame_id_out)
int32_t	RegisterRtpHeaderExtension (RTPExtensionType type, uint8_t id)
bool	IsRtpHeaderExtensionRegistered (RTPExtensionType type) const
int32_t	DeregisterRtpHeaderExtension (RTPExtensionType type)
bool	TimeToSendPacket (uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)
size_t	TimeToSendPadding (size_t bytes, const PacedPacketInfo &pacing_info)
int	SelectiveRetransmissions () const
int	SetSelectiveRetransmissions (uint8_t settings)
void	OnReceivedNack (const std::vector< uint16_t > &nack_sequence_numbers, int64_t avg_rtt)
void	SetStorePacketsStatus (bool enable, uint16_t number_to_store)
bool	StorePackets () const
int32_t	ReSendPacket (uint16_t packet_id, int64_t min_resend_time=0)
void	OnReceivedRtcpReportBlocks (const ReportBlockList &report_blocks)
void	SetRtxStatus (int mode)
int	RtxStatus () const
uint32_t	RtxSsrc () const
void	SetRtxSsrc (uint32_t ssrc)
void	SetRtxPayloadType (int payload_type, int associated_payload_type)
std::unique_ptr< RtpPacketToSend >	AllocatePacket () const
bool	AssignSequenceNumber (RtpPacketToSend *packet)
size_t	RtpHeaderLength () const
uint16_t	AllocateSequenceNumber (uint16_t packets_to_send)
size_t	MaxRtpPacketSize () const
uint32_t	SSRC () const
rtc::Optional< uint32_t >	FlexfecSsrc () const
bool	SendToNetwork (std::unique_ptr< RtpPacketToSend > packet, StorageType storage, RtpPacketSender::Priority priority)
int32_t	SendTelephoneEvent (uint8_t key, uint16_t time_ms, uint8_t level)
RTC_DEPRECATED int32_t	SetAudioPacketSize (uint16_t packet_size_samples)
int32_t	SetAudioLevel (uint8_t level_d_bov)
RtpVideoCodecTypes	VideoCodecType () const
uint32_t	MaxConfiguredBitrateVideo () const
void	SetUlpfecConfig (int red_payload_type, int ulpfec_payload_type)
bool	SetFecParameters (const FecProtectionParams &delta_params, const FecProtectionParams &key_params)
void	RegisterRtpStatisticsCallback (StreamDataCountersCallback *callback)
StreamDataCountersCallback *	GetRtpStatisticsCallback () const
uint32_t	BitrateSent () const
void	SetRtpState (const RtpState &rtp_state)
RtpState	GetRtpState () const
void	SetRtxRtpState (const RtpState &rtp_state)
RtpState	GetRtxRtpState () const
	RTPSender (bool audio, Clock *clock, Transport *transport, RtpPacketSender *paced_sender, FlexfecSender *flexfec_sender, TransportSequenceNumberAllocator *sequence_number_allocator, TransportFeedbackObserver *transport_feedback_callback, BitrateStatisticsObserver *bitrate_callback, FrameCountObserver *frame_count_observer, SendSideDelayObserver *send_side_delay_observer, RtcEventLog *event_log, SendPacketObserver *send_packet_observer, RateLimiter *nack_rate_limiter, OverheadObserver *overhead_observer)
	~RTPSender ()
void	ProcessBitrate ()
uint16_t	ActualSendBitrateKbit () const
uint32_t	VideoBitrateSent () const
uint32_t	FecOverheadRate () const
uint32_t	NackOverheadRate () const
size_t	MaxPayloadSize () const
