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 CACHING_TEXT N_("Caching value (ms)")
133 #define CACHING_LONGTEXT N_( \
134 "Default caching value for outbound RTP streams. This " \
135 "value should be set in milliseconds." )
137 #define PROTO_TEXT N_("Transport protocol")
138 #define PROTO_LONGTEXT N_( \
139 "This selects which transport protocol to use for RTP." )
141 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
142 #define SRTP_KEY_LONGTEXT N_( \
143 "RTP packets will be integrity-protected and ciphered "\
144 "with this Secure RTP master shared secret key.")
146 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
147 #define SRTP_SALT_LONGTEXT N_( \
148 "Secure RTP requires a (non-secret) master salt value.")
150 static const char *const ppsz_protos[] = {
151 "dccp", "sctp", "tcp", "udp", "udplite",
154 static const char *const ppsz_protocols[] = {
155 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
158 #define RFC3016_TEXT N_("MP4A LATM")
159 #define RFC3016_LONGTEXT N_( \
160 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
162 static int Open ( vlc_object_t * );
163 static void Close( vlc_object_t * );
165 #define SOUT_CFG_PREFIX "sout-rtp-"
166 #define MAX_EMPTY_BLOCKS 200
169 set_shortname( N_("RTP"))
170 set_description( N_("RTP stream output") )
171 set_capability( "sout stream", 0 )
172 add_shortcut( "rtp" )
173 set_category( CAT_SOUT )
174 set_subcategory( SUBCAT_SOUT_STREAM )
176 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
177 DEST_LONGTEXT, true )
178 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
180 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
182 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
185 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
186 NAME_LONGTEXT, true )
187 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
188 DESC_LONGTEXT, true )
189 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
191 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
192 EMAIL_LONGTEXT, true )
193 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
194 PHONE_LONGTEXT, true )
196 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
197 PROTO_LONGTEXT, false )
198 change_string_list( ppsz_protos, ppsz_protocols, NULL )
199 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
200 PORT_LONGTEXT, true )
201 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
202 PORT_AUDIO_LONGTEXT, true )
203 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
204 PORT_VIDEO_LONGTEXT, true )
206 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
208 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
209 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
210 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
211 CACHING_TEXT, CACHING_LONGTEXT, true )
214 add_string( SOUT_CFG_PREFIX "key", "", NULL,
215 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
216 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
217 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
220 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT,
221 RFC3016_LONGTEXT, false )
223 set_callbacks( Open, Close )
226 /*****************************************************************************
227 * Exported prototypes
228 *****************************************************************************/
229 static const char *const ppsz_sout_options[] = {
230 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
231 "sap", "description", "url", "email", "phone",
232 "proto", "rtcp-mux", "caching", "key", "salt",
236 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
237 static int Del ( sout_stream_t *, sout_stream_id_t * );
238 static int Send( sout_stream_t *, sout_stream_id_t *,
240 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
241 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
242 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
245 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
246 static void* ThreadSend( vlc_object_t *p_this );
247 static void *rtp_listen_thread( void * );
249 static void SDPHandleUrl( sout_stream_t *, const char * );
251 static int SapSetup( sout_stream_t *p_stream );
252 static int FileSetup( sout_stream_t *p_stream );
253 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
255 struct sout_stream_sys_t
259 vlc_mutex_t lock_sdp;
266 session_descriptor_t *p_session;
269 httpd_host_t *p_httpd_host;
270 httpd_file_t *p_httpd_file;
276 char *psz_destination;
277 uint32_t payload_bitmap;
279 uint16_t i_port_audio;
280 uint16_t i_port_video;
286 /* in case we do TS/PS over rtp */
288 sout_access_out_t *p_grab;
294 sout_stream_id_t **es;
297 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
299 typedef struct rtp_sink_t
305 struct sout_stream_id_t
309 sout_stream_t *p_stream;
312 uint8_t i_payload_type;
324 /* Packetizer specific fields */
327 srtp_session_t *srtp;
329 pf_rtp_packetizer_t pf_packetize;
332 vlc_mutex_t lock_sink;
335 rtsp_stream_id_t *rtsp_id;
341 block_fifo_t *p_fifo;
345 /*****************************************************************************
347 *****************************************************************************/
348 static int Open( vlc_object_t *p_this )
350 sout_stream_t *p_stream = (sout_stream_t*)p_this;
351 sout_instance_t *p_sout = p_stream->p_sout;
352 sout_stream_sys_t *p_sys = NULL;
353 config_chain_t *p_cfg = NULL;
