RTPStreamer object. More...

Collaboration diagram for RTP Streamer:

Data Structures
struct	GF_RTPStreamerConfig

Typedefs
typedef void(*	gf_rtcp_rr_callback) (void *cbk, u32 ssrc, u32 rtt_ms, u64 jitter_rtp_ts, u32 loss_rate)

Functions
GF_RTPStreamer *	gf_rtp_streamer_new (u32 streamType, u32 codecid, u32 timeScale, const char ip_dest, u16 port, u32 MTU, u8 TTL, const char ifce_addr, u32 flags, const u8 *dsi, u32 dsi_len, u32 PayloadType, u32 sample_rate, u32 nb_ch, Bool is_crypted, u32 IV_length, u32 KI_length, u32 MinSize, u32 MaxSize, u32 avgTS, u32 maxDTSDelta, u32 const_dur, u32 bandwidth, u32 max_ptime, u32 au_sn_len, Bool for_rtsp)
	RTP Streamer constructor with extended parameters.

GF_RTPStreamer *	gf_rtp_streamer_new_ex (const GF_RTPStreamerConfig *cfg, Bool for_rtsp)
	RTP Streamer constructor with extended parameters.

void	gf_rtp_streamer_del (GF_RTPStreamer *streamer)
	RTP file streamer destructor.

GF_Err	gf_rtp_streamer_append_sdp (GF_RTPStreamer rtp, u16 ESID, const u8 dsi, u32 dsi_len, char KMS_URI, char *out_sdp_buffer)
	gets the SDP file

GF_Err	gf_rtp_streamer_append_sdp_extended (GF_RTPStreamer rtp, u16 ESID, const u8 dsi, u32 dsi_len, const u8 dsi_enh, u32 dsi_enh_len, char KMS_URI, u32 width, u32 height, u32 tw, u32 th, s32 tx, s32 ty, s16 tl, u32 nb_channels, Bool for_rtsp, char **out_sdp_buffer)
	gets the SDP file

GF_Err	gf_rtp_streamer_send_au (GF_RTPStreamer rtp, u8 data, u32 size, u64 cts, u64 dts, Bool is_rap)

GF_Err	gf_rtp_streamer_send_au_with_sn (GF_RTPStreamer rtp, u8 data, u32 size, u64 cts, u64 dts, Bool is_rap, u32 inc_au_sn)

GF_Err	gf_rtp_streamer_send_data (GF_RTPStreamer streamer, u8 data, u32 size, u32 fullsize, u64 cts, u64 dts, Bool is_rap, Bool au_start, Bool au_end, u32 au_sn, u32 duration, u32 sampleDescriptionIndex)

char *	gf_rtp_streamer_format_sdp_header (char app_name, char ip_dest, char session_name, char iod64)

void	gf_rtp_streamer_disable_auto_rtcp (GF_RTPStreamer *streamer)

GF_Err	gf_rtp_streamer_send_bye (GF_RTPStreamer *streamer)

GF_Err	gf_rtp_streamer_send_rtcp (GF_RTPStreamer *streamer, Bool force_ts, u32 rtp_ts, u32 force_ntp_type, u32 ntp_sec, u32 ntp_frac)

u8	gf_rtp_streamer_get_payload_type (GF_RTPStreamer *streamer)

GF_Err	gf_rtp_streamer_init_rtsp (GF_RTPStreamer streamer, u32 path_mtu, GF_RTSPTransport tr, const char *ifce_addr)

u16	gf_rtp_streamer_get_next_rtp_sn (GF_RTPStreamer *streamer)

GF_Err	gf_rtp_streamer_set_interleave_callbacks (GF_RTPStreamer streamer, GF_Err(RTP_TCPCallback)(void cbk1, void cbk2, Bool is_rtcp, u8 pck, u32 pck_size), void cbk1, void *cbk2)

GF_Err	gf_rtp_streamer_read_rtcp (GF_RTPStreamer streamer, gf_rtcp_rr_callback rtcp_cbk, void udta)

u32	gf_rtp_streamer_get_ssrc (GF_RTPStreamer *streamer)

u32	gf_rtp_streamer_get_timescale (GF_RTPStreamer *streamer)

u32	gf_rtp_streamer_get_codecid (GF_RTPStreamer *streamer)

Detailed Description

RTPStreamer object.