int32_t	RegisterPayload (const char *payload_name, const int8_t payload_type, const uint32_t frequency, const size_t channels, const uint32_t rate)
int32_t	DeRegisterSendPayload (const int8_t payload_type)
void	SetSendPayloadType (int8_t payload_type)
int8_t	SendPayloadType () const
void	SetSendingMediaStatus (bool enabled)
bool	SendingMedia () const
void	GetDataCounters (StreamDataCounters *rtp_stats, StreamDataCounters *rtx_stats) const
uint32_t	TimestampOffset () const
void	SetTimestampOffset (uint32_t timestamp)
void	SetSSRC (uint32_t ssrc)
uint16_t	SequenceNumber () const
void	SetSequenceNumber (uint16_t seq)
void	SetCsrcs (const std::vector< uint32_t > &csrcs)
void	SetMaxRtpPacketSize (size_t max_packet_size)
bool	SendOutgoingData (FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_header, uint32_t *transport_frame_id_out)
int32_t	RegisterRtpHeaderExtension (RTPExtensionType type, uint8_t id)
bool	IsRtpHeaderExtensionRegistered (RTPExtensionType type) const
int32_t	DeregisterRtpHeaderExtension (RTPExtensionType type)
bool	TimeToSendPacket (uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)
size_t	TimeToSendPadding (size_t bytes, const PacedPacketInfo &pacing_info)
int	SelectiveRetransmissions () const
int	SetSelectiveRetransmissions (uint8_t settings)
void	OnReceivedNack (const std::vector< uint16_t > &nack_sequence_numbers, int64_t avg_rtt)
void	SetStorePacketsStatus (bool enable, uint16_t number_to_store)
bool	StorePackets () const
int32_t	ReSendPacket (uint16_t packet_id, int64_t min_resend_time=0)
void	OnReceivedRtcpReportBlocks (const ReportBlockList &report_blocks)
void	SetRtxStatus (int mode)
int	RtxStatus () const
uint32_t	RtxSsrc () const
void	SetRtxSsrc (uint32_t ssrc)
void	SetRtxPayloadType (int payload_type, int associated_payload_type)
std::unique_ptr< RtpPacketToSend >	AllocatePacket () const
bool	AssignSequenceNumber (RtpPacketToSend *packet)
size_t	RtpHeaderLength () const
uint16_t	AllocateSequenceNumber (uint16_t packets_to_send)
size_t	MaxRtpPacketSize () const
uint32_t	SSRC () const
rtc::Optional< uint32_t >	FlexfecSsrc () const
bool	SendToNetwork (std::unique_ptr< RtpPacketToSend > packet, StorageType storage, RtpPacketSender::Priority priority)
int32_t	SendTelephoneEvent (uint8_t key, uint16_t time_ms, uint8_t level)
RTC_DEPRECATED int32_t	SetAudioPacketSize (uint16_t packet_size_samples)
int32_t	SetAudioLevel (uint8_t level_d_bov)
RtpVideoCodecTypes	VideoCodecType () const
uint32_t	MaxConfiguredBitrateVideo () const
void	SetUlpfecConfig (int red_payload_type, int ulpfec_payload_type)
bool	SetFecParameters (const FecProtectionParams &delta_params, const FecProtectionParams &key_params)
void	RegisterRtpStatisticsCallback (StreamDataCountersCallback *callback)
StreamDataCountersCallback *	GetRtpStatisticsCallback () const
uint32_t	BitrateSent () const
void	SetRtpState (const RtpState &rtp_state)
RtpState	GetRtpState () const
void	SetRtxRtpState (const RtpState &rtp_state)
RtpState	GetRtxRtpState () const

