1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
32 #include <vlc_common.h>
33 #include <vlc_plugin.h>
35 #include <vlc_block.h>
37 #include <vlc_httpd.h>
39 #include <vlc_network.h>
40 #include <vlc_charset.h>
41 #include <vlc_strings.h>
47 # include <sys/types.h>
50 # include <sys/stat.h>
52 #ifdef HAVE_LINUX_DCCP_H
53 # include <linux/dccp.h>
56 # define IPPROTO_DCCP 33
58 #ifndef IPPROTO_UDPLITE
59 # define IPPROTO_UDPLITE 136
66 /*****************************************************************************
68 *****************************************************************************/
70 #define DEST_TEXT N_("Destination")
71 #define DEST_LONGTEXT N_( \
72 "This is the output URL that will be used." )
73 #define SDP_TEXT N_("SDP")
74 #define SDP_LONGTEXT N_( \
75 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
76 "session will be made available. You must use an url: http://location to " \
77 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
78 "for the SDP to be announced via SAP." )
79 #define SAP_TEXT N_("SAP announcing")
80 #define SAP_LONGTEXT N_("Announce this session with SAP.")
81 #define MUX_TEXT N_("Muxer")
82 #define MUX_LONGTEXT N_( \
83 "This allows you to specify the muxer used for the streaming output. " \
84 "Default is to use no muxer (standard RTP stream)." )
86 #define NAME_TEXT N_("Session name")
87 #define NAME_LONGTEXT N_( \
88 "This is the name of the session that will be announced in the SDP " \
89 "(Session Descriptor)." )
90 #define DESC_TEXT N_("Session description")
91 #define DESC_LONGTEXT N_( \
92 "This allows you to give a short description with details about the stream, " \
93 "that will be announced in the SDP (Session Descriptor)." )
94 #define URL_TEXT N_("Session URL")
95 #define URL_LONGTEXT N_( \
96 "This allows you to give an URL with more details about the stream " \
97 "(often the website of the streaming organization), that will " \
98 "be announced in the SDP (Session Descriptor)." )
99 #define EMAIL_TEXT N_("Session email")
100 #define EMAIL_LONGTEXT N_( \
101 "This allows you to give a contact mail address for the stream, that will " \
102 "be announced in the SDP (Session Descriptor)." )
103 #define PHONE_TEXT N_("Session phone number")
104 #define PHONE_LONGTEXT N_( \
105 "This allows you to give a contact telephone number for the stream, that will " \
106 "be announced in the SDP (Session Descriptor)." )
108 #define PORT_TEXT N_("Port")
109 #define PORT_LONGTEXT N_( \
110 "This allows you to specify the base port for the RTP streaming." )
111 #define PORT_AUDIO_TEXT N_("Audio port")
112 #define PORT_AUDIO_LONGTEXT N_( \
113 "This allows you to specify the default audio port for the RTP streaming." )
114 #define PORT_VIDEO_TEXT N_("Video port")
115 #define PORT_VIDEO_LONGTEXT N_( \
116 "This allows you to specify the default video port for the RTP streaming." )
118 #define TTL_TEXT N_("Hop limit (TTL)")
119 #define TTL_LONGTEXT N_( \
120 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
121 "the multicast packets sent by the stream output (-1 = use operating " \
122 "system built-in default).")
124 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
125 #define RTCP_MUX_LONGTEXT N_( \
126 "This sends and receives RTCP packet multiplexed over the same port " \
129 #define PROTO_TEXT N_("Transport protocol")
130 #define PROTO_LONGTEXT N_( \
131 "This selects which transport protocol to use for RTP." )
133 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
134 #define SRTP_KEY_LONGTEXT N_( \
135 "RTP packets will be integrity-protected and ciphered "\
136 "with this Secure RTP master shared secret key.")
138 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
139 #define SRTP_SALT_LONGTEXT N_( \
140 "Secure RTP requires a (non-secret) master salt value.")
142 static const char *const ppsz_protos[] = {
143 "dccp", "sctp", "tcp", "udp", "udplite",
146 static const char *const ppsz_protocols[] = {
147 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
150 #define RFC3016_TEXT N_("MP4A LATM")
151 #define RFC3016_LONGTEXT N_( \
152 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
154 static int Open ( vlc_object_t * );
155 static void Close( vlc_object_t * );
157 #define SOUT_CFG_PREFIX "sout-rtp-"
158 #define MAX_EMPTY_BLOCKS 200
161 set_shortname( N_("RTP"));
162 set_description( N_("RTP stream output") );
163 set_capability( "sout stream", 0 );
164 add_shortcut( "rtp" );
165 set_category( CAT_SOUT );
166 set_subcategory( SUBCAT_SOUT_STREAM );
168 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
169 DEST_LONGTEXT, true );
171 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
172 SDP_LONGTEXT, true );
173 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
174 MUX_LONGTEXT, true );
175 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
178 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
179 NAME_LONGTEXT, true );
180 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
181 DESC_LONGTEXT, true );
182 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
183 URL_LONGTEXT, true );
184 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
185 EMAIL_LONGTEXT, true );
186 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
187 PHONE_LONGTEXT, true );
189 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
190 PROTO_LONGTEXT, false );
191 change_string_list( ppsz_protos, ppsz_protocols, NULL );
192 add_integer( SOUT_CFG_PREFIX "port", 50004, NULL, PORT_TEXT,
193 PORT_LONGTEXT, true );
194 add_integer( SOUT_CFG_PREFIX "port-audio", 50000, NULL, PORT_AUDIO_TEXT,
195 PORT_AUDIO_LONGTEXT, true );
196 add_integer( SOUT_CFG_PREFIX "port-video", 50002, NULL, PORT_VIDEO_TEXT,
197 PORT_VIDEO_LONGTEXT, true );
199 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
200 TTL_LONGTEXT, true );
201 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
202 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );
204 add_string( SOUT_CFG_PREFIX "key", "", NULL,
205 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
206 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
207 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );
209 add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
210 RFC3016_LONGTEXT, false );
212 set_callbacks( Open, Close );
215 /*****************************************************************************
216 * Exported prototypes
217 *****************************************************************************/
218 static const char *const ppsz_sout_options[] = {
219 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
220 "sap", "description", "url", "email", "phone",
221 "proto", "rtcp-mux", "key", "salt",
225 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
226 static int Del ( sout_stream_t *, sout_stream_id_t * );
227 static int Send( sout_stream_t *, sout_stream_id_t *,
229 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
230 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
231 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
234 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
235 static void* ThreadSend( vlc_object_t *p_this );
237 static void SDPHandleUrl( sout_stream_t *, const char * );
239 static int SapSetup( sout_stream_t *p_stream );
240 static int FileSetup( sout_stream_t *p_stream );
241 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
243 struct sout_stream_sys_t
247 vlc_mutex_t lock_sdp;
250 bool b_export_sdp_file;
255 session_descriptor_t *p_session;
258 httpd_host_t *p_httpd_host;
259 httpd_file_t *p_httpd_file;
265 char *psz_destination;
267 uint16_t i_port_audio;
268 uint16_t i_port_video;
272 /* when need to use a private one or when using muxer */
273 unsigned i_payload_type:7;
276 /* in case we do TS/PS over rtp */
278 sout_access_out_t *p_grab;
284 sout_stream_id_t **es;
287 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
289 typedef struct rtp_sink_t
295 struct sout_stream_id_t
299 sout_stream_t *p_stream;
302 uint8_t i_payload_type;
314 /* Packetizer specific fields */
316 srtp_session_t *srtp;
317 pf_rtp_packetizer_t pf_packetize;
320 vlc_mutex_t lock_sink;
323 rtsp_stream_id_t *rtsp_id;
326 block_fifo_t *p_fifo;
330 /*****************************************************************************
332 *****************************************************************************/
333 static int Open( vlc_object_t *p_this )
335 sout_stream_t *p_stream = (sout_stream_t*)p_this;
336 sout_instance_t *p_sout = p_stream->p_sout;
337 sout_stream_sys_t *p_sys = NULL;
338 config_chain_t *p_cfg = NULL;
342 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
343 ppsz_sout_options, p_stream->p_cfg );
345 p_sys = malloc( sizeof( sout_stream_sys_t ) );
349 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
351 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
352 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
353 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
354 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
356 p_sys->psz_sdp_file = NULL;
358 if( p_sys->i_port_audio == p_sys->i_port_video )
360 msg_Err( p_stream, "audio and video port cannot be the same" );
361 p_sys->i_port_audio = 0;
362 p_sys->i_port_video = 0;
365 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
367 if( !strcmp( p_cfg->psz_name, "sdp" )
368 && ( p_cfg->psz_value != NULL )
369 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
377 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
380 if( !strncasecmp( psz, "rtsp:", 5 ) )
386 /* Transport protocol */
387 p_sys->proto = IPPROTO_UDP;
388 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
390 if ((psz == NULL) || !strcasecmp (psz, "udp"))
391 (void)0; /* default */
393 if (!strcasecmp (psz, "dccp"))
395 p_sys->proto = IPPROTO_DCCP;
396 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
400 if (!strcasecmp (psz, "sctp"))
402 p_sys->proto = IPPROTO_TCP;
403 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
408 if (!strcasecmp (psz, "tcp"))
410 p_sys->proto = IPPROTO_TCP;
411 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
415 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
416 p_sys->proto = IPPROTO_UDPLITE;
418 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
421 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
423 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
425 msg_Err( p_stream, "missing destination and not in RTSP mode" );
430 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
431 if( p_sys->i_ttl == -1 )
433 /* Normally, we should let the default hop limit up to the core,
434 * but we have to know it to build our SDP properly, which is why
435 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
437 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
440 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
442 p_sys->i_payload_type = 96;
446 p_sys->psz_sdp = NULL;
448 p_sys->b_export_sap = false;
449 p_sys->b_export_sdp_file = false;
450 p_sys->p_session = NULL;
452 p_sys->p_httpd_host = NULL;
453 p_sys->p_httpd_file = NULL;
455 p_stream->p_sys = p_sys;
457 vlc_mutex_init( &p_sys->lock_sdp );
458 vlc_mutex_init( &p_sys->lock_es );
460 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
463 sout_stream_id_t *id;
465 /* Check muxer type */
466 if( strncasecmp( psz, "ps", 2 )
467 && strncasecmp( psz, "mpeg1", 5 )
468 && strncasecmp( psz, "ts", 2 ) )
470 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
472 vlc_mutex_destroy( &p_sys->lock_sdp );
473 vlc_mutex_destroy( &p_sys->lock_es );
478 p_sys->p_grab = GrabberCreate( p_stream );
479 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
482 if( p_sys->p_mux == NULL )
484 msg_Err( p_stream, "cannot create muxer" );
485 sout_AccessOutDelete( p_sys->p_grab );
486 vlc_mutex_destroy( &p_sys->lock_sdp );
487 vlc_mutex_destroy( &p_sys->lock_es );
492 id = Add( p_stream, NULL );
495 sout_MuxDelete( p_sys->p_mux );
496 sout_AccessOutDelete( p_sys->p_grab );
497 vlc_mutex_destroy( &p_sys->lock_sdp );
498 vlc_mutex_destroy( &p_sys->lock_es );
503 p_sys->packet = NULL;
505 p_stream->pf_add = MuxAdd;
506 p_stream->pf_del = MuxDel;
507 p_stream->pf_send = MuxSend;
512 p_sys->p_grab = NULL;
514 p_stream->pf_add = Add;
515 p_stream->pf_del = Del;
516 p_stream->pf_send = Send;
519 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
520 SDPHandleUrl( p_stream, "sap" );
522 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
525 config_chain_t *p_cfg;
527 SDPHandleUrl( p_stream, psz );
529 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
531 if( !strcmp( p_cfg->psz_name, "sdp" ) )
533 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
536 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
537 if( !strcmp( p_cfg->psz_value, psz ) )
540 SDPHandleUrl( p_stream, p_cfg->psz_value );
546 /* update p_sout->i_out_pace_nocontrol */
547 p_stream->p_sout->i_out_pace_nocontrol++;
552 /*****************************************************************************
554 *****************************************************************************/
555 static void Close( vlc_object_t * p_this )
557 sout_stream_t *p_stream = (sout_stream_t*)p_this;
558 sout_stream_sys_t *p_sys = p_stream->p_sys;
560 /* update p_sout->i_out_pace_nocontrol */
561 p_stream->p_sout->i_out_pace_nocontrol--;
565 assert( p_sys->i_es == 1 );
566 Del( p_stream, p_sys->es[0] );
568 sout_MuxDelete( p_sys->p_mux );
569 sout_AccessOutDelete( p_sys->p_grab );
572 block_Release( p_sys->packet );
574 if( p_sys->b_export_sap )
577 SapSetup( p_stream );
581 if( p_sys->rtsp != NULL )
582 RtspUnsetup( p_sys->rtsp );
584 vlc_mutex_destroy( &p_sys->lock_sdp );
585 vlc_mutex_destroy( &p_sys->lock_es );
587 if( p_sys->p_httpd_file )
588 httpd_FileDelete( p_sys->p_httpd_file );
590 if( p_sys->p_httpd_host )
591 httpd_HostDelete( p_sys->p_httpd_host );
593 free( p_sys->psz_sdp );
595 if( p_sys->b_export_sdp_file )
598 unlink( p_sys->psz_sdp_file );
600 free( p_sys->psz_sdp_file );
602 free( p_sys->psz_destination );
606 /*****************************************************************************
608 *****************************************************************************/
609 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
611 sout_stream_sys_t *p_sys = p_stream->p_sys;
614 vlc_UrlParse( &url, psz_url, 0 );
615 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
617 if( p_sys->p_httpd_file )
619 msg_Err( p_stream, "you can use sdp=http:// only once" );
623 if( HttpSetup( p_stream, &url ) )
625 msg_Err( p_stream, "cannot export SDP as HTTP" );
628 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
630 if( p_sys->rtsp != NULL )
632 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
636 /* FIXME test if destination is multicast or no destination at all */
637 p_sys->rtsp = RtspSetup( p_stream, &url );
638 if( p_sys->rtsp == NULL )
640 msg_Err( p_stream, "cannot export SDP as RTSP" );
643 if( p_sys->p_mux != NULL )
645 sout_stream_id_t *id = p_sys->es[0];
646 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
647 p_sys->psz_destination, p_sys->i_ttl,
648 id->i_port, id->i_port + 1 );
651 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
652 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
654 p_sys->b_export_sap = true;
655 SapSetup( p_stream );
657 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
659 if( p_sys->b_export_sdp_file )
661 msg_Err( p_stream, "you can use sdp=file:// only once" );
664 p_sys->b_export_sdp_file = true;
665 psz_url = &psz_url[5];
666 if( psz_url[0] == '/' && psz_url[1] == '/' )
668 p_sys->psz_sdp_file = strdup( psz_url );
672 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
677 vlc_UrlClean( &url );
680 /*****************************************************************************
682 *****************************************************************************/
684 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
686 const sout_stream_sys_t *p_sys = p_stream->p_sys;
688 struct sockaddr_storage dst;
692 * When we have a fixed destination (typically when we do multicast),
693 * we need to put the actual port numbers in the SDP.
