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>
50 # include <sys/types.h>
53 # include <sys/stat.h>
55 #ifdef HAVE_LINUX_DCCP_H
56 # include <linux/dccp.h>
59 # define IPPROTO_DCCP 33
61 #ifndef IPPROTO_UDPLITE
62 # define IPPROTO_UDPLITE 136
69 /*****************************************************************************
71 *****************************************************************************/
73 #define DEST_TEXT N_("Destination")
74 #define DEST_LONGTEXT N_( \
75 "This is the output URL that will be used." )
76 #define SDP_TEXT N_("SDP")
77 #define SDP_LONGTEXT N_( \
78 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
79 "session will be made available. You must use an url: http://location to " \
80 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
81 "for the SDP to be announced via SAP." )
82 #define SAP_TEXT N_("SAP announcing")
83 #define SAP_LONGTEXT N_("Announce this session with SAP.")
84 #define MUX_TEXT N_("Muxer")
85 #define MUX_LONGTEXT N_( \
86 "This allows you to specify the muxer used for the streaming output. " \
87 "Default is to use no muxer (standard RTP stream)." )
89 #define NAME_TEXT N_("Session name")
90 #define NAME_LONGTEXT N_( \
91 "This is the name of the session that will be announced in the SDP " \
92 "(Session Descriptor)." )
93 #define DESC_TEXT N_("Session description")
94 #define DESC_LONGTEXT N_( \
95 "This allows you to give a short description with details about the stream, " \
96 "that will be announced in the SDP (Session Descriptor)." )
97 #define URL_TEXT N_("Session URL")
98 #define URL_LONGTEXT N_( \
99 "This allows you to give an URL with more details about the stream " \
100 "(often the website of the streaming organization), that will " \
101 "be announced in the SDP (Session Descriptor)." )
102 #define EMAIL_TEXT N_("Session email")
103 #define EMAIL_LONGTEXT N_( \
104 "This allows you to give a contact mail address for the stream, that will " \
105 "be announced in the SDP (Session Descriptor)." )
106 #define PHONE_TEXT N_("Session phone number")
107 #define PHONE_LONGTEXT N_( \
108 "This allows you to give a contact telephone number for the stream, that will " \
109 "be announced in the SDP (Session Descriptor)." )
111 #define PORT_TEXT N_("Port")
112 #define PORT_LONGTEXT N_( \
113 "This allows you to specify the base port for the RTP streaming." )
114 #define PORT_AUDIO_TEXT N_("Audio port")
115 #define PORT_AUDIO_LONGTEXT N_( \
116 "This allows you to specify the default audio port for the RTP streaming." )
117 #define PORT_VIDEO_TEXT N_("Video port")
118 #define PORT_VIDEO_LONGTEXT N_( \
119 "This allows you to specify the default video port for the RTP streaming." )
121 #define TTL_TEXT N_("Hop limit (TTL)")
122 #define TTL_LONGTEXT N_( \
123 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
124 "the multicast packets sent by the stream output (-1 = use operating " \
125 "system built-in default).")
127 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
128 #define RTCP_MUX_LONGTEXT N_( \
129 "This sends and receives RTCP packet multiplexed over the same port " \
132 #define PROTO_TEXT N_("Transport protocol")
133 #define PROTO_LONGTEXT N_( \
134 "This selects which transport protocol to use for RTP." )
136 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
137 #define SRTP_KEY_LONGTEXT N_( \
138 "RTP packets will be integrity-protected and ciphered "\
139 "with this Secure RTP master shared secret key.")
141 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
142 #define SRTP_SALT_LONGTEXT N_( \
143 "Secure RTP requires a (non-secret) master salt value.")