Protected Member Functions
int32_t	CheckPayloadType (int8_t payload_type, RtpVideoCodecTypes *video_type)
int32_t	CheckPayloadType (int8_t payload_type, RtpVideoCodecTypes *video_type)

Constructor & Destructor Documentation

RTPSender() [1/2]

webrtc::RTPSender::RTPSender	(	bool	audio,
		Clock *	clock,
		Transport *	transport,
		RtpPacketSender *	paced_sender,
		FlexfecSender *	flexfec_sender,
		TransportSequenceNumberAllocator *	sequence_number_allocator,
		TransportFeedbackObserver *	transport_feedback_callback,
		BitrateStatisticsObserver *	bitrate_callback,
		FrameCountObserver *	frame_count_observer,
		SendSideDelayObserver *	send_side_delay_observer,
		RtcEventLog *	event_log,
		SendPacketObserver *	send_packet_observer,
		RateLimiter *	nack_rate_limiter,
		OverheadObserver *	overhead_observer
	)

~RTPSender() [1/2]

webrtc::RTPSender::~RTPSender

(

)

RTPSender() [2/2]

webrtc::RTPSender::RTPSender	(	bool	audio,
		Clock *	clock,
		Transport *	transport,
		RtpPacketSender *	paced_sender,
		FlexfecSender *	flexfec_sender,
		TransportSequenceNumberAllocator *	sequence_number_allocator,
		TransportFeedbackObserver *	transport_feedback_callback,
		BitrateStatisticsObserver *	bitrate_callback,
		FrameCountObserver *	frame_count_observer,
		SendSideDelayObserver *	send_side_delay_observer,
		RtcEventLog *	event_log,
		SendPacketObserver *	send_packet_observer,
		RateLimiter *	nack_rate_limiter,
		OverheadObserver *	overhead_observer
	)

~RTPSender() [2/2]

webrtc::RTPSender::~RTPSender

(

)

Member Function Documentation

ActualSendBitrateKbit() [1/2]

uint16_t webrtc::RTPSender::ActualSendBitrateKbit

(

)

const

ActualSendBitrateKbit() [2/2]

uint16_t webrtc::RTPSender::ActualSendBitrateKbit

(

)

const

AllocatePacket() [1/2]

std::unique_ptr< RtpPacketToSend > webrtc::RTPSender::AllocatePacket

(

)

const

AllocatePacket() [2/2]

std::unique_ptr<RtpPacketToSend> webrtc::RTPSender::AllocatePacket

(

)

const

AllocateSequenceNumber() [1/2]

uint16_t webrtc::RTPSender::AllocateSequenceNumber

(

uint16_t

packets_to_send

)

AllocateSequenceNumber() [2/2]

uint16_t webrtc::RTPSender::AllocateSequenceNumber

(

uint16_t

packets_to_send

)

AssignSequenceNumber() [1/2]

bool webrtc::RTPSender::AssignSequenceNumber

(

RtpPacketToSend *

packet

)

AssignSequenceNumber() [2/2]

bool webrtc::RTPSender::AssignSequenceNumber

(

RtpPacketToSend *

packet

)

BitrateSent() [1/2]

uint32_t webrtc::RTPSender::BitrateSent

(

)

const

BitrateSent() [2/2]

uint32_t webrtc::RTPSender::BitrateSent

(

)

const

CheckPayloadType() [1/2]

int32_t webrtc::RTPSender::CheckPayloadType ( int8_t  payload_type, RtpVideoCodecTypes *  video_type  )

protected

CheckPayloadType() [2/2]

int32_t webrtc::RTPSender::CheckPayloadType ( int8_t  payload_type, RtpVideoCodecTypes *  video_type  )

protected

DeregisterRtpHeaderExtension() [1/2]

int32_t webrtc::RTPSender::DeregisterRtpHeaderExtension

(

RTPExtensionType

type

)

DeregisterRtpHeaderExtension() [2/2]

int32_t webrtc::RTPSender::DeregisterRtpHeaderExtension

(

RTPExtensionType

type

)

DeRegisterSendPayload() [1/2]

int32_t webrtc::RTPSender::DeRegisterSendPayload

(

const int8_t

payload_type

)

DeRegisterSendPayload() [2/2]

int32_t webrtc::RTPSender::DeRegisterSendPayload

(

const int8_t

payload_type

)

