1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
32 #include <vlc_common.h>
33 #include <vlc_plugin.h>
35 #include <vlc_block.h>
37 #include <vlc_httpd.h>
39 #include <vlc_network.h>
40 #include <vlc_charset.h>
41 #include <vlc_strings.h>
47 # include <sys/types.h>
50 # include <sys/stat.h>
52 #ifdef HAVE_LINUX_DCCP_H
53 # include <linux/dccp.h>
56 # define IPPROTO_DCCP 33
58 #ifndef IPPROTO_UDPLITE
59 # define IPPROTO_UDPLITE 136
66 /*****************************************************************************
68 *****************************************************************************/
70 #define DEST_TEXT N_("Destination")
71 #define DEST_LONGTEXT N_( \
72 "This is the output URL that will be used." )
73 #define SDP_TEXT N_("SDP")
74 #define SDP_LONGTEXT N_( \
75 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
76 "session will be made available. You must use an url: http://location to " \
77 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
78 "for the SDP to be announced via SAP." )
79 #define SAP_TEXT N_("SAP announcing")
80 #define SAP_LONGTEXT N_("Announce this session with SAP.")
81 #define MUX_TEXT N_("Muxer")
82 #define MUX_LONGTEXT N_( \
83 "This allows you to specify the muxer used for the streaming output. " \
84 "Default is to use no muxer (standard RTP stream)." )
86 #define NAME_TEXT N_("Session name")
87 #define NAME_LONGTEXT N_( \
88 "This is the name of the session that will be announced in the SDP " \
89 "(Session Descriptor)." )
90 #define DESC_TEXT N_("Session description")
91 #define DESC_LONGTEXT N_( \
92 "This allows you to give a short description with details about the stream, " \
93 "that will be announced in the SDP (Session Descriptor)." )
94 #define URL_TEXT N_("Session URL")
95 #define URL_LONGTEXT N_( \
96 "This allows you to give an URL with more details about the stream " \
97 "(often the website of the streaming organization), that will " \
98 "be announced in the SDP (Session Descriptor)." )
99 #define EMAIL_TEXT N_("Session email")
100 #define EMAIL_LONGTEXT N_( \
101 "This allows you to give a contact mail address for the stream, that will " \
102 "be announced in the SDP (Session Descriptor)." )
103 #define PHONE_TEXT N_("Session phone number")
104 #define PHONE_LONGTEXT N_( \
105 "This allows you to give a contact telephone number for the stream, that will " \
106 "be announced in the SDP (Session Descriptor)." )
108 #define PORT_TEXT N_("Port")
109 #define PORT_LONGTEXT N_( \
110 "This allows you to specify the base port for the RTP streaming." )
111 #define PORT_AUDIO_TEXT N_("Audio port")
112 #define PORT_AUDIO_LONGTEXT N_( \
113 "This allows you to specify the default audio port for the RTP streaming." )
114 #define PORT_VIDEO_TEXT N_("Video port")
115 #define PORT_VIDEO_LONGTEXT N_( \
116 "This allows you to specify the default video port for the RTP streaming." )
118 #define TTL_TEXT N_("Hop limit (TTL)")
119 #define TTL_LONGTEXT N_( \
120 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
121 "the multicast packets sent by the stream output (0 = use operating " \
122 "system built-in default).")
124 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
125 #define RTCP_MUX_LONGTEXT N_( \
126 "This sends and receives RTCP packet multiplexed over the same port " \
129 #define PROTO_TEXT N_("Transport protocol")
130 #define PROTO_LONGTEXT N_( \
131 "This selects which transport protocol to use for RTP." )
133 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
134 #define SRTP_KEY_LONGTEXT N_( \
135 "RTP packets will be integrity-protected and ciphered "\
136 "with this Secure RTP master shared secret key.")
138 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
139 #define SRTP_SALT_LONGTEXT N_( \
140 "Secure RTP requires a (non-secret) master salt value.")
