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>
48 # include <sys/types.h>
51 # include <sys/stat.h>
53 #ifdef HAVE_LINUX_DCCP_H
54 # include <linux/dccp.h>
57 # define IPPROTO_DCCP 33
59 #ifndef IPPROTO_UDPLITE
60 # define IPPROTO_UDPLITE 136
67 /*****************************************************************************
69 *****************************************************************************/
71 #define DEST_TEXT N_("Destination")
72 #define DEST_LONGTEXT N_( \
73 "This is the output URL that will be used." )
74 #define SDP_TEXT N_("SDP")
75 #define SDP_LONGTEXT N_( \
76 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
77 "session will be made available. You must use an url: http://location to " \
78 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
79 "for the SDP to be announced via SAP." )
80 #define SAP_TEXT N_("SAP announcing")
81 #define SAP_LONGTEXT N_("Announce this session with SAP.")
82 #define MUX_TEXT N_("Muxer")
83 #define MUX_LONGTEXT N_( \
84 "This allows you to specify the muxer used for the streaming output. " \
85 "Default is to use no muxer (standard RTP stream)." )
87 #define NAME_TEXT N_("Session name")
88 #define NAME_LONGTEXT N_( \
89 "This is the name of the session that will be announced in the SDP " \
90 "(Session Descriptor)." )
91 #define DESC_TEXT N_("Session description")
92 #define DESC_LONGTEXT N_( \
93 "This allows you to give a short description with details about the stream, " \
94 "that will be announced in the SDP (Session Descriptor)." )
95 #define URL_TEXT N_("Session URL")
96 #define URL_LONGTEXT N_( \
97 "This allows you to give an URL with more details about the stream " \
98 "(often the website of the streaming organization), that will " \
99 "be announced in the SDP (Session Descriptor)." )
100 #define EMAIL_TEXT N_("Session email")
101 #define EMAIL_LONGTEXT N_( \
102 "This allows you to give a contact mail address for the stream, that will " \
103 "be announced in the SDP (Session Descriptor)." )
104 #define PHONE_TEXT N_("Session phone number")
105 #define PHONE_LONGTEXT N_( \
106 "This allows you to give a contact telephone number for the stream, that will " \
107 "be announced in the SDP (Session Descriptor)." )
109 #define PORT_TEXT N_("Port")
110 #define PORT_LONGTEXT N_( \
111 "This allows you to specify the base port for the RTP streaming." )
112 #define PORT_AUDIO_TEXT N_("Audio port")
113 #define PORT_AUDIO_LONGTEXT N_( \
114 "This allows you to specify the default audio port for the RTP streaming." )
115 #define PORT_VIDEO_TEXT N_("Video port")
116 #define PORT_VIDEO_LONGTEXT N_( \
117 "This allows you to specify the default video port for the RTP streaming." )
119 #define TTL_TEXT N_("Hop limit (TTL)")
120 #define TTL_LONGTEXT N_( \
121 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
122 "the multicast packets sent by the stream output (-1 = use operating " \
123 "system built-in default).")
125 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
126 #define RTCP_MUX_LONGTEXT N_( \
127 "This sends and receives RTCP packet multiplexed over the same port " \
130 #define PROTO_TEXT N_("Transport protocol")
131 #define PROTO_LONGTEXT N_( \
132 "This selects which transport protocol to use for RTP." )
134 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
135 #define SRTP_KEY_LONGTEXT N_( \
136 "RTP packets will be integrity-protected and ciphered "\
137 "with this Secure RTP master shared secret key.")
139 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
140 #define SRTP_SALT_LONGTEXT N_( \
141 "Secure RTP requires a (non-secret) master salt value.")
