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>
41 #include <vlc_strings.h>
50 # include <sys/types.h>
53 #ifdef HAVE_ARPA_INET_H
54 # include <arpa/inet.h>
56 #ifdef HAVE_LINUX_DCCP_H
57 # include <linux/dccp.h>
60 # define IPPROTO_DCCP 33
62 #ifndef IPPROTO_UDPLITE
63 # define IPPROTO_UDPLITE 136
70 /*****************************************************************************
72 *****************************************************************************/
74 #define DEST_TEXT N_("Destination")
75 #define DEST_LONGTEXT N_( \
76 "This is the output URL that will be used." )
77 #define SDP_TEXT N_("SDP")
78 #define SDP_LONGTEXT N_( \
79 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
80 "session will be made available. You must use an url: http://location to " \
81 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
82 "for the SDP to be announced via SAP." )
83 #define SAP_TEXT N_("SAP announcing")
84 #define SAP_LONGTEXT N_("Announce this session with SAP.")
85 #define MUX_TEXT N_("Muxer")
86 #define MUX_LONGTEXT N_( \
87 "This allows you to specify the muxer used for the streaming output. " \
88 "Default is to use no muxer (standard RTP stream)." )
90 #define NAME_TEXT N_("Session name")
91 #define NAME_LONGTEXT N_( \
92 "This is the name of the session that will be announced in the SDP " \
93 "(Session Descriptor)." )
94 #define DESC_TEXT N_("Session description")
95 #define DESC_LONGTEXT N_( \
96 "This allows you to give a short description with details about the stream, " \
97 "that will be announced in the SDP (Session Descriptor)." )
98 #define URL_TEXT N_("Session URL")
99 #define URL_LONGTEXT N_( \
100 "This allows you to give an URL with more details about the stream " \
101 "(often the website of the streaming organization), that will " \
102 "be announced in the SDP (Session Descriptor)." )
103 #define EMAIL_TEXT N_("Session email")
104 #define EMAIL_LONGTEXT N_( \
105 "This allows you to give a contact mail address for the stream, that will " \
106 "be announced in the SDP (Session Descriptor)." )
107 #define PHONE_TEXT N_("Session phone number")
108 #define PHONE_LONGTEXT N_( \
109 "This allows you to give a contact telephone number for the stream, that will " \
110 "be announced in the SDP (Session Descriptor)." )
112 #define PORT_TEXT N_("Port")
113 #define PORT_LONGTEXT N_( \
114 "This allows you to specify the base port for the RTP streaming." )
115 #define PORT_AUDIO_TEXT N_("Audio port")
116 #define PORT_AUDIO_LONGTEXT N_( \
117 "This allows you to specify the default audio port for the RTP streaming." )
118 #define PORT_VIDEO_TEXT N_("Video port")
119 #define PORT_VIDEO_LONGTEXT N_( \
120 "This allows you to specify the default video port for the RTP streaming." )
122 #define TTL_TEXT N_("Hop limit (TTL)")
123 #define TTL_LONGTEXT N_( \
124 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
125 "the multicast packets sent by the stream output (-1 = use operating " \
126 "system built-in default).")
128 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
129 #define RTCP_MUX_LONGTEXT N_( \
130 "This sends and receives RTCP packet multiplexed over the same port " \
133 #define CACHING_TEXT N_("Caching value (ms)")
134 #define CACHING_LONGTEXT N_( \
135 "Default caching value for outbound RTP streams. This " \
136 "value should be set in milliseconds." )
138 #define PROTO_TEXT N_("Transport protocol")
139 #define PROTO_LONGTEXT N_( \
140 "This selects which transport protocol to use for RTP." )
142 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
143 #define SRTP_KEY_LONGTEXT N_( \
144 "RTP packets will be integrity-protected and ciphered "\
145 "with this Secure RTP master shared secret key.")
147 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
148 #define SRTP_SALT_LONGTEXT N_( \
149 "Secure RTP requires a (non-secret) master salt value.")