142 static const char *const ppsz_protos[] = {
143 "dccp", "sctp", "tcp", "udp", "udplite",
146 static const char *const ppsz_protocols[] = {
147 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
150 #define RFC3016_TEXT N_("MP4A LATM")
151 #define RFC3016_LONGTEXT N_( \
152 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
154 static int Open ( vlc_object_t * );
155 static void Close( vlc_object_t * );
157 #define SOUT_CFG_PREFIX "sout-rtp-"
158 #define MAX_EMPTY_BLOCKS 200
161 set_shortname( N_("RTP"));
162 set_description( N_("RTP stream output") );
163 set_capability( "sout stream", 0 );
164 add_shortcut( "rtp" );
165 set_category( CAT_SOUT );
166 set_subcategory( SUBCAT_SOUT_STREAM );
168 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
169 DEST_LONGTEXT, true );
171 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
172 SDP_LONGTEXT, true );
173 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
174 MUX_LONGTEXT, true );
175 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
178 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
179 NAME_LONGTEXT, true );
180 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
181 DESC_LONGTEXT, true );
182 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
183 URL_LONGTEXT, true );
184 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
185 EMAIL_LONGTEXT, true );
186 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
187 PHONE_LONGTEXT, true );
189 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
190 PROTO_LONGTEXT, false );
191 change_string_list( ppsz_protos, ppsz_protocols, NULL );
192 add_integer( SOUT_CFG_PREFIX "port", 50004, NULL, PORT_TEXT,
193 PORT_LONGTEXT, true );
194 add_integer( SOUT_CFG_PREFIX "port-audio", 50000, NULL, PORT_AUDIO_TEXT,
195 PORT_AUDIO_LONGTEXT, true );
196 add_integer( SOUT_CFG_PREFIX "port-video", 50002, NULL, PORT_VIDEO_TEXT,
197 PORT_VIDEO_LONGTEXT, true );
199 add_integer( SOUT_CFG_PREFIX "ttl", 0, NULL, TTL_TEXT,
200 TTL_LONGTEXT, true );
201 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
202 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );
204 add_string( SOUT_CFG_PREFIX "key", "", NULL,
205 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
206 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
207 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );
209 add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
210 RFC3016_LONGTEXT, false );
212 set_callbacks( Open, Close );
215 /*****************************************************************************
216 * Exported prototypes
217 *****************************************************************************/
218 static const char *const ppsz_sout_options[] = {
219 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
220 "sap", "description", "url", "email", "phone",
221 "proto", "rtcp-mux", "key", "salt",
225 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
226 static int Del ( sout_stream_t *, sout_stream_id_t * );
227 static int Send( sout_stream_t *, sout_stream_id_t *,
229 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
230 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
231 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
234 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
235 static void* ThreadSend( vlc_object_t *p_this );
237 static void SDPHandleUrl( sout_stream_t *, const char * );
239 static int SapSetup( sout_stream_t *p_stream );
240 static int FileSetup( sout_stream_t *p_stream );
241 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
243 struct sout_stream_sys_t
247 vlc_mutex_t lock_sdp;
250 bool b_export_sdp_file;
255 session_descriptor_t *p_session;
258 httpd_host_t *p_httpd_host;
259 httpd_file_t *p_httpd_file;
265 char *psz_destination;
269 uint16_t i_port_audio;
270 uint16_t i_port_video;
274 /* when need to use a private one or when using muxer */
277 /* in case we do TS/PS over rtp */
279 sout_access_out_t *p_grab;
285 sout_stream_id_t **es;
288 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
290 typedef struct rtp_sink_t
296 struct sout_stream_id_t
300 sout_stream_t *p_stream;
303 uint8_t i_payload_type;
315 /* Packetizer specific fields */
317 srtp_session_t *srtp;
318 pf_rtp_packetizer_t pf_packetize;
321 vlc_mutex_t lock_sink;
324 rtsp_stream_id_t *rtsp_id;
327 block_fifo_t *p_fifo;
331 /*****************************************************************************
333 *****************************************************************************/
334 static int Open( vlc_object_t *p_this )
336 sout_stream_t *p_stream = (sout_stream_t*)p_this;
337 sout_instance_t *p_sout = p_stream->p_sout;
338 sout_stream_sys_t *p_sys = NULL;
339 config_chain_t *p_cfg = NULL;
343 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
344 ppsz_sout_options, p_stream->p_cfg );
346 p_sys = malloc( sizeof( sout_stream_sys_t ) );
350 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
352 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
353 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
354 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
355 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
357 p_sys->psz_sdp_file = NULL;
359 if( p_sys->i_port_audio == p_sys->i_port_video )
361 msg_Err( p_stream, "audio and video port cannot be the same" );
362 p_sys->i_port_audio = 0;
363 p_sys->i_port_video = 0;
366 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
368 if( !strcmp( p_cfg->psz_name, "sdp" )
369 && ( p_cfg->psz_value != NULL )
370 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
378 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
381 if( !strncasecmp( psz, "rtsp:", 5 ) )
387 /* Transport protocol */
388 p_sys->proto = IPPROTO_UDP;
389 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
391 if ((psz == NULL) || !strcasecmp (psz, "udp"))
392 (void)0; /* default */
394 if (!strcasecmp (psz, "dccp"))
396 p_sys->proto = IPPROTO_DCCP;
397 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
401 if (!strcasecmp (psz, "sctp"))
403 p_sys->proto = IPPROTO_TCP;
404 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
409 if (!strcasecmp (psz, "tcp"))
411 p_sys->proto = IPPROTO_TCP;
412 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
416 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
417 p_sys->proto = IPPROTO_UDPLITE;
419 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
422 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
424 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
426 msg_Err( p_stream, "missing destination and not in RTSP mode" );
431 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
432 if( p_sys->i_ttl == 0 )
434 /* Normally, we should let the default hop limit up to the core,
435 * but we have to know it to build our SDP properly, which is why
436 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
438 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
441 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
443 p_sys->i_payload_type = 96;
447 p_sys->psz_sdp = NULL;
449 p_sys->b_export_sap = false;
450 p_sys->b_export_sdp_file = false;
451 p_sys->p_session = NULL;
453 p_sys->p_httpd_host = NULL;
454 p_sys->p_httpd_file = NULL;
456 p_stream->p_sys = p_sys;
458 vlc_mutex_init( &p_sys->lock_sdp );
459 vlc_mutex_init( &p_sys->lock_es );
461 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
464 sout_stream_id_t *id;
466 /* Check muxer type */
467 if( strncasecmp( psz, "ps", 2 )
468 && strncasecmp( psz, "mpeg1", 5 )
469 && strncasecmp( psz, "ts", 2 ) )
471 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
473 vlc_mutex_destroy( &p_sys->lock_sdp );
474 vlc_mutex_destroy( &p_sys->lock_es );
479 p_sys->p_grab = GrabberCreate( p_stream );
480 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
483 if( p_sys->p_mux == NULL )
485 msg_Err( p_stream, "cannot create muxer" );
486 sout_AccessOutDelete( p_sys->p_grab );
487 vlc_mutex_destroy( &p_sys->lock_sdp );
488 vlc_mutex_destroy( &p_sys->lock_es );
493 id = Add( p_stream, NULL );
496 sout_MuxDelete( p_sys->p_mux );
497 sout_AccessOutDelete( p_sys->p_grab );
498 vlc_mutex_destroy( &p_sys->lock_sdp );
499 vlc_mutex_destroy( &p_sys->lock_es );
504 p_sys->packet = NULL;
506 p_stream->pf_add = MuxAdd;
507 p_stream->pf_del = MuxDel;
508 p_stream->pf_send = MuxSend;
513 p_sys->p_grab = NULL;
515 p_stream->pf_add = Add;
516 p_stream->pf_del = Del;
517 p_stream->pf_send = Send;
520 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
521 SDPHandleUrl( p_stream, "sap" );
523 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
526 config_chain_t *p_cfg;
528 SDPHandleUrl( p_stream, psz );
530 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
532 if( !strcmp( p_cfg->psz_name, "sdp" ) )
534 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
537 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
538 if( !strcmp( p_cfg->psz_value, psz ) )
541 SDPHandleUrl( p_stream, p_cfg->psz_value );
547 /* update p_sout->i_out_pace_nocontrol */
548 p_stream->p_sout->i_out_pace_nocontrol++;
553 /*****************************************************************************
555 *****************************************************************************/
556 static void Close( vlc_object_t * p_this )
558 sout_stream_t *p_stream = (sout_stream_t*)p_this;
559 sout_stream_sys_t *p_sys = p_stream->p_sys;
561 /* update p_sout->i_out_pace_nocontrol */
562 p_stream->p_sout->i_out_pace_nocontrol--;
566 assert( p_sys->i_es == 1 );
567 Del( p_stream, p_sys->es[0] );
569 sout_MuxDelete( p_sys->p_mux );
570 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->b_export_sdp_file )
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 )
641 msg_Err( p_stream, "cannot export SDP as RTSP" );
644 if( p_sys->p_mux != NULL )
646 sout_stream_id_t *id = p_sys->es[0];
647 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
648 p_sys->psz_destination, p_sys->i_ttl,
649 id->i_port, id->i_port + 1 );
652 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
653 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
655 p_sys->b_export_sap = true;
656 SapSetup( p_stream );
658 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
660 if( p_sys->b_export_sdp_file )
662 msg_Err( p_stream, "you can use sdp=file:// only once" );
665 p_sys->b_export_sdp_file = true;
666 psz_url = &psz_url[5];
667 if( psz_url[0] == '/' && psz_url[1] == '/' )
669 p_sys->psz_sdp_file = strdup( psz_url );
673 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
678 vlc_UrlClean( &url );
681 /*****************************************************************************
683 *****************************************************************************/
685 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
687 const sout_stream_sys_t *p_sys = p_stream->p_sys;
689 struct sockaddr_storage dst;
693 * When we have a fixed destination (typically when we do multicast),
694 * we need to put the actual port numbers in the SDP.
695 * When there is no fixed destination, we only support RTSP unicast
696 * on-demand setup, so we should rather let the clients decide which ports
698 * When there is both a fixed destination and RTSP unicast, we need to
699 * put port numbers used by the fixed destination, otherwise the SDP would
700 * become totally incorrect for multicast use. It should be noted that
701 * port numbers from SDP with RTSP are only "recommendation" from the
702 * server to the clients (per RFC2326), so only broken clients will fail
703 * to handle this properly. There is no solution but to use two differents
704 * output chain with two different RTSP URLs if you need to handle this
709 if( p_sys->psz_destination != NULL )
713 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
714 dstlen = sizeof( dst );
715 if( p_sys->es[0]->listen_fd != NULL )
716 getsockname( p_sys->es[0]->listen_fd[0],
717 (struct sockaddr *)&dst, &dstlen );
719 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
720 (struct sockaddr *)&dst, &dstlen );
726 /* Dummy destination address for RTSP */
727 memset (&dst, 0, sizeof( struct sockaddr_in ) );
728 dst.ss_family = AF_INET;
732 dstlen = sizeof( struct sockaddr_in );
735 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
736 NULL, 0, (struct sockaddr *)&dst, dstlen );
737 if( psz_sdp == NULL )
740 /* TODO: a=source-filter */
741 if( p_sys->rtcp_mux )
742 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
744 if( rtsp_url != NULL )
745 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
747 /* FIXME: locking?! */
748 for( i = 0; i < p_sys->i_es; i++ )
750 sout_stream_id_t *id = p_sys->es[i];
751 const char *mime_major; /* major MIME type */
752 const char *proto = "RTP/AVP"; /* protocol */
757 mime_major = "video";
760 mime_major = "audio";
769 if( rtsp_url == NULL )
771 switch( p_sys->proto )
776 proto = "TCP/RTP/AVP";
779 proto = "DCCP/RTP/AVP";
781 case IPPROTO_UDPLITE:
786 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
787 id->i_payload_type, false, id->i_bitrate,
788 id->psz_enc, id->i_clock_rate, id->i_channels,
791 if( rtsp_url != NULL )
793 assert( strlen( rtsp_url ) > 0 );
794 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
795 sdp_AddAttribute ( &psz_sdp, "control",
796 addslash ? "%s/trackID=%u" : "%strackID=%u",
801 if( id->listen_fd != NULL )
802 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
803 if( p_sys->proto == IPPROTO_DCCP )
804 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
805 "SC:RTP%c", toupper( mime_major[0] ) );
812 /*****************************************************************************
814 *****************************************************************************/
816 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
818 static const char hex[16] = "0123456789abcdef";
821 for( i = 0; i < i_data; i++ )
823 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
824 s[2*i+1] = hex[(p_data[i] )&0xf];
830 * Shrink the MTU down to a fixed packetization time (for audio).
833 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
835 /* Samples per second */
836 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
837 bytes *= id->i_channels;
840 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
841 id->i_mtu = 12 + spl;
842 else /* MTU is too small for ptime, align to a sample boundary */
843 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
846 /** Add an ES as a new RTP stream */
847 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
849 /* NOTE: As a special case, if we use a non-RTP
850 * mux (TS/PS), then p_fmt is NULL. */
851 sout_stream_sys_t *p_sys = p_stream->p_sys;
852 sout_stream_id_t *id;
853 int i_port, cscov = -1;
856 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
859 vlc_object_attach( id, p_stream );
861 /* Choose the port */
866 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
868 i_port = p_sys->i_port_audio;
869 p_sys->i_port_audio = 0;
872 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
874 i_port = p_sys->i_port_video;
875 p_sys->i_port_video = 0;
880 if( p_sys->i_port != p_sys->i_port_audio
881 && p_sys->i_port != p_sys->i_port_video )
883 i_port = p_sys->i_port;
890 id->p_stream = p_stream;
892 id->i_sequence = rand()&0xffff;
893 id->i_payload_type = p_sys->i_payload_type;
894 id->ssrc[0] = rand()&0xff;
895 id->ssrc[1] = rand()&0xff;
896 id->ssrc[2] = rand()&0xff;
897 id->ssrc[3] = rand()&0xff;
901 id->i_clock_rate = 90000; /* most common case for video */
906 id->i_cat = p_fmt->i_cat;
907 if( p_fmt->i_cat == AUDIO_ES )
909 id->i_clock_rate = p_fmt->audio.i_rate;
910 id->i_channels = p_fmt->audio.i_channels;
912 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
916 id->i_cat = VIDEO_ES;