145 static const char *const ppsz_protos[] = {
146 "dccp", "sctp", "tcp", "udp", "udplite",
149 static const char *const ppsz_protocols[] = {
150 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
153 #define RFC3016_TEXT N_("MP4A LATM")
154 #define RFC3016_LONGTEXT N_( \
155 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
157 static int Open ( vlc_object_t * );
158 static void Close( vlc_object_t * );
160 #define SOUT_CFG_PREFIX "sout-rtp-"
161 #define MAX_EMPTY_BLOCKS 200
164 set_shortname( N_("RTP"))
165 set_description( N_("RTP stream output") )
166 set_capability( "sout stream", 0 )
167 add_shortcut( "rtp" )
168 set_category( CAT_SOUT )
169 set_subcategory( SUBCAT_SOUT_STREAM )
171 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
172 DEST_LONGTEXT, true )
173 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
175 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
177 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
180 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
181 NAME_LONGTEXT, true )
182 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
183 DESC_LONGTEXT, true )
184 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
186 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
187 EMAIL_LONGTEXT, true )
188 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
189 PHONE_LONGTEXT, true )
191 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
192 PROTO_LONGTEXT, false )
193 change_string_list( ppsz_protos, ppsz_protocols, NULL )
194 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
195 PORT_LONGTEXT, true )
196 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
197 PORT_AUDIO_LONGTEXT, true )
198 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
199 PORT_VIDEO_LONGTEXT, true )
201 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
203 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
204 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
207 add_string( SOUT_CFG_PREFIX "key", "", NULL,
208 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
209 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
210 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
213 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT,
214 RFC3016_LONGTEXT, false )
216 set_callbacks( Open, Close )
219 /*****************************************************************************
220 * Exported prototypes
221 *****************************************************************************/
222 static const char *const ppsz_sout_options[] = {
223 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
224 "sap", "description", "url", "email", "phone",
225 "proto", "rtcp-mux", "key", "salt",
229 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
230 static int Del ( sout_stream_t *, sout_stream_id_t * );
231 static int Send( sout_stream_t *, sout_stream_id_t *,
233 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
234 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
235 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
238 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
239 static void* ThreadSend( vlc_object_t *p_this );
240 static void *rtp_listen_thread( void * );
242 static void SDPHandleUrl( sout_stream_t *, const char * );
244 static int SapSetup( sout_stream_t *p_stream );
245 static int FileSetup( sout_stream_t *p_stream );
246 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
248 struct sout_stream_sys_t
252 vlc_mutex_t lock_sdp;
259 session_descriptor_t *p_session;
262 httpd_host_t *p_httpd_host;
263 httpd_file_t *p_httpd_file;
269 char *psz_destination;
270 uint32_t payload_bitmap;
272 uint16_t i_port_audio;
273 uint16_t i_port_video;
279 /* in case we do TS/PS over rtp */
281 sout_access_out_t *p_grab;
287 sout_stream_id_t **es;
290 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
292 typedef struct rtp_sink_t
298 struct sout_stream_id_t
302 sout_stream_t *p_stream;
305 uint8_t i_payload_type;
317 /* Packetizer specific fields */
320 srtp_session_t *srtp;
322 pf_rtp_packetizer_t pf_packetize;
325 vlc_mutex_t lock_sink;
328 rtsp_stream_id_t *rtsp_id;
334 block_fifo_t *p_fifo;
338 /*****************************************************************************
340 *****************************************************************************/
341 static int Open( vlc_object_t *p_this )
343 sout_stream_t *p_stream = (sout_stream_t*)p_this;
344 sout_instance_t *p_sout = p_stream->p_sout;
345 sout_stream_sys_t *p_sys = NULL;
346 config_chain_t *p_cfg = NULL;
350 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
351 ppsz_sout_options, p_stream->p_cfg );
353 p_sys = malloc( sizeof( sout_stream_sys_t ) );
357 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
359 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
360 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
361 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
362 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
364 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
366 msg_Err( p_stream, "audio and video RTP port must be distinct" );
367 free( p_sys->psz_destination );
372 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
374 if( !strcmp( p_cfg->psz_name, "sdp" )
375 && ( p_cfg->psz_value != NULL )
376 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
384 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
387 if( !strncasecmp( psz, "rtsp:", 5 ) )
393 /* Transport protocol */
394 p_sys->proto = IPPROTO_UDP;
395 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
397 if ((psz == NULL) || !strcasecmp (psz, "udp"))
398 (void)0; /* default */
400 if (!strcasecmp (psz, "dccp"))
402 p_sys->proto = IPPROTO_DCCP;
403 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
407 if (!strcasecmp (psz, "sctp"))
409 p_sys->proto = IPPROTO_TCP;
410 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
415 if (!strcasecmp (psz, "tcp"))
417 p_sys->proto = IPPROTO_TCP;
418 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
422 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
423 p_sys->proto = IPPROTO_UDPLITE;
