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 );
570 sout_MuxDelete( p_sys->p_mux );
571 Del( p_stream, p_sys->es[0] );
572 sout_AccessOutDelete( p_sys->p_grab );
576 block_Release( p_sys->packet );
578 if( p_sys->b_export_sap )
581 SapSetup( p_stream );
585 if( p_sys->rtsp != NULL )
586 RtspUnsetup( p_sys->rtsp );
588 vlc_mutex_destroy( &p_sys->lock_sdp );
589 vlc_mutex_destroy( &p_sys->lock_es );
591 if( p_sys->p_httpd_file )
592 httpd_FileDelete( p_sys->p_httpd_file );
594 if( p_sys->p_httpd_host )
595 httpd_HostDelete( p_sys->p_httpd_host );
597 free( p_sys->psz_sdp );
599 if( p_sys->b_export_sdp_file )
602 unlink( p_sys->psz_sdp_file );
604 free( p_sys->psz_sdp_file );
606 free( p_sys->psz_destination );
610 /*****************************************************************************
612 *****************************************************************************/
613 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
615 sout_stream_sys_t *p_sys = p_stream->p_sys;
618 vlc_UrlParse( &url, psz_url, 0 );
619 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
621 if( p_sys->p_httpd_file )
623 msg_Err( p_stream, "you can use sdp=http:// only once" );
627 if( HttpSetup( p_stream, &url ) )
629 msg_Err( p_stream, "cannot export SDP as HTTP" );
632 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
634 if( p_sys->rtsp != NULL )
636 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
640 /* FIXME test if destination is multicast or no destination at all */
641 p_sys->rtsp = RtspSetup( p_stream, &url );
642 if( p_sys->rtsp == NULL )
643 msg_Err( p_stream, "cannot export SDP as RTSP" );
645 if( p_sys->p_mux != NULL )
647 sout_stream_id_t *id = p_sys->es[0];
648 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
649 p_sys->psz_destination, p_sys->i_ttl,
650 id->i_port, id->i_port + 1 );
653 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
654 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
656 p_sys->b_export_sap = true;
657 SapSetup( p_stream );
659 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
661 if( p_sys->b_export_sdp_file )
663 msg_Err( p_stream, "you can use sdp=file:// only once" );
666 p_sys->b_export_sdp_file = true;
667 psz_url = &psz_url[5];
668 if( psz_url[0] == '/' && psz_url[1] == '/' )
670 p_sys->psz_sdp_file = strdup( psz_url );
671 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
672 FileSetup( p_stream );
676 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
681 vlc_UrlClean( &url );
684 /*****************************************************************************
686 *****************************************************************************/
688 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
690 const sout_stream_sys_t *p_sys = p_stream->p_sys;
692 struct sockaddr_storage dst;
696 * When we have a fixed destination (typically when we do multicast),
697 * we need to put the actual port numbers in the SDP.
698 * When there is no fixed destination, we only support RTSP unicast
699 * on-demand setup, so we should rather let the clients decide which ports
701 * When there is both a fixed destination and RTSP unicast, we need to
702 * put port numbers used by the fixed destination, otherwise the SDP would
703 * become totally incorrect for multicast use. It should be noted that
704 * port numbers from SDP with RTSP are only "recommendation" from the
705 * server to the clients (per RFC2326), so only broken clients will fail
706 * to handle this properly. There is no solution but to use two differents
707 * output chain with two different RTSP URLs if you need to handle this
712 if( p_sys->psz_destination != NULL )
716 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
717 dstlen = sizeof( dst );
718 if( p_sys->es[0]->listen_fd != NULL )
719 getsockname( p_sys->es[0]->listen_fd[0],
720 (struct sockaddr *)&dst, &dstlen );
722 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
723 (struct sockaddr *)&dst, &dstlen );
729 /* Dummy destination address for RTSP */
730 memset (&dst, 0, sizeof( struct sockaddr_in ) );
731 dst.ss_family = AF_INET;
735 dstlen = sizeof( struct sockaddr_in );
738 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
739 NULL, 0, (struct sockaddr *)&dst, dstlen );
740 if( psz_sdp == NULL )
743 /* TODO: a=source-filter */
744 if( p_sys->rtcp_mux )
745 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
747 if( rtsp_url != NULL )
748 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
750 /* FIXME: locking?! */
751 for( i = 0; i < p_sys->i_es; i++ )
753 sout_stream_id_t *id = p_sys->es[i];
754 const char *mime_major; /* major MIME type */
755 const char *proto = "RTP/AVP"; /* protocol */
760 mime_major = "video";
763 mime_major = "audio";
772 if( rtsp_url == NULL )
774 switch( p_sys->proto )
779 proto = "TCP/RTP/AVP";
782 proto = "DCCP/RTP/AVP";
784 case IPPROTO_UDPLITE:
789 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
790 id->i_payload_type, false, id->i_bitrate,
791 id->psz_enc, id->i_clock_rate, id->i_channels,
794 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
795 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
797 if( rtsp_url != NULL )
799 assert( strlen( rtsp_url ) > 0 );
800 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
801 sdp_AddAttribute ( &psz_sdp, "control",
802 addslash ? "%s/trackID=%u" : "%strackID=%u",
807 if( id->listen_fd != NULL )
808 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
809 if( p_sys->proto == IPPROTO_DCCP )
810 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
811 "SC:RTP%c", toupper( mime_major[0] ) );
818 /*****************************************************************************
820 *****************************************************************************/
822 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
824 static const char hex[16] = "0123456789abcdef";
827 for( i = 0; i < i_data; i++ )
829 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
830 s[2*i+1] = hex[(p_data[i] )&0xf];
836 * Shrink the MTU down to a fixed packetization time (for audio).
839 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
841 /* Samples per second */
842 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
843 bytes *= id->i_channels;
846 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
847 id->i_mtu = 12 + spl;
848 else /* MTU is too small for ptime, align to a sample boundary */
849 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
852 /** Add an ES as a new RTP stream */
853 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
855 /* NOTE: As a special case, if we use a non-RTP
856 * mux (TS/PS), then p_fmt is NULL. */
857 sout_stream_sys_t *p_sys = p_stream->p_sys;
858 sout_stream_id_t *id;
862 if (0xffffffff == p_sys->payload_bitmap)
864 msg_Err (p_stream, "too many RTP elementary streams");
868 /* Choose the port */
873 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
874 i_port = p_sys->i_port_audio;
876 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
877 i_port = p_sys->i_port_video;
879 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
880 * only one thread that can *modify* the ES table. The ES lock protects
881 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
882 for (int i = 0; i_port && (i < p_sys->i_es); i++)
883 if (i_port == p_sys->es[i]->i_port)
884 i_port = 0; /* Port already in use! */
885 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
889 msg_Err (p_stream, "too many RTP elementary streams");
893 for (int i = 0; i_port && (i < p_sys->i_es); i++)
894 if (p == p_sys->es[i]->i_port)
898 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
901 vlc_object_attach( id, p_stream );
903 id->p_stream = p_stream;
905 /* Look for free dymanic payload type */
906 id->i_payload_type = 96;
907 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
908 id->i_payload_type++;
909 assert (id->i_payload_type < 128);
911 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
912 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
916 id->i_clock_rate = 90000; /* most common case for video */
921 id->i_cat = p_fmt->i_cat;
922 if( p_fmt->i_cat == AUDIO_ES )
924 id->i_clock_rate = p_fmt->audio.i_rate;
925 id->i_channels = p_fmt->audio.i_channels;
927 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
931 id->i_cat = VIDEO_ES;
935 id->i_mtu = config_GetInt( p_stream, "mtu" );
936 if( id->i_mtu <= 12 + 16 )
937 id->i_mtu = 576 - 20 - 8; /* pessimistic */
938 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
941 id->pf_packetize = NULL;
943 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
946 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
947 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
948 if (id->srtp == NULL)
954 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
955 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
960 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
963 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
966 vlc_mutex_init( &id->lock_sink );
971 id->listen_fd = NULL;
974 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
976 if( p_sys->psz_destination != NULL )
977 switch( p_sys->proto )
984 case VIDEO_ES: code = "RTPV"; break;
985 case AUDIO_ES: code = "RTPARTPV"; break;
986 case SPU_ES: code = "RTPTRTPV"; break;
987 default: code = "RTPORTPV"; break;
989 var_SetString (p_stream, "dccp-service", code);
992 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
993 p_sys->psz_destination, i_port,
995 if( id->listen_fd == NULL )
997 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1004 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1005 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1006 i_port, ttl, p_sys->proto );
1009 msg_Err( p_stream, "cannot create RTP socket" );
1012 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1018 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1020 if( psz == NULL ) /* Uho! */
1023 if( strncmp( psz, "ts", 2 ) == 0 )
1025 id->i_payload_type = 33;
1026 id->psz_enc = "MP2T";
1030 id->psz_enc = "MP2P";
1035 switch( p_fmt->i_codec )
1037 case VLC_CODEC_MULAW:
1038 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1039 id->i_payload_type = 0;
1040 id->psz_enc = "PCMU";
1041 id->pf_packetize = rtp_packetize_split;
1042 rtp_set_ptime (id, 20, 1);
1044 case VLC_CODEC_ALAW:
1045 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1046 id->i_payload_type = 8;
1047 id->psz_enc = "PCMA";
1048 id->pf_packetize = rtp_packetize_split;
1049 rtp_set_ptime (id, 20, 1);
1051 case VLC_CODEC_S16B:
1052 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1054 id->i_payload_type = 11;
1056 else if( p_fmt->audio.i_channels == 2 &&
1057 p_fmt->audio.i_rate == 44100 )
1059 id->i_payload_type = 10;
1061 id->psz_enc = "L16";
1062 id->pf_packetize = rtp_packetize_split;
1063 rtp_set_ptime (id, 20, 2);
1067 id->pf_packetize = rtp_packetize_split;
1068 rtp_set_ptime (id, 20, 1);
1070 case VLC_CODEC_MPGA:
1071 id->i_payload_type = 14;
1072 id->psz_enc = "MPA";
1073 id->i_clock_rate = 90000; /* not 44100 */
1074 id->pf_packetize = rtp_packetize_mpa;