357 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
358 ppsz_sout_options, p_stream->p_cfg );
360 p_sys = malloc( sizeof( sout_stream_sys_t ) );
364 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
366 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
367 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
368 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
369 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
371 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
373 msg_Err( p_stream, "audio and video RTP port must be distinct" );
374 free( p_sys->psz_destination );
379 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
381 if( !strcmp( p_cfg->psz_name, "sdp" )
382 && ( p_cfg->psz_value != NULL )
383 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
391 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
394 if( !strncasecmp( psz, "rtsp:", 5 ) )
400 /* Transport protocol */
401 p_sys->proto = IPPROTO_UDP;
402 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
404 if ((psz == NULL) || !strcasecmp (psz, "udp"))
405 (void)0; /* default */
407 if (!strcasecmp (psz, "dccp"))
409 p_sys->proto = IPPROTO_DCCP;
410 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
414 if (!strcasecmp (psz, "sctp"))
416 p_sys->proto = IPPROTO_TCP;
417 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
422 if (!strcasecmp (psz, "tcp"))
424 p_sys->proto = IPPROTO_TCP;
425 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
429 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
430 p_sys->proto = IPPROTO_UDPLITE;
432 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
435 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
437 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
439 msg_Err( p_stream, "missing destination and not in RTSP mode" );
444 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
445 if( p_sys->i_ttl == -1 )
447 /* Normally, we should let the default hop limit up to the core,
448 * but we have to know it to build our SDP properly, which is why
449 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
451 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
454 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
456 p_sys->payload_bitmap = 0;
460 p_sys->psz_sdp = NULL;
462 p_sys->b_export_sap = false;
463 p_sys->p_session = NULL;
464 p_sys->psz_sdp_file = NULL;
466 p_sys->p_httpd_host = NULL;
467 p_sys->p_httpd_file = NULL;
469 p_stream->p_sys = p_sys;
471 vlc_mutex_init( &p_sys->lock_sdp );
472 vlc_mutex_init( &p_sys->lock_es );
474 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
477 sout_stream_id_t *id;
479 /* Check muxer type */
480 if( strncasecmp( psz, "ps", 2 )
481 && strncasecmp( psz, "mpeg1", 5 )
482 && strncasecmp( psz, "ts", 2 ) )
484 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
486 vlc_mutex_destroy( &p_sys->lock_sdp );
487 vlc_mutex_destroy( &p_sys->lock_es );
488 free( p_sys->psz_destination );
493 p_sys->p_grab = GrabberCreate( p_stream );
494 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
497 if( p_sys->p_mux == NULL )
499 msg_Err( p_stream, "cannot create muxer" );
500 sout_AccessOutDelete( p_sys->p_grab );
501 vlc_mutex_destroy( &p_sys->lock_sdp );
502 vlc_mutex_destroy( &p_sys->lock_es );
503 free( p_sys->psz_destination );
508 id = Add( p_stream, NULL );
511 sout_MuxDelete( p_sys->p_mux );
512 sout_AccessOutDelete( p_sys->p_grab );
513 vlc_mutex_destroy( &p_sys->lock_sdp );
514 vlc_mutex_destroy( &p_sys->lock_es );
515 free( p_sys->psz_destination );
520 p_sys->packet = NULL;
522 p_stream->pf_add = MuxAdd;
523 p_stream->pf_del = MuxDel;
524 p_stream->pf_send = MuxSend;
529 p_sys->p_grab = NULL;
531 p_stream->pf_add = Add;
532 p_stream->pf_del = Del;
533 p_stream->pf_send = Send;
536 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
537 SDPHandleUrl( p_stream, "sap" );
539 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
542 config_chain_t *p_cfg;
544 SDPHandleUrl( p_stream, psz );
546 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
548 if( !strcmp( p_cfg->psz_name, "sdp" ) )
550 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
553 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
554 if( !strcmp( p_cfg->psz_value, psz ) )
557 SDPHandleUrl( p_stream, p_cfg->psz_value );
563 /* update p_sout->i_out_pace_nocontrol */
564 p_stream->p_sout->i_out_pace_nocontrol++;
569 /*****************************************************************************
571 *****************************************************************************/
572 static void Close( vlc_object_t * p_this )
574 sout_stream_t *p_stream = (sout_stream_t*)p_this;
575 sout_stream_sys_t *p_sys = p_stream->p_sys;
577 /* update p_sout->i_out_pace_nocontrol */
578 p_stream->p_sout->i_out_pace_nocontrol--;
582 assert( p_sys->i_es == 1 );
584 sout_MuxDelete( p_sys->p_mux );
585 Del( p_stream, p_sys->es[0] );
586 sout_AccessOutDelete( p_sys->p_grab );
590 block_Release( p_sys->packet );
592 if( p_sys->b_export_sap )
595 SapSetup( p_stream );
599 if( p_sys->rtsp != NULL )
600 RtspUnsetup( p_sys->rtsp );
602 vlc_mutex_destroy( &p_sys->lock_sdp );
603 vlc_mutex_destroy( &p_sys->lock_es );
605 if( p_sys->p_httpd_file )
606 httpd_FileDelete( p_sys->p_httpd_file );
608 if( p_sys->p_httpd_host )
609 httpd_HostDelete( p_sys->p_httpd_host );
611 free( p_sys->psz_sdp );
613 if( p_sys->psz_sdp_file != NULL )
616 unlink( p_sys->psz_sdp_file );