425 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
428 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
430 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
432 msg_Err( p_stream, "missing destination and not in RTSP mode" );
437 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
438 if( p_sys->i_ttl == -1 )
440 /* Normally, we should let the default hop limit up to the core,
441 * but we have to know it to build our SDP properly, which is why
442 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
444 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
447 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
449 p_sys->payload_bitmap = 0;
453 p_sys->psz_sdp = NULL;
455 p_sys->b_export_sap = false;
456 p_sys->p_session = NULL;
457 p_sys->psz_sdp_file = NULL;
459 p_sys->p_httpd_host = NULL;
460 p_sys->p_httpd_file = NULL;
462 p_stream->p_sys = p_sys;
464 vlc_mutex_init( &p_sys->lock_sdp );
465 vlc_mutex_init( &p_sys->lock_es );
467 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
470 sout_stream_id_t *id;
472 /* Check muxer type */
473 if( strncasecmp( psz, "ps", 2 )
474 && strncasecmp( psz, "mpeg1", 5 )
475 && strncasecmp( psz, "ts", 2 ) )
477 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
479 vlc_mutex_destroy( &p_sys->lock_sdp );
480 vlc_mutex_destroy( &p_sys->lock_es );
481 free( p_sys->psz_destination );
486 p_sys->p_grab = GrabberCreate( p_stream );
487 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
490 if( p_sys->p_mux == NULL )
492 msg_Err( p_stream, "cannot create muxer" );
493 sout_AccessOutDelete( p_sys->p_grab );
494 vlc_mutex_destroy( &p_sys->lock_sdp );
495 vlc_mutex_destroy( &p_sys->lock_es );
496 free( p_sys->psz_destination );
501 id = Add( p_stream, NULL );
504 sout_MuxDelete( p_sys->p_mux );
505 sout_AccessOutDelete( p_sys->p_grab );
506 vlc_mutex_destroy( &p_sys->lock_sdp );
507 vlc_mutex_destroy( &p_sys->lock_es );
508 free( p_sys->psz_destination );
513 p_sys->packet = NULL;
515 p_stream->pf_add = MuxAdd;
516 p_stream->pf_del = MuxDel;
517 p_stream->pf_send = MuxSend;
522 p_sys->p_grab = NULL;
524 p_stream->pf_add = Add;
525 p_stream->pf_del = Del;
526 p_stream->pf_send = Send;
529 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
530 SDPHandleUrl( p_stream, "sap" );
532 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
535 config_chain_t *p_cfg;
537 SDPHandleUrl( p_stream, psz );
539 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
541 if( !strcmp( p_cfg->psz_name, "sdp" ) )
543 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
546 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
547 if( !strcmp( p_cfg->psz_value, psz ) )
550 SDPHandleUrl( p_stream, p_cfg->psz_value );
556 /* update p_sout->i_out_pace_nocontrol */
557 p_stream->p_sout->i_out_pace_nocontrol++;
562 /*****************************************************************************
564 *****************************************************************************/
565 static void Close( vlc_object_t * p_this )
567 sout_stream_t *p_stream = (sout_stream_t*)p_this;
568 sout_stream_sys_t *p_sys = p_stream->p_sys;
570 /* update p_sout->i_out_pace_nocontrol */
571 p_stream->p_sout->i_out_pace_nocontrol--;
575 assert( p_sys->i_es == 1 );
577 sout_MuxDelete( p_sys->p_mux );
578 Del( p_stream, p_sys->es[0] );
579 sout_AccessOutDelete( p_sys->p_grab );
583 block_Release( p_sys->packet );
585 if( p_sys->b_export_sap )
588 SapSetup( p_stream );
592 if( p_sys->rtsp != NULL )
593 RtspUnsetup( p_sys->rtsp );
595 vlc_mutex_destroy( &p_sys->lock_sdp );
596 vlc_mutex_destroy( &p_sys->lock_es );
598 if( p_sys->p_httpd_file )
599 httpd_FileDelete( p_sys->p_httpd_file );
601 if( p_sys->p_httpd_host )
602 httpd_HostDelete( p_sys->p_httpd_host );
604 free( p_sys->psz_sdp );
606 if( p_sys->psz_sdp_file != NULL )
609 unlink( p_sys->psz_sdp_file );
611 free( p_sys->psz_sdp_file );
613 free( p_sys->psz_destination );
617 /*****************************************************************************
619 *****************************************************************************/
620 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
622 sout_stream_sys_t *p_sys = p_stream->p_sys;
625 vlc_UrlParse( &url, psz_url, 0 );
626 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
628 if( p_sys->p_httpd_file )
630 msg_Err( p_stream, "you can use sdp=http:// only once" );
634 if( HttpSetup( p_stream, &url ) )
636 msg_Err( p_stream, "cannot export SDP as HTTP" );
639 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
641 if( p_sys->rtsp != NULL )
643 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
647 /* FIXME test if destination is multicast or no destination at all */
648 p_sys->rtsp = RtspSetup( p_stream, &url );
649 if( p_sys->rtsp == NULL )
650 msg_Err( p_stream, "cannot export SDP as RTSP" );
652 if( p_sys->p_mux != NULL )
654 sout_stream_id_t *id = p_sys->es[0];
655 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
656 p_sys->psz_destination, p_sys->i_ttl,
657 id->i_port, id->i_port + 1 );
660 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
661 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
663 p_sys->b_export_sap = true;
664 SapSetup( p_stream );
666 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
668 if( p_sys->psz_sdp_file != NULL )
670 msg_Err( p_stream, "you can use sdp=file:// only once" );
673 psz_url = &psz_url[5];
674 if( psz_url[0] == '/' && psz_url[1] == '/' )
676 p_sys->psz_sdp_file = strdup( psz_url );
677 if( p_sys->psz_sdp_file == NULL )
679 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
680 FileSetup( p_stream );
684 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
689 vlc_UrlClean( &url );
692 /*****************************************************************************
694 *****************************************************************************/
696 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
698 const sout_stream_sys_t *p_sys = p_stream->p_sys;
700 struct sockaddr_storage dst;
704 * When we have a fixed destination (typically when we do multicast),
705 * we need to put the actual port numbers in the SDP.
706 * When there is no fixed destination, we only support RTSP unicast
707 * on-demand setup, so we should rather let the clients decide which ports
709 * When there is both a fixed destination and RTSP unicast, we need to
710 * put port numbers used by the fixed destination, otherwise the SDP would
711 * become totally incorrect for multicast use. It should be noted that
712 * port numbers from SDP with RTSP are only "recommendation" from the
713 * server to the clients (per RFC2326), so only broken clients will fail
714 * to handle this properly. There is no solution but to use two differents
715 * output chain with two different RTSP URLs if you need to handle this
720 if( p_sys->psz_destination != NULL )
724 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
725 dstlen = sizeof( dst );
726 if( p_sys->es[0]->listen.fd != NULL )
727 getsockname( p_sys->es[0]->listen.fd[0],
728 (struct sockaddr *)&dst, &dstlen );
730 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
731 (struct sockaddr *)&dst, &dstlen );
737 /* Dummy destination address for RTSP */
738 memset (&dst, 0, sizeof( struct sockaddr_in ) );
739 dst.ss_family = AF_INET;
743 dstlen = sizeof( struct sockaddr_in );
746 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
747 NULL, 0, (struct sockaddr *)&dst, dstlen );
748 if( psz_sdp == NULL )
751 /* TODO: a=source-filter */
752 if( p_sys->rtcp_mux )
753 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
755 if( rtsp_url != NULL )
756 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
758 /* FIXME: locking?! */
759 for( i = 0; i < p_sys->i_es; i++ )
761 sout_stream_id_t *id = p_sys->es[i];
762 const char *mime_major; /* major MIME type */
763 const char *proto = "RTP/AVP"; /* protocol */
768 mime_major = "video";
771 mime_major = "audio";
780 if( rtsp_url == NULL )
782 switch( p_sys->proto )
787 proto = "TCP/RTP/AVP";
790 proto = "DCCP/RTP/AVP";
792 case IPPROTO_UDPLITE:
797 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
798 id->i_payload_type, false, id->i_bitrate,
799 id->psz_enc, id->i_clock_rate, id->i_channels,
802 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
803 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
805 if( rtsp_url != NULL )
807 assert( strlen( rtsp_url ) > 0 );
808 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
809 sdp_AddAttribute ( &psz_sdp, "control",
810 addslash ? "%s/trackID=%u" : "%strackID=%u",
815 if( id->listen.fd != NULL )
816 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
817 if( p_sys->proto == IPPROTO_DCCP )
818 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
819 "SC:RTP%c", toupper( mime_major[0] ) );
826 /*****************************************************************************
828 *****************************************************************************/
830 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
832 static const char hex[16] = "0123456789abcdef";
835 for( i = 0; i < i_data; i++ )
837 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
838 s[2*i+1] = hex[(p_data[i] )&0xf];
844 * Shrink the MTU down to a fixed packetization time (for audio).
847 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
849 /* Samples per second */
850 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
851 bytes *= id->i_channels;
854 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
855 id->i_mtu = 12 + spl;
856 else /* MTU is too small for ptime, align to a sample boundary */
857 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
860 /** Add an ES as a new RTP stream */
861 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
863 /* NOTE: As a special case, if we use a non-RTP
864 * mux (TS/PS), then p_fmt is NULL. */
865 sout_stream_sys_t *p_sys = p_stream->p_sys;
866 sout_stream_id_t *id;
869 if (0xffffffff == p_sys->payload_bitmap)
871 msg_Err (p_stream, "too many RTP elementary streams");
875 /* Choose the port */
880 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
881 i_port = p_sys->i_port_audio;
883 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
884 i_port = p_sys->i_port_video;
886 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
887 * only one thread that can *modify* the ES table. The ES lock protects
888 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
889 for (int i = 0; i_port && (i < p_sys->i_es); i++)
890 if (i_port == p_sys->es[i]->i_port)
891 i_port = 0; /* Port already in use! */
892 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
896 msg_Err (p_stream, "too many RTP elementary streams");
900 for (int i = 0; i_port && (i < p_sys->i_es); i++)
901 if (p == p_sys->es[i]->i_port)
905 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
908 vlc_object_attach( id, p_stream );
910 id->p_stream = p_stream;
912 /* Look for free dymanic payload type */
913 id->i_payload_type = 96;
914 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
915 id->i_payload_type++;
916 assert (id->i_payload_type < 128);
918 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
919 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
923 id->i_clock_rate = 90000; /* most common case for video */
928 id->i_cat = p_fmt->i_cat;
929 if( p_fmt->i_cat == AUDIO_ES )
931 id->i_clock_rate = p_fmt->audio.i_rate;
932 id->i_channels = p_fmt->audio.i_channels;
934 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
938 id->i_cat = VIDEO_ES;
942 id->i_mtu = config_GetInt( p_stream, "mtu" );
943 if( id->i_mtu <= 12 + 16 )
944 id->i_mtu = 576 - 20 - 8; /* pessimistic */
945 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
947 id->pf_packetize = NULL;
952 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
955 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
956 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
957 if (id->srtp == NULL)
963 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
964 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
969 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
972 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
976 vlc_mutex_init( &id->lock_sink );
981 id->listen.fd = NULL;
984 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
986 if( p_sys->psz_destination != NULL )
987 switch( p_sys->proto )
994 case VIDEO_ES: code = "RTPV"; break;
995 case AUDIO_ES: code = "RTPARTPV"; break;
996 case SPU_ES: code = "RTPTRTPV"; break;
997 default: code = "RTPORTPV"; break;
999 var_SetString (p_stream, "dccp-service", code);
1000 } /* fall through */
1002 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1003 p_sys->psz_destination, i_port,
1005 if( id->listen.fd == NULL )
1007 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1010 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1011 VLC_THREAD_PRIORITY_LOW ) )
1013 net_ListenClose( id->listen.fd );
1014 id->listen.fd = NULL;
1021 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1022 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1023 i_port, ttl, p_sys->proto );
1026 msg_Err( p_stream, "cannot create RTP socket" );
1029 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1035 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1037 if( psz == NULL ) /* Uho! */
1040 if( strncmp( psz, "ts", 2 ) == 0 )
1042 id->i_payload_type = 33;
1043 id->psz_enc = "MP2T";
1047 id->psz_enc = "MP2P";
1052 switch( p_fmt->i_codec )
1054 case VLC_CODEC_MULAW:
1055 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1056 id->i_payload_type = 0;
1057 id->psz_enc = "PCMU";
1058 id->pf_packetize = rtp_packetize_split;
1059 rtp_set_ptime (id, 20, 1);
1061 case VLC_CODEC_ALAW:
1062 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1063 id->i_payload_type = 8;
1064 id->psz_enc = "PCMA";
1065 id->pf_packetize = rtp_packetize_split;
1066 rtp_set_ptime (id, 20, 1);
1068 case VLC_CODEC_S16B:
1069 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1071 id->i_payload_type = 11;
1073 else if( p_fmt->audio.i_channels == 2 &&
1074 p_fmt->audio.i_rate == 44100 )
1076 id->i_payload_type = 10;
1078 id->psz_enc = "L16";
1079 id->pf_packetize = rtp_packetize_split;
1080 rtp_set_ptime (id, 20, 2);
1084 id->pf_packetize = rtp_packetize_split;
1085 rtp_set_ptime (id, 20, 1);
1087 case VLC_CODEC_MPGA:
1088 id->i_payload_type = 14;
1089 id->psz_enc = "MPA";
1090 id->i_clock_rate = 90000; /* not 44100 */
1091 id->pf_packetize = rtp_packetize_mpa;
1093 case VLC_CODEC_MPGV:
1094 id->i_payload_type = 32;
1095 id->psz_enc = "MPV";
1096 id->pf_packetize = rtp_packetize_mpv;
1098 case VLC_CODEC_ADPCM_G726:
1099 switch( p_fmt->i_bitrate / 1000 )
1102 id->psz_enc = "G726-16";
1103 id->pf_packetize = rtp_packetize_g726_16;
1106 id->psz_enc = "G726-24";
1107 id->pf_packetize = rtp_packetize_g726_24;
1110 id->psz_enc = "G726-32";
1111 id->pf_packetize = rtp_packetize_g726_32;
1114 id->psz_enc = "G726-40";
1115 id->pf_packetize = rtp_packetize_g726_40;
1118 msg_Err( p_stream, "cannot add this stream (unsupported "
1119 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1124 id->psz_enc = "ac3";
1125 id->pf_packetize = rtp_packetize_ac3;
1127 case VLC_CODEC_H263:
1128 id->psz_enc = "H263-1998";
1129 id->pf_packetize = rtp_packetize_h263;
1131 case VLC_CODEC_H264:
1132 id->psz_enc = "H264";
1133 id->pf_packetize = rtp_packetize_h264;
1134 id->psz_fmtp = NULL;
1136 if( p_fmt->i_extra > 0 )
1138 uint8_t *p_buffer = p_fmt->p_extra;
1139 int i_buffer = p_fmt->i_extra;
1140 char *p_64_sps = NULL;
1141 char *p_64_pps = NULL;
1144 while( i_buffer > 4 &&
1145 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1146 p_buffer[2] == 0 && p_buffer[3] == 1 )
1148 const int i_nal_type = p_buffer[4]&0x1f;
1152 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1155 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1157 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1159 /* we found another startcode */
1164 if( i_nal_type == 7 )
1166 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1167 sprintf_hexa( hexa, &p_buffer[5], 3 );
1169 else if( i_nal_type == 8 )
1171 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1177 if( p_64_sps && p_64_pps &&
1178 ( asprintf( &id->psz_fmtp,
1179 "packetization-mode=1;profile-level-id=%s;"
1180 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1181 p_64_pps ) == -1 ) )
1182 id->psz_fmtp = NULL;
1187 id->psz_fmtp = strdup( "packetization-mode=1" );
1190 case VLC_CODEC_MP4V:
1192 char hexa[2*p_fmt->i_extra +1];
1194 id->psz_enc = "MP4V-ES";
1195 id->pf_packetize = rtp_packetize_split;
1196 if( p_fmt->i_extra > 0 )
1198 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1199 if( asprintf( &id->psz_fmtp,
1200 "profile-level-id=3; config=%s;", hexa ) == -1 )
1201 id->psz_fmtp = NULL;
1205 case VLC_CODEC_MP4A:
1209 char hexa[2*p_fmt->i_extra +1];
1211 id->psz_enc = "mpeg4-generic";
1212 id->pf_packetize = rtp_packetize_mp4a;
1213 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1214 if( asprintf( &id->psz_fmtp,
1215 "streamtype=5; profile-level-id=15; "
1216 "mode=AAC-hbr; config=%s; SizeLength=13; "
1217 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1219 id->psz_fmtp = NULL;
1225 unsigned char config[6];
1226 unsigned int aacsrates[15] = {
1227 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1228 16000, 12000, 11025, 8000, 7350, 0, 0 };
1230 for( i = 0; i < 15; i++ )
1231 if( p_fmt->audio.i_rate == aacsrates[i] )
1237 config[3]=p_fmt->audio.i_channels<<4;
1241 id->psz_enc = "MP4A-LATM";
1242 id->pf_packetize = rtp_packetize_mp4a_latm;
1243 sprintf_hexa( hexa, config, 6 );
1244 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1245 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1246 id->psz_fmtp = NULL;
1250 case VLC_CODEC_AMR_NB:
1251 id->psz_enc = "AMR";
1252 id->psz_fmtp = strdup( "octet-align=1" );
1253 id->pf_packetize = rtp_packetize_amr;
1255 case VLC_CODEC_AMR_WB:
1256 id->psz_enc = "AMR-WB";
1257 id->psz_fmtp = strdup( "octet-align=1" );
1258 id->pf_packetize = rtp_packetize_amr;
1260 case VLC_CODEC_SPEEX:
1261 id->psz_enc = "SPEEX";
1262 id->pf_packetize = rtp_packetize_spx;
1264 case VLC_CODEC_ITU_T140:
1265 id->psz_enc = "t140" ;
1266 id->i_clock_rate = 1000;
1267 id->pf_packetize = rtp_packetize_t140;
1271 msg_Err( p_stream, "cannot add this stream (unsupported "
1272 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1275 if (id->i_payload_type >= 96)
1276 /* Mark dynamic payload type in use */
1277 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1279 #if 0 /* No payload formats sets this at the moment */
1282 cscov += 8 /* UDP */ + 12 /* RTP */;
1284 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1287 if( p_sys->rtsp != NULL )
1288 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1289 GetDWBE( id->ssrc ),
1290 p_sys->psz_destination,
1291 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1293 id->p_fifo = block_FifoNew();
1294 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1295 VLC_THREAD_PRIORITY_HIGHEST ) )
1298 /* Update p_sys context */
1299 vlc_mutex_lock( &p_sys->lock_es );
1300 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1301 vlc_mutex_unlock( &p_sys->lock_es );
1303 psz_sdp = SDPGenerate( p_stream, NULL );
1305 vlc_mutex_lock( &p_sys->lock_sdp );
1306 free( p_sys->psz_sdp );
1307 p_sys->psz_sdp = psz_sdp;
1308 vlc_mutex_unlock( &p_sys->lock_sdp );
1310 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1312 /* Update SDP (sap/file) */
1313 if( p_sys->b_export_sap ) SapSetup( p_stream );
1314 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1319 Del( p_stream, id );
1323 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1325 sout_stream_sys_t *p_sys = p_stream->p_sys;
1327 if( id->p_fifo != NULL )
1329 vlc_object_kill( id );
1330 vlc_thread_join( id );
1331 block_FifoRelease( id->p_fifo );
1334 vlc_mutex_lock( &p_sys->lock_es );
1335 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1336 vlc_mutex_unlock( &p_sys->lock_es );
1338 /* Release dynamic payload type */
1339 if (id->i_payload_type >= 96)
1340 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1342 free( id->psz_fmtp );
1345 RtspDelId( p_sys->rtsp, id->rtsp_id );
1347 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1348 if( id->listen.fd != NULL )
1350 vlc_cancel( id->listen.thread );
1351 vlc_join( id->listen.thread, NULL );
1352 net_ListenClose( id->listen.fd );
1355 if( id->srtp != NULL )
1356 srtp_destroy( id->srtp );
1359 vlc_mutex_destroy( &id->lock_sink );
1361 /* Update SDP (sap/file) */
1362 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1363 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1365 vlc_object_detach( id );
1366 vlc_object_release( id );
1370 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1375 assert( p_stream->p_sys->p_mux == NULL );
1378 while( p_buffer != NULL )
1380 p_next = p_buffer->p_next;
1381 if( id->pf_packetize( id, p_buffer ) )
1384 block_Release( p_buffer );
1390 /****************************************************************************
1392 ****************************************************************************/
1393 static int SapSetup( sout_stream_t *p_stream )
1395 sout_stream_sys_t *p_sys = p_stream->p_sys;
1396 sout_instance_t *p_sout = p_stream->p_sout;
1398 /* Remove the previous session */
1399 if( p_sys->p_session != NULL)
1401 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1402 p_sys->p_session = NULL;
1405 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1407 announce_method_t *p_method = sout_SAPMethod();
1408 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1410 p_sys->psz_destination,
1412 sout_MethodRelease( p_method );
1418 /****************************************************************************
1420 ****************************************************************************/
1421 static int FileSetup( sout_stream_t *p_stream )
1423 sout_stream_sys_t *p_sys = p_stream->p_sys;
1426 if( p_sys->psz_sdp == NULL )
1427 return VLC_EGENERIC; /* too early */
1429 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1431 msg_Err( p_stream, "cannot open file '%s' (%m)",
1432 p_sys->psz_sdp_file );
1433 return VLC_EGENERIC;
1436 fputs( p_sys->psz_sdp, f );
1442 /****************************************************************************
1444 ****************************************************************************/
1445 static int HttpCallback( httpd_file_sys_t *p_args,
1446 httpd_file_t *, uint8_t *p_request,
1447 uint8_t **pp_data, int *pi_data );
1449 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1451 sout_stream_sys_t *p_sys = p_stream->p_sys;
1453 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1454 url->i_port > 0 ? url->i_port : 80 );
1455 if( p_sys->p_httpd_host )
1457 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1458 url->psz_path ? url->psz_path : "/",
1461 HttpCallback, (void*)p_sys );
1463 if( p_sys->p_httpd_file == NULL )
1465 return VLC_EGENERIC;
1470 static int HttpCallback( httpd_file_sys_t *p_args,
1471 httpd_file_t *f, uint8_t *p_request,
1472 uint8_t **pp_data, int *pi_data )
1474 VLC_UNUSED(f); VLC_UNUSED(p_request);
1475 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1477 vlc_mutex_lock( &p_sys->lock_sdp );
1478 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1480 *pi_data = strlen( p_sys->psz_sdp );
1481 *pp_data = malloc( *pi_data );
1482 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1489 vlc_mutex_unlock( &p_sys->lock_sdp );
1494 /****************************************************************************
1496 ****************************************************************************/
1497 static void* ThreadSend( vlc_object_t *p_this )
1500 # define ECONNREFUSED WSAECONNREFUSED
1501 # define ENOPROTOOPT WSAENOPROTOOPT
1502 # define EHOSTUNREACH WSAEHOSTUNREACH
1503 # define ENETUNREACH WSAENETUNREACH
1504 # define ENETDOWN WSAENETDOWN
1505 # define ENOBUFS WSAENOBUFS
1506 # define EAGAIN WSAEWOULDBLOCK
1507 # define EWOULDBLOCK WSAEWOULDBLOCK
1509 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1510 unsigned i_caching = id->i_caching;
1514 block_t *out = block_FifoGet( id->p_fifo );
1515 block_cleanup_push (out);
1519 { /* FIXME: this is awfully inefficient */
1520 size_t len = out->i_buffer;
1521 out = block_Realloc( out, 0, len + 10 );
1522 out->i_buffer = len;
1524 int canc = vlc_savecancel ();
1525 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1526 vlc_restorecancel (canc);
1530 msg_Dbg( id, "SRTP sending error: %m" );
1531 block_Release( out );
1535 out->i_buffer = len;
1539 mwait (out->i_dts + i_caching);
1544 ssize_t len = out->i_buffer;
1545 int canc = vlc_savecancel ();
1547 vlc_mutex_lock( &id->lock_sink );
1548 unsigned deadc = 0; /* How many dead sockets? */
1549 int deadv[id->sinkc]; /* Dead sockets list */
1551 for( int i = 0; i < id->sinkc; i++ )
1554 if( !id->srtp ) /* FIXME: SRTCP support */
1556 SendRTCP( id->sinkv[i].rtcp, out );
1558 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1562 /* Soft errors (e.g. ICMP): */
1563 case ECONNREFUSED: /* Port unreachable */
1566 case EPROTO: /* Protocol unreachable */
1568 case EHOSTUNREACH: /* Host unreachable */
1569 case ENETUNREACH: /* Network unreachable */
1570 case ENETDOWN: /* Entire network down */
1571 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1572 /* Transient congestion: */
1573 case ENOMEM: /* out of socket buffers */
1576 #if (EAGAIN != EWOULDBLOCK)
1582 deadv[deadc++] = id->sinkv[i].rtp_fd;
1584 vlc_mutex_unlock( &id->lock_sink );
1585 block_Release( out );
1587 for( unsigned i = 0; i < deadc; i++ )
1589 msg_Dbg( id, "removing socket %d", deadv[i] );
1590 rtp_del_sink( id, deadv[i] );
1592 vlc_restorecancel (canc);
1598 /* This thread dequeues incoming connections (DCCP streaming) */
1599 static void *rtp_listen_thread( void *data )
1601 sout_stream_id_t *id = data;
1603 assert( id->listen.fd != NULL );
1607 int fd = net_Accept( id, id->listen.fd );
1610 int canc = vlc_savecancel( );
1611 rtp_add_sink( id, fd, true );
1612 vlc_restorecancel( canc );
1619 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1621 rtp_sink_t sink = { fd, NULL };
1622 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1624 if( sink.rtcp == NULL )
1625 msg_Err( id, "RTCP failed!" );
1627 vlc_mutex_lock( &id->lock_sink );
1628 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1629 vlc_mutex_unlock( &id->lock_sink );
1633 void rtp_del_sink( sout_stream_id_t *id, int fd )
1635 rtp_sink_t sink = { fd, NULL };
1637 /* NOTE: must be safe to use if fd is not included */
1638 vlc_mutex_lock( &id->lock_sink );
1639 for( int i = 0; i < id->sinkc; i++ )
1641 if (id->sinkv[i].rtp_fd == fd)
1643 sink = id->sinkv[i];
1644 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1648 vlc_mutex_unlock( &id->lock_sink );
1650 CloseRTCP( sink.rtcp );
1651 net_Close( sink.rtp_fd );
1654 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1656 /* This will return values for the next packet.
1657 * Accounting for caching would not be totally trivial. */
1658 return id->i_sequence;
1661 /* FIXME: this is pretty bad - if we remove and then insert an ES
1662 * the number will get unsynched from inside RTSP */
1663 unsigned rtp_get_num( const sout_stream_id_t *id )
1665 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1668 vlc_mutex_lock( &p_sys->lock_es );
1669 for( i = 0; i < p_sys->i_es; i++ )
1671 if( id == p_sys->es[i] )
1674 vlc_mutex_unlock( &p_sys->lock_es );
1680 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1681 int b_marker, int64_t i_pts )
1683 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
1685 out->p_buffer[0] = 0x80;
1686 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1687 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1688 out->p_buffer[3] = ( id->i_sequence )&0xff;
1689 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1690 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1691 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1692 out->p_buffer[7] = ( i_timestamp )&0xff;
1694 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1700 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1702 block_FifoPut( id->p_fifo, out );
1706 * @return configured max RTP payload size (including payload type-specific
1707 * headers, excluding RTP and transport headers)
1709 size_t rtp_mtu (const sout_stream_id_t *id)
1711 return id->i_mtu - 12;
1714 /*****************************************************************************
1716 *****************************************************************************/
1718 /** Add an ES to a non-RTP muxed stream */
1719 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1721 sout_input_t *p_input;
1722 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1723 assert( p_mux != NULL );
1725 p_input = sout_MuxAddStream( p_mux, p_fmt );
1726 if( p_input == NULL )
1728 msg_Err( p_stream, "cannot add this stream to the muxer" );
1732 return (sout_stream_id_t *)p_input;
1736 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1739 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1740 assert( p_mux != NULL );
1742 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1747 /** Remove an ES from a non-RTP muxed stream */
1748 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1750 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1751 assert( p_mux != NULL );
1753 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1758 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1759 const block_t *p_buffer )
1761 sout_stream_sys_t *p_sys = p_stream->p_sys;
1762 sout_stream_id_t *id = p_sys->es[0];
1764 int64_t i_dts = p_buffer->i_dts;
1766 uint8_t *p_data = p_buffer->p_buffer;
1767 size_t i_data = p_buffer->i_buffer;
1768 size_t i_max = id->i_mtu - 12;
1770 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1776 /* output complete packet */
1777 if( p_sys->packet &&
1778 p_sys->packet->i_buffer + i_data > i_max )
1780 rtp_packetize_send( id, p_sys->packet );
1781 p_sys->packet = NULL;
1784 if( p_sys->packet == NULL )
1786 /* allocate a new packet */
1787 p_sys->packet = block_New( p_stream, id->i_mtu );
1788 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1789 p_sys->packet->i_dts = i_dts;
1790 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1791 i_dts += p_sys->packet->i_length;
1794 i_size = __MIN( i_data,
1795 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1797 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1800 p_sys->packet->i_buffer += i_size;
1809 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1812 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1818 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1820 p_next = p_buffer->p_next;
1821 block_Release( p_buffer );
1829 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1831 sout_access_out_t *p_grab;
1833 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1834 if( p_grab == NULL )
1837 p_grab->p_module = NULL;
1838 p_grab->psz_access = strdup( "grab" );
1839 p_grab->p_cfg = NULL;
1840 p_grab->psz_path = strdup( "" );
1841 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1842 p_grab->pf_seek = NULL;
1843 p_grab->pf_write = AccessOutGrabberWrite;
1844 vlc_object_attach( p_grab, p_stream );