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;
304 uint32_t i_timestamp;
306 uint8_t i_payload_type;
318 /* Packetizer specific fields */
321 srtp_session_t *srtp;
323 pf_rtp_packetizer_t pf_packetize;
326 vlc_mutex_t lock_sink;
329 rtsp_stream_id_t *rtsp_id;
335 block_fifo_t *p_fifo;
339 /*****************************************************************************
341 *****************************************************************************/
342 static int Open( vlc_object_t *p_this )
344 sout_stream_t *p_stream = (sout_stream_t*)p_this;
345 sout_instance_t *p_sout = p_stream->p_sout;
346 sout_stream_sys_t *p_sys = NULL;
347 config_chain_t *p_cfg = NULL;
351 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
352 ppsz_sout_options, p_stream->p_cfg );
354 p_sys = malloc( sizeof( sout_stream_sys_t ) );
358 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
360 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
361 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
362 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
363 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
365 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
367 msg_Err( p_stream, "audio and video RTP port must be distinct" );
368 free( p_sys->psz_destination );
373 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
375 if( !strcmp( p_cfg->psz_name, "sdp" )
376 && ( p_cfg->psz_value != NULL )
377 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
385 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
388 if( !strncasecmp( psz, "rtsp:", 5 ) )
394 /* Transport protocol */
395 p_sys->proto = IPPROTO_UDP;
396 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
398 if ((psz == NULL) || !strcasecmp (psz, "udp"))
399 (void)0; /* default */
401 if (!strcasecmp (psz, "dccp"))
403 p_sys->proto = IPPROTO_DCCP;
404 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
408 if (!strcasecmp (psz, "sctp"))
410 p_sys->proto = IPPROTO_TCP;
411 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
416 if (!strcasecmp (psz, "tcp"))
418 p_sys->proto = IPPROTO_TCP;
419 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
423 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
424 p_sys->proto = IPPROTO_UDPLITE;
426 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
429 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
431 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
433 msg_Err( p_stream, "missing destination and not in RTSP mode" );
438 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
439 if( p_sys->i_ttl == -1 )
441 /* Normally, we should let the default hop limit up to the core,
442 * but we have to know it to build our SDP properly, which is why
443 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
445 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
448 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
450 p_sys->payload_bitmap = 0;
454 p_sys->psz_sdp = NULL;
456 p_sys->b_export_sap = false;
457 p_sys->p_session = NULL;
458 p_sys->psz_sdp_file = NULL;
460 p_sys->p_httpd_host = NULL;
461 p_sys->p_httpd_file = NULL;
463 p_stream->p_sys = p_sys;
465 vlc_mutex_init( &p_sys->lock_sdp );
466 vlc_mutex_init( &p_sys->lock_es );
468 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
471 sout_stream_id_t *id;
473 /* Check muxer type */
474 if( strncasecmp( psz, "ps", 2 )
475 && strncasecmp( psz, "mpeg1", 5 )
476 && strncasecmp( psz, "ts", 2 ) )
478 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
480 vlc_mutex_destroy( &p_sys->lock_sdp );
481 vlc_mutex_destroy( &p_sys->lock_es );
482 free( p_sys->psz_destination );
487 p_sys->p_grab = GrabberCreate( p_stream );
488 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
491 if( p_sys->p_mux == NULL )
493 msg_Err( p_stream, "cannot create muxer" );
494 sout_AccessOutDelete( p_sys->p_grab );
495 vlc_mutex_destroy( &p_sys->lock_sdp );
496 vlc_mutex_destroy( &p_sys->lock_es );
497 free( p_sys->psz_destination );
502 id = Add( p_stream, NULL );
505 sout_MuxDelete( p_sys->p_mux );
506 sout_AccessOutDelete( p_sys->p_grab );
507 vlc_mutex_destroy( &p_sys->lock_sdp );
508 vlc_mutex_destroy( &p_sys->lock_es );
509 free( p_sys->psz_destination );