618 free( p_sys->psz_sdp_file );
620 free( p_sys->psz_destination );
624 /*****************************************************************************
626 *****************************************************************************/
627 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
629 sout_stream_sys_t *p_sys = p_stream->p_sys;
632 vlc_UrlParse( &url, psz_url, 0 );
633 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
635 if( p_sys->p_httpd_file )
637 msg_Err( p_stream, "you can use sdp=http:// only once" );
641 if( HttpSetup( p_stream, &url ) )
643 msg_Err( p_stream, "cannot export SDP as HTTP" );
646 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
648 if( p_sys->rtsp != NULL )
650 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
654 /* FIXME test if destination is multicast or no destination at all */
655 p_sys->rtsp = RtspSetup( p_stream, &url );
656 if( p_sys->rtsp == NULL )
657 msg_Err( p_stream, "cannot export SDP as RTSP" );
659 if( p_sys->p_mux != NULL )
661 sout_stream_id_t *id = p_sys->es[0];
662 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
663 p_sys->psz_destination, p_sys->i_ttl,
664 id->i_port, id->i_port + 1 );
667 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
668 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
670 p_sys->b_export_sap = true;
671 SapSetup( p_stream );
673 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
675 if( p_sys->psz_sdp_file != NULL )
677 msg_Err( p_stream, "you can use sdp=file:// only once" );
680 psz_url = &psz_url[5];
681 if( psz_url[0] == '/' && psz_url[1] == '/' )
683 p_sys->psz_sdp_file = strdup( psz_url );
684 if( p_sys->psz_sdp_file == NULL )
686 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
687 FileSetup( p_stream );
691 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
696 vlc_UrlClean( &url );
699 /*****************************************************************************
701 *****************************************************************************/
703 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
705 const sout_stream_sys_t *p_sys = p_stream->p_sys;
707 struct sockaddr_storage dst;
711 * When we have a fixed destination (typically when we do multicast),
712 * we need to put the actual port numbers in the SDP.
713 * When there is no fixed destination, we only support RTSP unicast
714 * on-demand setup, so we should rather let the clients decide which ports
716 * When there is both a fixed destination and RTSP unicast, we need to
717 * put port numbers used by the fixed destination, otherwise the SDP would
718 * become totally incorrect for multicast use. It should be noted that
719 * port numbers from SDP with RTSP are only "recommendation" from the
720 * server to the clients (per RFC2326), so only broken clients will fail
721 * to handle this properly. There is no solution but to use two differents
722 * output chain with two different RTSP URLs if you need to handle this
727 if( p_sys->psz_destination != NULL )
731 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
732 dstlen = sizeof( dst );
733 if( p_sys->es[0]->listen.fd != NULL )
734 getsockname( p_sys->es[0]->listen.fd[0],
735 (struct sockaddr *)&dst, &dstlen );
737 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
738 (struct sockaddr *)&dst, &dstlen );
744 /* Dummy destination address for RTSP */
745 memset (&dst, 0, sizeof( struct sockaddr_in ) );
746 dst.ss_family = AF_INET;
750 dstlen = sizeof( struct sockaddr_in );
753 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
754 NULL, 0, (struct sockaddr *)&dst, dstlen );
755 if( psz_sdp == NULL )
758 /* TODO: a=source-filter */
759 if( p_sys->rtcp_mux )
760 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
762 if( rtsp_url != NULL )
763 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
765 /* FIXME: locking?! */
766 for( i = 0; i < p_sys->i_es; i++ )
768 sout_stream_id_t *id = p_sys->es[i];
769 const char *mime_major; /* major MIME type */
770 const char *proto = "RTP/AVP"; /* protocol */
775 mime_major = "video";
778 mime_major = "audio";
787 if( rtsp_url == NULL )
789 switch( p_sys->proto )
794 proto = "TCP/RTP/AVP";
797 proto = "DCCP/RTP/AVP";
799 case IPPROTO_UDPLITE:
804 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
805 id->i_payload_type, false, id->i_bitrate,
806 id->psz_enc, id->i_clock_rate, id->i_channels,
809 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
810 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
812 if( rtsp_url != NULL )
814 assert( strlen( rtsp_url ) > 0 );
815 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
816 sdp_AddAttribute ( &psz_sdp, "control",
817 addslash ? "%s/trackID=%u" : "%strackID=%u",
822 if( id->listen.fd != NULL )
823 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
824 if( p_sys->proto == IPPROTO_DCCP )
825 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
826 "SC:RTP%c", toupper( mime_major[0] ) );
833 /*****************************************************************************
835 *****************************************************************************/
837 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
839 static const char hex[16] = "0123456789abcdef";
842 for( i = 0; i < i_data; i++ )
844 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
845 s[2*i+1] = hex[(p_data[i] )&0xf];
851 * Shrink the MTU down to a fixed packetization time (for audio).
854 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
856 /* Samples per second */
857 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
858 bytes *= id->i_channels;
861 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
862 id->i_mtu = 12 + spl;
863 else /* MTU is too small for ptime, align to a sample boundary */
864 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
867 /** Add an ES as a new RTP stream */
868 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
870 /* NOTE: As a special case, if we use a non-RTP
871 * mux (TS/PS), then p_fmt is NULL. */
872 sout_stream_sys_t *p_sys = p_stream->p_sys;
873 sout_stream_id_t *id;
876 if (0xffffffff == p_sys->payload_bitmap)
878 msg_Err (p_stream, "too many RTP elementary streams");
882 /* Choose the port */
887 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
888 i_port = p_sys->i_port_audio;
890 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
891 i_port = p_sys->i_port_video;
893 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
894 * only one thread that can *modify* the ES table. The ES lock protects
895 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
896 for (int i = 0; i_port && (i < p_sys->i_es); i++)
897 if (i_port == p_sys->es[i]->i_port)
898 i_port = 0; /* Port already in use! */
899 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
903 msg_Err (p_stream, "too many RTP elementary streams");
907 for (int i = 0; i_port && (i < p_sys->i_es); i++)
908 if (p == p_sys->es[i]->i_port)
912 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
915 vlc_object_attach( id, p_stream );
917 id->p_stream = p_stream;
919 /* Look for free dymanic payload type */
920 id->i_payload_type = 96;
921 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
922 id->i_payload_type++;
923 assert (id->i_payload_type < 128);
925 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
926 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
930 id->i_clock_rate = 90000; /* most common case for video */
935 id->i_cat = p_fmt->i_cat;
936 if( p_fmt->i_cat == AUDIO_ES )
938 id->i_clock_rate = p_fmt->audio.i_rate;
939 id->i_channels = p_fmt->audio.i_channels;
941 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
945 id->i_cat = VIDEO_ES;
949 id->i_mtu = config_GetInt( p_stream, "mtu" );
950 if( id->i_mtu <= 12 + 16 )
951 id->i_mtu = 576 - 20 - 8; /* pessimistic */
952 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
954 id->pf_packetize = NULL;
959 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
962 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
963 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
964 if (id->srtp == NULL)
970 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
971 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
976 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
979 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
983 vlc_mutex_init( &id->lock_sink );
988 id->listen.fd = NULL;
991 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
993 if( p_sys->psz_destination != NULL )
994 switch( p_sys->proto )
1001 case VIDEO_ES: code = "RTPV"; break;
1002 case AUDIO_ES: code = "RTPARTPV"; break;
1003 case SPU_ES: code = "RTPTRTPV"; break;
1004 default: code = "RTPORTPV"; break;
1006 var_SetString (p_stream, "dccp-service", code);
1007 } /* fall through */
1009 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1010 p_sys->psz_destination, i_port,
1012 if( id->listen.fd == NULL )
1014 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1017 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1018 VLC_THREAD_PRIORITY_LOW ) )
1020 net_ListenClose( id->listen.fd );
1021 id->listen.fd = NULL;
1028 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1029 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1030 i_port, ttl, p_sys->proto );
1033 msg_Err( p_stream, "cannot create RTP socket" );
1036 /* Ignore any unexpected incoming packet (including RTCP-RR
1037 * packets in case of rtcp-mux) */
1038 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1040 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1046 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1048 if( psz == NULL ) /* Uho! */
1051 if( strncmp( psz, "ts", 2 ) == 0 )
1053 id->i_payload_type = 33;
1054 id->psz_enc = "MP2T";
1058 id->psz_enc = "MP2P";
1063 switch( p_fmt->i_codec )
1065 case VLC_CODEC_MULAW:
1066 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1067 id->i_payload_type = 0;
1068 id->psz_enc = "PCMU";