151 static const char *const ppsz_protos[] = {
152 "dccp", "sctp", "tcp", "udp", "udplite",
155 static const char *const ppsz_protocols[] = {
156 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
159 #define RFC3016_TEXT N_("MP4A LATM")
160 #define RFC3016_LONGTEXT N_( \
161 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
163 static int Open ( vlc_object_t * );
164 static void Close( vlc_object_t * );
166 #define SOUT_CFG_PREFIX "sout-rtp-"
167 #define MAX_EMPTY_BLOCKS 200
170 set_shortname( N_("RTP"))
171 set_description( N_("RTP stream output") )
172 set_capability( "sout stream", 0 )
173 add_shortcut( "rtp" )
174 set_category( CAT_SOUT )
175 set_subcategory( SUBCAT_SOUT_STREAM )
177 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
178 DEST_LONGTEXT, true )
179 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
181 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
183 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
186 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
187 NAME_LONGTEXT, true )
188 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
189 DESC_LONGTEXT, true )
190 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
192 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
193 EMAIL_LONGTEXT, true )
194 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
195 PHONE_LONGTEXT, true )
197 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
198 PROTO_LONGTEXT, false )
199 change_string_list( ppsz_protos, ppsz_protocols, NULL )
200 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
201 PORT_LONGTEXT, true )
202 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
203 PORT_AUDIO_LONGTEXT, true )
204 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
205 PORT_VIDEO_LONGTEXT, true )
207 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
209 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
210 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
211 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
212 CACHING_TEXT, CACHING_LONGTEXT, true )
215 add_string( SOUT_CFG_PREFIX "key", "",
216 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
217 add_string( SOUT_CFG_PREFIX "salt", "",
218 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
221 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT,
222 RFC3016_LONGTEXT, false )
224 set_callbacks( Open, Close )
227 /*****************************************************************************
228 * Exported prototypes
229 *****************************************************************************/
230 static const char *const ppsz_sout_options[] = {
231 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
232 "sap", "description", "url", "email", "phone",
233 "proto", "rtcp-mux", "caching", "key", "salt",
237 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
238 static int Del ( sout_stream_t *, sout_stream_id_t * );
239 static int Send( sout_stream_t *, sout_stream_id_t *,
241 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
242 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
243 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
246 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
247 static void* ThreadSend( void * );
248 static void *rtp_listen_thread( void * );
250 static void SDPHandleUrl( sout_stream_t *, const char * );
252 static int SapSetup( sout_stream_t *p_stream );
253 static int FileSetup( sout_stream_t *p_stream );
254 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
256 struct sout_stream_sys_t
260 vlc_mutex_t lock_sdp;
267 session_descriptor_t *p_session;
270 httpd_host_t *p_httpd_host;
271 httpd_file_t *p_httpd_file;
276 /* RTSP NPT and timestamp computations */
277 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
278 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
279 int64_t i_pts_offset; /* matches actual PTS to prediction */
283 char *psz_destination;
284 uint32_t payload_bitmap;
286 uint16_t i_port_audio;
287 uint16_t i_port_video;
293 /* in case we do TS/PS over rtp */
295 sout_access_out_t *p_grab;
301 sout_stream_id_t **es;
304 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
306 typedef struct rtp_sink_t
312 struct sout_stream_id_t
314 sout_stream_t *p_stream;
317 uint8_t i_payload_type;
319 uint32_t i_ts_offset;
323 uint16_t i_seq_sent_next;
334 /* Packetizer specific fields */
337 srtp_session_t *srtp;
339 pf_rtp_packetizer_t pf_packetize;
343 vlc_mutex_t lock_sink;
346 rtsp_stream_id_t *rtsp_id;
352 block_fifo_t *p_fifo;
356 /*****************************************************************************
358 *****************************************************************************/
359 static int Open( vlc_object_t *p_this )
361 sout_stream_t *p_stream = (sout_stream_t*)p_this;
362 sout_instance_t *p_sout = p_stream->p_sout;
363 sout_stream_sys_t *p_sys = NULL;
364 config_chain_t *p_cfg = NULL;
368 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
369 ppsz_sout_options, p_stream->p_cfg );
371 p_sys = malloc( sizeof( sout_stream_sys_t ) );
375 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
377 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
378 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
379 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
380 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
382 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
384 msg_Err( p_stream, "audio and video RTP port must be distinct" );
385 free( p_sys->psz_destination );
390 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
392 if( !strcmp( p_cfg->psz_name, "sdp" )
393 && ( p_cfg->psz_value != NULL )
394 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
402 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
405 if( !strncasecmp( psz, "rtsp:", 5 ) )
411 /* Transport protocol */
412 p_sys->proto = IPPROTO_UDP;
413 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
415 if ((psz == NULL) || !strcasecmp (psz, "udp"))
416 (void)0; /* default */
418 if (!strcasecmp (psz, "dccp"))
420 p_sys->proto = IPPROTO_DCCP;
421 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
425 if (!strcasecmp (psz, "sctp"))
427 p_sys->proto = IPPROTO_TCP;
428 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
433 if (!strcasecmp (psz, "tcp"))
435 p_sys->proto = IPPROTO_TCP;
436 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
440 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
441 p_sys->proto = IPPROTO_UDPLITE;
443 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
446 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
448 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
450 msg_Err( p_stream, "missing destination and not in RTSP mode" );
455 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
456 if( p_sys->i_ttl == -1 )
458 /* Normally, we should let the default hop limit up to the core,
459 * but we have to know it to write our RTSP headers properly,
460 * which is why we ask the core. FIXME: broken when neither
461 * sout-rtp-ttl nor ttl are set. */
462 p_sys->i_ttl = var_InheritInteger( p_stream, "ttl" );
465 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
467 /* NPT=0 time will be determined when we packetize the first packet
468 * (of any ES). But we want to be able to report rtptime in RTSP
469 * without waiting. So until then, we use an arbitrary reference
470 * PTS for timestamp computations, and then actual PTS will catch
471 * up using offsets. */
472 p_sys->i_npt_zero = VLC_TS_INVALID;
473 p_sys->i_pts_zero = mdate(); /* arbitrary value, could probably be
475 p_sys->payload_bitmap = 0xFFFFFFFF;
479 p_sys->psz_sdp = NULL;
481 p_sys->b_export_sap = false;
482 p_sys->p_session = NULL;
483 p_sys->psz_sdp_file = NULL;
485 p_sys->p_httpd_host = NULL;
486 p_sys->p_httpd_file = NULL;
488 p_stream->p_sys = p_sys;
490 vlc_mutex_init( &p_sys->lock_sdp );
491 vlc_mutex_init( &p_sys->lock_ts );
492 vlc_mutex_init( &p_sys->lock_es );
494 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
497 sout_stream_id_t *id;
499 /* Check muxer type */
500 if( strncasecmp( psz, "ps", 2 )
501 && strncasecmp( psz, "mpeg1", 5 )
502 && strncasecmp( psz, "ts", 2 ) )
504 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
506 vlc_mutex_destroy( &p_sys->lock_sdp );
507 vlc_mutex_destroy( &p_sys->lock_es );
508 free( p_sys->psz_destination );
513 p_sys->p_grab = GrabberCreate( p_stream );
514 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
517 if( p_sys->p_mux == NULL )
519 msg_Err( p_stream, "cannot create muxer" );
520 sout_AccessOutDelete( p_sys->p_grab );
521 vlc_mutex_destroy( &p_sys->lock_sdp );
522 vlc_mutex_destroy( &p_sys->lock_es );
523 free( p_sys->psz_destination );
528 id = Add( p_stream, NULL );
531 sout_MuxDelete( p_sys->p_mux );
532 sout_AccessOutDelete( p_sys->p_grab );
533 vlc_mutex_destroy( &p_sys->lock_sdp );
534 vlc_mutex_destroy( &p_sys->lock_es );
535 free( p_sys->psz_destination );
540 p_sys->packet = NULL;
542 p_stream->pf_add = MuxAdd;
543 p_stream->pf_del = MuxDel;
544 p_stream->pf_send = MuxSend;
549 p_sys->p_grab = NULL;
551 p_stream->pf_add = Add;
552 p_stream->pf_del = Del;
553 p_stream->pf_send = Send;
556 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
557 SDPHandleUrl( p_stream, "sap" );
559 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
562 config_chain_t *p_cfg;
564 SDPHandleUrl( p_stream, psz );
566 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
568 if( !strcmp( p_cfg->psz_name, "sdp" ) )
570 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
573 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
574 if( !strcmp( p_cfg->psz_value, psz ) )
577 SDPHandleUrl( p_stream, p_cfg->psz_value );
583 /* update p_sout->i_out_pace_nocontrol */
584 p_stream->p_sout->i_out_pace_nocontrol++;
589 /*****************************************************************************
591 *****************************************************************************/
592 static void Close( vlc_object_t * p_this )
594 sout_stream_t *p_stream = (sout_stream_t*)p_this;
595 sout_stream_sys_t *p_sys = p_stream->p_sys;
597 /* update p_sout->i_out_pace_nocontrol */
598 p_stream->p_sout->i_out_pace_nocontrol--;
602 assert( p_sys->i_es == 1 );
604 sout_MuxDelete( p_sys->p_mux );
605 Del( p_stream, p_sys->es[0] );
606 sout_AccessOutDelete( p_sys->p_grab );
610 block_Release( p_sys->packet );
612 if( p_sys->b_export_sap )
615 SapSetup( p_stream );
619 if( p_sys->rtsp != NULL )
620 RtspUnsetup( p_sys->rtsp );
622 vlc_mutex_destroy( &p_sys->lock_sdp );
623 vlc_mutex_destroy( &p_sys->lock_ts );
624 vlc_mutex_destroy( &p_sys->lock_es );
626 if( p_sys->p_httpd_file )
627 httpd_FileDelete( p_sys->p_httpd_file );
629 if( p_sys->p_httpd_host )
630 httpd_HostDelete( p_sys->p_httpd_host );
632 free( p_sys->psz_sdp );
634 if( p_sys->psz_sdp_file != NULL )
637 unlink( p_sys->psz_sdp_file );
639 free( p_sys->psz_sdp_file );
641 free( p_sys->psz_destination );
645 /*****************************************************************************
647 *****************************************************************************/
648 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
650 sout_stream_sys_t *p_sys = p_stream->p_sys;
653 vlc_UrlParse( &url, psz_url, 0 );
654 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
656 if( p_sys->p_httpd_file )
658 msg_Err( p_stream, "you can use sdp=http:// only once" );
662 if( HttpSetup( p_stream, &url ) )
664 msg_Err( p_stream, "cannot export SDP as HTTP" );
667 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
669 if( p_sys->rtsp != NULL )
671 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
675 /* FIXME test if destination is multicast or no destination at all */
676 p_sys->rtsp = RtspSetup( p_stream, &url );
677 if( p_sys->rtsp == NULL )
678 msg_Err( p_stream, "cannot export SDP as RTSP" );
680 if( p_sys->p_mux != NULL )
682 sout_stream_id_t *id = p_sys->es[0];
683 rtsp_stream_id_t *rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
684 p_sys->psz_destination, p_sys->i_ttl,
685 id->i_port, id->i_port + 1 );
686 vlc_mutex_lock( &p_sys->lock_es );
687 id->rtsp_id = rtsp_id;
688 vlc_mutex_unlock( &p_sys->lock_es );
691 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
692 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
694 p_sys->b_export_sap = true;
695 SapSetup( p_stream );
697 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
699 if( p_sys->psz_sdp_file != NULL )
701 msg_Err( p_stream, "you can use sdp=file:// only once" );
704 p_sys->psz_sdp_file = make_path( psz_url );
705 if( p_sys->psz_sdp_file == NULL )
707 FileSetup( p_stream );
711 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
716 vlc_UrlClean( &url );
719 /*****************************************************************************
721 *****************************************************************************/
723 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
725 sout_stream_sys_t *p_sys = p_stream->p_sys;
726 char *psz_sdp = NULL;
727 struct sockaddr_storage dst;
731 * When we have a fixed destination (typically when we do multicast),
732 * we need to put the actual port numbers in the SDP.