514 p_sys->packet = NULL;
516 p_stream->pf_add = MuxAdd;
517 p_stream->pf_del = MuxDel;
518 p_stream->pf_send = MuxSend;
523 p_sys->p_grab = NULL;
525 p_stream->pf_add = Add;
526 p_stream->pf_del = Del;
527 p_stream->pf_send = Send;
530 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
531 SDPHandleUrl( p_stream, "sap" );
533 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
536 config_chain_t *p_cfg;
538 SDPHandleUrl( p_stream, psz );
540 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
542 if( !strcmp( p_cfg->psz_name, "sdp" ) )
544 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
547 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
548 if( !strcmp( p_cfg->psz_value, psz ) )
551 SDPHandleUrl( p_stream, p_cfg->psz_value );
557 /* update p_sout->i_out_pace_nocontrol */
558 p_stream->p_sout->i_out_pace_nocontrol++;
563 /*****************************************************************************
565 *****************************************************************************/
566 static void Close( vlc_object_t * p_this )
568 sout_stream_t *p_stream = (sout_stream_t*)p_this;
569 sout_stream_sys_t *p_sys = p_stream->p_sys;
571 /* update p_sout->i_out_pace_nocontrol */
572 p_stream->p_sout->i_out_pace_nocontrol--;
576 assert( p_sys->i_es == 1 );
578 sout_MuxDelete( p_sys->p_mux );
579 Del( p_stream, p_sys->es[0] );
580 sout_AccessOutDelete( p_sys->p_grab );
584 block_Release( p_sys->packet );
586 if( p_sys->b_export_sap )
589 SapSetup( p_stream );
593 if( p_sys->rtsp != NULL )
594 RtspUnsetup( p_sys->rtsp );
596 vlc_mutex_destroy( &p_sys->lock_sdp );
597 vlc_mutex_destroy( &p_sys->lock_es );
599 if( p_sys->p_httpd_file )
600 httpd_FileDelete( p_sys->p_httpd_file );
602 if( p_sys->p_httpd_host )
603 httpd_HostDelete( p_sys->p_httpd_host );
605 free( p_sys->psz_sdp );
607 if( p_sys->psz_sdp_file != NULL )
610 unlink( p_sys->psz_sdp_file );
612 free( p_sys->psz_sdp_file );
614 free( p_sys->psz_destination );
618 /*****************************************************************************
620 *****************************************************************************/
621 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
623 sout_stream_sys_t *p_sys = p_stream->p_sys;
626 vlc_UrlParse( &url, psz_url, 0 );
627 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
629 if( p_sys->p_httpd_file )
631 msg_Err( p_stream, "you can use sdp=http:// only once" );
635 if( HttpSetup( p_stream, &url ) )
637 msg_Err( p_stream, "cannot export SDP as HTTP" );
640 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
642 if( p_sys->rtsp != NULL )
644 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
648 /* FIXME test if destination is multicast or no destination at all */
649 p_sys->rtsp = RtspSetup( p_stream, &url );
650 if( p_sys->rtsp == NULL )
651 msg_Err( p_stream, "cannot export SDP as RTSP" );
653 if( p_sys->p_mux != NULL )
655 sout_stream_id_t *id = p_sys->es[0];
656 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
657 p_sys->psz_destination, p_sys->i_ttl,
658 id->i_port, id->i_port + 1 );
661 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
662 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
664 p_sys->b_export_sap = true;
665 SapSetup( p_stream );
667 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
669 if( p_sys->psz_sdp_file != NULL )
671 msg_Err( p_stream, "you can use sdp=file:// only once" );
674 psz_url = &psz_url[5];
675 if( psz_url[0] == '/' && psz_url[1] == '/' )
677 p_sys->psz_sdp_file = strdup( psz_url );
678 if( p_sys->psz_sdp_file == NULL )
680 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
681 FileSetup( p_stream );
685 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
690 vlc_UrlClean( &url );
693 /*****************************************************************************
695 *****************************************************************************/
697 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
699 const sout_stream_sys_t *p_sys = p_stream->p_sys;
701 struct sockaddr_storage dst;
705 * When we have a fixed destination (typically when we do multicast),
706 * we need to put the actual port numbers in the SDP.
707 * When there is no fixed destination, we only support RTSP unicast
708 * on-demand setup, so we should rather let the clients decide which ports
710 * When there is both a fixed destination and RTSP unicast, we need to
711 * put port numbers used by the fixed destination, otherwise the SDP would
712 * become totally incorrect for multicast use. It should be noted that
713 * port numbers from SDP with RTSP are only "recommendation" from the
714 * server to the clients (per RFC2326), so only broken clients will fail
715 * to handle this properly. There is no solution but to use two differents
716 * output chain with two different RTSP URLs if you need to handle this
721 if( p_sys->psz_destination != NULL )
725 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
726 dstlen = sizeof( dst );
727 if( p_sys->es[0]->listen.fd != NULL )
728 getsockname( p_sys->es[0]->listen.fd[0],
729 (struct sockaddr *)&dst, &dstlen );
731 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
732 (struct sockaddr *)&dst, &dstlen );
738 /* Dummy destination address for RTSP */
739 memset (&dst, 0, sizeof( struct sockaddr_in ) );
740 dst.ss_family = AF_INET;
744 dstlen = sizeof( struct sockaddr_in );
747 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
748 NULL, 0, (struct sockaddr *)&dst, dstlen );
749 if( psz_sdp == NULL )
752 /* TODO: a=source-filter */
753 if( p_sys->rtcp_mux )
754 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
756 if( rtsp_url != NULL )
757 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
759 /* FIXME: locking?! */
760 for( i = 0; i < p_sys->i_es; i++ )
762 sout_stream_id_t *id = p_sys->es[i];
763 const char *mime_major; /* major MIME type */
764 const char *proto = "RTP/AVP"; /* protocol */
769 mime_major = "video";
772 mime_major = "audio";
781 if( rtsp_url == NULL )
783 switch( p_sys->proto )
788 proto = "TCP/RTP/AVP";
791 proto = "DCCP/RTP/AVP";
793 case IPPROTO_UDPLITE:
798 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
799 id->i_payload_type, false, id->i_bitrate,
800 id->psz_enc, id->i_clock_rate, id->i_channels,
803 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
804 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
806 if( rtsp_url != NULL )
808 assert( strlen( rtsp_url ) > 0 );
809 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
810 sdp_AddAttribute ( &psz_sdp, "control",
811 addslash ? "%s/trackID=%u" : "%strackID=%u",
816 if( id->listen.fd != NULL )
817 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
818 if( p_sys->proto == IPPROTO_DCCP )
819 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
820 "SC:RTP%c", toupper( mime_major[0] ) );
827 /*****************************************************************************
829 *****************************************************************************/
831 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
833 static const char hex[16] = "0123456789abcdef";
836 for( i = 0; i < i_data; i++ )
838 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
839 s[2*i+1] = hex[(p_data[i] )&0xf];
845 * Shrink the MTU down to a fixed packetization time (for audio).
848 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
850 /* Samples per second */
851 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
852 bytes *= id->i_channels;
855 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
856 id->i_mtu = 12 + spl;
857 else /* MTU is too small for ptime, align to a sample boundary */
858 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
861 /** Add an ES as a new RTP stream */
862 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
864 /* NOTE: As a special case, if we use a non-RTP
865 * mux (TS/PS), then p_fmt is NULL. */
866 sout_stream_sys_t *p_sys = p_stream->p_sys;
867 sout_stream_id_t *id;
870 if (0xffffffff == p_sys->payload_bitmap)
872 msg_Err (p_stream, "too many RTP elementary streams");
876 /* Choose the port */
881 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
882 i_port = p_sys->i_port_audio;
884 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
885 i_port = p_sys->i_port_video;
887 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
888 * only one thread that can *modify* the ES table. The ES lock protects
889 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
890 for (int i = 0; i_port && (i < p_sys->i_es); i++)
891 if (i_port == p_sys->es[i]->i_port)
892 i_port = 0; /* Port already in use! */
893 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
897 msg_Err (p_stream, "too many RTP elementary streams");
901 for (int i = 0; i_port && (i < p_sys->i_es); i++)
902 if (p == p_sys->es[i]->i_port)
906 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
909 vlc_object_attach( id, p_stream );
911 id->p_stream = p_stream;
913 id->i_timestamp = 0; /* It will be filled when the first packet is sent */
915 /* Look for free dymanic payload type */
916 id->i_payload_type = 96;
917 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
918 id->i_payload_type++;
919 assert (id->i_payload_type < 128);
921 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
922 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
926 id->i_clock_rate = 90000; /* most common case for video */
931 id->i_cat = p_fmt->i_cat;
932 if( p_fmt->i_cat == AUDIO_ES )
934 id->i_clock_rate = p_fmt->audio.i_rate;
935 id->i_channels = p_fmt->audio.i_channels;
937 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
941 id->i_cat = VIDEO_ES;
945 id->i_mtu = config_GetInt( p_stream, "mtu" );
946 if( id->i_mtu <= 12 + 16 )
947 id->i_mtu = 576 - 20 - 8; /* pessimistic */
948 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
950 id->pf_packetize = NULL;
955 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
958 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
959 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
960 if (id->srtp == NULL)
966 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
967 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
972 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
975 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
979 vlc_mutex_init( &id->lock_sink );
984 id->listen.fd = NULL;
987 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
989 if( p_sys->psz_destination != NULL )
990 switch( p_sys->proto )
997 case VIDEO_ES: code = "RTPV"; break;
998 case AUDIO_ES: code = "RTPARTPV"; break;
999 case SPU_ES: code = "RTPTRTPV"; break;
1000 default: code = "RTPORTPV"; break;
1002 var_SetString (p_stream, "dccp-service", code);
1003 } /* fall through */
1005 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1006 p_sys->psz_destination, i_port,
1008 if( id->listen.fd == NULL )
1010 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1013 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1014 VLC_THREAD_PRIORITY_LOW ) )
1016 net_ListenClose( id->listen.fd );
1017 id->listen.fd = NULL;
1024 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1025 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1026 i_port, ttl, p_sys->proto );
1029 msg_Err( p_stream, "cannot create RTP socket" );
1032 /* Ignore any unexpected incoming packet (including RTCP-RR
1033 * packets in case of rtcp-mux) */
1034 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1036 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1042 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1044 if( psz == NULL ) /* Uho! */
1047 if( strncmp( psz, "ts", 2 ) == 0 )
1049 id->i_payload_type = 33;
1050 id->psz_enc = "MP2T";
1054 id->psz_enc = "MP2P";
1059 switch( p_fmt->i_codec )
1061 case VLC_CODEC_MULAW:
1062 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1063 id->i_payload_type = 0;
1064 id->psz_enc = "PCMU";
1065 id->pf_packetize = rtp_packetize_split;
1066 rtp_set_ptime (id, 20, 1);
1068 case VLC_CODEC_ALAW:
1069 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1070 id->i_payload_type = 8;
1071 id->psz_enc = "PCMA";
1072 id->pf_packetize = rtp_packetize_split;
1073 rtp_set_ptime (id, 20, 1);
1075 case VLC_CODEC_S16B:
1076 case VLC_CODEC_S16L:
1077 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1079 id->i_payload_type = 11;
1081 else if( p_fmt->audio.i_channels == 2 &&
1082 p_fmt->audio.i_rate == 44100 )
1084 id->i_payload_type = 10;
1086 id->psz_enc = "L16";
1087 if( p_fmt->i_codec == VLC_CODEC_S16B )
1088 id->pf_packetize = rtp_packetize_split;
1090 id->pf_packetize = rtp_packetize_swab;
1091 rtp_set_ptime (id, 20, 2);
1095 id->pf_packetize = rtp_packetize_split;
1096 rtp_set_ptime (id, 20, 1);
1098 case VLC_CODEC_MPGA:
1099 id->i_payload_type = 14;
1100 id->psz_enc = "MPA";
1101 id->i_clock_rate = 90000; /* not 44100 */
1102 id->pf_packetize = rtp_packetize_mpa;
1104 case VLC_CODEC_MPGV:
1105 id->i_payload_type = 32;
1106 id->psz_enc = "MPV";
1107 id->pf_packetize = rtp_packetize_mpv;
1109 case VLC_CODEC_ADPCM_G726:
1110 switch( p_fmt->i_bitrate / 1000 )
1113 id->psz_enc = "G726-16";
1114 id->pf_packetize = rtp_packetize_g726_16;
1117 id->psz_enc = "G726-24";
1118 id->pf_packetize = rtp_packetize_g726_24;
1121 id->psz_enc = "G726-32";
1122 id->pf_packetize = rtp_packetize_g726_32;
1125 id->psz_enc = "G726-40";
1126 id->pf_packetize = rtp_packetize_g726_40;
1129 msg_Err( p_stream, "cannot add this stream (unsupported "
1130 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1135 id->psz_enc = "ac3";
1136 id->pf_packetize = rtp_packetize_ac3;
1138 case VLC_CODEC_H263:
1139 id->psz_enc = "H263-1998";
1140 id->pf_packetize = rtp_packetize_h263;
1142 case VLC_CODEC_H264:
1143 id->psz_enc = "H264";
1144 id->pf_packetize = rtp_packetize_h264;
1145 id->psz_fmtp = NULL;
1147 if( p_fmt->i_extra > 0 )
1149 uint8_t *p_buffer = p_fmt->p_extra;
1150 int i_buffer = p_fmt->i_extra;
1151 char *p_64_sps = NULL;
1152 char *p_64_pps = NULL;
1155 while( i_buffer > 4 &&
1156 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1157 p_buffer[2] == 0 && p_buffer[3] == 1 )
1159 const int i_nal_type = p_buffer[4]&0x1f;
1163 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1166 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1168 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1170 /* we found another startcode */
1175 if( i_nal_type == 7 )
1177 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1178 sprintf_hexa( hexa, &p_buffer[5], 3 );
1180 else if( i_nal_type == 8 )
1182 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1188 if( p_64_sps && p_64_pps &&
1189 ( asprintf( &id->psz_fmtp,
1190 "packetization-mode=1;profile-level-id=%s;"
1191 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1192 p_64_pps ) == -1 ) )
1193 id->psz_fmtp = NULL;
1198 id->psz_fmtp = strdup( "packetization-mode=1" );
1201 case VLC_CODEC_MP4V:
1203 char hexa[2*p_fmt->i_extra +1];
1205 id->psz_enc = "MP4V-ES";
1206 id->pf_packetize = rtp_packetize_split;
1207 if( p_fmt->i_extra > 0 )
1209 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1210 if( asprintf( &id->psz_fmtp,
1211 "profile-level-id=3; config=%s;", hexa ) == -1 )
1212 id->psz_fmtp = NULL;
1216 case VLC_CODEC_MP4A:
1220 char hexa[2*p_fmt->i_extra +1];
1222 id->psz_enc = "mpeg4-generic";
1223 id->pf_packetize = rtp_packetize_mp4a;
1224 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1225 if( asprintf( &id->psz_fmtp,
1226 "streamtype=5; profile-level-id=15; "
1227 "mode=AAC-hbr; config=%s; SizeLength=13; "
1228 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1230 id->psz_fmtp = NULL;
1236 unsigned char config[6];
1237 unsigned int aacsrates[15] = {
1238 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1239 16000, 12000, 11025, 8000, 7350, 0, 0 };
1241 for( i = 0; i < 15; i++ )
1242 if( p_fmt->audio.i_rate == aacsrates[i] )
1248 config[3]=p_fmt->audio.i_channels<<4;
1252 id->psz_enc = "MP4A-LATM";
1253 id->pf_packetize = rtp_packetize_mp4a_latm;
1254 sprintf_hexa( hexa, config, 6 );
1255 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1256 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1257 id->psz_fmtp = NULL;
1261 case VLC_CODEC_AMR_NB:
1262 id->psz_enc = "AMR";
1263 id->psz_fmtp = strdup( "octet-align=1" );
1264 id->pf_packetize = rtp_packetize_amr;
1266 case VLC_CODEC_AMR_WB:
1267 id->psz_enc = "AMR-WB";
1268 id->psz_fmtp = strdup( "octet-align=1" );
1269 id->pf_packetize = rtp_packetize_amr;
1271 case VLC_CODEC_SPEEX:
1272 id->psz_enc = "SPEEX";
1273 id->pf_packetize = rtp_packetize_spx;
1275 case VLC_CODEC_ITU_T140:
1276 id->psz_enc = "t140" ;
1277 id->i_clock_rate = 1000;
1278 id->pf_packetize = rtp_packetize_t140;
1282 msg_Err( p_stream, "cannot add this stream (unsupported "
1283 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1286 if (id->i_payload_type >= 96)
1287 /* Mark dynamic payload type in use */