1069 id->pf_packetize = rtp_packetize_split;
1070 rtp_set_ptime (id, 20, 1);
1072 case VLC_CODEC_ALAW:
1073 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1074 id->i_payload_type = 8;
1075 id->psz_enc = "PCMA";
1076 id->pf_packetize = rtp_packetize_split;
1077 rtp_set_ptime (id, 20, 1);
1079 case VLC_CODEC_S16B:
1080 case VLC_CODEC_S16L:
1081 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1083 id->i_payload_type = 11;
1085 else if( p_fmt->audio.i_channels == 2 &&
1086 p_fmt->audio.i_rate == 44100 )
1088 id->i_payload_type = 10;
1090 id->psz_enc = "L16";
1091 if( p_fmt->i_codec == VLC_CODEC_S16B )
1092 id->pf_packetize = rtp_packetize_split;
1094 id->pf_packetize = rtp_packetize_swab;
1095 rtp_set_ptime (id, 20, 2);
1099 id->pf_packetize = rtp_packetize_split;
1100 rtp_set_ptime (id, 20, 1);
1102 case VLC_CODEC_MPGA:
1103 id->i_payload_type = 14;
1104 id->psz_enc = "MPA";
1105 id->i_clock_rate = 90000; /* not 44100 */
1106 id->pf_packetize = rtp_packetize_mpa;
1108 case VLC_CODEC_MPGV:
1109 id->i_payload_type = 32;
1110 id->psz_enc = "MPV";
1111 id->pf_packetize = rtp_packetize_mpv;
1113 case VLC_CODEC_ADPCM_G726:
1114 switch( p_fmt->i_bitrate / 1000 )
1117 id->psz_enc = "G726-16";
1118 id->pf_packetize = rtp_packetize_g726_16;
1121 id->psz_enc = "G726-24";
1122 id->pf_packetize = rtp_packetize_g726_24;
1125 id->psz_enc = "G726-32";
1126 id->pf_packetize = rtp_packetize_g726_32;
1129 id->psz_enc = "G726-40";
1130 id->pf_packetize = rtp_packetize_g726_40;
1133 msg_Err( p_stream, "cannot add this stream (unsupported "
1134 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1139 id->psz_enc = "ac3";
1140 id->pf_packetize = rtp_packetize_ac3;
1142 case VLC_CODEC_H263:
1143 id->psz_enc = "H263-1998";
1144 id->pf_packetize = rtp_packetize_h263;
1146 case VLC_CODEC_H264:
1147 id->psz_enc = "H264";
1148 id->pf_packetize = rtp_packetize_h264;
1149 id->psz_fmtp = NULL;
1151 if( p_fmt->i_extra > 0 )
1153 uint8_t *p_buffer = p_fmt->p_extra;
1154 int i_buffer = p_fmt->i_extra;
1155 char *p_64_sps = NULL;
1156 char *p_64_pps = NULL;
1159 while( i_buffer > 4 &&
1160 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1161 p_buffer[2] == 0 && p_buffer[3] == 1 )
1163 const int i_nal_type = p_buffer[4]&0x1f;
1167 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1170 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1172 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1174 /* we found another startcode */
1179 if( i_nal_type == 7 )
1181 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1182 sprintf_hexa( hexa, &p_buffer[5], 3 );
1184 else if( i_nal_type == 8 )
1186 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1192 if( p_64_sps && p_64_pps &&
1193 ( asprintf( &id->psz_fmtp,
1194 "packetization-mode=1;profile-level-id=%s;"
1195 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1196 p_64_pps ) == -1 ) )
1197 id->psz_fmtp = NULL;
1202 id->psz_fmtp = strdup( "packetization-mode=1" );
1205 case VLC_CODEC_MP4V:
1207 char hexa[2*p_fmt->i_extra +1];
1209 id->psz_enc = "MP4V-ES";
1210 id->pf_packetize = rtp_packetize_split;
1211 if( p_fmt->i_extra > 0 )
1213 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1214 if( asprintf( &id->psz_fmtp,
1215 "profile-level-id=3; config=%s;", hexa ) == -1 )
1216 id->psz_fmtp = NULL;
1220 case VLC_CODEC_MP4A:
1224 char hexa[2*p_fmt->i_extra +1];
1226 id->psz_enc = "mpeg4-generic";
1227 id->pf_packetize = rtp_packetize_mp4a;
1228 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1229 if( asprintf( &id->psz_fmtp,
1230 "streamtype=5; profile-level-id=15; "
1231 "mode=AAC-hbr; config=%s; SizeLength=13; "
1232 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1234 id->psz_fmtp = NULL;
1240 unsigned char config[6];
1241 unsigned int aacsrates[15] = {
1242 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1243 16000, 12000, 11025, 8000, 7350, 0, 0 };
1245 for( i = 0; i < 15; i++ )
1246 if( p_fmt->audio.i_rate == aacsrates[i] )
1252 config[3]=p_fmt->audio.i_channels<<4;
1256 id->psz_enc = "MP4A-LATM";
1257 id->pf_packetize = rtp_packetize_mp4a_latm;
1258 sprintf_hexa( hexa, config, 6 );
1259 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1260 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1261 id->psz_fmtp = NULL;
1265 case VLC_CODEC_AMR_NB:
1266 id->psz_enc = "AMR";
1267 id->psz_fmtp = strdup( "octet-align=1" );
1268 id->pf_packetize = rtp_packetize_amr;
1270 case VLC_CODEC_AMR_WB:
1271 id->psz_enc = "AMR-WB";
1272 id->psz_fmtp = strdup( "octet-align=1" );