694 * When there is no fixed destination, we only support RTSP unicast
695 * on-demand setup, so we should rather let the clients decide which ports
697 * When there is both a fixed destination and RTSP unicast, we need to
698 * put port numbers used by the fixed destination, otherwise the SDP would
699 * become totally incorrect for multicast use. It should be noted that
700 * port numbers from SDP with RTSP are only "recommendation" from the
701 * server to the clients (per RFC2326), so only broken clients will fail
702 * to handle this properly. There is no solution but to use two differents
703 * output chain with two different RTSP URLs if you need to handle this
708 if( p_sys->psz_destination != NULL )
712 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
713 dstlen = sizeof( dst );
714 if( p_sys->es[0]->listen_fd != NULL )
715 getsockname( p_sys->es[0]->listen_fd[0],
716 (struct sockaddr *)&dst, &dstlen );
718 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
719 (struct sockaddr *)&dst, &dstlen );
725 /* Dummy destination address for RTSP */
726 memset (&dst, 0, sizeof( struct sockaddr_in ) );
727 dst.ss_family = AF_INET;
731 dstlen = sizeof( struct sockaddr_in );
734 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
735 NULL, 0, (struct sockaddr *)&dst, dstlen );
736 if( psz_sdp == NULL )
739 /* TODO: a=source-filter */
740 if( p_sys->rtcp_mux )
741 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
743 if( rtsp_url != NULL )
744 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
746 /* FIXME: locking?! */
747 for( i = 0; i < p_sys->i_es; i++ )
749 sout_stream_id_t *id = p_sys->es[i];
750 const char *mime_major; /* major MIME type */
751 const char *proto = "RTP/AVP"; /* protocol */
756 mime_major = "video";
759 mime_major = "audio";
768 if( rtsp_url == NULL )
770 switch( p_sys->proto )
775 proto = "TCP/RTP/AVP";
778 proto = "DCCP/RTP/AVP";
780 case IPPROTO_UDPLITE:
785 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
786 id->i_payload_type, false, id->i_bitrate,
787 id->psz_enc, id->i_clock_rate, id->i_channels,
790 if( rtsp_url != NULL )
792 assert( strlen( rtsp_url ) > 0 );
793 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
794 sdp_AddAttribute ( &psz_sdp, "control",
795 addslash ? "%s/trackID=%u" : "%strackID=%u",
800 if( id->listen_fd != NULL )
801 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
802 if( p_sys->proto == IPPROTO_DCCP )
803 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
804 "SC:RTP%c", toupper( mime_major[0] ) );
811 /*****************************************************************************
813 *****************************************************************************/
815 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
817 static const char hex[16] = "0123456789abcdef";
820 for( i = 0; i < i_data; i++ )
822 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
823 s[2*i+1] = hex[(p_data[i] )&0xf];
829 * Shrink the MTU down to a fixed packetization time (for audio).
832 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
834 /* Samples per second */
835 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
836 bytes *= id->i_channels;
839 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
840 id->i_mtu = 12 + spl;
841 else /* MTU is too small for ptime, align to a sample boundary */
842 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
845 /** Add an ES as a new RTP stream */
846 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
848 /* NOTE: As a special case, if we use a non-RTP
849 * mux (TS/PS), then p_fmt is NULL. */
850 sout_stream_sys_t *p_sys = p_stream->p_sys;
851 sout_stream_id_t *id;
852 int i_port, cscov = -1;
855 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
858 vlc_object_attach( id, p_stream );
860 /* Choose the port */
865 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
867 i_port = p_sys->i_port_audio;
868 p_sys->i_port_audio = 0;
871 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
873 i_port = p_sys->i_port_video;
874 p_sys->i_port_video = 0;
879 if( p_sys->i_port != p_sys->i_port_audio
880 && p_sys->i_port != p_sys->i_port_video )
882 i_port = p_sys->i_port;
889 id->p_stream = p_stream;
891 id->i_sequence = rand()&0xffff;
892 id->i_payload_type = p_sys->i_payload_type;
893 id->ssrc[0] = rand()&0xff;
894 id->ssrc[1] = rand()&0xff;
895 id->ssrc[2] = rand()&0xff;
896 id->ssrc[3] = rand()&0xff;
900 id->i_clock_rate = 90000; /* most common case for video */
905 id->i_cat = p_fmt->i_cat;
906 if( p_fmt->i_cat == AUDIO_ES )
908 id->i_clock_rate = p_fmt->audio.i_rate;
909 id->i_channels = p_fmt->audio.i_channels;
911 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
915 id->i_cat = VIDEO_ES;
919 id->i_mtu = config_GetInt( p_stream, "mtu" );
920 if( id->i_mtu <= 12 + 16 )
921 id->i_mtu = 576 - 20 - 8; /* pessimistic */
922 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
925 id->pf_packetize = NULL;
927 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
930 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
931 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
932 if (id->srtp == NULL)
938 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
939 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
944 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
947 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
950 vlc_mutex_init( &id->lock_sink );
955 id->listen_fd = NULL;
958 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
960 if( p_sys->psz_destination != NULL )
961 switch( p_sys->proto )
968 case VIDEO_ES: code = "RTPV"; break;
969 case AUDIO_ES: code = "RTPARTPV"; break;
970 case SPU_ES: code = "RTPTRPTV"; break;
971 default: code = "RTPORTPV"; break;
973 var_SetString (p_stream, "dccp-service", code);
976 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
977 p_sys->psz_destination, i_port,
979 if( id->listen_fd == NULL )
981 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
988 int ttl = (p_sys->i_ttl > 0) ? p_sys->i_ttl : -1;
989 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
990 i_port, ttl, p_sys->proto );
993 msg_Err( p_stream, "cannot create RTP socket" );
996 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1002 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1004 if( psz == NULL ) /* Uho! */
1007 if( strncmp( psz, "ts", 2 ) == 0 )
1009 id->i_payload_type = 33;
1010 id->psz_enc = "MP2T";
1014 id->psz_enc = "MP2P";
1019 switch( p_fmt->i_codec )
1021 case VLC_FOURCC( 'u', 'l', 'a', 'w' ):
1022 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1023 id->i_payload_type = 0;
1024 id->psz_enc = "PCMU";
1025 id->pf_packetize = rtp_packetize_split;
1026 rtp_set_ptime (id, 20, 1);
1028 case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
1029 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1030 id->i_payload_type = 8;
1031 id->psz_enc = "PCMA";
1032 id->pf_packetize = rtp_packetize_split;
1033 rtp_set_ptime (id, 20, 1);
1035 case VLC_FOURCC( 's', '1', '6', 'b' ):
1036 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1038 id->i_payload_type = 11;
1040 else if( p_fmt->audio.i_channels == 2 &&
1041 p_fmt->audio.i_rate == 44100 )
1043 id->i_payload_type = 10;
1045 id->psz_enc = "L16";
1046 id->pf_packetize = rtp_packetize_split;
1047 rtp_set_ptime (id, 20, 2);
1049 case VLC_FOURCC( 'u', '8', ' ', ' ' ):
1051 id->pf_packetize = rtp_packetize_split;
1052 rtp_set_ptime (id, 20, 1);
1054 case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
1055 case VLC_FOURCC( 'm', 'p', '3', ' ' ):
1056 id->i_payload_type = 14;
1057 id->psz_enc = "MPA";
1058 id->i_clock_rate = 90000; /* not 44100 */
1059 id->pf_packetize = rtp_packetize_mpa;
1061 case VLC_FOURCC( 'm', 'p', 'g', 'v' ):
1062 id->i_payload_type = 32;
1063 id->psz_enc = "MPV";
1064 id->pf_packetize = rtp_packetize_mpv;
1066 case VLC_FOURCC( 'G', '7', '2', '6' ):
1067 case VLC_FOURCC( 'g', '7', '2', '6' ):
1068 switch( p_fmt->i_bitrate / 1000 )
1071 id->psz_enc = "G726-16";
1072 id->pf_packetize = rtp_packetize_g726_16;
1075 id->psz_enc = "G726-24";
1076 id->pf_packetize = rtp_packetize_g726_24;
1079 id->psz_enc = "G726-32";
1080 id->pf_packetize = rtp_packetize_g726_32;
1083 id->psz_enc = "G726-40";
1084 id->pf_packetize = rtp_packetize_g726_40;
1088 case VLC_FOURCC( 'a', '5', '2', ' ' ):
1089 id->psz_enc = "ac3";
1090 id->pf_packetize = rtp_packetize_ac3;
1092 case VLC_FOURCC( 'H', '2', '6', '3' ):
1093 id->psz_enc = "H263-1998";
1094 id->pf_packetize = rtp_packetize_h263;
1096 case VLC_FOURCC( 'h', '2', '6', '4' ):
1097 id->psz_enc = "H264";
1098 id->pf_packetize = rtp_packetize_h264;
1099 id->psz_fmtp = NULL;
1101 if( p_fmt->i_extra > 0 )
1103 uint8_t *p_buffer = p_fmt->p_extra;
1104 int i_buffer = p_fmt->i_extra;
1105 char *p_64_sps = NULL;
1106 char *p_64_pps = NULL;
1109 while( i_buffer > 4 &&
1110 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1111 p_buffer[2] == 0 && p_buffer[3] == 1 )
1113 const int i_nal_type = p_buffer[4]&0x1f;
1117 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1120 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1122 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1124 /* we found another startcode */
1129 if( i_nal_type == 7 )
1131 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1132 sprintf_hexa( hexa, &p_buffer[5], 3 );
1134 else if( i_nal_type == 8 )
1136 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1142 if( p_64_sps && p_64_pps &&
1143 ( asprintf( &id->psz_fmtp,
1144 "packetization-mode=1;profile-level-id=%s;"
1145 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1146 p_64_pps ) == -1 ) )
1147 id->psz_fmtp = NULL;
1152 id->psz_fmtp = strdup( "packetization-mode=1" );
1155 case VLC_FOURCC( 'm', 'p', '4', 'v' ):
1157 char hexa[2*p_fmt->i_extra +1];
1159 id->psz_enc = "MP4V-ES";
1160 id->pf_packetize = rtp_packetize_split;
1161 if( p_fmt->i_extra > 0 )
1163 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1164 if( asprintf( &id->psz_fmtp,
1165 "profile-level-id=3; config=%s;", hexa ) == -1 )
1166 id->psz_fmtp = NULL;
1170 case VLC_FOURCC( 'm', 'p', '4', 'a' ):
1174 char hexa[2*p_fmt->i_extra +1];
1176 id->psz_enc = "mpeg4-generic";
1177 id->pf_packetize = rtp_packetize_mp4a;
1178 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1179 if( asprintf( &id->psz_fmtp,
1180 "streamtype=5; profile-level-id=15; "
1181 "mode=AAC-hbr; config=%s; SizeLength=13; "
1182 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1184 id->psz_fmtp = NULL;
1190 unsigned char config[6];
1191 unsigned int aacsrates[15] = {
1192 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1193 16000, 12000, 11025, 8000, 7350, 0, 0 };
1195 for( i = 0; i < 15; i++ )
1196 if( p_fmt->audio.i_rate == aacsrates[i] )
1202 config[3]=p_fmt->audio.i_channels<<4;
1206 id->psz_enc = "MP4A-LATM";
1207 id->pf_packetize = rtp_packetize_mp4a_latm;
1208 sprintf_hexa( hexa, config, 6 );
1209 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1210 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1211 id->psz_fmtp = NULL;
1215 case VLC_FOURCC( 's', 'a', 'm', 'r' ):
1216 id->psz_enc = "AMR";
1217 id->psz_fmtp = strdup( "octet-align=1" );
1218 id->pf_packetize = rtp_packetize_amr;
1220 case VLC_FOURCC( 's', 'a', 'w', 'b' ):
1221 id->psz_enc = "AMR-WB";
1222 id->psz_fmtp = strdup( "octet-align=1" );
1223 id->pf_packetize = rtp_packetize_amr;
1225 case VLC_FOURCC( 's', 'p', 'x', ' ' ):
1226 id->i_payload_type = p_sys->i_payload_type++;
1227 id->psz_enc = "SPEEX";
1228 id->pf_packetize = rtp_packetize_spx;
1230 case VLC_FOURCC( 't', '1', '4', '0' ):
1231 id->psz_enc = "t140" ;
1232 id->i_clock_rate = 1000;
1233 id->pf_packetize = rtp_packetize_t140;
1237 msg_Err( p_stream, "cannot add this stream (unsupported "
1238 "codec:%4.4s)", (char*)&p_fmt->i_codec );
1243 cscov += 8 /* UDP */ + 12 /* RTP */;
1245 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1247 if( id->i_payload_type == p_sys->i_payload_type )
1248 p_sys->i_payload_type++;
1250 if( p_sys->rtsp != NULL )
1251 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1252 GetDWBE( id->ssrc ),
1253 p_sys->psz_destination,
1254 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1256 id->p_fifo = block_FifoNew();
1257 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1258 VLC_THREAD_PRIORITY_HIGHEST, false ) )
1261 /* Update p_sys context */
1262 vlc_mutex_lock( &p_sys->lock_es );
1263 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1264 vlc_mutex_unlock( &p_sys->lock_es );
1266 psz_sdp = SDPGenerate( p_stream, NULL );
1268 vlc_mutex_lock( &p_sys->lock_sdp );
1269 free( p_sys->psz_sdp );
1270 p_sys->psz_sdp = psz_sdp;
1271 vlc_mutex_unlock( &p_sys->lock_sdp );
1273 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1275 /* Update SDP (sap/file) */
1276 if( p_sys->b_export_sap ) SapSetup( p_stream );
1277 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1282 Del( p_stream, id );
1286 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1288 sout_stream_sys_t *p_sys = p_stream->p_sys;
1290 if( id->p_fifo != NULL )
1292 vlc_object_kill( id );
1293 vlc_thread_join( id );
1294 block_FifoRelease( id->p_fifo );
1297 vlc_mutex_lock( &p_sys->lock_es );
1298 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1299 vlc_mutex_unlock( &p_sys->lock_es );
1302 if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
1303 p_sys->i_port_audio = id->i_port;
1304 if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
1305 p_sys->i_port_video = id->i_port;
1307 free( id->psz_fmtp );
1310 RtspDelId( p_sys->rtsp, id->rtsp_id );
1312 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1313 if( id->listen_fd != NULL )
1314 net_ListenClose( id->listen_fd );
1315 if( id->srtp != NULL )
1316 srtp_destroy( id->srtp );
1318 vlc_mutex_destroy( &id->lock_sink );
1320 /* Update SDP (sap/file) */
1321 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1322 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1324 vlc_object_detach( id );
1325 vlc_object_release( id );
1329 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1334 assert( p_stream->p_sys->p_mux == NULL );
1337 while( p_buffer != NULL )
1339 p_next = p_buffer->p_next;
1340 if( id->pf_packetize( id, p_buffer ) )
1343 block_Release( p_buffer );
1349 /****************************************************************************
1351 ****************************************************************************/
1352 static int SapSetup( sout_stream_t *p_stream )
1354 sout_stream_sys_t *p_sys = p_stream->p_sys;
1355 sout_instance_t *p_sout = p_stream->p_sout;
1357 /* Remove the previous session */
1358 if( p_sys->p_session != NULL)
1360 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1361 p_sys->p_session = NULL;
1364 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1366 announce_method_t *p_method = sout_SAPMethod();
1367 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1369 p_sys->psz_destination,
1371 sout_MethodRelease( p_method );
1377 /****************************************************************************
1379 ****************************************************************************/
1380 static int FileSetup( sout_stream_t *p_stream )
1382 sout_stream_sys_t *p_sys = p_stream->p_sys;
1385 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1387 msg_Err( p_stream, "cannot open file '%s' (%m)",
1388 p_sys->psz_sdp_file );
1389 return VLC_EGENERIC;
1392 fputs( p_sys->psz_sdp, f );
1398 /****************************************************************************
1400 ****************************************************************************/
1401 static int HttpCallback( httpd_file_sys_t *p_args,
1402 httpd_file_t *, uint8_t *p_request,
1403 uint8_t **pp_data, int *pi_data );
1405 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1407 sout_stream_sys_t *p_sys = p_stream->p_sys;
1409 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1410 url->i_port > 0 ? url->i_port : 80 );
1411 if( p_sys->p_httpd_host )
1413 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1414 url->psz_path ? url->psz_path : "/",
1417 HttpCallback, (void*)p_sys );
1419 if( p_sys->p_httpd_file == NULL )
1421 return VLC_EGENERIC;
1426 static int HttpCallback( httpd_file_sys_t *p_args,
1427 httpd_file_t *f, uint8_t *p_request,
1428 uint8_t **pp_data, int *pi_data )
1430 VLC_UNUSED(f); VLC_UNUSED(p_request);
1431 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1433 vlc_mutex_lock( &p_sys->lock_sdp );
1434 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1436 *pi_data = strlen( p_sys->psz_sdp );
1437 *pp_data = malloc( *pi_data );
1438 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1445 vlc_mutex_unlock( &p_sys->lock_sdp );
1450 /****************************************************************************
1452 ****************************************************************************/
1453 static void* ThreadSend( vlc_object_t *p_this )
1455 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1456 unsigned i_caching = id->i_caching;
1460 block_t *out = block_FifoGet( id->p_fifo );
1461 block_cleanup_push (out);
1464 { /* FIXME: this is awfully inefficient */
1465 size_t len = out->i_buffer;
1466 out = block_Realloc( out, 0, len + 10 );
1467 out->i_buffer = len;
1469 int canc = vlc_savecancel ();
1470 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1471 vlc_restorecancel (canc);
1475 msg_Dbg( id, "SRTP sending error: %m" );
1476 block_Release( out );
1480 out->i_buffer = len;
1484 mwait (out->i_dts + i_caching);
1489 ssize_t len = out->i_buffer;
1490 int canc = vlc_savecancel ();
1492 vlc_mutex_lock( &id->lock_sink );
1493 unsigned deadc = 0; /* How many dead sockets? */
1494 int deadv[id->sinkc]; /* Dead sockets list */
1496 for( int i = 0; i < id->sinkc; i++ )
1498 if( !id->srtp ) /* FIXME: SRTCP support */
1499 SendRTCP( id->sinkv[i].rtcp, out );
1501 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1503 /* Retry sending to root out soft-errors */
1504 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1507 deadv[deadc++] = id->sinkv[i].rtp_fd;
1509 vlc_mutex_unlock( &id->lock_sink );
1510 block_Release( out );
1512 for( unsigned i = 0; i < deadc; i++ )
1514 msg_Dbg( id, "removing socket %d", deadv[i] );
1515 rtp_del_sink( id, deadv[i] );
1518 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1519 while( id->listen_fd != NULL )
1521 int fd = net_Accept( id, id->listen_fd, 0 );
1524 msg_Dbg( id, "adding socket %d", fd );
1525 rtp_add_sink( id, fd, true );
1527 vlc_restorecancel (canc);
1532 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1534 rtp_sink_t sink = { fd, NULL };
1535 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1537 if( sink.rtcp == NULL )
1538 msg_Err( id, "RTCP failed!" );
1540 vlc_mutex_lock( &id->lock_sink );
1541 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1542 vlc_mutex_unlock( &id->lock_sink );
1546 void rtp_del_sink( sout_stream_id_t *id, int fd )
1548 rtp_sink_t sink = { fd, NULL };
1550 /* NOTE: must be safe to use if fd is not included */
1551 vlc_mutex_lock( &id->lock_sink );
1552 for( int i = 0; i < id->sinkc; i++ )
1554 if (id->sinkv[i].rtp_fd == fd)
1556 sink = id->sinkv[i];
1557 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1561 vlc_mutex_unlock( &id->lock_sink );
1563 CloseRTCP( sink.rtcp );
1564 net_Close( sink.rtp_fd );
1567 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1569 /* This will return values for the next packet.
1570 * Accounting for caching would not be totally trivial. */
1571 return id->i_sequence;
1574 /* FIXME: this is pretty bad - if we remove and then insert an ES
1575 * the number will get unsynched from inside RTSP */
1576 unsigned rtp_get_num( const sout_stream_id_t *id )
1578 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1581 vlc_mutex_lock( &p_sys->lock_es );
1582 for( i = 0; i < p_sys->i_es; i++ )
1584 if( id == p_sys->es[i] )
1587 vlc_mutex_unlock( &p_sys->lock_es );
1593 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1594 int b_marker, int64_t i_pts )
1596 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
1598 out->p_buffer[0] = 0x80;
1599 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1600 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1601 out->p_buffer[3] = ( id->i_sequence )&0xff;
1602 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1603 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1604 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1605 out->p_buffer[7] = ( i_timestamp )&0xff;
1607 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1613 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1615 block_FifoPut( id->p_fifo, out );
1619 * @return configured max RTP payload size (including payload type-specific
1620 * headers, excluding RTP and transport headers)
1622 size_t rtp_mtu (const sout_stream_id_t *id)
1624 return id->i_mtu - 12;
1627 /*****************************************************************************
1629 *****************************************************************************/
1631 /** Add an ES to a non-RTP muxed stream */
1632 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1634 sout_input_t *p_input;
1635 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1636 assert( p_mux != NULL );
1638 p_input = sout_MuxAddStream( p_mux, p_fmt );
1639 if( p_input == NULL )
1641 msg_Err( p_stream, "cannot add this stream to the muxer" );
1645 return (sout_stream_id_t *)p_input;
1649 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1652 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1653 assert( p_mux != NULL );
1655 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1660 /** Remove an ES from a non-RTP muxed stream */
1661 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1663 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1664 assert( p_mux != NULL );
1666 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1671 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1672 const block_t *p_buffer )
1674 sout_stream_sys_t *p_sys = p_stream->p_sys;
1675 sout_stream_id_t *id = p_sys->es[0];
1677 int64_t i_dts = p_buffer->i_dts;
1679 uint8_t *p_data = p_buffer->p_buffer;
1680 size_t i_data = p_buffer->i_buffer;
1681 size_t i_max = id->i_mtu - 12;
1683 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1689 /* output complete packet */
1690 if( p_sys->packet &&
1691 p_sys->packet->i_buffer + i_data > i_max )
1693 rtp_packetize_send( id, p_sys->packet );
1694 p_sys->packet = NULL;
1697 if( p_sys->packet == NULL )
1699 /* allocate a new packet */
1700 p_sys->packet = block_New( p_stream, id->i_mtu );
1701 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1702 p_sys->packet->i_dts = i_dts;
1703 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1704 i_dts += p_sys->packet->i_length;
1707 i_size = __MIN( i_data,
1708 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1710 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1713 p_sys->packet->i_buffer += i_size;
1722 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1725 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1731 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1733 p_next = p_buffer->p_next;
1734 block_Release( p_buffer );
1742 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1744 sout_access_out_t *p_grab;
1746 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1747 if( p_grab == NULL )
1750 p_grab->p_module = NULL;
1751 p_grab->psz_access = strdup( "grab" );
1752 p_grab->p_cfg = NULL;
1753 p_grab->psz_path = strdup( "" );
1754 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1755 p_grab->pf_seek = NULL;
1756 p_grab->pf_write = AccessOutGrabberWrite;
1757 vlc_object_attach( p_grab, p_stream );