FecOverheadRate() [1/2]

uint32_t webrtc::RTPSender::FecOverheadRate

(

)

const

FecOverheadRate() [2/2]

uint32_t webrtc::RTPSender::FecOverheadRate

(

)

const

FlexfecSsrc() [1/2]

rtc::Optional< uint32_t > webrtc::RTPSender::FlexfecSsrc

(

)

const

FlexfecSsrc() [2/2]

rtc::Optional<uint32_t> webrtc::RTPSender::FlexfecSsrc

(

)

const

GetDataCounters() [1/2]

void webrtc::RTPSender::GetDataCounters	(	StreamDataCounters *	rtp_stats,
		StreamDataCounters *	rtx_stats
	)		const

GetDataCounters() [2/2]

void webrtc::RTPSender::GetDataCounters	(	StreamDataCounters *	rtp_stats,
		StreamDataCounters *	rtx_stats
	)		const

GetRtpState() [1/2]

RtpState webrtc::RTPSender::GetRtpState

(

)

const

GetRtpState() [2/2]

RtpState webrtc::RTPSender::GetRtpState

(

)

const

GetRtpStatisticsCallback() [1/2]

StreamDataCountersCallback * webrtc::RTPSender::GetRtpStatisticsCallback

(

)

const

GetRtpStatisticsCallback() [2/2]

StreamDataCountersCallback* webrtc::RTPSender::GetRtpStatisticsCallback

(

)

const

GetRtxRtpState() [1/2]

RtpState webrtc::RTPSender::GetRtxRtpState

(

)

const

GetRtxRtpState() [2/2]

RtpState webrtc::RTPSender::GetRtxRtpState

(

)

const

IsRtpHeaderExtensionRegistered() [1/2]

bool webrtc::RTPSender::IsRtpHeaderExtensionRegistered

(

RTPExtensionType

type

)

const

IsRtpHeaderExtensionRegistered() [2/2]

bool webrtc::RTPSender::IsRtpHeaderExtensionRegistered

(

RTPExtensionType

type

)

const

MaxConfiguredBitrateVideo() [1/2]

uint32_t webrtc::RTPSender::MaxConfiguredBitrateVideo

(

)

const

MaxConfiguredBitrateVideo() [2/2]

uint32_t webrtc::RTPSender::MaxConfiguredBitrateVideo

(

)

const

MaxPayloadSize() [1/2]

size_t webrtc::RTPSender::MaxPayloadSize

(

)

const

MaxPayloadSize() [2/2]

size_t webrtc::RTPSender::MaxPayloadSize

(

)

const

MaxRtpPacketSize() [1/2]

size_t webrtc::RTPSender::MaxRtpPacketSize

(

)

const

MaxRtpPacketSize() [2/2]

size_t webrtc::RTPSender::MaxRtpPacketSize

(

)

const

NackOverheadRate() [1/2]

uint32_t webrtc::RTPSender::NackOverheadRate

(

)

const

NackOverheadRate() [2/2]

uint32_t webrtc::RTPSender::NackOverheadRate

(

)

const

OnReceivedNack() [1/2]

void webrtc::RTPSender::OnReceivedNack	(	const std::vector< uint16_t > &	nack_sequence_numbers,
		int64_t	avg_rtt
	)

OnReceivedNack() [2/2]

void webrtc::RTPSender::OnReceivedNack	(	const std::vector< uint16_t > &	nack_sequence_numbers,
		int64_t	avg_rtt
	)

OnReceivedRtcpReportBlocks() [1/2]

void webrtc::RTPSender::OnReceivedRtcpReportBlocks

(

const ReportBlockList &

report_blocks

)

OnReceivedRtcpReportBlocks() [2/2]

void webrtc::RTPSender::OnReceivedRtcpReportBlocks

(

const ReportBlockList &

report_blocks

)

ProcessBitrate() [1/2]

void webrtc::RTPSender::ProcessBitrate

(

)

ProcessBitrate() [2/2]

void webrtc::RTPSender::ProcessBitrate

(

)

RegisterPayload() [1/2]

int32_t webrtc::RTPSender::RegisterPayload	(	const char *	payload_name,
		const int8_t	payload_type,
		const uint32_t	frequency,
		const size_t	channels,
		const uint32_t	rate
	)