1273 id->pf_packetize = rtp_packetize_amr;
1275 case VLC_CODEC_SPEEX:
1276 id->psz_enc = "SPEEX";
1277 id->pf_packetize = rtp_packetize_spx;
1279 case VLC_CODEC_ITU_T140:
1280 id->psz_enc = "t140" ;
1281 id->i_clock_rate = 1000;
1282 id->pf_packetize = rtp_packetize_t140;
1286 msg_Err( p_stream, "cannot add this stream (unsupported "
1287 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1290 if (id->i_payload_type >= 96)
1291 /* Mark dynamic payload type in use */
1292 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1294 #if 0 /* No payload formats sets this at the moment */
1297 cscov += 8 /* UDP */ + 12 /* RTP */;
1299 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1302 if( p_sys->rtsp != NULL )
1303 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1304 GetDWBE( id->ssrc ),
1305 p_sys->psz_destination,
1306 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1308 id->p_fifo = block_FifoNew();
1309 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1310 VLC_THREAD_PRIORITY_HIGHEST ) )
1313 /* Update p_sys context */
1314 vlc_mutex_lock( &p_sys->lock_es );
1315 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1316 vlc_mutex_unlock( &p_sys->lock_es );
1318 psz_sdp = SDPGenerate( p_stream, NULL );
1320 vlc_mutex_lock( &p_sys->lock_sdp );
1321 free( p_sys->psz_sdp );
1322 p_sys->psz_sdp = psz_sdp;
1323 vlc_mutex_unlock( &p_sys->lock_sdp );
1325 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1327 /* Update SDP (sap/file) */
1328 if( p_sys->b_export_sap ) SapSetup( p_stream );
1329 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1334 Del( p_stream, id );
1338 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1340 sout_stream_sys_t *p_sys = p_stream->p_sys;
1342 if( id->p_fifo != NULL )
1344 vlc_object_kill( id );
1345 vlc_thread_join( id );
1346 block_FifoRelease( id->p_fifo );
1349 vlc_mutex_lock( &p_sys->lock_es );
1350 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1351 vlc_mutex_unlock( &p_sys->lock_es );
1353 /* Release dynamic payload type */
1354 if (id->i_payload_type >= 96)
1355 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1357 free( id->psz_fmtp );
1360 RtspDelId( p_sys->rtsp, id->rtsp_id );
1362 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1363 if( id->listen.fd != NULL )
1365 vlc_cancel( id->listen.thread );
1366 vlc_join( id->listen.thread, NULL );
1367 net_ListenClose( id->listen.fd );
1370 if( id->srtp != NULL )
1371 srtp_destroy( id->srtp );
1374 vlc_mutex_destroy( &id->lock_sink );
1376 /* Update SDP (sap/file) */
1377 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1378 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1380 vlc_object_detach( id );
1381 vlc_object_release( id );
1385 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1390 assert( p_stream->p_sys->p_mux == NULL );
1393 while( p_buffer != NULL )
1395 p_next = p_buffer->p_next;
1396 if( id->pf_packetize( id, p_buffer ) )
1399 block_Release( p_buffer );
1405 /****************************************************************************
1407 ****************************************************************************/
1408 static int SapSetup( sout_stream_t *p_stream )
1410 sout_stream_sys_t *p_sys = p_stream->p_sys;
1411 sout_instance_t *p_sout = p_stream->p_sout;
1413 /* Remove the previous session */
1414 if( p_sys->p_session != NULL)
1416 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1417 p_sys->p_session = NULL;
1420 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1422 announce_method_t *p_method = sout_SAPMethod();
1423 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1425 p_sys->psz_destination,
1427 sout_MethodRelease( p_method );
1433 /****************************************************************************
1435 ****************************************************************************/
1436 static int FileSetup( sout_stream_t *p_stream )
1438 sout_stream_sys_t *p_sys = p_stream->p_sys;
1441 if( p_sys->psz_sdp == NULL )
1442 return VLC_EGENERIC; /* too early */
1444 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1446 msg_Err( p_stream, "cannot open file '%s' (%m)",
1447 p_sys->psz_sdp_file );
1448 return VLC_EGENERIC;
1451 fputs( p_sys->psz_sdp, f );
1457 /****************************************************************************
1459 ****************************************************************************/
1460 static int HttpCallback( httpd_file_sys_t *p_args,
1461 httpd_file_t *, uint8_t *p_request,