This section documents the RTP streamer object of the GPAC framework.

Data Structure Documentation

◆ GF_RTPStreamerConfig

struct GF_RTPStreamerConfig

RTP streamer configuration

Data Fields
u32	streamType	type of the stream (GF_STREAM_* as defined in <gpac/constants.h>
u32	codecid	codec ID for the stream (GF_CODEC_* as defined in <gpac/constants.h>)
u32	timeScale	unit to express timestamps of access units
const char *	ip_dest	IP address of the destination.
u16	port	port number of the destination
u32	MTU	Maximum Transmission Unit size to use.
u8	TTL	Time To Leave.
const char *	ifce_addr	IP of the local interface to use (may be NULL)
u32	flags	set of RTP flags passed to the streamer
const u8 *	dsi	MPEG-4 Decoder Specific Info for the stream.
u32	dsi_len	length of the dsi parameter
u32	PayloadType	RTP payload type.
u32	sample_rate	audio sample rate
u32	nb_ch	number of channels in audio streams
Bool	is_crypted	indicating if the stream is crypted
u32	IV_length	length of the Initialisation Vector used for ISMA encryption
u32	KI_length	length of the key index
u32	MinSize	minimum AU size, 0 if unknown
u32	MaxSize
u32	avgTS	average TS delta in timeScale, 0 if unknown
u32	maxDTSDelta	maximum DTS delta in timeScale, 0 if unknown
u32	const_dur	constant duration in timeScale, 0 if unknown
u32	bandwidth	bandwidth, 0 if unknown
u32	max_ptime	maximum packet duration in timeScale, 0 if unknown
u32	au_sn_len	length of the MPEG-4 SL descriptor AU sequence number field, 0 if unknown
const char *	netcap_id	ID of netcap configuration to use, may be null (see gpac -h netcap)

Typedef Documentation

◆ gf_rtcp_rr_callback

typedef void(* gf_rtcp_rr_callback) (void *cbk, u32 ssrc, u32 rtt_ms, u64 jitter_rtp_ts, u32 loss_rate)

callback function for procesing RTCP receiver reports

Parameters

cbk	user data passed to gf_rtp_streamer_read_rtcp
ssrc	ssrc for this report, 0 if same as ssrc of channel
rtt_ms	round-trip time estimate in ms
jitter_rtp_ts	inter-arrival jitter in RTP channel timescale
loss_rate	loss rate in per-thousands

Function Documentation

◆ gf_rtp_streamer_new()

GF_RTPStreamer * gf_rtp_streamer_new	(	u32	streamType,
		u32	codecid,
		u32	timeScale,
		const char *	ip_dest,
		u16	port,
		u32	MTU,
		u8	TTL,
		const char *	ifce_addr,
		u32	flags,
		const u8 *	dsi,
		u32	dsi_len,
		u32	PayloadType,
		u32	sample_rate,
		u32	nb_ch,
		Bool	is_crypted,
		u32	IV_length,
		u32	KI_length,
		u32	MinSize,
		u32	MaxSize,
		u32	avgTS,
		u32	maxDTSDelta,
		u32	const_dur,
		u32	bandwidth,
		u32	max_ptime,
		u32	au_sn_len,
		Bool	for_rtsp
	)

RTP Streamer constructor with extended parameters.

