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	CACHING_TEXT N_("Caching value (ms)")
#define	CACHING_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 *)
	Probes and initializes.
static void	Close (vlc_object_t *)
	Releases resources.
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 (void *)
static void *	rtp_listen_thread (void *)
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 (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).
uint32_t	rtp_compute_ts (const sout_stream_id_t *id, int64_t i_pts)
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, uint16_t seq)
void	rtp_del_sink (sout_stream_id_t *id, int fd)
uint16_t	rtp_get_seq (sout_stream_id_t *id)
int64_t	rtp_get_ts (const sout_stream_t *p_stream)
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)
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 CACHING_LONGTEXT

Value:

N_( \
    "Default caching value for outbound RTP streams. This " \
    "value should be set in milliseconds." )

#define CACHING_TEXT N_("Caching value (ms)")

#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 (-1 = 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]`

References AccessOutGrabberWriteBuffer(), block_Release(), block_t::p_next, sout_access_out_t::p_sys, and VLC_SUCCESS.

Referenced by GrabberCreate().

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

References __MIN, block_New, sout_stream_sys_t::es, block_t::i_buffer, block_t::i_dts, block_t::i_length, sout_stream_id_t::i_mtu, block_t::p_buffer, sout_stream_t::p_sys, sout_stream_sys_t::packet, rtp_packetize_common(), rtp_packetize_send(), and VLC_SUCCESS.

Referenced by AccessOutGrabberWrite().

static sout_stream_id_t * Add	(	sout_stream_t *	p_stream,
		es_format_t *	p_fmt
	)			`[static]`

Add an ES as a new RTP stream.

static void Close ( vlc_object_t * p_this ) [static]

Releases resources.

Close the module.

Destroys the X11 window.

Disconnect from the X server.

Close a SDL video output.

It destroys an OpenGL vout display.

Terminate a vout display created by Open.

It destroyes a Direct3D vout display.

Terminate an output method created by Open.

Release the drawable.

Close a libcaca video output.

Close a aa video output method.

Close the filter.

Terminate a splitter module.

This function closes a clone video splitter module.

Common close function.

Releases allocate resources.

Close: Destructor.

Parameters:

	p_this	pointer to this filter object
	p_this,:	the filter object

static int Del	(	sout_stream_t *	p_stream,
		sout_stream_id_t *	id
	)			`[static]`

static int FileSetup ( sout_stream_t * p_stream ) [static]

References msg_Err, sout_stream_t::p_sys, sout_stream_sys_t::psz_sdp, sout_stream_sys_t::psz_sdp_file, VLC_EGENERIC, vlc_fopen(), and VLC_SUCCESS.

Referenced by Add(), Del(), and SDPHandleUrl().

static sout_access_out_t * GrabberCreate ( sout_stream_t * p_sout ) [static]

References AccessOutGrabberWrite(), sout_access_out_t::p_cfg, sout_access_out_t::p_module, sout_stream_t::p_sout, sout_access_out_t::p_sys, sout_access_out_t::pf_seek, sout_access_out_t::pf_write, sout_access_out_t::psz_access, sout_access_out_t::psz_path, strdup(), vlc_object_attach, and vlc_object_create.

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

References HttpCallback(), httpd_FileNew(), httpd_HostNew(), vlc_url_t::i_port, sout_stream_sys_t::p_httpd_file, sout_stream_sys_t::p_httpd_host, sout_stream_t::p_sys, vlc_url_t::psz_host, vlc_url_t::psz_path, VLC_EGENERIC, VLC_OBJECT, and VLC_SUCCESS.

Referenced by SDPHandleUrl().

static sout_stream_id_t * MuxAdd	(	sout_stream_t *	p_stream,
		es_format_t *	p_fmt
	)			`[static]`

Add an ES to a non-RTP muxed stream.

References msg_Err, sout_stream_sys_t::p_mux, sout_stream_t::p_sys, and sout_MuxAddStream().

static int MuxDel	(	sout_stream_t *	p_stream,
		sout_stream_id_t *	id
	)			`[static]`

Remove an ES from a non-RTP muxed stream.

References sout_stream_sys_t::p_mux, sout_stream_t::p_sys, sout_MuxDeleteStream(), and VLC_SUCCESS.

static int MuxSend	(	sout_stream_t *	p_stream,
		sout_stream_id_t *	id,
		block_t *	p_buffer
	)			`[static]`

References sout_stream_sys_t::p_mux, sout_stream_t::p_sys, sout_MuxSendBuffer(), and VLC_SUCCESS.

static int Open ( vlc_object_t * p_this ) [static]

Probes and initializes.

Status of this demuxer: Real Media format -----------------.

Basic chorus/flanger/delay audio filter This implements a variable delay filter for VLC.

Todo:

Improve CDDB support (non-blocking, cache, .

Open the module.

Create an X11 window.

Probe the X server.

This function initializes SDL vout method.

It creates an OpenGL vout display.

This function allocates and initialize the DirectX vout display.

It creates a Direct3D vout display.

This function allocates and initializes a FB vout method.

Find the drawable set by libvlc application.

This function initializes libcaca vout method.

This function allocates and initializes a aa vout method.

This function allocates and initializes a Wall splitter module.

Open the filter.

This function allocates and initializes a Clone splitter module.

Open: initialize and create stuff.

Connect to the sftp server and ask for a file.

Parameters:

p_this,:

the vlc_object

Returns:: VLC_SUCCESS if everything was fine

Parameters:

	p_this
	p_this,:	the filter object

Returns:: VLC_SUCCESS or vlc error codes

Todo:: Reinstate meta codec name

int rtp_add_sink	(	sout_stream_id_t *	id,
		int	fd,
		bool	rtcp_mux,
		uint16_t *	seq
	)

References INSERT_ELEM, sout_stream_id_t::lock_sink, msg_Err, OpenRTCP(), sout_stream_id_t::p_stream, rtp_sink_t::rtcp, sout_stream_id_t::sinkc, sout_stream_id_t::sinkv, vlc_mutex_lock(), vlc_mutex_unlock(), VLC_OBJECT, and VLC_SUCCESS.

Referenced by Add(), rtp_listen_thread(), and RtspHandler().

uint32_t rtp_compute_ts	(	const sout_stream_id_t *	id,
		int64_t	i_pts
	)

References CLOCK_FREQ, and sout_stream_id_t::i_clock_rate.

Referenced by Add(), rtp_packetize_common(), and RtspHandler().

void rtp_del_sink	(	sout_stream_id_t *	id,
		int	fd
	)

References CloseRTCP(), sout_stream_id_t::lock_sink, net_Close, REMOVE_ELEM, rtp_sink_t::rtcp, rtp_sink_t::rtp_fd, sout_stream_id_t::sinkc, sout_stream_id_t::sinkv, vlc_mutex_lock(), and vlc_mutex_unlock().

Referenced by Del(), RtspClientDel(), RtspDelId(), RtspHandler(), and ThreadSend().

uint16_t rtp_get_seq ( sout_stream_id_t * id )

References sout_stream_id_t::lock_sink, vlc_mutex_lock(), and vlc_mutex_unlock().

Referenced by RtspHandler().

int64_t rtp_get_ts ( const sout_stream_t * p_stream )

References sout_stream_sys_t::i_npt_zero, sout_stream_sys_t::i_pts_zero, sout_stream_sys_t::lock_ts, mdate(), sout_stream_t::p_sys, vlc_mutex_lock(), vlc_mutex_unlock(), and VLC_TS_INVALID.

Referenced by RtspHandler().

static void * rtp_listen_thread ( void * data ) [static]

References sout_stream_id_t::fd, sout_stream_id_t::listen, net_Accept, sout_stream_id_t::p_stream, rtp_add_sink(), vlc_restorecancel(), and vlc_savecancel().

Referenced by Add().

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)

Referenced by rtp_packetize_ac3(), rtp_packetize_amr(), rtp_packetize_g726(), rtp_packetize_h263(), rtp_packetize_h264_nal(), rtp_packetize_mp4a(), rtp_packetize_mp4a_latm(), rtp_packetize_mpa(), rtp_packetize_mpv(), rtp_packetize_split(), rtp_packetize_spx(), rtp_packetize_swab(), rtp_packetize_t140(), and rtp_set_ptime().

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
	)

References block_FifoPut(), and sout_stream_id_t::p_fifo.

Referenced by AccessOutGrabberWriteBuffer(), rtp_packetize_ac3(), rtp_packetize_amr(), rtp_packetize_g726(), rtp_packetize_h263(), rtp_packetize_h264_nal(), rtp_packetize_mp4a(), rtp_packetize_mp4a_latm(), rtp_packetize_mpa(), rtp_packetize_mpv(), rtp_packetize_split(), rtp_packetize_spx(), rtp_packetize_swab(), and rtp_packetize_t140().

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).

References sout_stream_id_t::i_mtu, and rtp_mtu().

Referenced by Add().

static int SapSetup ( sout_stream_t * p_stream ) [static]

References sout_stream_sys_t::i_es, sout_stream_sys_t::p_mux, sout_stream_sys_t::p_session, sout_stream_t::p_sout, sout_stream_t::p_sys, sout_stream_sys_t::psz_destination, sout_stream_sys_t::psz_sdp, sout_AnnounceRegisterSDP, sout_AnnounceUnRegister, sout_MethodRelease(), sout_SAPMethod(), and VLC_SUCCESS.

Referenced by Add(), Del(), and SDPHandleUrl().

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

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

static int Send	(	sout_stream_t *	p_stream,
		sout_stream_id_t *	id,
		block_t *	p_buffer
	)			`[static]`

References block_Release(), sout_stream_sys_t::p_mux, block_t::p_next, sout_stream_t::p_sys, sout_stream_id_t::pf_packetize, and VLC_SUCCESS.

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

References hex().

static void * ThreadSend ( void * data ) [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", "caching", "key", "salt",
    "mp4a-latm", NULL
}

rtp.c File Reference

Data Structures

Defines

Typedefs

Functions

Variables

Define Documentation

Typedef Documentation

Function Documentation

Variable Documentation