143 static const char *const ppsz_protos[] = {
144 "dccp", "sctp", "tcp", "udp", "udplite",
147 static const char *const ppsz_protocols[] = {
148 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
151 #define RFC3016_TEXT N_("MP4A LATM")
152 #define RFC3016_LONGTEXT N_( \
153 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
155 static int Open ( vlc_object_t * );
156 static void Close( vlc_object_t * );
158 #define SOUT_CFG_PREFIX "sout-rtp-"
159 #define MAX_EMPTY_BLOCKS 200
162 set_shortname( N_("RTP"))
163 set_description( N_("RTP stream output") )
164 set_capability( "sout stream", 0 )
165 add_shortcut( "rtp" )
166 set_category( CAT_SOUT )
167 set_subcategory( SUBCAT_SOUT_STREAM )
169 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
170 DEST_LONGTEXT, true )
171 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
173 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
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,
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", 5004, NULL, PORT_TEXT,
193 PORT_LONGTEXT, true )
194 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
195 PORT_AUDIO_LONGTEXT, true )
196 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
197 PORT_VIDEO_LONGTEXT, true )
199 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
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;
254 session_descriptor_t *p_session;
257 httpd_host_t *p_httpd_host;
258 httpd_file_t *p_httpd_file;
264 char *psz_destination;
265 uint32_t payload_bitmap;
267 uint16_t i_port_audio;
268 uint16_t i_port_video;
274 /* in case we do TS/PS over rtp */
276 sout_access_out_t *p_grab;
282 sout_stream_id_t **es;
285 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
287 typedef struct rtp_sink_t
293 struct sout_stream_id_t
297 sout_stream_t *p_stream;
300 uint8_t i_payload_type;
312 /* Packetizer specific fields */
314 srtp_session_t *srtp;
315 pf_rtp_packetizer_t pf_packetize;
318 vlc_mutex_t lock_sink;
321 rtsp_stream_id_t *rtsp_id;
324 block_fifo_t *p_fifo;
328 /*****************************************************************************
330 *****************************************************************************/
331 static int Open( vlc_object_t *p_this )
333 sout_stream_t *p_stream = (sout_stream_t*)p_this;
334 sout_instance_t *p_sout = p_stream->p_sout;
335 sout_stream_sys_t *p_sys = NULL;
336 config_chain_t *p_cfg = NULL;
340 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
341 ppsz_sout_options, p_stream->p_cfg );
343 p_sys = malloc( sizeof( sout_stream_sys_t ) );
347 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
349 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
350 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
351 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
352 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
354 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
356 msg_Err( p_stream, "audio and video RTP port must be distinct" );
357 free( p_sys->psz_destination );
362 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
364 if( !strcmp( p_cfg->psz_name, "sdp" )
365 && ( p_cfg->psz_value != NULL )
366 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
374 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
377 if( !strncasecmp( psz, "rtsp:", 5 ) )
383 /* Transport protocol */
384 p_sys->proto = IPPROTO_UDP;
385 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
387 if ((psz == NULL) || !strcasecmp (psz, "udp"))
388 (void)0; /* default */
390 if (!strcasecmp (psz, "dccp"))
392 p_sys->proto = IPPROTO_DCCP;
393 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
397 if (!strcasecmp (psz, "sctp"))
399 p_sys->proto = IPPROTO_TCP;
400 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
405 if (!strcasecmp (psz, "tcp"))
407 p_sys->proto = IPPROTO_TCP;
408 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
412 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
413 p_sys->proto = IPPROTO_UDPLITE;
415 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
418 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
420 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
422 msg_Err( p_stream, "missing destination and not in RTSP mode" );
427 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
428 if( p_sys->i_ttl == -1 )
430 /* Normally, we should let the default hop limit up to the core,
431 * but we have to know it to build our SDP properly, which is why
432 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
434 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
437 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
439 p_sys->payload_bitmap = 0;
443 p_sys->psz_sdp = NULL;
445 p_sys->b_export_sap = false;
446 p_sys->p_session = NULL;
447 p_sys->psz_sdp_file = NULL;
449 p_sys->p_httpd_host = NULL;
450 p_sys->p_httpd_file = NULL;
452 p_stream->p_sys = p_sys;
454 vlc_mutex_init( &p_sys->lock_sdp );
455 vlc_mutex_init( &p_sys->lock_es );
457 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
460 sout_stream_id_t *id;
462 /* Check muxer type */
463 if( strncasecmp( psz, "ps", 2 )
464 && strncasecmp( psz, "mpeg1", 5 )
465 && strncasecmp( psz, "ts", 2 ) )
467 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
469 vlc_mutex_destroy( &p_sys->lock_sdp );
470 vlc_mutex_destroy( &p_sys->lock_es );
471 free( p_sys->psz_destination );
476 p_sys->p_grab = GrabberCreate( p_stream );
477 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
480 if( p_sys->p_mux == NULL )
482 msg_Err( p_stream, "cannot create muxer" );
483 sout_AccessOutDelete( p_sys->p_grab );
484 vlc_mutex_destroy( &p_sys->lock_sdp );
485 vlc_mutex_destroy( &p_sys->lock_es );
486 free( p_sys->psz_destination );
491 id = Add( p_stream, NULL );
494 sout_MuxDelete( p_sys->p_mux );
495 sout_AccessOutDelete( p_sys->p_grab );
496 vlc_mutex_destroy( &p_sys->lock_sdp );
497 vlc_mutex_destroy( &p_sys->lock_es );
498 free( p_sys->psz_destination );
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 );