RegisterPayload() [2/2]

int32_t webrtc::RTPSender::RegisterPayload	(	const char *	payload_name,
		const int8_t	payload_type,
		const uint32_t	frequency,
		const size_t	channels,
		const uint32_t	rate
	)

RegisterRtpHeaderExtension() [1/2]

int32_t webrtc::RTPSender::RegisterRtpHeaderExtension	(	RTPExtensionType	type,
		uint8_t	id
	)

RegisterRtpHeaderExtension() [2/2]

int32_t webrtc::RTPSender::RegisterRtpHeaderExtension	(	RTPExtensionType	type,
		uint8_t	id
	)

RegisterRtpStatisticsCallback() [1/2]

void webrtc::RTPSender::RegisterRtpStatisticsCallback

(

StreamDataCountersCallback *

callback

)

RegisterRtpStatisticsCallback() [2/2]

void webrtc::RTPSender::RegisterRtpStatisticsCallback

(

StreamDataCountersCallback *

callback

)

ReSendPacket() [1/2]

int32_t webrtc::RTPSender::ReSendPacket	(	uint16_t	packet_id,
		int64_t	min_resend_time = 0
	)

ReSendPacket() [2/2]

int32_t webrtc::RTPSender::ReSendPacket	(	uint16_t	packet_id,
		int64_t	min_resend_time = 0
	)

RtpHeaderLength() [1/2]

size_t webrtc::RTPSender::RtpHeaderLength

(

)

const

RtpHeaderLength() [2/2]

size_t webrtc::RTPSender::RtpHeaderLength

(

)

const

RtxSsrc() [1/2]

uint32_t webrtc::RTPSender::RtxSsrc

(

)

const

RtxSsrc() [2/2]

uint32_t webrtc::RTPSender::RtxSsrc

(

)

const

RtxStatus() [1/2]

int webrtc::RTPSender::RtxStatus

(

)

const

RtxStatus() [2/2]

int webrtc::RTPSender::RtxStatus

(

)

const

SelectiveRetransmissions() [1/2]

int webrtc::RTPSender::SelectiveRetransmissions

(

)

const

SelectiveRetransmissions() [2/2]

int webrtc::RTPSender::SelectiveRetransmissions

(

)

const

SendingMedia() [1/2]

bool webrtc::RTPSender::SendingMedia

(

)

const

SendingMedia() [2/2]

bool webrtc::RTPSender::SendingMedia

(

)

const

SendOutgoingData() [1/2]

bool webrtc::RTPSender::SendOutgoingData	(	FrameType	frame_type,
		int8_t	payload_type,
		uint32_t	timestamp,
		int64_t	capture_time_ms,
		const uint8_t *	payload_data,
		size_t	payload_size,
		const RTPFragmentationHeader *	fragmentation,
		const RTPVideoHeader *	rtp_header,
		uint32_t *	transport_frame_id_out
	)

SendOutgoingData() [2/2]

bool webrtc::RTPSender::SendOutgoingData	(	FrameType	frame_type,
		int8_t	payload_type,
		uint32_t	timestamp,
		int64_t	capture_time_ms,
		const uint8_t *	payload_data,
		size_t	payload_size,
		const RTPFragmentationHeader *	fragmentation,
		const RTPVideoHeader *	rtp_header,
		uint32_t *	transport_frame_id_out
	)

SendPayloadType() [1/2]

int8_t webrtc::RTPSender::SendPayloadType

(

)

const

SendPayloadType() [2/2]

int8_t webrtc::RTPSender::SendPayloadType

(

)

const

SendTelephoneEvent() [1/2]

int32_t webrtc::RTPSender::SendTelephoneEvent	(	uint8_t	key,
		uint16_t	time_ms,
		uint8_t	level
	)

SendTelephoneEvent() [2/2]

int32_t webrtc::RTPSender::SendTelephoneEvent	(	uint8_t	key,
		uint16_t	time_ms,
		uint8_t	level
	)