Constructs a new RTP file streamer

Parameters

streamType	type of the stream (GF_STREAM_* as defined in <gpac/constants.h>)
codecid	codec ID for the stream (GF_CODEC_* as defined in <gpac/constants.h>)
timeScale	unit to express timestamps of access units
ip_dest	IP address of the destination
port	port number of the destination
MTU	Maximum Transmission Unit size to use
TTL	Time To Leave
ifce_addr	IP of the local interface to use (may be NULL)
flags	set of RTP flags passed to the streamer
dsi	MPEG-4 Decoder Specific Info for the stream
dsi_len	length of the dsi parameter
PayloadType	RTP payload type
sample_rate	audio sample rate
nb_ch	number of channels in audio streams
is_crypted	Boolean indicating if the stream is crypted
IV_length	length of the Initialisation Vector used for encryption
KI_length	length of the key index
MinSize	minimum AU size, 0 if unknown
MaxSize	maximum AU size, 0 if unknown
avgTS	average TS delta in timeScale, 0 if unknown
maxDTSDelta	maximum DTS delta in timeScale, 0 if unknown
const_dur	constant duration in timeScale, 0 if unknown
bandwidth	bandwidth, 0 if unknown
max_ptime	maximum packet duration in timeScale, 0 if unknown
au_sn_len	length of the MPEG-4 SL descriptor AU sequence number field, 0 if unknown
for_rtsp	indicates this is an RTP channel in an RTSP session, RTP channel will not be created, use gf_rtp_streamer_init_rtsp

Returns: a new RTP streamer, or NULL of error or not supported

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◆ gf_rtp_streamer_new_ex()

GF_RTPStreamer * gf_rtp_streamer_new_ex	(	const GF_RTPStreamerConfig *	cfg,
		Bool	for_rtsp
	)

RTP Streamer constructor with extended parameters.

Constructs a new RTP file streamer

Parameters

cfg	configuration of the streamer
for_rtsp	indicates this is an RTP channel in an RTSP session, RTP channel will not be created, use gf_rtp_streamer_init_rtsp

Returns: a new RTP streamer, or NULL of error or not supported

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◆ gf_rtp_streamer_del()

void gf_rtp_streamer_del ( GF_RTPStreamer * streamer )

RTP file streamer destructor.

Destructs an RTP file streamer

Parameters

streamer the target RTP streamer

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◆ gf_rtp_streamer_append_sdp()

GF_Err gf_rtp_streamer_append_sdp	(	GF_RTPStreamer *	rtp,
		u16	ESID,
		const u8 *	dsi,
		u32	dsi_len,
		char *	KMS_URI,
		char **	out_sdp_buffer
	)

gets the SDP file

Gets the SDP asscoiated with all media in the streaming session (only media parts are returned)

Parameters

rtp	the target RTP streamer
ESID	The MPEG-4 elementary stream id of the stream to process
dsi	The decoder specific info data
dsi_len	length of the decoder specific info data
KMS_URI	URI of the Key Management System
out_sdp_buffer	location to the SDP buffer to allocate and fill

Returns: error if any

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◆ gf_rtp_streamer_append_sdp_extended()

GF_Err gf_rtp_streamer_append_sdp_extended	(	GF_RTPStreamer *	rtp,
		u16	ESID,
		const u8 *	dsi,
		u32	dsi_len,
		const u8 *	dsi_enh,
		u32	dsi_enh_len,
		char *	KMS_URI,
		u32	width,
		u32	height,
		u32	tw,
		u32	th,
		s32	tx,
		s32	ty,
		s16	tl,
		u32	nb_channels,
		Bool	for_rtsp,
		char **	out_sdp_buffer
	)

gets the SDP file

Gets the SDP asscoiated with all media in the streaming session (only media parts are returned)

Parameters

rtp	the target RTP streamer
ESID	The MPEG-4 elementary stream id of the stream to process
dsi	decoder specific info data
dsi_len	length of the decoder specific info data
dsi_enh	enhancement layer decoder specific info data
dsi_enh_len	length of enhancement layer decoder specific info data
KMS_URI	URI of the Key Management System
width	video width
height	video height
tw	text window width
th	text window height
tx	text window horizontal offset
ty	text window vertical offset
tl	text window z-index
nb_channels	number of audio channels, 0 if unknown
for_rtsp	if GF_TRUE, produces the SDP for an RTSP describe (no port info)
out_sdp_buffer	location to the SDP buffer to allocate and fill

Returns: error if any

defined(GPAC_DISABLE_ISOM) && !defined(GPAC_DISABLE_STREAMING)

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◆ gf_rtp_streamer_send_au()

GF_Err gf_rtp_streamer_send_au	(	GF_RTPStreamer *	rtp,
		u8 *	data,
		u32	size,
		u64	cts,
		u64	dts,
		Bool	is_rap
	)

sends a full Access Unit over RTP

Parameters

rtp	the target RTP streamer
data	AU payload
size	AU payload size
cts	composition timestamp in timeScale unit
dts	decoding timestamp in timeScale unit
is_rap	indicates if the AU is a random access

Returns: error if any

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◆ gf_rtp_streamer_send_au_with_sn()

GF_Err gf_rtp_streamer_send_au_with_sn	(	GF_RTPStreamer *	rtp,
		u8 *	data,
		u32	size,
		u64	cts,
		u64	dts,
		Bool	is_rap,
		u32	inc_au_sn
	)

sends a full Access Unit over RTP

Parameters

rtp	the target RTP streamer
data	AU payload
size	AU payload size
cts	composition timestamp in timeScale unit
dts	decoding timestamp in timeScale unit
is_rap	indicates if the AU is a random access
inc_au_sn	increments the AU sequence number by the given value before packetizing (used for MPEG-4 systems carousel)

Returns: error if any

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◆ gf_rtp_streamer_send_data()

GF_Err gf_rtp_streamer_send_data	(	GF_RTPStreamer *	streamer,
		u8 *	data,
		u32	size,
		u32	fullsize,
		u64	cts,
		u64	dts,
		Bool	is_rap,
		Bool	au_start,
		Bool	au_end,
		u32	au_sn,
		u32	duration,
		u32	sampleDescriptionIndex
	)

sends a full or partial Access Unit over RTP

Parameters

streamer	the target RTP streamer
data	AU payload
size	AU payload size
fullsize	the size of the complete AU
cts	composition timestamp in timeScale unit
dts	decoding timestamp in timeScale unit
is_rap	indicates if the AU is a random access
au_start	indicates if this chunk is the start of the AU
au_end	indicates if this chunk is the end of the AU
au_sn	indicates the AU sequence number
duration	indicates the AU duration
sampleDescriptionIndex	indicates the ISOBMFF sampleDescriptionIndex for this AU (needed for 3GPP timed text)

Returns: error if any

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◆ gf_rtp_streamer_format_sdp_header()

char * gf_rtp_streamer_format_sdp_header	(	char *	app_name,
		char *	ip_dest,
		char *	session_name,
		char *	iod64
	)

formats a generic SDP header

Parameters

app_name	application name, may be NULL
ip_dest	destination IP address, shall NOT be NULL
session_name	session name, may be NULL
iod64	base64 encoded MPEG-4 IOD, may be NULL

Returns: the SDP header - shall be destroyed by caller

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◆ gf_rtp_streamer_disable_auto_rtcp()

void gf_rtp_streamer_disable_auto_rtcp ( GF_RTPStreamer * streamer )

disables RTCP keepalive

Parameters

streamer the target RTP streamer

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◆ gf_rtp_streamer_send_bye()

GF_Err gf_rtp_streamer_send_bye ( GF_RTPStreamer * streamer )

sends RTCP bye packet

Parameters

streamer the target RTP streamer

Returns: error if any

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◆ gf_rtp_streamer_send_rtcp()

GF_Err gf_rtp_streamer_send_rtcp	(	GF_RTPStreamer *	streamer,
		Bool	force_ts,
		u32	rtp_ts,
		u32	force_ntp_type,
		u32	ntp_sec,
		u32	ntp_frac
	)

sends RTCP server report

Parameters

streamer	the target RTP streamer
force_ts	if GF_TRUE, forces using the indicatet ts
rtp_ts	the force RTP timestamp to use
force_ntp_type	if 0, computes NTP while sending. If 1 or 2, uses ntp_sec and ntp_frac for report. If 2, forces sending reports right away
ntp_sec	NTP seconds
ntp_frac	NTP fractional part

Returns: error if any

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◆ gf_rtp_streamer_get_payload_type()

u8 gf_rtp_streamer_get_payload_type ( GF_RTPStreamer * streamer )

gets associated payload type

Parameters

streamer the target RTP streamer

Returns: the payload type

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◆ gf_rtp_streamer_init_rtsp()

GF_Err gf_rtp_streamer_init_rtsp	(	GF_RTPStreamer *	streamer,
		u32	path_mtu,
		GF_RTSPTransport *	tr,
		const char *	ifce_addr
	)

initializes the RTP channel for RTSP setup

Parameters

streamer	the target RTP streamer
path_mtu	MTU path size in bytes
tr	the RTSP transport description
ifce_addr	IP address of network interface to use

Returns: error if any

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◆ gf_rtp_streamer_get_next_rtp_sn()

u16 gf_rtp_streamer_get_next_rtp_sn ( GF_RTPStreamer * streamer )

gets the sequence number of the next RTP packet to be sent

Parameters

streamer the target RTP streamer

Returns: the sequence number

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◆ gf_rtp_streamer_set_interleave_callbacks()

GF_Err gf_rtp_streamer_set_interleave_callbacks	(	GF_RTPStreamer *	streamer,
		GF_Err()(void cbk1, void cbk2, Bool is_rtcp, u8 pck, u32 pck_size)	RTP_TCPCallback,
		void *	cbk1,
		void *	cbk2
	)

sets callback functions for RTP over RTSP sending

Parameters

streamer	the target RTP streamer
RTP_TCPCallback	the callback function
cbk1	first opaque data passed to callback function
cbk2	second opaque data passed to callback function

Returns: error if any

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◆ gf_rtp_streamer_read_rtcp()

GF_Err gf_rtp_streamer_read_rtcp	(	GF_RTPStreamer *	streamer,
		gf_rtcp_rr_callback	rtcp_cbk,
		void *	udta
	)

process rtcp reports if any

Parameters

streamer	the target RTP streamer
rtcp_cbk	callback function for RTCP reports
udta	user data to use for callback function

Returns: error if any, GF_EOS if empty

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◆ gf_rtp_streamer_get_ssrc()

u32 gf_rtp_streamer_get_ssrc ( GF_RTPStreamer * streamer )

gets ssrc of this streamer

Parameters

streamer the target RTP streamer

Returns: ssrc ID

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◆ gf_rtp_streamer_get_timescale()

u32 gf_rtp_streamer_get_timescale ( GF_RTPStreamer * streamer )

gets rtp timescale of this streamer

Parameters

streamer the target RTP streamer

Returns: timescale

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◆ gf_rtp_streamer_get_codecid()

u32 gf_rtp_streamer_get_codecid ( GF_RTPStreamer * streamer )

gets codecid of this streamer

Parameters

streamer the target RTP streamer

Returns: CodecID

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Data Structures

Typedefs

Functions

Detailed Description

Data Structure Documentation

◆ GF_RTPStreamerConfig

Typedef Documentation

◆ gf_rtcp_rr_callback

Function Documentation

◆ gf_rtp_streamer_new()

◆ gf_rtp_streamer_new_ex()

◆ gf_rtp_streamer_del()

◆ gf_rtp_streamer_append_sdp()

◆ gf_rtp_streamer_append_sdp_extended()

◆ gf_rtp_streamer_send_au()

◆ gf_rtp_streamer_send_au_with_sn()

◆ gf_rtp_streamer_send_data()

◆ gf_rtp_streamer_format_sdp_header()

◆ gf_rtp_streamer_disable_auto_rtcp()

◆ gf_rtp_streamer_send_bye()

◆ gf_rtp_streamer_send_rtcp()

◆ gf_rtp_streamer_get_payload_type()

◆ gf_rtp_streamer_init_rtsp()

◆ gf_rtp_streamer_get_next_rtp_sn()

◆ gf_rtp_streamer_set_interleave_callbacks()

◆ gf_rtp_streamer_read_rtcp()

◆ gf_rtp_streamer_get_ssrc()

◆ gf_rtp_streamer_get_timescale()

◆ gf_rtp_streamer_get_codecid()