1076 case VLC_CODEC_MPGV:
1077 id->i_payload_type = 32;
1078 id->psz_enc = "MPV";
1079 id->pf_packetize = rtp_packetize_mpv;
1081 case VLC_CODEC_ADPCM_G726:
1082 switch( p_fmt->i_bitrate / 1000 )
1085 id->psz_enc = "G726-16";
1086 id->pf_packetize = rtp_packetize_g726_16;
1089 id->psz_enc = "G726-24";
1090 id->pf_packetize = rtp_packetize_g726_24;
1093 id->psz_enc = "G726-32";
1094 id->pf_packetize = rtp_packetize_g726_32;
1097 id->psz_enc = "G726-40";
1098 id->pf_packetize = rtp_packetize_g726_40;
1101 msg_Err( p_stream, "cannot add this stream (unsupported "
1102 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1107 id->psz_enc = "ac3";
1108 id->pf_packetize = rtp_packetize_ac3;
1110 case VLC_CODEC_H263:
1111 id->psz_enc = "H263-1998";
1112 id->pf_packetize = rtp_packetize_h263;
1114 case VLC_CODEC_H264:
1115 id->psz_enc = "H264";
1116 id->pf_packetize = rtp_packetize_h264;
1117 id->psz_fmtp = NULL;
1119 if( p_fmt->i_extra > 0 )
1121 uint8_t *p_buffer = p_fmt->p_extra;
1122 int i_buffer = p_fmt->i_extra;
1123 char *p_64_sps = NULL;
1124 char *p_64_pps = NULL;
1127 while( i_buffer > 4 &&
1128 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1129 p_buffer[2] == 0 && p_buffer[3] == 1 )
1131 const int i_nal_type = p_buffer[4]&0x1f;
1135 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1138 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1140 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1142 /* we found another startcode */
1147 if( i_nal_type == 7 )
1149 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1150 sprintf_hexa( hexa, &p_buffer[5], 3 );
1152 else if( i_nal_type == 8 )
1154 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1160 if( p_64_sps && p_64_pps &&
1161 ( asprintf( &id->psz_fmtp,
1162 "packetization-mode=1;profile-level-id=%s;"
1163 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1164 p_64_pps ) == -1 ) )
1165 id->psz_fmtp = NULL;
1170 id->psz_fmtp = strdup( "packetization-mode=1" );
1173 case VLC_CODEC_MP4V:
1175 char hexa[2*p_fmt->i_extra +1];
1177 id->psz_enc = "MP4V-ES";
1178 id->pf_packetize = rtp_packetize_split;
1179 if( p_fmt->i_extra > 0 )
1181 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1182 if( asprintf( &id->psz_fmtp,
1183 "profile-level-id=3; config=%s;", hexa ) == -1 )
1184 id->psz_fmtp = NULL;
1188 case VLC_CODEC_MP4A:
1192 char hexa[2*p_fmt->i_extra +1];
1194 id->psz_enc = "mpeg4-generic";
1195 id->pf_packetize = rtp_packetize_mp4a;
1196 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1197 if( asprintf( &id->psz_fmtp,
1198 "streamtype=5; profile-level-id=15; "
1199 "mode=AAC-hbr; config=%s; SizeLength=13; "
1200 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1202 id->psz_fmtp = NULL;
1208 unsigned char config[6];
1209 unsigned int aacsrates[15] = {
1210 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1211 16000, 12000, 11025, 8000, 7350, 0, 0 };
1213 for( i = 0; i < 15; i++ )
1214 if( p_fmt->audio.i_rate == aacsrates[i] )
1220 config[3]=p_fmt->audio.i_channels<<4;
1224 id->psz_enc = "MP4A-LATM";
1225 id->pf_packetize = rtp_packetize_mp4a_latm;
1226 sprintf_hexa( hexa, config, 6 );
1227 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1228 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1229 id->psz_fmtp = NULL;
1233 case VLC_CODEC_AMR_NB:
1234 id->psz_enc = "AMR";
1235 id->psz_fmtp = strdup( "octet-align=1" );
1236 id->pf_packetize = rtp_packetize_amr;
1238 case VLC_CODEC_AMR_WB:
1239 id->psz_enc = "AMR-WB";
1240 id->psz_fmtp = strdup( "octet-align=1" );
1241 id->pf_packetize = rtp_packetize_amr;
1243 case VLC_CODEC_SPEEX:
1244 id->psz_enc = "SPEEX";
1245 id->pf_packetize = rtp_packetize_spx;
1247 case VLC_CODEC_ITU_T140:
1248 id->psz_enc = "t140" ;
1249 id->i_clock_rate = 1000;
1250 id->pf_packetize = rtp_packetize_t140;
1254 msg_Err( p_stream, "cannot add this stream (unsupported "
1255 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1258 if (id->i_payload_type >= 96)
1259 /* Mark dynamic payload type in use */
1260 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1262 #if 0 /* No payload formats sets this at the moment */
1264 cscov += 8 /* UDP */ + 12 /* RTP */;
1266 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1269 if( p_sys->rtsp != NULL )
1270 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1271 GetDWBE( id->ssrc ),
1272 p_sys->psz_destination,
1273 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1275 id->p_fifo = block_FifoNew();
1276 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1277 VLC_THREAD_PRIORITY_HIGHEST ) )
1280 /* Update p_sys context */
1281 vlc_mutex_lock( &p_sys->lock_es );
1282 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1283 vlc_mutex_unlock( &p_sys->lock_es );
1285 psz_sdp = SDPGenerate( p_stream, NULL );
1287 vlc_mutex_lock( &p_sys->lock_sdp );
1288 free( p_sys->psz_sdp );
1289 p_sys->psz_sdp = psz_sdp;
1290 vlc_mutex_unlock( &p_sys->lock_sdp );
1292 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1294 /* Update SDP (sap/file) */
1295 if( p_sys->b_export_sap ) SapSetup( p_stream );
1296 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1301 Del( p_stream, id );
1305 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1307 sout_stream_sys_t *p_sys = p_stream->p_sys;
1309 if( id->p_fifo != NULL )
1311 vlc_object_kill( id );
1312 vlc_thread_join( id );
1313 block_FifoRelease( id->p_fifo );
1316 vlc_mutex_lock( &p_sys->lock_es );
1317 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1318 vlc_mutex_unlock( &p_sys->lock_es );
1320 /* Release dynamic payload type */
1321 if (id->i_payload_type >= 96)
1322 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1324 free( id->psz_fmtp );
1327 RtspDelId( p_sys->rtsp, id->rtsp_id );
1329 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1330 if( id->listen_fd != NULL )
1331 net_ListenClose( id->listen_fd );
1332 if( id->srtp != NULL )
1333 srtp_destroy( id->srtp );
1335 vlc_mutex_destroy( &id->lock_sink );
1337 /* Update SDP (sap/file) */
1338 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1339 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1341 vlc_object_detach( id );
1342 vlc_object_release( id );
1346 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1351 assert( p_stream->p_sys->p_mux == NULL );
1354 while( p_buffer != NULL )
1356 p_next = p_buffer->p_next;
1357 if( id->pf_packetize( id, p_buffer ) )
1360 block_Release( p_buffer );
1366 /****************************************************************************
1368 ****************************************************************************/
1369 static int SapSetup( sout_stream_t *p_stream )
1371 sout_stream_sys_t *p_sys = p_stream->p_sys;
1372 sout_instance_t *p_sout = p_stream->p_sout;
1374 /* Remove the previous session */
1375 if( p_sys->p_session != NULL)
1377 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1378 p_sys->p_session = NULL;
1381 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1383 announce_method_t *p_method = sout_SAPMethod();
1384 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1386 p_sys->psz_destination,
1388 sout_MethodRelease( p_method );
1394 /****************************************************************************
1396 ****************************************************************************/
1397 static int FileSetup( sout_stream_t *p_stream )
1399 sout_stream_sys_t *p_sys = p_stream->p_sys;
1402 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1404 msg_Err( p_stream, "cannot open file '%s' (%m)",
1405 p_sys->psz_sdp_file );
1406 return VLC_EGENERIC;
1409 fputs( p_sys->psz_sdp, f );
1415 /****************************************************************************
1417 ****************************************************************************/
1418 static int HttpCallback( httpd_file_sys_t *p_args,
1419 httpd_file_t *, uint8_t *p_request,
1420 uint8_t **pp_data, int *pi_data );
1422 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1424 sout_stream_sys_t *p_sys = p_stream->p_sys;
1426 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1427 url->i_port > 0 ? url->i_port : 80 );
1428 if( p_sys->p_httpd_host )
1430 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1431 url->psz_path ? url->psz_path : "/",
1434 HttpCallback, (void*)p_sys );
1436 if( p_sys->p_httpd_file == NULL )
1438 return VLC_EGENERIC;
1443 static int HttpCallback( httpd_file_sys_t *p_args,
1444 httpd_file_t *f, uint8_t *p_request,
1445 uint8_t **pp_data, int *pi_data )
1447 VLC_UNUSED(f); VLC_UNUSED(p_request);
1448 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1450 vlc_mutex_lock( &p_sys->lock_sdp );
1451 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1453 *pi_data = strlen( p_sys->psz_sdp );
1454 *pp_data = malloc( *pi_data );
1455 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1462 vlc_mutex_unlock( &p_sys->lock_sdp );
1467 /****************************************************************************
1469 ****************************************************************************/
1470 static void* ThreadSend( vlc_object_t *p_this )
1473 # define ECONNREFUSED WSAECONNREFUSED
1474 # define ENOPROTOOPT WSAENOPROTOOPT
1475 # define EHOSTUNREACH WSAEHOSTUNREACH
1476 # define ENETUNREACH WSAENETUNREACH
1477 # define ENETDOWN WSAENETDOWN
1478 # define ENOBUFS WSAENOBUFS
1479 # define EAGAIN WSAEWOULDBLOCK
1480 # define EWOULDBLOCK WSAEWOULDBLOCK
1482 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1483 unsigned i_caching = id->i_caching;
1487 block_t *out = block_FifoGet( id->p_fifo );
1488 block_cleanup_push (out);
1491 { /* FIXME: this is awfully inefficient */
1492 size_t len = out->i_buffer;
1493 out = block_Realloc( out, 0, len + 10 );
1494 out->i_buffer = len;
1496 int canc = vlc_savecancel ();
1497 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1498 vlc_restorecancel (canc);
1502 msg_Dbg( id, "SRTP sending error: %m" );
1503 block_Release( out );
1507 out->i_buffer = len;
1511 mwait (out->i_dts + i_caching);
1516 ssize_t len = out->i_buffer;
1517 int canc = vlc_savecancel ();
1519 vlc_mutex_lock( &id->lock_sink );
1520 unsigned deadc = 0; /* How many dead sockets? */
1521 int deadv[id->sinkc]; /* Dead sockets list */
1523 for( int i = 0; i < id->sinkc; i++ )
1525 if( !id->srtp ) /* FIXME: SRTCP support */
1526 SendRTCP( id->sinkv[i].rtcp, out );
1528 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1532 /* Soft errors (e.g. ICMP): */
1533 case ECONNREFUSED: /* Port unreachable */
1536 case EPROTO: /* Protocol unreachable */
1538 case EHOSTUNREACH: /* Host unreachable */
1539 case ENETUNREACH: /* Network unreachable */
1540 case ENETDOWN: /* Entire network down */
1541 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1542 /* Transient congestion: */
1543 case ENOMEM: /* out of socket buffers */
1546 #if (EAGAIN != EWOULDBLOCK)
1552 deadv[deadc++] = id->sinkv[i].rtp_fd;
1554 vlc_mutex_unlock( &id->lock_sink );
1555 block_Release( out );
1557 for( unsigned i = 0; i < deadc; i++ )
1559 msg_Dbg( id, "removing socket %d", deadv[i] );
1560 rtp_del_sink( id, deadv[i] );
1563 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1564 while( id->listen_fd != NULL )
1566 int fd = net_Accept( id, id->listen_fd, 0 );
1569 msg_Dbg( id, "adding socket %d", fd );
1570 rtp_add_sink( id, fd, true );
1572 vlc_restorecancel (canc);
1577 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1579 rtp_sink_t sink = { fd, NULL };
1580 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1582 if( sink.rtcp == NULL )
1583 msg_Err( id, "RTCP failed!" );
1585 vlc_mutex_lock( &id->lock_sink );
1586 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1587 vlc_mutex_unlock( &id->lock_sink );
1591 void rtp_del_sink( sout_stream_id_t *id, int fd )
1593 rtp_sink_t sink = { fd, NULL };
1595 /* NOTE: must be safe to use if fd is not included */
1596 vlc_mutex_lock( &id->lock_sink );
1597 for( int i = 0; i < id->sinkc; i++ )
1599 if (id->sinkv[i].rtp_fd == fd)
1601 sink = id->sinkv[i];
1602 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1606 vlc_mutex_unlock( &id->lock_sink );
1608 CloseRTCP( sink.rtcp );
1609 net_Close( sink.rtp_fd );
1612 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1614 /* This will return values for the next packet.
1615 * Accounting for caching would not be totally trivial. */
1616 return id->i_sequence;
1619 /* FIXME: this is pretty bad - if we remove and then insert an ES
1620 * the number will get unsynched from inside RTSP */
1621 unsigned rtp_get_num( const sout_stream_id_t *id )
1623 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1626 vlc_mutex_lock( &p_sys->lock_es );
1627 for( i = 0; i < p_sys->i_es; i++ )
1629 if( id == p_sys->es[i] )
1632 vlc_mutex_unlock( &p_sys->lock_es );
1638 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1639 int b_marker, int64_t i_pts )
1641 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
1643 out->p_buffer[0] = 0x80;
1644 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1645 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1646 out->p_buffer[3] = ( id->i_sequence )&0xff;
1647 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1648 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1649 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1650 out->p_buffer[7] = ( i_timestamp )&0xff;
1652 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1658 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1660 block_FifoPut( id->p_fifo, out );
1664 * @return configured max RTP payload size (including payload type-specific
1665 * headers, excluding RTP and transport headers)
1667 size_t rtp_mtu (const sout_stream_id_t *id)
1669 return id->i_mtu - 12;
1672 /*****************************************************************************
1674 *****************************************************************************/
1676 /** Add an ES to a non-RTP muxed stream */
1677 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1679 sout_input_t *p_input;
1680 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1681 assert( p_mux != NULL );
1683 p_input = sout_MuxAddStream( p_mux, p_fmt );
1684 if( p_input == NULL )
1686 msg_Err( p_stream, "cannot add this stream to the muxer" );
1690 return (sout_stream_id_t *)p_input;
1694 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1697 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1698 assert( p_mux != NULL );
1700 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1705 /** Remove an ES from a non-RTP muxed stream */
1706 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1708 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1709 assert( p_mux != NULL );
1711 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1716 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1717 const block_t *p_buffer )
1719 sout_stream_sys_t *p_sys = p_stream->p_sys;
1720 sout_stream_id_t *id = p_sys->es[0];
1722 int64_t i_dts = p_buffer->i_dts;
1724 uint8_t *p_data = p_buffer->p_buffer;
1725 size_t i_data = p_buffer->i_buffer;
1726 size_t i_max = id->i_mtu - 12;
1728 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1734 /* output complete packet */
1735 if( p_sys->packet &&
1736 p_sys->packet->i_buffer + i_data > i_max )
1738 rtp_packetize_send( id, p_sys->packet );
1739 p_sys->packet = NULL;
1742 if( p_sys->packet == NULL )
1744 /* allocate a new packet */
1745 p_sys->packet = block_New( p_stream, id->i_mtu );
1746 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1747 p_sys->packet->i_dts = i_dts;
1748 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1749 i_dts += p_sys->packet->i_length;
1752 i_size = __MIN( i_data,
1753 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1755 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1758 p_sys->packet->i_buffer += i_size;
1767 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1770 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1776 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1778 p_next = p_buffer->p_next;
1779 block_Release( p_buffer );
1787 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1789 sout_access_out_t *p_grab;
1791 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1792 if( p_grab == NULL )
1795 p_grab->p_module = NULL;
1796 p_grab->psz_access = strdup( "grab" );
1797 p_grab->p_cfg = NULL;
1798 p_grab->psz_path = strdup( "" );
1799 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1800 p_grab->pf_seek = NULL;
1801 p_grab->pf_write = AccessOutGrabberWrite;
1802 vlc_object_attach( p_grab, p_stream );