SendToNetwork() [1/2]

bool webrtc::RTPSender::SendToNetwork	(	std::unique_ptr< RtpPacketToSend >	packet,
		StorageType	storage,
		RtpPacketSender::Priority	priority
	)

SendToNetwork() [2/2]

bool webrtc::RTPSender::SendToNetwork	(	std::unique_ptr< RtpPacketToSend >	packet,
		StorageType	storage,
		RtpPacketSender::Priority	priority
	)

SequenceNumber() [1/2]

uint16_t webrtc::RTPSender::SequenceNumber

(

)

const

SequenceNumber() [2/2]

uint16_t webrtc::RTPSender::SequenceNumber

(

)

const

SetAudioLevel() [1/2]

int32_t webrtc::RTPSender::SetAudioLevel

(

uint8_t

level_d_bov

)

SetAudioLevel() [2/2]

int32_t webrtc::RTPSender::SetAudioLevel

(

uint8_t

level_d_bov

)

SetAudioPacketSize() [1/2]

RTC_DEPRECATED int32_t webrtc::RTPSender::SetAudioPacketSize

(

uint16_t

packet_size_samples

)

SetAudioPacketSize() [2/2]

int32_t webrtc::RTPSender::SetAudioPacketSize

(

uint16_t

packet_size_samples

)

SetCsrcs() [1/2]

void webrtc::RTPSender::SetCsrcs

(

const std::vector< uint32_t > &

csrcs

)

SetCsrcs() [2/2]

void webrtc::RTPSender::SetCsrcs

(

const std::vector< uint32_t > &

csrcs

)

SetFecParameters() [1/2]

bool webrtc::RTPSender::SetFecParameters	(	const FecProtectionParams &	delta_params,
		const FecProtectionParams &	key_params
	)

SetFecParameters() [2/2]

bool webrtc::RTPSender::SetFecParameters	(	const FecProtectionParams &	delta_params,
		const FecProtectionParams &	key_params
	)

SetMaxRtpPacketSize() [1/2]

void webrtc::RTPSender::SetMaxRtpPacketSize

(

size_t

max_packet_size

)

SetMaxRtpPacketSize() [2/2]

void webrtc::RTPSender::SetMaxRtpPacketSize

(

size_t

max_packet_size

)

SetRtpState() [1/2]

void webrtc::RTPSender::SetRtpState

(

const RtpState &

rtp_state

)

SetRtpState() [2/2]

void webrtc::RTPSender::SetRtpState

(

const RtpState &

rtp_state

)

SetRtxPayloadType() [1/2]

void webrtc::RTPSender::SetRtxPayloadType	(	int	payload_type,
		int	associated_payload_type
	)

SetRtxPayloadType() [2/2]

void webrtc::RTPSender::SetRtxPayloadType	(	int	payload_type,
		int	associated_payload_type
	)

SetRtxRtpState() [1/2]

void webrtc::RTPSender::SetRtxRtpState

(

const RtpState &

rtp_state

)

SetRtxRtpState() [2/2]

void webrtc::RTPSender::SetRtxRtpState

(

const RtpState &

rtp_state

)

SetRtxSsrc() [1/2]

void webrtc::RTPSender::SetRtxSsrc

(

uint32_t

ssrc

)

SetRtxSsrc() [2/2]

void webrtc::RTPSender::SetRtxSsrc

(

uint32_t

ssrc

)

SetRtxStatus() [1/2]

void webrtc::RTPSender::SetRtxStatus

(

int

mode

)

SetRtxStatus() [2/2]

void webrtc::RTPSender::SetRtxStatus

(

int

mode

)

SetSelectiveRetransmissions() [1/2]

int webrtc::RTPSender::SetSelectiveRetransmissions

(

uint8_t

settings

)

SetSelectiveRetransmissions() [2/2]

int webrtc::RTPSender::SetSelectiveRetransmissions

(

uint8_t

settings

)

SetSendingMediaStatus() [1/2]

void webrtc::RTPSender::SetSendingMediaStatus

(

bool

enabled

)