567 sout_MuxDelete( p_sys->p_mux );
568 Del( p_stream, p_sys->es[0] );
569 sout_AccessOutDelete( p_sys->p_grab );
573 block_Release( p_sys->packet );
575 if( p_sys->b_export_sap )
578 SapSetup( p_stream );
582 if( p_sys->rtsp != NULL )
583 RtspUnsetup( p_sys->rtsp );
585 vlc_mutex_destroy( &p_sys->lock_sdp );
586 vlc_mutex_destroy( &p_sys->lock_es );
588 if( p_sys->p_httpd_file )
589 httpd_FileDelete( p_sys->p_httpd_file );
591 if( p_sys->p_httpd_host )
592 httpd_HostDelete( p_sys->p_httpd_host );
594 free( p_sys->psz_sdp );
596 if( p_sys->psz_sdp_file != NULL )
599 unlink( p_sys->psz_sdp_file );
601 free( p_sys->psz_sdp_file );
603 free( p_sys->psz_destination );
607 /*****************************************************************************
609 *****************************************************************************/
610 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
612 sout_stream_sys_t *p_sys = p_stream->p_sys;
615 vlc_UrlParse( &url, psz_url, 0 );
616 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
618 if( p_sys->p_httpd_file )
620 msg_Err( p_stream, "you can use sdp=http:// only once" );
624 if( HttpSetup( p_stream, &url ) )
626 msg_Err( p_stream, "cannot export SDP as HTTP" );
629 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
631 if( p_sys->rtsp != NULL )
633 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
637 /* FIXME test if destination is multicast or no destination at all */
638 p_sys->rtsp = RtspSetup( p_stream, &url );
639 if( p_sys->rtsp == NULL )
640 msg_Err( p_stream, "cannot export SDP as RTSP" );
642 if( p_sys->p_mux != NULL )
644 sout_stream_id_t *id = p_sys->es[0];
645 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
646 p_sys->psz_destination, p_sys->i_ttl,
647 id->i_port, id->i_port + 1 );
650 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
651 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
653 p_sys->b_export_sap = true;
654 SapSetup( p_stream );
656 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
658 if( p_sys->psz_sdp_file != NULL )
660 msg_Err( p_stream, "you can use sdp=file:// only once" );
663 psz_url = &psz_url[5];
664 if( psz_url[0] == '/' && psz_url[1] == '/' )
666 p_sys->psz_sdp_file = strdup( psz_url );
667 if( p_sys->psz_sdp_file == NULL )
669 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
670 FileSetup( p_stream );
674 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
679 vlc_UrlClean( &url );
682 /*****************************************************************************
684 *****************************************************************************/
686 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
688 const sout_stream_sys_t *p_sys = p_stream->p_sys;
690 struct sockaddr_storage dst;
694 * When we have a fixed destination (typically when we do multicast),
695 * we need to put the actual port numbers in the SDP.
696 * When there is no fixed destination, we only support RTSP unicast
697 * on-demand setup, so we should rather let the clients decide which ports
699 * When there is both a fixed destination and RTSP unicast, we need to
700 * put port numbers used by the fixed destination, otherwise the SDP would
701 * become totally incorrect for multicast use. It should be noted that
702 * port numbers from SDP with RTSP are only "recommendation" from the
703 * server to the clients (per RFC2326), so only broken clients will fail
704 * to handle this properly. There is no solution but to use two differents
705 * output chain with two different RTSP URLs if you need to handle this
710 if( p_sys->psz_destination != NULL )
714 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
715 dstlen = sizeof( dst );
716 if( p_sys->es[0]->listen_fd != NULL )
717 getsockname( p_sys->es[0]->listen_fd[0],
718 (struct sockaddr *)&dst, &dstlen );
720 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
721 (struct sockaddr *)&dst, &dstlen );
727 /* Dummy destination address for RTSP */
728 memset (&dst, 0, sizeof( struct sockaddr_in ) );
729 dst.ss_family = AF_INET;
733 dstlen = sizeof( struct sockaddr_in );
736 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
737 NULL, 0, (struct sockaddr *)&dst, dstlen );
738 if( psz_sdp == NULL )
741 /* TODO: a=source-filter */
742 if( p_sys->rtcp_mux )
743 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
745 if( rtsp_url != NULL )
746 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
748 /* FIXME: locking?! */
749 for( i = 0; i < p_sys->i_es; i++ )
751 sout_stream_id_t *id = p_sys->es[i];
752 const char *mime_major; /* major MIME type */
753 const char *proto = "RTP/AVP"; /* protocol */
758 mime_major = "video";
761 mime_major = "audio";
770 if( rtsp_url == NULL )
772 switch( p_sys->proto )
777 proto = "TCP/RTP/AVP";
780 proto = "DCCP/RTP/AVP";
782 case IPPROTO_UDPLITE:
787 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
788 id->i_payload_type, false, id->i_bitrate,
789 id->psz_enc, id->i_clock_rate, id->i_channels,
792 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
793 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
795 if( rtsp_url != NULL )
797 assert( strlen( rtsp_url ) > 0 );
798 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
799 sdp_AddAttribute ( &psz_sdp, "control",
800 addslash ? "%s/trackID=%u" : "%strackID=%u",
805 if( id->listen_fd != NULL )
806 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
807 if( p_sys->proto == IPPROTO_DCCP )
808 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
809 "SC:RTP%c", toupper( mime_major[0] ) );
816 /*****************************************************************************
818 *****************************************************************************/
820 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
822 static const char hex[16] = "0123456789abcdef";
825 for( i = 0; i < i_data; i++ )
827 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
828 s[2*i+1] = hex[(p_data[i] )&0xf];
834 * Shrink the MTU down to a fixed packetization time (for audio).
837 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
839 /* Samples per second */
840 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
841 bytes *= id->i_channels;
844 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
845 id->i_mtu = 12 + spl;
846 else /* MTU is too small for ptime, align to a sample boundary */
847 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
850 /** Add an ES as a new RTP stream */
851 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
853 /* NOTE: As a special case, if we use a non-RTP
854 * mux (TS/PS), then p_fmt is NULL. */
855 sout_stream_sys_t *p_sys = p_stream->p_sys;
856 sout_stream_id_t *id;
860 if (0xffffffff == p_sys->payload_bitmap)
862 msg_Err (p_stream, "too many RTP elementary streams");
866 /* Choose the port */
871 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
872 i_port = p_sys->i_port_audio;
874 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
875 i_port = p_sys->i_port_video;
877 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
878 * only one thread that can *modify* the ES table. The ES lock protects
879 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
880 for (int i = 0; i_port && (i < p_sys->i_es); i++)
881 if (i_port == p_sys->es[i]->i_port)
882 i_port = 0; /* Port already in use! */
883 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
887 msg_Err (p_stream, "too many RTP elementary streams");
891 for (int i = 0; i_port && (i < p_sys->i_es); i++)
892 if (p == p_sys->es[i]->i_port)
896 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
899 vlc_object_attach( id, p_stream );
901 id->p_stream = p_stream;
903 /* Look for free dymanic payload type */
904 id->i_payload_type = 96;
905 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
906 id->i_payload_type++;
907 assert (id->i_payload_type < 128);
909 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
910 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
914 id->i_clock_rate = 90000; /* most common case for video */
919 id->i_cat = p_fmt->i_cat;
920 if( p_fmt->i_cat == AUDIO_ES )
922 id->i_clock_rate = p_fmt->audio.i_rate;
923 id->i_channels = p_fmt->audio.i_channels;
925 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
929 id->i_cat = VIDEO_ES;
933 id->i_mtu = config_GetInt( p_stream, "mtu" );
934 if( id->i_mtu <= 12 + 16 )
935 id->i_mtu = 576 - 20 - 8; /* pessimistic */
936 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
939 id->pf_packetize = NULL;
941 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
944 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
945 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
946 if (id->srtp == NULL)
952 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
953 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
958 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
961 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
964 vlc_mutex_init( &id->lock_sink );
969 id->listen_fd = NULL;
972 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
974 if( p_sys->psz_destination != NULL )
975 switch( p_sys->proto )
982 case VIDEO_ES: code = "RTPV"; break;
983 case AUDIO_ES: code = "RTPARTPV"; break;
984 case SPU_ES: code = "RTPTRTPV"; break;
985 default: code = "RTPORTPV"; break;
987 var_SetString (p_stream, "dccp-service", code);
990 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
991 p_sys->psz_destination, i_port,
993 if( id->listen_fd == NULL )
995 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1002 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1003 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1004 i_port, ttl, p_sys->proto );
1007 msg_Err( p_stream, "cannot create RTP socket" );
1010 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1016 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1018 if( psz == NULL ) /* Uho! */
1021 if( strncmp( psz, "ts", 2 ) == 0 )
1023 id->i_payload_type = 33;
1024 id->psz_enc = "MP2T";
1028 id->psz_enc = "MP2P";
1033 switch( p_fmt->i_codec )
1035 case VLC_CODEC_MULAW:
1036 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1037 id->i_payload_type = 0;
1038 id->psz_enc = "PCMU";
1039 id->pf_packetize = rtp_packetize_split;
1040 rtp_set_ptime (id, 20, 1);
1042 case VLC_CODEC_ALAW:
1043 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1044 id->i_payload_type = 8;
1045 id->psz_enc = "PCMA";
1046 id->pf_packetize = rtp_packetize_split;
1047 rtp_set_ptime (id, 20, 1);
1049 case VLC_CODEC_S16B:
1050 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1052 id->i_payload_type = 11;
1054 else if( p_fmt->audio.i_channels == 2 &&
1055 p_fmt->audio.i_rate == 44100 )
1057 id->i_payload_type = 10;
1059 id->psz_enc = "L16";
1060 id->pf_packetize = rtp_packetize_split;
1061 rtp_set_ptime (id, 20, 2);
1065 id->pf_packetize = rtp_packetize_split;
1066 rtp_set_ptime (id, 20, 1);
1068 case VLC_CODEC_MPGA:
1069 id->i_payload_type = 14;
1070 id->psz_enc = "MPA";
1071 id->i_clock_rate = 90000; /* not 44100 */
1072 id->pf_packetize = rtp_packetize_mpa;
1074 case VLC_CODEC_MPGV:
1075 id->i_payload_type = 32;
1076 id->psz_enc = "MPV";
1077 id->pf_packetize = rtp_packetize_mpv;
1079 case VLC_CODEC_ADPCM_G726:
1080 switch( p_fmt->i_bitrate / 1000 )
1083 id->psz_enc = "G726-16";
1084 id->pf_packetize = rtp_packetize_g726_16;
1087 id->psz_enc = "G726-24";
1088 id->pf_packetize = rtp_packetize_g726_24;
1091 id->psz_enc = "G726-32";
1092 id->pf_packetize = rtp_packetize_g726_32;
1095 id->psz_enc = "G726-40";
1096 id->pf_packetize = rtp_packetize_g726_40;
1099 msg_Err( p_stream, "cannot add this stream (unsupported "
1100 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1105 id->psz_enc = "ac3";
1106 id->pf_packetize = rtp_packetize_ac3;
1108 case VLC_CODEC_H263:
1109 id->psz_enc = "H263-1998";
1110 id->pf_packetize = rtp_packetize_h263;
1112 case VLC_CODEC_H264:
1113 id->psz_enc = "H264";
1114 id->pf_packetize = rtp_packetize_h264;
1115 id->psz_fmtp = NULL;
1117 if( p_fmt->i_extra > 0 )
1119 uint8_t *p_buffer = p_fmt->p_extra;
1120 int i_buffer = p_fmt->i_extra;
1121 char *p_64_sps = NULL;
1122 char *p_64_pps = NULL;
1125 while( i_buffer > 4 &&
1126 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1127 p_buffer[2] == 0 && p_buffer[3] == 1 )
1129 const int i_nal_type = p_buffer[4]&0x1f;
1133 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1136 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1138 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1140 /* we found another startcode */
1145 if( i_nal_type == 7 )
1147 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1148 sprintf_hexa( hexa, &p_buffer[5], 3 );
1150 else if( i_nal_type == 8 )
1152 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1158 if( p_64_sps && p_64_pps &&
1159 ( asprintf( &id->psz_fmtp,
1160 "packetization-mode=1;profile-level-id=%s;"
1161 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1162 p_64_pps ) == -1 ) )
1163 id->psz_fmtp = NULL;
1168 id->psz_fmtp = strdup( "packetization-mode=1" );
1171 case VLC_CODEC_MP4V:
1173 char hexa[2*p_fmt->i_extra +1];
1175 id->psz_enc = "MP4V-ES";
1176 id->pf_packetize = rtp_packetize_split;
1177 if( p_fmt->i_extra > 0 )
1179 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1180 if( asprintf( &id->psz_fmtp,
1181 "profile-level-id=3; config=%s;", hexa ) == -1 )
1182 id->psz_fmtp = NULL;
1186 case VLC_CODEC_MP4A:
1190 char hexa[2*p_fmt->i_extra +1];
1192 id->psz_enc = "mpeg4-generic";
1193 id->pf_packetize = rtp_packetize_mp4a;
1194 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1195 if( asprintf( &id->psz_fmtp,
1196 "streamtype=5; profile-level-id=15; "
1197 "mode=AAC-hbr; config=%s; SizeLength=13; "
1198 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1200 id->psz_fmtp = NULL;
1206 unsigned char config[6];
1207 unsigned int aacsrates[15] = {
1208 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1209 16000, 12000, 11025, 8000, 7350, 0, 0 };
1211 for( i = 0; i < 15; i++ )
1212 if( p_fmt->audio.i_rate == aacsrates[i] )
1218 config[3]=p_fmt->audio.i_channels<<4;
1222 id->psz_enc = "MP4A-LATM";
1223 id->pf_packetize = rtp_packetize_mp4a_latm;
1224 sprintf_hexa( hexa, config, 6 );
1225 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1226 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1227 id->psz_fmtp = NULL;
1231 case VLC_CODEC_AMR_NB:
1232 id->psz_enc = "AMR";
1233 id->psz_fmtp = strdup( "octet-align=1" );
1234 id->pf_packetize = rtp_packetize_amr;
1236 case VLC_CODEC_AMR_WB:
1237 id->psz_enc = "AMR-WB";
1238 id->psz_fmtp = strdup( "octet-align=1" );
1239 id->pf_packetize = rtp_packetize_amr;
1241 case VLC_CODEC_SPEEX:
1242 id->psz_enc = "SPEEX";
1243 id->pf_packetize = rtp_packetize_spx;
1245 case VLC_CODEC_ITU_T140:
1246 id->psz_enc = "t140" ;
1247 id->i_clock_rate = 1000;
1248 id->pf_packetize = rtp_packetize_t140;
1252 msg_Err( p_stream, "cannot add this stream (unsupported "
1253 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1256 if (id->i_payload_type >= 96)
1257 /* Mark dynamic payload type in use */
1258 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1260 #if 0 /* No payload formats sets this at the moment */
1262 cscov += 8 /* UDP */ + 12 /* RTP */;
1264 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1267 if( p_sys->rtsp != NULL )
1268 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1269 GetDWBE( id->ssrc ),
1270 p_sys->psz_destination,
1271 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1273 id->p_fifo = block_FifoNew();
1274 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1275 VLC_THREAD_PRIORITY_HIGHEST ) )
1278 /* Update p_sys context */
1279 vlc_mutex_lock( &p_sys->lock_es );
1280 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1281 vlc_mutex_unlock( &p_sys->lock_es );
1283 psz_sdp = SDPGenerate( p_stream, NULL );
1285 vlc_mutex_lock( &p_sys->lock_sdp );
1286 free( p_sys->psz_sdp );
1287 p_sys->psz_sdp = psz_sdp;
1288 vlc_mutex_unlock( &p_sys->lock_sdp );
1290 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1292 /* Update SDP (sap/file) */
1293 if( p_sys->b_export_sap ) SapSetup( p_stream );
1294 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1299 Del( p_stream, id );
1303 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1305 sout_stream_sys_t *p_sys = p_stream->p_sys;
1307 if( id->p_fifo != NULL )
1309 vlc_object_kill( id );
1310 vlc_thread_join( id );
1311 block_FifoRelease( id->p_fifo );
1314 vlc_mutex_lock( &p_sys->lock_es );
1315 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1316 vlc_mutex_unlock( &p_sys->lock_es );
1318 /* Release dynamic payload type */
1319 if (id->i_payload_type >= 96)
1320 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1322 free( id->psz_fmtp );
1325 RtspDelId( p_sys->rtsp, id->rtsp_id );
1327 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1328 if( id->listen_fd != NULL )
1329 net_ListenClose( id->listen_fd );
1330 if( id->srtp != NULL )
1331 srtp_destroy( id->srtp );
1333 vlc_mutex_destroy( &id->lock_sink );
1335 /* Update SDP (sap/file) */
1336 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1337 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1339 vlc_object_detach( id );
1340 vlc_object_release( id );
1344 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1349 assert( p_stream->p_sys->p_mux == NULL );
1352 while( p_buffer != NULL )
1354 p_next = p_buffer->p_next;
1355 if( id->pf_packetize( id, p_buffer ) )
1358 block_Release( p_buffer );
1364 /****************************************************************************
1366 ****************************************************************************/
1367 static int SapSetup( sout_stream_t *p_stream )
1369 sout_stream_sys_t *p_sys = p_stream->p_sys;
1370 sout_instance_t *p_sout = p_stream->p_sout;
1372 /* Remove the previous session */
1373 if( p_sys->p_session != NULL)
1375 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1376 p_sys->p_session = NULL;
1379 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1381 announce_method_t *p_method = sout_SAPMethod();
1382 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1384 p_sys->psz_destination,
1386 sout_MethodRelease( p_method );
1392 /****************************************************************************
1394 ****************************************************************************/
1395 static int FileSetup( sout_stream_t *p_stream )
1397 sout_stream_sys_t *p_sys = p_stream->p_sys;
1400 if( p_sys->psz_sdp == NULL )
1401 return VLC_EGENERIC; /* too early */
1403 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1405 msg_Err( p_stream, "cannot open file '%s' (%m)",
1406 p_sys->psz_sdp_file );
1407 return VLC_EGENERIC;
1410 fputs( p_sys->psz_sdp, f );
1416 /****************************************************************************
1418 ****************************************************************************/
1419 static int HttpCallback( httpd_file_sys_t *p_args,
1420 httpd_file_t *, uint8_t *p_request,
1421 uint8_t **pp_data, int *pi_data );
1423 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1425 sout_stream_sys_t *p_sys = p_stream->p_sys;
1427 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1428 url->i_port > 0 ? url->i_port : 80 );
1429 if( p_sys->p_httpd_host )
1431 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1432 url->psz_path ? url->psz_path : "/",
1435 HttpCallback, (void*)p_sys );
1437 if( p_sys->p_httpd_file == NULL )
1439 return VLC_EGENERIC;
1444 static int HttpCallback( httpd_file_sys_t *p_args,
1445 httpd_file_t *f, uint8_t *p_request,
1446 uint8_t **pp_data, int *pi_data )
1448 VLC_UNUSED(f); VLC_UNUSED(p_request);
1449 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1451 vlc_mutex_lock( &p_sys->lock_sdp );
1452 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1454 *pi_data = strlen( p_sys->psz_sdp );
1455 *pp_data = malloc( *pi_data );
1456 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1463 vlc_mutex_unlock( &p_sys->lock_sdp );
1468 /****************************************************************************
1470 ****************************************************************************/
1471 static void* ThreadSend( vlc_object_t *p_this )
1474 # define ECONNREFUSED WSAECONNREFUSED
1475 # define ENOPROTOOPT WSAENOPROTOOPT
1476 # define EHOSTUNREACH WSAEHOSTUNREACH
1477 # define ENETUNREACH WSAENETUNREACH
1478 # define ENETDOWN WSAENETDOWN
1479 # define ENOBUFS WSAENOBUFS
1480 # define EAGAIN WSAEWOULDBLOCK
1481 # define EWOULDBLOCK WSAEWOULDBLOCK
1483 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1484 unsigned i_caching = id->i_caching;
1488 block_t *out = block_FifoGet( id->p_fifo );
1489 block_cleanup_push (out);
1492 { /* FIXME: this is awfully inefficient */
1493 size_t len = out->i_buffer;
1494 out = block_Realloc( out, 0, len + 10 );
1495 out->i_buffer = len;
1497 int canc = vlc_savecancel ();
1498 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1499 vlc_restorecancel (canc);
1503 msg_Dbg( id, "SRTP sending error: %m" );
1504 block_Release( out );
1508 out->i_buffer = len;
1512 mwait (out->i_dts + i_caching);
1517 ssize_t len = out->i_buffer;
1518 int canc = vlc_savecancel ();
1520 vlc_mutex_lock( &id->lock_sink );
1521 unsigned deadc = 0; /* How many dead sockets? */
1522 int deadv[id->sinkc]; /* Dead sockets list */
1524 for( int i = 0; i < id->sinkc; i++ )
1526 if( !id->srtp ) /* FIXME: SRTCP support */
1527 SendRTCP( id->sinkv[i].rtcp, out );
1529 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1533 /* Soft errors (e.g. ICMP): */
1534 case ECONNREFUSED: /* Port unreachable */
1537 case EPROTO: /* Protocol unreachable */
1539 case EHOSTUNREACH: /* Host unreachable */
1540 case ENETUNREACH: /* Network unreachable */
1541 case ENETDOWN: /* Entire network down */
1542 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1543 /* Transient congestion: */
1544 case ENOMEM: /* out of socket buffers */
1547 #if (EAGAIN != EWOULDBLOCK)
1553 deadv[deadc++] = id->sinkv[i].rtp_fd;
1555 vlc_mutex_unlock( &id->lock_sink );
1556 block_Release( out );
1558 for( unsigned i = 0; i < deadc; i++ )
1560 msg_Dbg( id, "removing socket %d", deadv[i] );
1561 rtp_del_sink( id, deadv[i] );
1564 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1565 while( id->listen_fd != NULL )
1567 int fd = net_Accept( id, id->listen_fd, 0 );
1570 msg_Dbg( id, "adding socket %d", fd );
1571 rtp_add_sink( id, fd, true );
1573 vlc_restorecancel (canc);
1578 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1580 rtp_sink_t sink = { fd, NULL };
1581 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1583 if( sink.rtcp == NULL )
1584 msg_Err( id, "RTCP failed!" );
1586 vlc_mutex_lock( &id->lock_sink );
1587 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1588 vlc_mutex_unlock( &id->lock_sink );
1592 void rtp_del_sink( sout_stream_id_t *id, int fd )
1594 rtp_sink_t sink = { fd, NULL };
1596 /* NOTE: must be safe to use if fd is not included */
1597 vlc_mutex_lock( &id->lock_sink );
1598 for( int i = 0; i < id->sinkc; i++ )
1600 if (id->sinkv[i].rtp_fd == fd)
1602 sink = id->sinkv[i];
1603 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1607 vlc_mutex_unlock( &id->lock_sink );
1609 CloseRTCP( sink.rtcp );
1610 net_Close( sink.rtp_fd );
1613 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1615 /* This will return values for the next packet.
1616 * Accounting for caching would not be totally trivial. */
1617 return id->i_sequence;
1620 /* FIXME: this is pretty bad - if we remove and then insert an ES
1621 * the number will get unsynched from inside RTSP */
1622 unsigned rtp_get_num( const sout_stream_id_t *id )
1624 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1627 vlc_mutex_lock( &p_sys->lock_es );
1628 for( i = 0; i < p_sys->i_es; i++ )
1630 if( id == p_sys->es[i] )
1633 vlc_mutex_unlock( &p_sys->lock_es );
1639 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1640 int b_marker, int64_t i_pts )
1642 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
1644 out->p_buffer[0] = 0x80;
1645 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1646 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1647 out->p_buffer[3] = ( id->i_sequence )&0xff;
1648 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1649 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1650 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1651 out->p_buffer[7] = ( i_timestamp )&0xff;
1653 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1659 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1661 block_FifoPut( id->p_fifo, out );
1665 * @return configured max RTP payload size (including payload type-specific
1666 * headers, excluding RTP and transport headers)
1668 size_t rtp_mtu (const sout_stream_id_t *id)
1670 return id->i_mtu - 12;
1673 /*****************************************************************************
1675 *****************************************************************************/
1677 /** Add an ES to a non-RTP muxed stream */
1678 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1680 sout_input_t *p_input;
1681 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1682 assert( p_mux != NULL );
1684 p_input = sout_MuxAddStream( p_mux, p_fmt );
1685 if( p_input == NULL )
1687 msg_Err( p_stream, "cannot add this stream to the muxer" );
1691 return (sout_stream_id_t *)p_input;
1695 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1698 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1699 assert( p_mux != NULL );
1701 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1706 /** Remove an ES from a non-RTP muxed stream */
1707 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1709 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1710 assert( p_mux != NULL );
1712 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1717 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1718 const block_t *p_buffer )
1720 sout_stream_sys_t *p_sys = p_stream->p_sys;
1721 sout_stream_id_t *id = p_sys->es[0];
1723 int64_t i_dts = p_buffer->i_dts;
1725 uint8_t *p_data = p_buffer->p_buffer;
1726 size_t i_data = p_buffer->i_buffer;
1727 size_t i_max = id->i_mtu - 12;
1729 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1735 /* output complete packet */
1736 if( p_sys->packet &&
1737 p_sys->packet->i_buffer + i_data > i_max )
1739 rtp_packetize_send( id, p_sys->packet );
1740 p_sys->packet = NULL;
1743 if( p_sys->packet == NULL )
1745 /* allocate a new packet */
1746 p_sys->packet = block_New( p_stream, id->i_mtu );
1747 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1748 p_sys->packet->i_dts = i_dts;
1749 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1750 i_dts += p_sys->packet->i_length;
1753 i_size = __MIN( i_data,
1754 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1756 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1759 p_sys->packet->i_buffer += i_size;
1768 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1771 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1777 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1779 p_next = p_buffer->p_next;
1780 block_Release( p_buffer );
1788 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1790 sout_access_out_t *p_grab;
1792 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1793 if( p_grab == NULL )
1796 p_grab->p_module = NULL;
1797 p_grab->psz_access = strdup( "grab" );
1798 p_grab->p_cfg = NULL;
1799 p_grab->psz_path = strdup( "" );
1800 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1801 p_grab->pf_seek = NULL;
1802 p_grab->pf_write = AccessOutGrabberWrite;
1803 vlc_object_attach( p_grab, p_stream );