1462 uint8_t **pp_data, int *pi_data );
1464 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1466 sout_stream_sys_t *p_sys = p_stream->p_sys;
1468 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1469 url->i_port > 0 ? url->i_port : 80 );
1470 if( p_sys->p_httpd_host )
1472 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1473 url->psz_path ? url->psz_path : "/",
1476 HttpCallback, (void*)p_sys );
1478 if( p_sys->p_httpd_file == NULL )
1480 return VLC_EGENERIC;
1485 static int HttpCallback( httpd_file_sys_t *p_args,
1486 httpd_file_t *f, uint8_t *p_request,
1487 uint8_t **pp_data, int *pi_data )
1489 VLC_UNUSED(f); VLC_UNUSED(p_request);
1490 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1492 vlc_mutex_lock( &p_sys->lock_sdp );
1493 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1495 *pi_data = strlen( p_sys->psz_sdp );
1496 *pp_data = malloc( *pi_data );
1497 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1504 vlc_mutex_unlock( &p_sys->lock_sdp );
1509 /****************************************************************************
1511 ****************************************************************************/
1512 static void* ThreadSend( vlc_object_t *p_this )
1515 # define ECONNREFUSED WSAECONNREFUSED
1516 # define ENOPROTOOPT WSAENOPROTOOPT
1517 # define EHOSTUNREACH WSAEHOSTUNREACH
1518 # define ENETUNREACH WSAENETUNREACH
1519 # define ENETDOWN WSAENETDOWN
1520 # define ENOBUFS WSAENOBUFS
1521 # define EAGAIN WSAEWOULDBLOCK
1522 # define EWOULDBLOCK WSAEWOULDBLOCK
1524 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1525 unsigned i_caching = id->i_caching;
1529 block_t *out = block_FifoGet( id->p_fifo );
1530 block_cleanup_push (out);
1534 { /* FIXME: this is awfully inefficient */
1535 size_t len = out->i_buffer;
1536 out = block_Realloc( out, 0, len + 10 );
1537 out->i_buffer = len;
1539 int canc = vlc_savecancel ();
1540 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1541 vlc_restorecancel (canc);
1545 msg_Dbg( id, "SRTP sending error: %m" );
1546 block_Release( out );
1550 out->i_buffer = len;
1554 mwait (out->i_dts + i_caching);
1559 ssize_t len = out->i_buffer;
1560 int canc = vlc_savecancel ();
1562 vlc_mutex_lock( &id->lock_sink );
1563 unsigned deadc = 0; /* How many dead sockets? */
1564 int deadv[id->sinkc]; /* Dead sockets list */
1566 for( int i = 0; i < id->sinkc; i++ )
1569 if( !id->srtp ) /* FIXME: SRTCP support */
1571 SendRTCP( id->sinkv[i].rtcp, out );
1573 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1577 /* Soft errors (e.g. ICMP): */
1578 case ECONNREFUSED: /* Port unreachable */
1581 case EPROTO: /* Protocol unreachable */
1583 case EHOSTUNREACH: /* Host unreachable */
1584 case ENETUNREACH: /* Network unreachable */
1585 case ENETDOWN: /* Entire network down */
1586 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1587 /* Transient congestion: */
1588 case ENOMEM: /* out of socket buffers */
1591 #if (EAGAIN != EWOULDBLOCK)
1597 deadv[deadc++] = id->sinkv[i].rtp_fd;
1599 vlc_mutex_unlock( &id->lock_sink );
1600 block_Release( out );
1602 for( unsigned i = 0; i < deadc; i++ )
1604 msg_Dbg( id, "removing socket %d", deadv[i] );
1605 rtp_del_sink( id, deadv[i] );
1607 vlc_restorecancel (canc);
1613 /* This thread dequeues incoming connections (DCCP streaming) */
1614 static void *rtp_listen_thread( void *data )
1616 sout_stream_id_t *id = data;
1618 assert( id->listen.fd != NULL );
1622 int fd = net_Accept( id, id->listen.fd );
1625 int canc = vlc_savecancel( );
1626 rtp_add_sink( id, fd, true );
1627 vlc_restorecancel( canc );
1634 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1636 rtp_sink_t sink = { fd, NULL };
1637 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1639 if( sink.rtcp == NULL )
1640 msg_Err( id, "RTCP failed!" );
1642 vlc_mutex_lock( &id->lock_sink );
1643 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1644 vlc_mutex_unlock( &id->lock_sink );
1648 void rtp_del_sink( sout_stream_id_t *id, int fd )
1650 rtp_sink_t sink = { fd, NULL };
1652 /* NOTE: must be safe to use if fd is not included */
1653 vlc_mutex_lock( &id->lock_sink );
1654 for( int i = 0; i < id->sinkc; i++ )
1656 if (id->sinkv[i].rtp_fd == fd)
1658 sink = id->sinkv[i];
1659 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1663 vlc_mutex_unlock( &id->lock_sink );
1665 CloseRTCP( sink.rtcp );
1666 net_Close( sink.rtp_fd );
1669 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1671 /* This will return values for the next packet.