SetSendingMediaStatus() [2/2]

void webrtc::RTPSender::SetSendingMediaStatus

(

bool

enabled

)

SetSendPayloadType() [1/2]

void webrtc::RTPSender::SetSendPayloadType

(

int8_t

payload_type

)

SetSendPayloadType() [2/2]

void webrtc::RTPSender::SetSendPayloadType

(

int8_t

payload_type

)

SetSequenceNumber() [1/2]

void webrtc::RTPSender::SetSequenceNumber

(

uint16_t

seq

)

SetSequenceNumber() [2/2]

void webrtc::RTPSender::SetSequenceNumber

(

uint16_t

seq

)

SetSSRC() [1/2]

void webrtc::RTPSender::SetSSRC

(

uint32_t

ssrc

)

SetSSRC() [2/2]

void webrtc::RTPSender::SetSSRC

(

uint32_t

ssrc

)

SetStorePacketsStatus() [1/2]

void webrtc::RTPSender::SetStorePacketsStatus	(	bool	enable,
		uint16_t	number_to_store
	)

SetStorePacketsStatus() [2/2]

void webrtc::RTPSender::SetStorePacketsStatus	(	bool	enable,
		uint16_t	number_to_store
	)

SetTimestampOffset() [1/2]

void webrtc::RTPSender::SetTimestampOffset

(

uint32_t

timestamp

)

SetTimestampOffset() [2/2]

void webrtc::RTPSender::SetTimestampOffset

(

uint32_t

timestamp

)

SetUlpfecConfig() [1/2]

void webrtc::RTPSender::SetUlpfecConfig	(	int	red_payload_type,
		int	ulpfec_payload_type
	)

SetUlpfecConfig() [2/2]

void webrtc::RTPSender::SetUlpfecConfig	(	int	red_payload_type,
		int	ulpfec_payload_type
	)

SSRC() [1/2]

uint32_t webrtc::RTPSender::SSRC

(

)

const

SSRC() [2/2]

uint32_t webrtc::RTPSender::SSRC

(

)

const

StorePackets() [1/2]

bool webrtc::RTPSender::StorePackets

(

)

const

StorePackets() [2/2]

bool webrtc::RTPSender::StorePackets

(

)

const

TimestampOffset() [1/2]

uint32_t webrtc::RTPSender::TimestampOffset

(

)

const

TimestampOffset() [2/2]

uint32_t webrtc::RTPSender::TimestampOffset

(

)

const

TimeToSendPacket() [1/2]

bool webrtc::RTPSender::TimeToSendPacket	(	uint32_t	ssrc,
		uint16_t	sequence_number,
		int64_t	capture_time_ms,
		bool	retransmission,
		const PacedPacketInfo &	pacing_info
	)

TimeToSendPacket() [2/2]

bool webrtc::RTPSender::TimeToSendPacket	(	uint32_t	ssrc,
		uint16_t	sequence_number,
		int64_t	capture_time_ms,
		bool	retransmission,
		const PacedPacketInfo &	pacing_info
	)

TimeToSendPadding() [1/2]

size_t webrtc::RTPSender::TimeToSendPadding	(	size_t	bytes,
		const PacedPacketInfo &	pacing_info
	)

TimeToSendPadding() [2/2]

size_t webrtc::RTPSender::TimeToSendPadding	(	size_t	bytes,
		const PacedPacketInfo &	pacing_info
	)

VideoBitrateSent() [1/2]

uint32_t webrtc::RTPSender::VideoBitrateSent

(

)

const

VideoBitrateSent() [2/2]

uint32_t webrtc::RTPSender::VideoBitrateSent

(

)

const

VideoCodecType() [1/2]

RtpVideoCodecTypes webrtc::RTPSender::VideoCodecType

(

)

const

VideoCodecType() [2/2]

RtpVideoCodecTypes webrtc::RTPSender::VideoCodecType

(

)

const

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/rtp_rtcp/source/rtp_sender.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc