rtp.c File Reference

Include dependency graph for rtp.c:

Data Structures
struct	sout_stream_sys_t
struct	rtp_sink_t
struct	sout_stream_id_t
Defines
#define	IPPROTO_DCCP   33
#define	IPPROTO_UDPLITE   136
#define	DEST_TEXT   N_("Destination")
#define	DEST_LONGTEXT
#define	SDP_TEXT   N_("SDP")
#define	SDP_LONGTEXT
#define	SAP_TEXT   N_("SAP announcing")
#define	SAP_LONGTEXT   N_("Announce this session with SAP.")
#define	MUX_TEXT   N_("Muxer")
#define	MUX_LONGTEXT
#define	NAME_TEXT   N_("Session name")
#define	NAME_LONGTEXT
#define	DESC_TEXT   N_("Session description")
#define	DESC_LONGTEXT
#define	URL_TEXT   N_("Session URL")
#define	URL_LONGTEXT
#define	EMAIL_TEXT   N_("Session email")
#define	EMAIL_LONGTEXT
#define	PHONE_TEXT   N_("Session phone number")
#define	PHONE_LONGTEXT
#define	PORT_TEXT   N_("Port")
#define	PORT_LONGTEXT
#define	PORT_AUDIO_TEXT   N_("Audio port")
#define	PORT_AUDIO_LONGTEXT
#define	PORT_VIDEO_TEXT   N_("Video port")
#define	PORT_VIDEO_LONGTEXT
#define	TTL_TEXT   N_("Hop limit (TTL)")
#define	TTL_LONGTEXT
#define	RTCP_MUX_TEXT   N_("RTP/RTCP multiplexing")
#define	RTCP_MUX_LONGTEXT
#define	PROTO_TEXT   N_("Transport protocol")
#define	PROTO_LONGTEXT
#define	SRTP_KEY_TEXT   N_("SRTP key (hexadecimal)")
#define	SRTP_KEY_LONGTEXT
#define	SRTP_SALT_TEXT   N_("SRTP salt (hexadecimal)")
#define	SRTP_SALT_LONGTEXT
#define	RFC3016_TEXT   N_("MP4A LATM")
#define	RFC3016_LONGTEXT
#define	SOUT_CFG_PREFIX   "sout-rtp-"
#define	MAX_EMPTY_BLOCKS   200
Typedefs
typedef int(*)	pf_rtp_packetizer_t (sout_stream_id_t *, block_t *)
Functions
static int	Open (vlc_object_t *)
static void	Close (vlc_object_t *)
int	vlc_entry__main (module_t *p_module)
const char *	vlc_entry_license__main (void)
static sout_stream_id_t *	Add (sout_stream_t *p_stream, es_format_t *p_fmt)
	Add an ES as a new RTP stream.
static int	Del (sout_stream_t *, sout_stream_id_t *)
static int	Send (sout_stream_t *, sout_stream_id_t *, block_t *)
static sout_stream_id_t *	MuxAdd (sout_stream_t *p_stream, es_format_t *p_fmt)
	Add an ES to a non-RTP muxed stream.
static int	MuxDel (sout_stream_t *p_stream, sout_stream_id_t *id)
	Remove an ES from a non-RTP muxed stream.
static int	MuxSend (sout_stream_t *, sout_stream_id_t *, block_t *)
static sout_access_out_t *	GrabberCreate (sout_stream_t *p_sout)
static void *	ThreadSend (vlc_object_t *p_this)
static void	SDPHandleUrl (sout_stream_t *, const char *)
static int	SapSetup (sout_stream_t *p_stream)
static int	FileSetup (sout_stream_t *p_stream)
static int	HttpSetup (sout_stream_t *p_stream, const vlc_url_t *)
char *	SDPGenerate (const sout_stream_t *p_stream, const char *rtsp_url)
static void	sprintf_hexa (char *s, uint8_t *p_data, int i_data)
static void	rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
	Shrink the MTU down to a fixed packetization time (for audio).
static int	HttpCallback (httpd_file_sys_t *p_args, httpd_file_t *, uint8_t *p_request, uint8_t **pp_data, int *pi_data)
	This function is the main HTTPD Callback used by the HTTP Interface.
int	rtp_add_sink (sout_stream_id_t *id, int fd, bool rtcp_mux)
void	rtp_del_sink (sout_stream_id_t *id, int fd)
uint16_t	rtp_get_seq (const sout_stream_id_t *id)
unsigned	rtp_get_num (const sout_stream_id_t *id)
void	rtp_packetize_common (sout_stream_id_t *id, block_t *out, int b_marker, int64_t i_pts)
void	rtp_packetize_send (sout_stream_id_t *id, block_t *out)
size_t	rtp_mtu (const sout_stream_id_t *id)
	Returns: configured max RTP payload size (including payload type-specific headers, excluding RTP and transport headers)
static ssize_t	AccessOutGrabberWriteBuffer (sout_stream_t *p_stream, const block_t *p_buffer)
static ssize_t	AccessOutGrabberWrite (sout_access_out_t *p_access, block_t *p_buffer)
Variables
static const char *const	ppsz_protos []
static const char *const	ppsz_protocols []
static const char *const	ppsz_sout_options []

---

Define Documentation

#define DESC_LONGTEXT

Value:

N_( \
    "This allows you to give a short description with details about the stream, " \
    "that will be announced in the SDP (Session Descriptor)." )

#define DESC_TEXT   N_("Session description")

#define DEST_LONGTEXT

Value:

N_( \
    "This is the output URL that will be used." )

#define DEST_TEXT   N_("Destination")

#define EMAIL_LONGTEXT

Value:

N_( \
    "This allows you to give a contact mail address for the stream, that will " \
    "be announced in the SDP (Session Descriptor)." )

#define EMAIL_TEXT   N_("Session email")

#define IPPROTO_DCCP   33

#define IPPROTO_UDPLITE   136

#define MAX_EMPTY_BLOCKS   200

#define MUX_LONGTEXT

Value:

N_( \
    "This allows you to specify the muxer used for the streaming output. " \
    "Default is to use no muxer (standard RTP stream)." )

#define MUX_TEXT   N_("Muxer")

#define NAME_LONGTEXT

Value:

N_( \
    "This is the name of the session that will be announced in the SDP " \
    "(Session Descriptor)." )

#define NAME_TEXT   N_("Session name")

#define PHONE_LONGTEXT

Value:

N_( \
    "This allows you to give a contact telephone number for the stream, that will " \
    "be announced in the SDP (Session Descriptor)." )

#define PHONE_TEXT   N_("Session phone number")

#define PORT_AUDIO_LONGTEXT

Value:

N_( \
    "This allows you to specify the default audio port for the RTP streaming." )

#define PORT_AUDIO_TEXT   N_("Audio port")

#define PORT_LONGTEXT

Value:

N_( \
    "This allows you to specify the base port for the RTP streaming." )

#define PORT_TEXT   N_("Port")

#define PORT_VIDEO_LONGTEXT

Value:

N_( \
    "This allows you to specify the default video port for the RTP streaming." )

#define PORT_VIDEO_TEXT   N_("Video port")

#define PROTO_LONGTEXT

Value:

N_( \
    "This selects which transport protocol to use for RTP." )

#define PROTO_TEXT   N_("Transport protocol")

#define RFC3016_LONGTEXT

Value:

N_( \
    "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )

#define RFC3016_TEXT   N_("MP4A LATM")

#define RTCP_MUX_LONGTEXT

Value:

N_( \
    "This sends and receives RTCP packet multiplexed over the same port " \
    "as RTP packets." )

#define RTCP_MUX_TEXT   N_("RTP/RTCP multiplexing")

#define SAP_LONGTEXT   N_("Announce this session with SAP.")

#define SAP_TEXT   N_("SAP announcing")

#define SDP_LONGTEXT

Value:

N_( \
    "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
    "session will be made available. You must use an url: http://location to " \
    "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
    "for the SDP to be announced via SAP." )

#define SDP_TEXT   N_("SDP")

#define SOUT_CFG_PREFIX   "sout-rtp-"

#define SRTP_KEY_LONGTEXT

Value:

N_( \
    "RTP packets will be integrity-protected and ciphered "\
    "with this Secure RTP master shared secret key.")

#define SRTP_KEY_TEXT   N_("SRTP key (hexadecimal)")

#define SRTP_SALT_LONGTEXT

Value:

N_( \
    "Secure RTP requires a (non-secret) master salt value.")

#define SRTP_SALT_TEXT   N_("SRTP salt (hexadecimal)")

#define TTL_LONGTEXT

Value:

N_( \
    "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
    "the multicast packets sent by the stream output (0 = use operating " \
    "system built-in default).")

#define TTL_TEXT   N_("Hop limit (TTL)")

#define URL_LONGTEXT

Value:

N_( \
    "This allows you to give an URL with more details about the stream " \
    "(often the website of the streaming organization), that will " \
    "be announced in the SDP (Session Descriptor)." )

#define URL_TEXT   N_("Session URL")

---

Typedef Documentation

typedef int(*) pf_rtp_packetizer_t(sout_stream_id_t *, block_t *)

---

Function Documentation

static ssize_t AccessOutGrabberWrite	(	sout_access_out_t *	p_access,
		block_t *	p_buffer
	)			[static]

static ssize_t AccessOutGrabberWriteBuffer	(	sout_stream_t *	p_stream,
		const block_t *	p_buffer
	)			[static]

static sout_stream_id_t * Add	(	sout_stream_t *	,
		es_format_t *
	)			[static]

Add an ES as a new RTP stream.

static void Close

(

vlc_object_t *

)

[static]

static int Del	(	sout_stream_t *	,
		sout_stream_id_t *
	)			[static]

static int FileSetup

(

sout_stream_t *

p_stream

)

[static]

static sout_access_out_t * GrabberCreate

(

sout_stream_t *

p_sout

)

[static]

static int HttpSetup	(	sout_stream_t *	p_stream,
		const vlc_url_t *
	)			[static]

static sout_stream_id_t * MuxAdd	(	sout_stream_t *	,
		es_format_t *
	)			[static]

Add an ES to a non-RTP muxed stream.

static int MuxDel	(	sout_stream_t *	,
		sout_stream_id_t *
	)			[static]

Remove an ES from a non-RTP muxed stream.

static int MuxSend	(	sout_stream_t *	,
		sout_stream_id_t *	,
		block_t *
	)			[static]

static int Open

(

vlc_object_t *

)

[static]

int rtp_add_sink	(	sout_stream_id_t *	id,
		int	fd,
		bool	rtcp_mux
	)

void rtp_del_sink	(	sout_stream_id_t *	id,
		int	fd
	)

unsigned rtp_get_num

(

const sout_stream_id_t *

id

)

uint16_t rtp_get_seq

(

const sout_stream_id_t *

id

)

size_t rtp_mtu

(

const sout_stream_id_t *

id

)

Returns:

configured max RTP payload size (including payload type-specific headers, excluding RTP and transport headers)

void rtp_packetize_common	(	sout_stream_id_t *	id,
		block_t *	out,
		int	b_marker,
		int64_t	i_pts
	)

void rtp_packetize_send	(	sout_stream_id_t *	id,
		block_t *	out
	)

static void rtp_set_ptime	(	sout_stream_id_t *	id,
		unsigned	ptime_ms,
		size_t	bytes
	)			[static]

Shrink the MTU down to a fixed packetization time (for audio).

static int SapSetup

(

sout_stream_t *

p_stream

)

[static]

char* SDPGenerate	(	const sout_stream_t *	p_stream,
		const char *	rtsp_url
	)

static void SDPHandleUrl	(	sout_stream_t *	,
		const char *
	)			[static]

static int Send	(	sout_stream_t *	,
		sout_stream_id_t *	,
		block_t *
	)			[static]

static void sprintf_hexa	(	char *	s,
		uint8_t *	p_data,
		int	i_data
	)			[static]

static void * ThreadSend

(

vlc_object_t *

p_this

)

[static]

int vlc_entry__main

(

module_t *

p_module

)

const char* vlc_entry_license__main

(

void

)

---

Variable Documentation

const char* const ppsz_protocols[] [static]

Initial value:

 {
    "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
}

const char* const ppsz_protos[] [static]

Initial value:

 {
    "dccp", "sctp", "tcp", "udp", "udplite",
}

const char* const ppsz_sout_options[] [static]

Initial value:

 {
    "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
    "sap", "description", "url", "email", "phone",
    "proto", "rtcp-mux", "key", "salt",
    "mp4a-latm", NULL
}

---