920 id->i_mtu = config_GetInt( p_stream, "mtu" );
921 if( id->i_mtu <= 12 + 16 )
922 id->i_mtu = 576 - 20 - 8; /* pessimistic */
923 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
926 id->pf_packetize = NULL;
928 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
931 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
932 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
933 if (id->srtp == NULL)
939 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
940 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
945 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
948 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
951 vlc_mutex_init( &id->lock_sink );
956 id->listen_fd = NULL;
959 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
961 if( p_sys->psz_destination != NULL )
962 switch( p_sys->proto )
969 case VIDEO_ES: code = "RTPV"; break;
970 case AUDIO_ES: code = "RTPARTPV"; break;
971 case SPU_ES: code = "RTPTRPTV"; break;
972 default: code = "RTPORTPV"; break;
974 var_SetString (p_stream, "dccp-service", code);
977 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
978 p_sys->psz_destination, i_port,
980 if( id->listen_fd == NULL )
982 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
989 int ttl = (p_sys->i_ttl > 0) ? p_sys->i_ttl : -1;
990 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
991 i_port, ttl, p_sys->proto );
994 msg_Err( p_stream, "cannot create RTP socket" );
997 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1003 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1005 if( psz == NULL ) /* Uho! */
1008 if( strncmp( psz, "ts", 2 ) == 0 )
1010 id->i_payload_type = 33;
1011 id->psz_enc = "MP2T";
1015 id->psz_enc = "MP2P";
1020 switch( p_fmt->i_codec )
1022 case VLC_FOURCC( 'u', 'l', 'a', 'w' ):
1023 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1024 id->i_payload_type = 0;
1025 id->psz_enc = "PCMU";
1026 id->pf_packetize = rtp_packetize_split;
1027 rtp_set_ptime (id, 20, 1);
1029 case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
1030 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1031 id->i_payload_type = 8;
1032 id->psz_enc = "PCMA";
1033 id->pf_packetize = rtp_packetize_split;
1034 rtp_set_ptime (id, 20, 1);
1036 case VLC_FOURCC( 's', '1', '6', 'b' ):
1037 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1039 id->i_payload_type = 11;
1041 else if( p_fmt->audio.i_channels == 2 &&
1042 p_fmt->audio.i_rate == 44100 )
1044 id->i_payload_type = 10;
1046 id->psz_enc = "L16";
1047 id->pf_packetize = rtp_packetize_split;
1048 rtp_set_ptime (id, 20, 2);
1050 case VLC_FOURCC( 'u', '8', ' ', ' ' ):
1052 id->pf_packetize = rtp_packetize_split;
1053 rtp_set_ptime (id, 20, 1);
1055 case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
1056 case VLC_FOURCC( 'm', 'p', '3', ' ' ):
1057 id->i_payload_type = 14;
1058 id->psz_enc = "MPA";
1059 id->i_clock_rate = 90000; /* not 44100 */
1060 id->pf_packetize = rtp_packetize_mpa;
1062 case VLC_FOURCC( 'm', 'p', 'g', 'v' ):
1063 id->i_payload_type = 32;
1064 id->psz_enc = "MPV";
1065 id->pf_packetize = rtp_packetize_mpv;
1067 case VLC_FOURCC( 'a', '5', '2', ' ' ):
1068 id->psz_enc = "ac3";
1069 id->pf_packetize = rtp_packetize_ac3;
1071 case VLC_FOURCC( 'H', '2', '6', '3' ):
1072 id->psz_enc = "H263-1998";
1073 id->pf_packetize = rtp_packetize_h263;
1075 case VLC_FOURCC( 'h', '2', '6', '4' ):
1076 id->psz_enc = "H264";
1077 id->pf_packetize = rtp_packetize_h264;
1078 id->psz_fmtp = NULL;
1080 if( p_fmt->i_extra > 0 )
1082 uint8_t *p_buffer = p_fmt->p_extra;
1083 int i_buffer = p_fmt->i_extra;
1084 char *p_64_sps = NULL;
1085 char *p_64_pps = NULL;
1088 while( i_buffer > 4 &&
1089 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1090 p_buffer[2] == 0 && p_buffer[3] == 1 )
1092 const int i_nal_type = p_buffer[4]&0x1f;
1096 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1099 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1101 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1103 /* we found another startcode */
1108 if( i_nal_type == 7 )
1110 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1111 sprintf_hexa( hexa, &p_buffer[5], 3 );
1113 else if( i_nal_type == 8 )
1115 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1121 if( p_64_sps && p_64_pps &&
1122 ( asprintf( &id->psz_fmtp,
1123 "packetization-mode=1;profile-level-id=%s;"
1124 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1125 p_64_pps ) == -1 ) )
1126 id->psz_fmtp = NULL;
1131 id->psz_fmtp = strdup( "packetization-mode=1" );
1134 case VLC_FOURCC( 'm', 'p', '4', 'v' ):
1136 char hexa[2*p_fmt->i_extra +1];
1138 id->psz_enc = "MP4V-ES";
1139 id->pf_packetize = rtp_packetize_split;
1140 if( p_fmt->i_extra > 0 )
1142 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1143 if( asprintf( &id->psz_fmtp,
