webrtc::RTPSenderAudio Class Reference

#include <rtp_sender_audio.h>

Public Member Functions
	RTPSenderAudio (Clock *clock, RTPSender *rtp_sender)
	~RTPSenderAudio ()
int32_t	RegisterAudioPayload (const char payloadName[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency, size_t channels, uint32_t rate, RtpUtility::Payload **payload)
bool	SendAudio (FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)
int32_t	SetAudioLevel (uint8_t level_dbov)
int32_t	SendTelephoneEvent (uint8_t key, uint16_t time_ms, uint8_t level)
	RTPSenderAudio (Clock *clock, RTPSender *rtp_sender)
	~RTPSenderAudio ()
int32_t	RegisterAudioPayload (const char payloadName[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency, size_t channels, uint32_t rate, RtpUtility::Payload **payload)
bool	SendAudio (FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)
int32_t	SetAudioLevel (uint8_t level_dbov)
int32_t	SendTelephoneEvent (uint8_t key, uint16_t time_ms, uint8_t level)

Protected Member Functions
bool	SendTelephoneEventPacket (bool ended, uint32_t dtmf_timestamp, uint16_t duration, bool marker_bit)
bool	MarkerBit (FrameType frame_type, int8_t payload_type)
bool	SendTelephoneEventPacket (bool ended, uint32_t dtmf_timestamp, uint16_t duration, bool marker_bit)
bool	MarkerBit (FrameType frame_type, int8_t payload_type)

Constructor & Destructor Documentation

RTPSenderAudio() [1/2]

webrtc::RTPSenderAudio::RTPSenderAudio	(	Clock *	clock,
		RTPSender *	rtp_sender
	)

~RTPSenderAudio() [1/2]

webrtc::RTPSenderAudio::~RTPSenderAudio

(

)

RTPSenderAudio() [2/2]

webrtc::RTPSenderAudio::RTPSenderAudio	(	Clock *	clock,
		RTPSender *	rtp_sender
	)

~RTPSenderAudio() [2/2]

webrtc::RTPSenderAudio::~RTPSenderAudio

(

)

Member Function Documentation

MarkerBit() [1/2]

bool webrtc::RTPSenderAudio::MarkerBit ( FrameType  frame_type, int8_t  payload_type  )

protected

MarkerBit() [2/2]

bool webrtc::RTPSenderAudio::MarkerBit ( FrameType  frame_type, int8_t  payload_type  )

protected

RegisterAudioPayload() [1/2]

int32_t webrtc::RTPSenderAudio::RegisterAudioPayload	(	const char	payloadName[RTP_PAYLOAD_NAME_SIZE],
		int8_t	payload_type,
		uint32_t	frequency,
		size_t	channels,
		uint32_t	rate,
		RtpUtility::Payload **	payload
	)

RegisterAudioPayload() [2/2]

int32_t webrtc::RTPSenderAudio::RegisterAudioPayload	(	const char	payloadName[RTP_PAYLOAD_NAME_SIZE],
		int8_t	payload_type,
		uint32_t	frequency,
		size_t	channels,
		uint32_t	rate,
		RtpUtility::Payload **	payload
	)

SendAudio() [1/2]

bool webrtc::RTPSenderAudio::SendAudio	(	FrameType	frame_type,
		int8_t	payload_type,
		uint32_t	capture_timestamp,
		const uint8_t *	payload_data,
		size_t	payload_size,
		const RTPFragmentationHeader *	fragmentation
	)

SendAudio() [2/2]

bool webrtc::RTPSenderAudio::SendAudio	(	FrameType	frame_type,
		int8_t	payload_type,
		uint32_t	capture_timestamp,
		const uint8_t *	payload_data,
		size_t	payload_size,
		const RTPFragmentationHeader *	fragmentation
	)

SendTelephoneEvent() [1/2]

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

SendTelephoneEvent() [2/2]

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

SendTelephoneEventPacket() [1/2]

bool webrtc::RTPSenderAudio::SendTelephoneEventPacket ( bool  ended, uint32_t  dtmf_timestamp, uint16_t  duration, bool  marker_bit  )

protected

SendTelephoneEventPacket() [2/2]

bool webrtc::RTPSenderAudio::SendTelephoneEventPacket ( bool  ended, uint32_t  dtmf_timestamp, uint16_t  duration, bool  marker_bit  )

protected

SetAudioLevel() [1/2]

int32_t webrtc::RTPSenderAudio::SetAudioLevel

(

uint8_t

level_dbov

)

SetAudioLevel() [2/2]

int32_t webrtc::RTPSenderAudio::SetAudioLevel

(

uint8_t

level_dbov

)

---

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_audio.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc