1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
32 #include <vlc_common.h>
33 #include <vlc_plugin.h>
35 #include <vlc_block.h>
37 #include <vlc_httpd.h>
39 #include <vlc_network.h>
40 #include <vlc_charset.h>
41 #include <vlc_strings.h>
48 # include <sys/types.h>
51 # include <sys/stat.h>
53 #ifdef HAVE_LINUX_DCCP_H
54 # include <linux/dccp.h>
57 # define IPPROTO_DCCP 33
59 #ifndef IPPROTO_UDPLITE
60 # define IPPROTO_UDPLITE 136
67 /*****************************************************************************
69 *****************************************************************************/
71 #define DEST_TEXT N_("Destination")
72 #define DEST_LONGTEXT N_( \
73 "This is the output URL that will be used." )
74 #define SDP_TEXT N_("SDP")
75 #define SDP_LONGTEXT N_( \
76 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
77 "session will be made available. You must use an url: http://location to " \
78 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
79 "for the SDP to be announced via SAP." )
80 #define SAP_TEXT N_("SAP announcing")
81 #define SAP_LONGTEXT N_("Announce this session with SAP.")
82 #define MUX_TEXT N_("Muxer")
83 #define MUX_LONGTEXT N_( \
84 "This allows you to specify the muxer used for the streaming output. " \
85 "Default is to use no muxer (standard RTP stream)." )
87 #define NAME_TEXT N_("Session name")
88 #define NAME_LONGTEXT N_( \
89 "This is the name of the session that will be announced in the SDP " \
90 "(Session Descriptor)." )
91 #define DESC_TEXT N_("Session description")
92 #define DESC_LONGTEXT N_( \
93 "This allows you to give a short description with details about the stream, " \
94 "that will be announced in the SDP (Session Descriptor)." )
95 #define URL_TEXT N_("Session URL")
96 #define URL_LONGTEXT N_( \
97 "This allows you to give an URL with more details about the stream " \
98 "(often the website of the streaming organization), that will " \
99 "be announced in the SDP (Session Descriptor)." )
100 #define EMAIL_TEXT N_("Session email")
101 #define EMAIL_LONGTEXT N_( \
102 "This allows you to give a contact mail address for the stream, that will " \
103 "be announced in the SDP (Session Descriptor)." )
104 #define PHONE_TEXT N_("Session phone number")
105 #define PHONE_LONGTEXT N_( \
106 "This allows you to give a contact telephone number for the stream, that will " \
107 "be announced in the SDP (Session Descriptor)." )
109 #define PORT_TEXT N_("Port")
110 #define PORT_LONGTEXT N_( \
111 "This allows you to specify the base port for the RTP streaming." )
112 #define PORT_AUDIO_TEXT N_("Audio port")
113 #define PORT_AUDIO_LONGTEXT N_( \
114 "This allows you to specify the default audio port for the RTP streaming." )
115 #define PORT_VIDEO_TEXT N_("Video port")
116 #define PORT_VIDEO_LONGTEXT N_( \
117 "This allows you to specify the default video port for the RTP streaming." )
119 #define TTL_TEXT N_("Hop limit (TTL)")
120 #define TTL_LONGTEXT N_( \
121 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
122 "the multicast packets sent by the stream output (-1 = use operating " \
123 "system built-in default).")
125 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
126 #define RTCP_MUX_LONGTEXT N_( \
127 "This sends and receives RTCP packet multiplexed over the same port " \
130 #define PROTO_TEXT N_("Transport protocol")
131 #define PROTO_LONGTEXT N_( \
132 "This selects which transport protocol to use for RTP." )
134 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
135 #define SRTP_KEY_LONGTEXT N_( \
136 "RTP packets will be integrity-protected and ciphered "\
137 "with this Secure RTP master shared secret key.")
139 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
140 #define SRTP_SALT_LONGTEXT N_( \
141 "Secure RTP requires a (non-secret) master salt value.")
143 static const char *const ppsz_protos[] = {
144 "dccp", "sctp", "tcp", "udp", "udplite",
147 static const char *const ppsz_protocols[] = {
148 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
151 #define RFC3016_TEXT N_("MP4A LATM")
152 #define RFC3016_LONGTEXT N_( \
153 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
155 static int Open ( vlc_object_t * );
156 static void Close( vlc_object_t * );
158 #define SOUT_CFG_PREFIX "sout-rtp-"
159 #define MAX_EMPTY_BLOCKS 200
162 set_shortname( N_("RTP"))
163 set_description( N_("RTP stream output") )
164 set_capability( "sout stream", 0 )
165 add_shortcut( "rtp" )
166 set_category( CAT_SOUT )
167 set_subcategory( SUBCAT_SOUT_STREAM )
169 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
170 DEST_LONGTEXT, true )
171 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
173 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
175 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
178 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
179 NAME_LONGTEXT, true )
180 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
181 DESC_LONGTEXT, true )
182 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
184 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
185 EMAIL_LONGTEXT, true )
186 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
187 PHONE_LONGTEXT, true )
189 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
190 PROTO_LONGTEXT, false )
191 change_string_list( ppsz_protos, ppsz_protocols, NULL )
192 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
193 PORT_LONGTEXT, true )
194 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
195 PORT_AUDIO_LONGTEXT, true )
196 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
197 PORT_VIDEO_LONGTEXT, true )
199 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
201 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
202 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
204 add_string( SOUT_CFG_PREFIX "key", "", NULL,
205 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
206 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
207 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
209 add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
210 RFC3016_LONGTEXT, false )
212 set_callbacks( Open, Close )
215 /*****************************************************************************
216 * Exported prototypes
217 *****************************************************************************/
218 static const char *const ppsz_sout_options[] = {
219 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
220 "sap", "description", "url", "email", "phone",
221 "proto", "rtcp-mux", "key", "salt",
225 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
226 static int Del ( sout_stream_t *, sout_stream_id_t * );
227 static int Send( sout_stream_t *, sout_stream_id_t *,
229 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
230 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
231 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
234 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
235 static void* ThreadSend( vlc_object_t *p_this );
237 static void SDPHandleUrl( sout_stream_t *, const char * );
239 static int SapSetup( sout_stream_t *p_stream );
240 static int FileSetup( sout_stream_t *p_stream );
241 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
243 struct sout_stream_sys_t
247 vlc_mutex_t lock_sdp;
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;
266 uint32_t payload_bitmap;
268 uint16_t i_port_audio;
269 uint16_t i_port_video;
275 /* in case we do TS/PS over rtp */
277 sout_access_out_t *p_grab;
283 sout_stream_id_t **es;
286 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
288 typedef struct rtp_sink_t
294 struct sout_stream_id_t
298 sout_stream_t *p_stream;
301 uint8_t i_payload_type;
313 /* Packetizer specific fields */
315 srtp_session_t *srtp;
316 pf_rtp_packetizer_t pf_packetize;
319 vlc_mutex_t lock_sink;
322 rtsp_stream_id_t *rtsp_id;
325 block_fifo_t *p_fifo;
329 /*****************************************************************************
331 *****************************************************************************/
332 static int Open( vlc_object_t *p_this )
334 sout_stream_t *p_stream = (sout_stream_t*)p_this;
335 sout_instance_t *p_sout = p_stream->p_sout;
336 sout_stream_sys_t *p_sys = NULL;
337 config_chain_t *p_cfg = NULL;
341 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
342 ppsz_sout_options, p_stream->p_cfg );
344 p_sys = malloc( sizeof( sout_stream_sys_t ) );
348 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
350 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
351 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
352 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
353 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
355 p_sys->psz_sdp_file = NULL;
357 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
359 msg_Err( p_stream, "audio and video RTP port must be distinct" );
360 free( p_sys->psz_destination );
365 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
367 if( !strcmp( p_cfg->psz_name, "sdp" )
368 && ( p_cfg->psz_value != NULL )
369 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
377 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
380 if( !strncasecmp( psz, "rtsp:", 5 ) )
386 /* Transport protocol */
387 p_sys->proto = IPPROTO_UDP;
388 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
390 if ((psz == NULL) || !strcasecmp (psz, "udp"))
391 (void)0; /* default */
393 if (!strcasecmp (psz, "dccp"))
395 p_sys->proto = IPPROTO_DCCP;
396 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
400 if (!strcasecmp (psz, "sctp"))
402 p_sys->proto = IPPROTO_TCP;
403 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
408 if (!strcasecmp (psz, "tcp"))
410 p_sys->proto = IPPROTO_TCP;
411 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
415 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
416 p_sys->proto = IPPROTO_UDPLITE;
418 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
421 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
423 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
425 msg_Err( p_stream, "missing destination and not in RTSP mode" );
430 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
431 if( p_sys->i_ttl == -1 )
433 /* Normally, we should let the default hop limit up to the core,
434 * but we have to know it to build our SDP properly, which is why
435 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
437 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
440 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
442 p_sys->payload_bitmap = 0;
446 p_sys->psz_sdp = NULL;
448 p_sys->b_export_sap = false;
449 p_sys->b_export_sdp_file = false;
450 p_sys->p_session = NULL;
452 p_sys->p_httpd_host = NULL;
453 p_sys->p_httpd_file = NULL;
455 p_stream->p_sys = p_sys;
457 vlc_mutex_init( &p_sys->lock_sdp );
458 vlc_mutex_init( &p_sys->lock_es );
460 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
463 sout_stream_id_t *id;
465 /* Check muxer type */
466 if( strncasecmp( psz, "ps", 2 )
467 && strncasecmp( psz, "mpeg1", 5 )
468 && strncasecmp( psz, "ts", 2 ) )
470 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
472 vlc_mutex_destroy( &p_sys->lock_sdp );
473 vlc_mutex_destroy( &p_sys->lock_es );
474 free( p_sys->psz_destination );
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 );
489 free( p_sys->psz_destination );
494 id = Add( p_stream, NULL );
497 sout_MuxDelete( p_sys->p_mux );
498 sout_AccessOutDelete( p_sys->p_grab );
499 vlc_mutex_destroy( &p_sys->lock_sdp );
500 vlc_mutex_destroy( &p_sys->lock_es );
501 free( p_sys->psz_destination );
506 p_sys->packet = NULL;
508 p_stream->pf_add = MuxAdd;
509 p_stream->pf_del = MuxDel;
510 p_stream->pf_send = MuxSend;
515 p_sys->p_grab = NULL;
517 p_stream->pf_add = Add;
518 p_stream->pf_del = Del;
519 p_stream->pf_send = Send;
522 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
523 SDPHandleUrl( p_stream, "sap" );
525 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
528 config_chain_t *p_cfg;
530 SDPHandleUrl( p_stream, psz );
532 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
534 if( !strcmp( p_cfg->psz_name, "sdp" ) )
536 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
539 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
540 if( !strcmp( p_cfg->psz_value, psz ) )
543 SDPHandleUrl( p_stream, p_cfg->psz_value );
549 /* update p_sout->i_out_pace_nocontrol */
550 p_stream->p_sout->i_out_pace_nocontrol++;
555 /*****************************************************************************
557 *****************************************************************************/
558 static void Close( vlc_object_t * p_this )
560 sout_stream_t *p_stream = (sout_stream_t*)p_this;
561 sout_stream_sys_t *p_sys = p_stream->p_sys;
563 /* update p_sout->i_out_pace_nocontrol */
564 p_stream->p_sout->i_out_pace_nocontrol--;
568 assert( p_sys->i_es == 1 );
569 Del( p_stream, p_sys->es[0] );
571 sout_MuxDelete( p_sys->p_mux );
572 sout_AccessOutDelete( p_sys->p_grab );
575 block_Release( p_sys->packet );
577 if( p_sys->b_export_sap )
580 SapSetup( p_stream );
584 if( p_sys->rtsp != NULL )
585 RtspUnsetup( p_sys->rtsp );
587 vlc_mutex_destroy( &p_sys->lock_sdp );
588 vlc_mutex_destroy( &p_sys->lock_es );
590 if( p_sys->p_httpd_file )
591 httpd_FileDelete( p_sys->p_httpd_file );
593 if( p_sys->p_httpd_host )
594 httpd_HostDelete( p_sys->p_httpd_host );
596 free( p_sys->psz_sdp );
598 if( p_sys->b_export_sdp_file )
601 unlink( p_sys->psz_sdp_file );
603 free( p_sys->psz_sdp_file );
605 free( p_sys->psz_destination );
609 /*****************************************************************************
611 *****************************************************************************/
612 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
614 sout_stream_sys_t *p_sys = p_stream->p_sys;
617 vlc_UrlParse( &url, psz_url, 0 );
618 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
620 if( p_sys->p_httpd_file )
622 msg_Err( p_stream, "you can use sdp=http:// only once" );
626 if( HttpSetup( p_stream, &url ) )
628 msg_Err( p_stream, "cannot export SDP as HTTP" );
631 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
633 if( p_sys->rtsp != NULL )
635 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
639 /* FIXME test if destination is multicast or no destination at all */
640 p_sys->rtsp = RtspSetup( p_stream, &url );
641 if( p_sys->rtsp == NULL )
642 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( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
792 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
794 if( rtsp_url != NULL )
796 assert( strlen( rtsp_url ) > 0 );
797 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
798 sdp_AddAttribute ( &psz_sdp, "control",
799 addslash ? "%s/trackID=%u" : "%strackID=%u",
804 if( id->listen_fd != NULL )
805 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
806 if( p_sys->proto == IPPROTO_DCCP )
807 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
808 "SC:RTP%c", toupper( mime_major[0] ) );
815 /*****************************************************************************
817 *****************************************************************************/
819 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
821 static const char hex[16] = "0123456789abcdef";
824 for( i = 0; i < i_data; i++ )
826 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
827 s[2*i+1] = hex[(p_data[i] )&0xf];
833 * Shrink the MTU down to a fixed packetization time (for audio).
836 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
838 /* Samples per second */
839 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
840 bytes *= id->i_channels;
843 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
844 id->i_mtu = 12 + spl;
845 else /* MTU is too small for ptime, align to a sample boundary */
846 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
849 /** Add an ES as a new RTP stream */
850 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
852 /* NOTE: As a special case, if we use a non-RTP
853 * mux (TS/PS), then p_fmt is NULL. */
854 sout_stream_sys_t *p_sys = p_stream->p_sys;
855 sout_stream_id_t *id;
856 int i_port, cscov = -1;
858 int i_port_audio_option = var_GetInteger( p_stream, "port-audio" );
859 int i_port_video_option = var_GetInteger( p_stream, "port-video" );
861 if (0xffffffff == p_sys->payload_bitmap)
863 msg_Err (p_stream, "too many RTP elementary streams");
867 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
870 vlc_object_attach( id, p_stream );
872 /* Choose the port */
877 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
879 i_port = p_sys->i_port_audio;
880 p_sys->i_port_audio = 0;
883 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
885 i_port = p_sys->i_port_video;
886 p_sys->i_port_video = 0;
891 if( p_sys->i_port != i_port_audio_option
892 && p_sys->i_port != i_port_video_option )
894 i_port = p_sys->i_port;
899 id->p_stream = p_stream;
901 /* Look for free dymanic payload type */
902 id->i_payload_type = 96;
903 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
904 id->i_payload_type++;
905 assert (id->i_payload_type < 128);
907 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
908 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
912 id->i_clock_rate = 90000; /* most common case for video */
917 id->i_cat = p_fmt->i_cat;
918 if( p_fmt->i_cat == AUDIO_ES )
920 id->i_clock_rate = p_fmt->audio.i_rate;
921 id->i_channels = p_fmt->audio.i_channels;
923 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
927 id->i_cat = VIDEO_ES;
931 id->i_mtu = config_GetInt( p_stream, "mtu" );
932 if( id->i_mtu <= 12 + 16 )
933 id->i_mtu = 576 - 20 - 8; /* pessimistic */
934 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
937 id->pf_packetize = NULL;
939 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
942 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
943 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
944 if (id->srtp == NULL)
950 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
951 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
956 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
959 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
962 vlc_mutex_init( &id->lock_sink );
967 id->listen_fd = NULL;
970 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
972 if( p_sys->psz_destination != NULL )
973 switch( p_sys->proto )
980 case VIDEO_ES: code = "RTPV"; break;
981 case AUDIO_ES: code = "RTPARTPV"; break;
982 case SPU_ES: code = "RTPTRTPV"; break;
983 default: code = "RTPORTPV"; break;
985 var_SetString (p_stream, "dccp-service", code);
988 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
989 p_sys->psz_destination, i_port,
991 if( id->listen_fd == NULL )
993 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1000 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1001 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1002 i_port, ttl, p_sys->proto );
1005 msg_Err( p_stream, "cannot create RTP socket" );
1008 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1014 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1016 if( psz == NULL ) /* Uho! */
1019 if( strncmp( psz, "ts", 2 ) == 0 )
1021 id->i_payload_type = 33;
1022 id->psz_enc = "MP2T";
1026 id->psz_enc = "MP2P";
1031 switch( p_fmt->i_codec )
1033 case VLC_FOURCC( 'u', 'l', 'a', 'w' ):
1034 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1035 id->i_payload_type = 0;
1036 id->psz_enc = "PCMU";
1037 id->pf_packetize = rtp_packetize_split;
1038 rtp_set_ptime (id, 20, 1);
1040 case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
1041 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1042 id->i_payload_type = 8;
1043 id->psz_enc = "PCMA";
1044 id->pf_packetize = rtp_packetize_split;
1045 rtp_set_ptime (id, 20, 1);
1047 case VLC_FOURCC( 's', '1', '6', 'b' ):
1048 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1050 id->i_payload_type = 11;
1052 else if( p_fmt->audio.i_channels == 2 &&
1053 p_fmt->audio.i_rate == 44100 )
1055 id->i_payload_type = 10;
1057 id->psz_enc = "L16";
1058 id->pf_packetize = rtp_packetize_split;
1059 rtp_set_ptime (id, 20, 2);
1061 case VLC_FOURCC( 'u', '8', ' ', ' ' ):
1063 id->pf_packetize = rtp_packetize_split;
1064 rtp_set_ptime (id, 20, 1);
1066 case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
1067 case VLC_FOURCC( 'm', 'p', '3', ' ' ):
1068 id->i_payload_type = 14;
1069 id->psz_enc = "MPA";
1070 id->i_clock_rate = 90000; /* not 44100 */
1071 id->pf_packetize = rtp_packetize_mpa;
1073 case VLC_FOURCC( 'm', 'p', 'g', 'v' ):
1074 id->i_payload_type = 32;
1075 id->psz_enc = "MPV";
1076 id->pf_packetize = rtp_packetize_mpv;
1078 case VLC_FOURCC( 'G', '7', '2', '6' ):
1079 case VLC_FOURCC( 'g', '7', '2', '6' ):
1080 switch( p_fmt->i_bitrate / 1000 )
1083 id->psz_enc = "G726-16";
1084 id->pf_packetize = rtp_packetize_g726_16;
1087 id->psz_enc = "G726-24";
1088 id->pf_packetize = rtp_packetize_g726_24;
1091 id->psz_enc = "G726-32";
1092 id->pf_packetize = rtp_packetize_g726_32;
1095 id->psz_enc = "G726-40";
1096 id->pf_packetize = rtp_packetize_g726_40;
1100 case VLC_FOURCC( 'a', '5', '2', ' ' ):
1101 id->psz_enc = "ac3";
1102 id->pf_packetize = rtp_packetize_ac3;
1104 case VLC_FOURCC( 'H', '2', '6', '3' ):
1105 id->psz_enc = "H263-1998";
1106 id->pf_packetize = rtp_packetize_h263;
1108 case VLC_FOURCC( 'h', '2', '6', '4' ):
1109 id->psz_enc = "H264";
1110 id->pf_packetize = rtp_packetize_h264;
1111 id->psz_fmtp = NULL;
1113 if( p_fmt->i_extra > 0 )
1115 uint8_t *p_buffer = p_fmt->p_extra;
1116 int i_buffer = p_fmt->i_extra;
1117 char *p_64_sps = NULL;
1118 char *p_64_pps = NULL;
1121 while( i_buffer > 4 &&
1122 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1123 p_buffer[2] == 0 && p_buffer[3] == 1 )
1125 const int i_nal_type = p_buffer[4]&0x1f;
1129 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1132 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1134 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1136 /* we found another startcode */
1141 if( i_nal_type == 7 )
1143 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1144 sprintf_hexa( hexa, &p_buffer[5], 3 );
1146 else if( i_nal_type == 8 )
1148 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1154 if( p_64_sps && p_64_pps &&
1155 ( asprintf( &id->psz_fmtp,
1156 "packetization-mode=1;profile-level-id=%s;"
1157 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1158 p_64_pps ) == -1 ) )
1159 id->psz_fmtp = NULL;
1164 id->psz_fmtp = strdup( "packetization-mode=1" );
1167 case VLC_FOURCC( 'm', 'p', '4', 'v' ):
1169 char hexa[2*p_fmt->i_extra +1];
1171 id->psz_enc = "MP4V-ES";
1172 id->pf_packetize = rtp_packetize_split;
1173 if( p_fmt->i_extra > 0 )
1175 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1176 if( asprintf( &id->psz_fmtp,
1177 "profile-level-id=3; config=%s;", hexa ) == -1 )
1178 id->psz_fmtp = NULL;
1182 case VLC_FOURCC( 'm', 'p', '4', 'a' ):
1186 char hexa[2*p_fmt->i_extra +1];
1188 id->psz_enc = "mpeg4-generic";
1189 id->pf_packetize = rtp_packetize_mp4a;
1190 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1191 if( asprintf( &id->psz_fmtp,
1192 "streamtype=5; profile-level-id=15; "
1193 "mode=AAC-hbr; config=%s; SizeLength=13; "
1194 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1196 id->psz_fmtp = NULL;
1202 unsigned char config[6];
1203 unsigned int aacsrates[15] = {
1204 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1205 16000, 12000, 11025, 8000, 7350, 0, 0 };
1207 for( i = 0; i < 15; i++ )
1208 if( p_fmt->audio.i_rate == aacsrates[i] )
1214 config[3]=p_fmt->audio.i_channels<<4;
1218 id->psz_enc = "MP4A-LATM";
1219 id->pf_packetize = rtp_packetize_mp4a_latm;
1220 sprintf_hexa( hexa, config, 6 );
1221 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1222 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1223 id->psz_fmtp = NULL;
1227 case VLC_FOURCC( 's', 'a', 'm', 'r' ):
1228 id->psz_enc = "AMR";
1229 id->psz_fmtp = strdup( "octet-align=1" );
1230 id->pf_packetize = rtp_packetize_amr;
1232 case VLC_FOURCC( 's', 'a', 'w', 'b' ):
1233 id->psz_enc = "AMR-WB";
1234 id->psz_fmtp = strdup( "octet-align=1" );
1235 id->pf_packetize = rtp_packetize_amr;
1237 case VLC_FOURCC( 's', 'p', 'x', ' ' ):
1238 id->psz_enc = "SPEEX";
1239 id->pf_packetize = rtp_packetize_spx;
1241 case VLC_FOURCC( 't', '1', '4', '0' ):
1242 id->psz_enc = "t140" ;
1243 id->i_clock_rate = 1000;
1244 id->pf_packetize = rtp_packetize_t140;
1248 msg_Err( p_stream, "cannot add this stream (unsupported "
1249 "codec:%4.4s)", (char*)&p_fmt->i_codec );
1252 if (id->i_payload_type >= 96)
1253 /* Mark dynamic payload type in use */
1254 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1256 #if 0 /* No payload formats sets this at the moment */
1258 cscov += 8 /* UDP */ + 12 /* RTP */;
1260 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1263 if( p_sys->rtsp != NULL )
1264 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1265 GetDWBE( id->ssrc ),
1266 p_sys->psz_destination,
1267 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1269 id->p_fifo = block_FifoNew();
1270 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1271 VLC_THREAD_PRIORITY_HIGHEST ) )
1274 /* Update p_sys context */
1275 vlc_mutex_lock( &p_sys->lock_es );
1276 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1277 vlc_mutex_unlock( &p_sys->lock_es );
1279 psz_sdp = SDPGenerate( p_stream, NULL );
1281 vlc_mutex_lock( &p_sys->lock_sdp );
1282 free( p_sys->psz_sdp );
1283 p_sys->psz_sdp = psz_sdp;
1284 vlc_mutex_unlock( &p_sys->lock_sdp );
1286 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1288 /* Update SDP (sap/file) */
1289 if( p_sys->b_export_sap ) SapSetup( p_stream );
1290 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1295 Del( p_stream, id );
1299 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1301 sout_stream_sys_t *p_sys = p_stream->p_sys;
1303 if( id->p_fifo != NULL )
1305 vlc_object_kill( id );
1306 vlc_thread_join( id );
1307 block_FifoRelease( id->p_fifo );
1310 vlc_mutex_lock( &p_sys->lock_es );
1311 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1312 vlc_mutex_unlock( &p_sys->lock_es );
1315 if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
1316 p_sys->i_port_audio = id->i_port;
1317 if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
1318 p_sys->i_port_video = id->i_port;
1319 /* Release dynamic payload type */
1320 if (id->i_payload_type >= 96)
1321 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1323 free( id->psz_fmtp );
1326 RtspDelId( p_sys->rtsp, id->rtsp_id );
1328 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1329 if( id->listen_fd != NULL )
1330 net_ListenClose( id->listen_fd );
1331 if( id->srtp != NULL )
1332 srtp_destroy( id->srtp );
1334 vlc_mutex_destroy( &id->lock_sink );
1336 /* Update SDP (sap/file) */
1337 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1338 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1340 vlc_object_detach( id );
1341 vlc_object_release( id );
1345 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1350 assert( p_stream->p_sys->p_mux == NULL );
1353 while( p_buffer != NULL )
1355 p_next = p_buffer->p_next;
1356 if( id->pf_packetize( id, p_buffer ) )
1359 block_Release( p_buffer );
1365 /****************************************************************************
1367 ****************************************************************************/
1368 static int SapSetup( sout_stream_t *p_stream )
1370 sout_stream_sys_t *p_sys = p_stream->p_sys;
1371 sout_instance_t *p_sout = p_stream->p_sout;
1373 /* Remove the previous session */
1374 if( p_sys->p_session != NULL)
1376 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1377 p_sys->p_session = NULL;
1380 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1382 announce_method_t *p_method = sout_SAPMethod();
1383 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1385 p_sys->psz_destination,
1387 sout_MethodRelease( p_method );
1393 /****************************************************************************
1395 ****************************************************************************/
1396 static int FileSetup( sout_stream_t *p_stream )
1398 sout_stream_sys_t *p_sys = p_stream->p_sys;
1401 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1403 msg_Err( p_stream, "cannot open file '%s' (%m)",
1404 p_sys->psz_sdp_file );
1405 return VLC_EGENERIC;
1408 fputs( p_sys->psz_sdp, f );
1414 /****************************************************************************
1416 ****************************************************************************/
1417 static int HttpCallback( httpd_file_sys_t *p_args,
1418 httpd_file_t *, uint8_t *p_request,
1419 uint8_t **pp_data, int *pi_data );
1421 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1423 sout_stream_sys_t *p_sys = p_stream->p_sys;
1425 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1426 url->i_port > 0 ? url->i_port : 80 );
1427 if( p_sys->p_httpd_host )
1429 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1430 url->psz_path ? url->psz_path : "/",
1433 HttpCallback, (void*)p_sys );
1435 if( p_sys->p_httpd_file == NULL )
1437 return VLC_EGENERIC;
1442 static int HttpCallback( httpd_file_sys_t *p_args,
1443 httpd_file_t *f, uint8_t *p_request,
1444 uint8_t **pp_data, int *pi_data )
1446 VLC_UNUSED(f); VLC_UNUSED(p_request);
1447 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1449 vlc_mutex_lock( &p_sys->lock_sdp );
1450 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1452 *pi_data = strlen( p_sys->psz_sdp );
1453 *pp_data = malloc( *pi_data );
1454 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1461 vlc_mutex_unlock( &p_sys->lock_sdp );
1466 /****************************************************************************
1468 ****************************************************************************/
1469 static void* ThreadSend( vlc_object_t *p_this )
1471 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1472 unsigned i_caching = id->i_caching;
1476 block_t *out = block_FifoGet( id->p_fifo );
1477 block_cleanup_push (out);
1480 { /* FIXME: this is awfully inefficient */
1481 size_t len = out->i_buffer;
1482 out = block_Realloc( out, 0, len + 10 );
1483 out->i_buffer = len;
1485 int canc = vlc_savecancel ();
1486 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1487 vlc_restorecancel (canc);
1491 msg_Dbg( id, "SRTP sending error: %m" );
1492 block_Release( out );
1496 out->i_buffer = len;
1500 mwait (out->i_dts + i_caching);
1505 ssize_t len = out->i_buffer;
1506 int canc = vlc_savecancel ();
1508 vlc_mutex_lock( &id->lock_sink );
1509 unsigned deadc = 0; /* How many dead sockets? */
1510 int deadv[id->sinkc]; /* Dead sockets list */
1512 for( int i = 0; i < id->sinkc; i++ )
1514 if( !id->srtp ) /* FIXME: SRTCP support */
1515 SendRTCP( id->sinkv[i].rtcp, out );
1517 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1519 /* Retry sending to root out soft-errors */
1520 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1523 deadv[deadc++] = id->sinkv[i].rtp_fd;
1525 vlc_mutex_unlock( &id->lock_sink );
1526 block_Release( out );
1528 for( unsigned i = 0; i < deadc; i++ )
1530 msg_Dbg( id, "removing socket %d", deadv[i] );
1531 rtp_del_sink( id, deadv[i] );
1534 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1535 while( id->listen_fd != NULL )
1537 int fd = net_Accept( id, id->listen_fd, 0 );
1540 msg_Dbg( id, "adding socket %d", fd );
1541 rtp_add_sink( id, fd, true );
1543 vlc_restorecancel (canc);
1548 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1550 rtp_sink_t sink = { fd, NULL };
1551 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1553 if( sink.rtcp == NULL )
1554 msg_Err( id, "RTCP failed!" );
1556 vlc_mutex_lock( &id->lock_sink );
1557 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1558 vlc_mutex_unlock( &id->lock_sink );
1562 void rtp_del_sink( sout_stream_id_t *id, int fd )
1564 rtp_sink_t sink = { fd, NULL };
1566 /* NOTE: must be safe to use if fd is not included */
1567 vlc_mutex_lock( &id->lock_sink );
1568 for( int i = 0; i < id->sinkc; i++ )
1570 if (id->sinkv[i].rtp_fd == fd)
1572 sink = id->sinkv[i];
1573 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1577 vlc_mutex_unlock( &id->lock_sink );
1579 CloseRTCP( sink.rtcp );
1580 net_Close( sink.rtp_fd );
1583 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1585 /* This will return values for the next packet.