1288 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1290 #if 0 /* No payload formats sets this at the moment */
1293 cscov += 8 /* UDP */ + 12 /* RTP */;
1295 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1298 if( p_sys->rtsp != NULL )
1299 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1300 GetDWBE( id->ssrc ),
1301 p_sys->psz_destination,
1302 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1304 id->p_fifo = block_FifoNew();
1305 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1306 VLC_THREAD_PRIORITY_HIGHEST ) )
1309 /* Update p_sys context */
1310 vlc_mutex_lock( &p_sys->lock_es );
1311 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1312 vlc_mutex_unlock( &p_sys->lock_es );
1314 psz_sdp = SDPGenerate( p_stream, NULL );
1316 vlc_mutex_lock( &p_sys->lock_sdp );
1317 free( p_sys->psz_sdp );
1318 p_sys->psz_sdp = psz_sdp;
1319 vlc_mutex_unlock( &p_sys->lock_sdp );
1321 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1323 /* Update SDP (sap/file) */
1324 if( p_sys->b_export_sap ) SapSetup( p_stream );
1325 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1330 Del( p_stream, id );
1334 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1336 sout_stream_sys_t *p_sys = p_stream->p_sys;
1338 if( id->p_fifo != NULL )
1340 vlc_object_kill( id );
1341 vlc_thread_join( id );
1342 block_FifoRelease( id->p_fifo );
1345 vlc_mutex_lock( &p_sys->lock_es );
1346 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1347 vlc_mutex_unlock( &p_sys->lock_es );
1349 /* Release dynamic payload type */
1350 if (id->i_payload_type >= 96)
1351 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1353 free( id->psz_fmtp );
1356 RtspDelId( p_sys->rtsp, id->rtsp_id );
1358 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1359 if( id->listen.fd != NULL )
1361 vlc_cancel( id->listen.thread );
1362 vlc_join( id->listen.thread, NULL );
1363 net_ListenClose( id->listen.fd );
1366 if( id->srtp != NULL )
1367 srtp_destroy( id->srtp );
1370 vlc_mutex_destroy( &id->lock_sink );
1372 /* Update SDP (sap/file) */
1373 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1374 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1376 vlc_object_detach( id );
1377 vlc_object_release( id );
1381 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1386 assert( p_stream->p_sys->p_mux == NULL );
1389 while( p_buffer != NULL )
1391 p_next = p_buffer->p_next;
1392 if( id->pf_packetize( id, p_buffer ) )
1395 block_Release( p_buffer );
1401 /****************************************************************************
1403 ****************************************************************************/
1404 static int SapSetup( sout_stream_t *p_stream )
1406 sout_stream_sys_t *p_sys = p_stream->p_sys;
1407 sout_instance_t *p_sout = p_stream->p_sout;
1409 /* Remove the previous session */
1410 if( p_sys->p_session != NULL)
1412 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1413 p_sys->p_session = NULL;
1416 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1418 announce_method_t *p_method = sout_SAPMethod();
1419 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1421 p_sys->psz_destination,
1423 sout_MethodRelease( p_method );
1429 /****************************************************************************
1431 ****************************************************************************/
1432 static int FileSetup( sout_stream_t *p_stream )
1434 sout_stream_sys_t *p_sys = p_stream->p_sys;
1437 if( p_sys->psz_sdp == NULL )
1438 return VLC_EGENERIC; /* too early */
1440 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1442 msg_Err( p_stream, "cannot open file '%s' (%m)",
1443 p_sys->psz_sdp_file );
1444 return VLC_EGENERIC;
1447 fputs( p_sys->psz_sdp, f );
1453 /****************************************************************************
1455 ****************************************************************************/
1456 static int HttpCallback( httpd_file_sys_t *p_args,
1457 httpd_file_t *, uint8_t *p_request,
1458 uint8_t **pp_data, int *pi_data );
1460 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1462 sout_stream_sys_t *p_sys = p_stream->p_sys;
1464 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1465 url->i_port > 0 ? url->i_port : 80 );
1466 if( p_sys->p_httpd_host )