1144 "profile-level-id=3; config=%s;", hexa ) == -1 )
1145 id->psz_fmtp = NULL;
1149 case VLC_FOURCC( 'm', 'p', '4', 'a' ):
1153 char hexa[2*p_fmt->i_extra +1];
1155 id->psz_enc = "mpeg4-generic";
1156 id->pf_packetize = rtp_packetize_mp4a;
1157 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1158 if( asprintf( &id->psz_fmtp,
1159 "streamtype=5; profile-level-id=15; "
1160 "mode=AAC-hbr; config=%s; SizeLength=13; "
1161 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1163 id->psz_fmtp = NULL;
1169 unsigned char config[6];
1170 unsigned int aacsrates[15] = {
1171 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1172 16000, 12000, 11025, 8000, 7350, 0, 0 };
1174 for( i = 0; i < 15; i++ )
1175 if( p_fmt->audio.i_rate == aacsrates[i] )
1181 config[3]=p_fmt->audio.i_channels<<4;
1185 id->psz_enc = "MP4A-LATM";
1186 id->pf_packetize = rtp_packetize_mp4a_latm;
1187 sprintf_hexa( hexa, config, 6 );
1188 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1189 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1190 id->psz_fmtp = NULL;
1194 case VLC_FOURCC( 's', 'a', 'm', 'r' ):
1195 id->psz_enc = "AMR";
1196 id->psz_fmtp = strdup( "octet-align=1" );
1197 id->pf_packetize = rtp_packetize_amr;
1199 case VLC_FOURCC( 's', 'a', 'w', 'b' ):
1200 id->psz_enc = "AMR-WB";
1201 id->psz_fmtp = strdup( "octet-align=1" );
1202 id->pf_packetize = rtp_packetize_amr;
1204 case VLC_FOURCC( 's', 'p', 'x', ' ' ):
1205 id->i_payload_type = p_sys->i_payload_type++;
1206 id->psz_enc = "SPEEX";
1207 id->pf_packetize = rtp_packetize_spx;
1209 case VLC_FOURCC( 't', '1', '4', '0' ):
1210 id->psz_enc = "t140" ;
1211 id->i_clock_rate = 1000;
1212 id->pf_packetize = rtp_packetize_t140;
1216 msg_Err( p_stream, "cannot add this stream (unsupported "
1217 "codec:%4.4s)", (char*)&p_fmt->i_codec );
1222 cscov += 8 /* UDP */ + 12 /* RTP */;
1224 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1226 if( id->i_payload_type == p_sys->i_payload_type )
1227 p_sys->i_payload_type++;
1229 if( p_sys->rtsp != NULL )
1230 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1231 GetDWBE( id->ssrc ),
1232 p_sys->psz_destination,
1233 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1235 id->p_fifo = block_FifoNew();
1236 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1237 VLC_THREAD_PRIORITY_HIGHEST, false ) )
1240 /* Update p_sys context */
1241 vlc_mutex_lock( &p_sys->lock_es );
1242 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1243 vlc_mutex_unlock( &p_sys->lock_es );
1245 psz_sdp = SDPGenerate( p_stream, NULL );
1247 vlc_mutex_lock( &p_sys->lock_sdp );
1248 free( p_sys->psz_sdp );
1249 p_sys->psz_sdp = psz_sdp;
1250 vlc_mutex_unlock( &p_sys->lock_sdp );
1252 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1254 /* Update SDP (sap/file) */
1255 if( p_sys->b_export_sap ) SapSetup( p_stream );
1256 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1261 Del( p_stream, id );
1265 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1267 sout_stream_sys_t *p_sys = p_stream->p_sys;
1269 if( id->p_fifo != NULL )
1271 vlc_object_kill( id );
1272 vlc_thread_join( id );
1273 block_FifoRelease( id->p_fifo );
1276 vlc_mutex_lock( &p_sys->lock_es );
1277 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1278 vlc_mutex_unlock( &p_sys->lock_es );
1281 if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
1282 p_sys->i_port_audio = id->i_port;
1283 if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
1284 p_sys->i_port_video = id->i_port;
1286 free( id->psz_fmtp );
1289 RtspDelId( p_sys->rtsp, id->rtsp_id );
1291 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1292 if( id->listen_fd != NULL )
1293 net_ListenClose( id->listen_fd );
1294 if( id->srtp != NULL )
1295 srtp_destroy( id->srtp );
1297 vlc_mutex_destroy( &id->lock_sink );
1299 /* Update SDP (sap/file) */
1300 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1301 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1303 vlc_object_detach( id );
1304 vlc_object_release( id );
1308 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1313 assert( p_stream->p_sys->p_mux == NULL );
1316 while( p_buffer != NULL )
1318 p_next = p_buffer->p_next;
1319 if( id->pf_packetize( id, p_buffer ) )
1322 block_Release( p_buffer );
1328 /****************************************************************************
1330 ****************************************************************************/
1331 static int SapSetup( sout_stream_t *p_stream )
1333 sout_stream_sys_t *p_sys = p_stream->p_sys;
1334 sout_instance_t *p_sout = p_stream->p_sout;
1336 /* Remove the previous session */
1337 if( p_sys->p_session != NULL)
1339 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1340 p_sys->p_session = NULL;