1586 * Accounting for caching would not be totally trivial. */
1587 return id->i_sequence;
1590 /* FIXME: this is pretty bad - if we remove and then insert an ES
1591 * the number will get unsynched from inside RTSP */
1592 unsigned rtp_get_num( const sout_stream_id_t *id )
1594 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1597 vlc_mutex_lock( &p_sys->lock_es );
1598 for( i = 0; i < p_sys->i_es; i++ )
1600 if( id == p_sys->es[i] )
1603 vlc_mutex_unlock( &p_sys->lock_es );
1609 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1610 int b_marker, int64_t i_pts )
1612 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
1614 out->p_buffer[0] = 0x80;
1615 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1616 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1617 out->p_buffer[3] = ( id->i_sequence )&0xff;
1618 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1619 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1620 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1621 out->p_buffer[7] = ( i_timestamp )&0xff;
1623 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1629 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1631 block_FifoPut( id->p_fifo, out );
1635 * @return configured max RTP payload size (including payload type-specific
1636 * headers, excluding RTP and transport headers)
1638 size_t rtp_mtu (const sout_stream_id_t *id)
1640 return id->i_mtu - 12;
1643 /*****************************************************************************
1645 *****************************************************************************/
1647 /** Add an ES to a non-RTP muxed stream */
1648 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1650 sout_input_t *p_input;
1651 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1652 assert( p_mux != NULL );
1654 p_input = sout_MuxAddStream( p_mux, p_fmt );
1655 if( p_input == NULL )
1657 msg_Err( p_stream, "cannot add this stream to the muxer" );
1661 return (sout_stream_id_t *)p_input;
1665 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1668 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1669 assert( p_mux != NULL );
1671 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1676 /** Remove an ES from a non-RTP muxed stream */
1677 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1679 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1680 assert( p_mux != NULL );
1682 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1687 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1688 const block_t *p_buffer )
1690 sout_stream_sys_t *p_sys = p_stream->p_sys;
1691 sout_stream_id_t *id = p_sys->es[0];
1693 int64_t i_dts = p_buffer->i_dts;
1695 uint8_t *p_data = p_buffer->p_buffer;
1696 size_t i_data = p_buffer->i_buffer;
1697 size_t i_max = id->i_mtu - 12;
1699 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1705 /* output complete packet */
1706 if( p_sys->packet &&
1707 p_sys->packet->i_buffer + i_data > i_max )
1709 rtp_packetize_send( id, p_sys->packet );
1710 p_sys->packet = NULL;
1713 if( p_sys->packet == NULL )
1715 /* allocate a new packet */
1716 p_sys->packet = block_New( p_stream, id->i_mtu );
1717 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1718 p_sys->packet->i_dts = i_dts;
1719 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1720 i_dts += p_sys->packet->i_length;
1723 i_size = __MIN( i_data,
1724 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1726 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1729 p_sys->packet->i_buffer += i_size;
1738 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1741 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1747 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1749 p_next = p_buffer->p_next;
1750 block_Release( p_buffer );
1758 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1760 sout_access_out_t *p_grab;
1762 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1763 if( p_grab == NULL )
1766 p_grab->p_module = NULL;
1767 p_grab->psz_access = strdup( "grab" );
1768 p_grab->p_cfg = NULL;
1769 p_grab->psz_path = strdup( "" );
1770 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1771 p_grab->pf_seek = NULL;
1772 p_grab->pf_write = AccessOutGrabberWrite;
1773 vlc_object_attach( p_grab, p_stream );