1468 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1469 url->psz_path ? url->psz_path : "/",
1472 HttpCallback, (void*)p_sys );
1474 if( p_sys->p_httpd_file == NULL )
1476 return VLC_EGENERIC;
1481 static int HttpCallback( httpd_file_sys_t *p_args,
1482 httpd_file_t *f, uint8_t *p_request,
1483 uint8_t **pp_data, int *pi_data )
1485 VLC_UNUSED(f); VLC_UNUSED(p_request);
1486 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1488 vlc_mutex_lock( &p_sys->lock_sdp );
1489 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1491 *pi_data = strlen( p_sys->psz_sdp );
1492 *pp_data = malloc( *pi_data );
1493 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1500 vlc_mutex_unlock( &p_sys->lock_sdp );
1505 /****************************************************************************
1507 ****************************************************************************/
1508 static void* ThreadSend( vlc_object_t *p_this )
1511 # define ECONNREFUSED WSAECONNREFUSED
1512 # define ENOPROTOOPT WSAENOPROTOOPT
1513 # define EHOSTUNREACH WSAEHOSTUNREACH
1514 # define ENETUNREACH WSAENETUNREACH
1515 # define ENETDOWN WSAENETDOWN
1516 # define ENOBUFS WSAENOBUFS
1517 # define EAGAIN WSAEWOULDBLOCK
1518 # define EWOULDBLOCK WSAEWOULDBLOCK
1520 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1521 unsigned i_caching = id->i_caching;
1525 block_t *out = block_FifoGet( id->p_fifo );
1526 block_cleanup_push (out);
1530 { /* FIXME: this is awfully inefficient */
1531 size_t len = out->i_buffer;
1532 out = block_Realloc( out, 0, len + 10 );
1533 out->i_buffer = len;
1535 int canc = vlc_savecancel ();
1536 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1537 vlc_restorecancel (canc);
1541 msg_Dbg( id, "SRTP sending error: %m" );
1542 block_Release( out );
1546 out->i_buffer = len;
1550 mwait (out->i_dts + i_caching);
1555 ssize_t len = out->i_buffer;
1556 int canc = vlc_savecancel ();
1558 vlc_mutex_lock( &id->lock_sink );
1559 unsigned deadc = 0; /* How many dead sockets? */
1560 int deadv[id->sinkc]; /* Dead sockets list */
1562 for( int i = 0; i < id->sinkc; i++ )
1565 if( !id->srtp ) /* FIXME: SRTCP support */
1567 SendRTCP( id->sinkv[i].rtcp, out );
1569 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1573 /* Soft errors (e.g. ICMP): */
1574 case ECONNREFUSED: /* Port unreachable */
1577 case EPROTO: /* Protocol unreachable */
1579 case EHOSTUNREACH: /* Host unreachable */
1580 case ENETUNREACH: /* Network unreachable */
1581 case ENETDOWN: /* Entire network down */
1582 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1583 /* Transient congestion: */
1584 case ENOMEM: /* out of socket buffers */
1587 #if (EAGAIN != EWOULDBLOCK)
1593 deadv[deadc++] = id->sinkv[i].rtp_fd;
1595 vlc_mutex_unlock( &id->lock_sink );
1596 block_Release( out );
1598 for( unsigned i = 0; i < deadc; i++ )
1600 msg_Dbg( id, "removing socket %d", deadv[i] );
1601 rtp_del_sink( id, deadv[i] );
1603 vlc_restorecancel (canc);
1609 /* This thread dequeues incoming connections (DCCP streaming) */
1610 static void *rtp_listen_thread( void *data )
1612 sout_stream_id_t *id = data;
1614 assert( id->listen.fd != NULL );
1618 int fd = net_Accept( id, id->listen.fd );
1621 int canc = vlc_savecancel( );
1622 rtp_add_sink( id, fd, true );
1623 vlc_restorecancel( canc );
1630 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1632 rtp_sink_t sink = { fd, NULL };
1633 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1635 if( sink.rtcp == NULL )
1636 msg_Err( id, "RTCP failed!" );
1638 vlc_mutex_lock( &id->lock_sink );
1639 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1640 vlc_mutex_unlock( &id->lock_sink );
1644 void rtp_del_sink( sout_stream_id_t *id, int fd )
1646 rtp_sink_t sink = { fd, NULL };
1648 /* NOTE: must be safe to use if fd is not included */
1649 vlc_mutex_lock( &id->lock_sink );
1650 for( int i = 0; i < id->sinkc; i++ )
1652 if (id->sinkv[i].rtp_fd == fd)
1654 sink = id->sinkv[i];
1655 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1659 vlc_mutex_unlock( &id->lock_sink );
1661 CloseRTCP( sink.rtcp );
1662 net_Close( sink.rtp_fd );
1665 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1667 /* This will return values for the next packet.
1668 * Accounting for caching would not be totally trivial. */
1669 return id->i_sequence;
1672 uint32_t rtp_get_ts( const sout_stream_id_t *id )
1674 /* ... and this will return the value for the last packet.
1675 * Lame, but close enough. */
1676 return id->i_timestamp;
1679 /* FIXME: this is pretty bad - if we remove and then insert an ES
1680 * the number will get unsynched from inside RTSP */
1681 unsigned rtp_get_num( const sout_stream_id_t *id )
1683 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1686 vlc_mutex_lock( &p_sys->lock_es );
1687 for( i = 0; i < p_sys->i_es; i++ )
1689 if( id == p_sys->es[i] )
1692 vlc_mutex_unlock( &p_sys->lock_es );
1698 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1699 int b_marker, int64_t i_pts )
1701 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
1703 out->p_buffer[0] = 0x80;
1704 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1705 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1706 out->p_buffer[3] = ( id->i_sequence )&0xff;
1707 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1708 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1709 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1710 out->p_buffer[7] = ( i_timestamp )&0xff;
1712 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1715 id->i_timestamp = i_timestamp;
1719 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1721 block_FifoPut( id->p_fifo, out );
1725 * @return configured max RTP payload size (including payload type-specific
1726 * headers, excluding RTP and transport headers)
1728 size_t rtp_mtu (const sout_stream_id_t *id)
1730 return id->i_mtu - 12;
1733 /*****************************************************************************
1735 *****************************************************************************/
1737 /** Add an ES to a non-RTP muxed stream */
1738 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1740 sout_input_t *p_input;
1741 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1742 assert( p_mux != NULL );
1744 p_input = sout_MuxAddStream( p_mux, p_fmt );
1745 if( p_input == NULL )
1747 msg_Err( p_stream, "cannot add this stream to the muxer" );
1751 return (sout_stream_id_t *)p_input;
1755 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1758 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1759 assert( p_mux != NULL );
1761 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1766 /** Remove an ES from a non-RTP muxed stream */
1767 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1769 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1770 assert( p_mux != NULL );
1772 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1777 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1778 const block_t *p_buffer )
1780 sout_stream_sys_t *p_sys = p_stream->p_sys;
1781 sout_stream_id_t *id = p_sys->es[0];
1783 int64_t i_dts = p_buffer->i_dts;
1785 uint8_t *p_data = p_buffer->p_buffer;
1786 size_t i_data = p_buffer->i_buffer;
1787 size_t i_max = id->i_mtu - 12;
1789 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1795 /* output complete packet */
1796 if( p_sys->packet &&
1797 p_sys->packet->i_buffer + i_data > i_max )
1799 rtp_packetize_send( id, p_sys->packet );
1800 p_sys->packet = NULL;
1803 if( p_sys->packet == NULL )
1805 /* allocate a new packet */
1806 p_sys->packet = block_New( p_stream, id->i_mtu );
1807 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1808 p_sys->packet->i_dts = i_dts;
1809 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1810 i_dts += p_sys->packet->i_length;
1813 i_size = __MIN( i_data,
1814 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1816 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1819 p_sys->packet->i_buffer += i_size;
1828 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1831 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1837 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1839 p_next = p_buffer->p_next;
1840 block_Release( p_buffer );
1848 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1850 sout_access_out_t *p_grab;
1852 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1853 if( p_grab == NULL )
1856 p_grab->p_module = NULL;
1857 p_grab->psz_access = strdup( "grab" );
1858 p_grab->p_cfg = NULL;
1859 p_grab->psz_path = strdup( "" );
1860 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1861 p_grab->pf_seek = NULL;
1862 p_grab->pf_write = AccessOutGrabberWrite;
1863 vlc_object_attach( p_grab, p_stream );