1343 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1345 announce_method_t *p_method = sout_SAPMethod();
1346 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1348 p_sys->psz_destination,
1350 sout_MethodRelease( p_method );
1356 /****************************************************************************
1358 ****************************************************************************/
1359 static int FileSetup( sout_stream_t *p_stream )
1361 sout_stream_sys_t *p_sys = p_stream->p_sys;
1364 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1366 msg_Err( p_stream, "cannot open file '%s' (%m)",
1367 p_sys->psz_sdp_file );
1368 return VLC_EGENERIC;
1371 fputs( p_sys->psz_sdp, f );
1377 /****************************************************************************
1379 ****************************************************************************/
1380 static int HttpCallback( httpd_file_sys_t *p_args,
1381 httpd_file_t *, uint8_t *p_request,
1382 uint8_t **pp_data, int *pi_data );
1384 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1386 sout_stream_sys_t *p_sys = p_stream->p_sys;
1388 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1389 url->i_port > 0 ? url->i_port : 80 );
1390 if( p_sys->p_httpd_host )
1392 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1393 url->psz_path ? url->psz_path : "/",
1396 HttpCallback, (void*)p_sys );
1398 if( p_sys->p_httpd_file == NULL )
1400 return VLC_EGENERIC;
1405 static int HttpCallback( httpd_file_sys_t *p_args,
1406 httpd_file_t *f, uint8_t *p_request,
1407 uint8_t **pp_data, int *pi_data )
1409 VLC_UNUSED(f); VLC_UNUSED(p_request);
1410 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1412 vlc_mutex_lock( &p_sys->lock_sdp );
1413 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1415 *pi_data = strlen( p_sys->psz_sdp );
1416 *pp_data = malloc( *pi_data );
1417 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1424 vlc_mutex_unlock( &p_sys->lock_sdp );
1429 /****************************************************************************
1431 ****************************************************************************/
1432 static void* ThreadSend( vlc_object_t *p_this )
1434 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1435 unsigned i_caching = id->i_caching;
1439 block_t *out = block_FifoGet( id->p_fifo );
1440 block_cleanup_push (out);
1443 { /* FIXME: this is awfully inefficient */
1444 size_t len = out->i_buffer;
1445 out = block_Realloc( out, 0, len + 10 );
1446 out->i_buffer = len;
1448 int canc = vlc_savecancel ();
1449 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1450 vlc_restorecancel (canc);
1454 msg_Dbg( id, "SRTP sending error: %m" );
1455 block_Release( out );
1459 out->i_buffer = len;
1463 mwait (out->i_dts + i_caching);
1468 ssize_t len = out->i_buffer;
1469 int canc = vlc_savecancel ();
1471 vlc_mutex_lock( &id->lock_sink );
1472 unsigned deadc = 0; /* How many dead sockets? */
1473 int deadv[id->sinkc]; /* Dead sockets list */
1475 for( int i = 0; i < id->sinkc; i++ )
1477 if( !id->srtp ) /* FIXME: SRTCP support */
1478 SendRTCP( id->sinkv[i].rtcp, out );
1480 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1482 /* Retry sending to root out soft-errors */
1483 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1486 deadv[deadc++] = id->sinkv[i].rtp_fd;
1488 vlc_mutex_unlock( &id->lock_sink );
1489 block_Release( out );
1491 for( unsigned i = 0; i < deadc; i++ )
1493 msg_Dbg( id, "removing socket %d", deadv[i] );
1494 rtp_del_sink( id, deadv[i] );
1497 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1498 while( id->listen_fd != NULL )
1500 int fd = net_Accept( id, id->listen_fd, 0 );
1503 msg_Dbg( id, "adding socket %d", fd );
1504 rtp_add_sink( id, fd, true );
1506 vlc_restorecancel (canc);
1511 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1513 rtp_sink_t sink = { fd, NULL };
1514 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1516 if( sink.rtcp == NULL )
1517 msg_Err( id, "RTCP failed!" );
1519 vlc_mutex_lock( &id->lock_sink );
1520 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1521 vlc_mutex_unlock( &id->lock_sink );
1525 void rtp_del_sink( sout_stream_id_t *id, int fd )
1527 rtp_sink_t sink = { fd, NULL };
1529 /* NOTE: must be safe to use if fd is not included */
1530 vlc_mutex_lock( &id->lock_sink );
1531 for( int i = 0; i < id->sinkc; i++ )
1533 if (id->sinkv[i].rtp_fd == fd)
1535 sink = id->sinkv[i];
1536 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1540 vlc_mutex_unlock( &id->lock_sink );
1542 CloseRTCP( sink.rtcp );
1543 net_Close( sink.rtp_fd );
1546 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1548 /* This will return values for the next packet.