1672 * Accounting for caching would not be totally trivial. */
1673 return id->i_sequence;
1676 /* FIXME: this is pretty bad - if we remove and then insert an ES
1677 * the number will get unsynched from inside RTSP */
1678 unsigned rtp_get_num( const sout_stream_id_t *id )
1680 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1683 vlc_mutex_lock( &p_sys->lock_es );
1684 for( i = 0; i < p_sys->i_es; i++ )
1686 if( id == p_sys->es[i] )
1689 vlc_mutex_unlock( &p_sys->lock_es );
1695 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1696 int b_marker, int64_t i_pts )
1698 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
1700 out->p_buffer[0] = 0x80;
1701 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1702 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1703 out->p_buffer[3] = ( id->i_sequence )&0xff;
1704 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1705 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1706 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1707 out->p_buffer[7] = ( i_timestamp )&0xff;
1709 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1715 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1717 block_FifoPut( id->p_fifo, out );
1721 * @return configured max RTP payload size (including payload type-specific
1722 * headers, excluding RTP and transport headers)
1724 size_t rtp_mtu (const sout_stream_id_t *id)
1726 return id->i_mtu - 12;
1729 /*****************************************************************************
1731 *****************************************************************************/
1733 /** Add an ES to a non-RTP muxed stream */
1734 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1736 sout_input_t *p_input;
1737 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1738 assert( p_mux != NULL );
1740 p_input = sout_MuxAddStream( p_mux, p_fmt );
1741 if( p_input == NULL )
1743 msg_Err( p_stream, "cannot add this stream to the muxer" );
1747 return (sout_stream_id_t *)p_input;
1751 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1754 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1755 assert( p_mux != NULL );
1757 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1762 /** Remove an ES from a non-RTP muxed stream */
1763 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1765 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1766 assert( p_mux != NULL );
1768 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1773 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1774 const block_t *p_buffer )
1776 sout_stream_sys_t *p_sys = p_stream->p_sys;
1777 sout_stream_id_t *id = p_sys->es[0];
1779 int64_t i_dts = p_buffer->i_dts;
1781 uint8_t *p_data = p_buffer->p_buffer;
1782 size_t i_data = p_buffer->i_buffer;
1783 size_t i_max = id->i_mtu - 12;
1785 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1791 /* output complete packet */
1792 if( p_sys->packet &&
1793 p_sys->packet->i_buffer + i_data > i_max )
1795 rtp_packetize_send( id, p_sys->packet );
1796 p_sys->packet = NULL;
1799 if( p_sys->packet == NULL )
1801 /* allocate a new packet */
1802 p_sys->packet = block_New( p_stream, id->i_mtu );
1803 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1804 p_sys->packet->i_dts = i_dts;
1805 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1806 i_dts += p_sys->packet->i_length;
1809 i_size = __MIN( i_data,
1810 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1812 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1815 p_sys->packet->i_buffer += i_size;
1824 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1827 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1833 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1835 p_next = p_buffer->p_next;
1836 block_Release( p_buffer );
1844 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1846 sout_access_out_t *p_grab;
1848 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1849 if( p_grab == NULL )
1852 p_grab->p_module = NULL;
1853 p_grab->psz_access = strdup( "grab" );
1854 p_grab->p_cfg = NULL;
1855 p_grab->psz_path = strdup( "" );
1856 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1857 p_grab->pf_seek = NULL;
1858 p_grab->pf_write = AccessOutGrabberWrite;
1859 vlc_object_attach( p_grab, p_stream );