1549 * Accounting for caching would not be totally trivial. */
1550 return id->i_sequence;
1553 /* FIXME: this is pretty bad - if we remove and then insert an ES
1554 * the number will get unsynched from inside RTSP */
1555 unsigned rtp_get_num( const sout_stream_id_t *id )
1557 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1560 vlc_mutex_lock( &p_sys->lock_es );
1561 for( i = 0; i < p_sys->i_es; i++ )
1563 if( id == p_sys->es[i] )
1566 vlc_mutex_unlock( &p_sys->lock_es );
1572 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1573 int b_marker, int64_t i_pts )
1575 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
1577 out->p_buffer[0] = 0x80;
1578 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1579 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1580 out->p_buffer[3] = ( id->i_sequence )&0xff;
1581 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1582 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1583 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1584 out->p_buffer[7] = ( i_timestamp )&0xff;
1586 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1592 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1594 block_FifoPut( id->p_fifo, out );
1598 * @return configured max RTP payload size (including payload type-specific
1599 * headers, excluding RTP and transport headers)
1601 size_t rtp_mtu (const sout_stream_id_t *id)
1603 return id->i_mtu - 12;
1606 /*****************************************************************************
1608 *****************************************************************************/
1610 /** Add an ES to a non-RTP muxed stream */
1611 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1613 sout_input_t *p_input;
1614 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1615 assert( p_mux != NULL );
1617 p_input = sout_MuxAddStream( p_mux, p_fmt );
1618 if( p_input == NULL )
1620 msg_Err( p_stream, "cannot add this stream to the muxer" );
1624 return (sout_stream_id_t *)p_input;
1628 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1631 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1632 assert( p_mux != NULL );
1634 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1639 /** Remove an ES from a non-RTP muxed stream */
1640 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1642 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1643 assert( p_mux != NULL );
1645 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1650 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1651 const block_t *p_buffer )
1653 sout_stream_sys_t *p_sys = p_stream->p_sys;
1654 sout_stream_id_t *id = p_sys->es[0];
1656 int64_t i_dts = p_buffer->i_dts;
1658 uint8_t *p_data = p_buffer->p_buffer;
1659 size_t i_data = p_buffer->i_buffer;
1660 size_t i_max = id->i_mtu - 12;
1662 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1668 /* output complete packet */
1669 if( p_sys->packet &&
1670 p_sys->packet->i_buffer + i_data > i_max )
1672 rtp_packetize_send( id, p_sys->packet );
1673 p_sys->packet = NULL;
1676 if( p_sys->packet == NULL )
1678 /* allocate a new packet */
1679 p_sys->packet = block_New( p_stream, id->i_mtu );
1680 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1681 p_sys->packet->i_dts = i_dts;
1682 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1683 i_dts += p_sys->packet->i_length;
1686 i_size = __MIN( i_data,
1687 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1689 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1692 p_sys->packet->i_buffer += i_size;
1701 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1704 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1710 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1712 p_next = p_buffer->p_next;
1713 block_Release( p_buffer );
1721 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1723 sout_access_out_t *p_grab;
1725 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1726 if( p_grab == NULL )
1729 p_grab->p_module = NULL;
1730 p_grab->p_sout = p_stream->p_sout;
1731 p_grab->psz_access = strdup( "grab" );
1732 p_grab->p_cfg = NULL;
1733 p_grab->psz_path = strdup( "" );
1734 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1735 p_grab->pf_seek = NULL;
1736 p_grab->pf_write = AccessOutGrabberWrite;
1737 vlc_object_attach( p_grab, p_stream );