733 * When there is no fixed destination, we only support RTSP unicast
734 * on-demand setup, so we should rather let the clients decide which ports
736 * When there is both a fixed destination and RTSP unicast, we need to
737 * put port numbers used by the fixed destination, otherwise the SDP would
738 * become totally incorrect for multicast use. It should be noted that
739 * port numbers from SDP with RTSP are only "recommendation" from the
740 * server to the clients (per RFC2326), so only broken clients will fail
741 * to handle this properly. There is no solution but to use two differents
742 * output chain with two different RTSP URLs if you need to handle this
747 vlc_mutex_lock( &p_sys->lock_es );
748 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
749 goto out; /* hmm... */
751 if( p_sys->psz_destination != NULL )
755 /* Oh boy, this is really ugly! */
756 dstlen = sizeof( dst );
757 if( p_sys->es[0]->listen.fd != NULL )
758 getsockname( p_sys->es[0]->listen.fd[0],
759 (struct sockaddr *)&dst, &dstlen );
761 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
762 (struct sockaddr *)&dst, &dstlen );
768 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
769 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
770 && rtsp_url[7] == '[';
772 /* Dummy destination address for RTSP */
773 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
774 : sizeof( struct sockaddr_in );
775 memset (&dst, 0, dstlen);
776 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
782 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
783 NULL, 0, (struct sockaddr *)&dst, dstlen );
784 if( psz_sdp == NULL )
787 /* TODO: a=source-filter */
788 if( p_sys->rtcp_mux )
789 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
791 if( rtsp_url != NULL )
792 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
794 const char *proto = "RTP/AVP"; /* protocol */
795 if( rtsp_url == NULL )
797 switch( p_sys->proto )
802 proto = "TCP/RTP/AVP";
805 proto = "DCCP/RTP/AVP";
807 case IPPROTO_UDPLITE:
812 for( i = 0; i < p_sys->i_es; i++ )
814 sout_stream_id_t *id = p_sys->es[i];
815 const char *mime_major; /* major MIME type */
820 mime_major = "video";
823 mime_major = "audio";
832 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
833 id->i_payload_type, false, id->i_bitrate,
834 id->psz_enc, id->i_clock_rate, id->i_channels,
837 /* cf RFC4566 §5.14 */
838 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
839 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
841 if( rtsp_url != NULL )
843 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
844 if( track_url != NULL )
846 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
852 if( id->listen.fd != NULL )
853 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
854 if( p_sys->proto == IPPROTO_DCCP )
855 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
856 "SC:RTP%c", toupper( mime_major[0] ) );
860 vlc_mutex_unlock( &p_sys->lock_es );
864 /*****************************************************************************
866 *****************************************************************************/
868 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
870 static const char hex[16] = "0123456789abcdef";
872 for( int i = 0; i < i_data; i++ )
874 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
875 s[2*i+1] = hex[(p_data[i] )&0xf];
881 * Shrink the MTU down to a fixed packetization time (for audio).
884 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
886 /* Samples per second */
887 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
888 bytes *= id->i_channels;
891 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
892 id->i_mtu = 12 + spl;
893 else /* MTU is too small for ptime, align to a sample boundary */
894 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
897 uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts )
899 /* NOTE: this plays nice with offsets because the calculations are
901 return i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
904 /** Add an ES as a new RTP stream */
905 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
907 /* NOTE: As a special case, if we use a non-RTP
908 * mux (TS/PS), then p_fmt is NULL. */
909 sout_stream_sys_t *p_sys = p_stream->p_sys;
912 if (0 == p_sys->payload_bitmap)
914 msg_Err (p_stream, "too many RTP elementary streams");
918 sout_stream_id_t *id = malloc( sizeof( *id ) );
919 if( unlikely(id == NULL) )
921 id->p_stream = p_stream;
923 /* Look for free dymanic payload type */
924 id->i_payload_type = 96 + clz32 (p_sys->payload_bitmap);
925 assert (id->i_payload_type < 128);
927 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
928 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
932 id->i_clock_rate = 90000; /* most common case for video */
936 id->i_cat = p_fmt->i_cat;
937 if( p_fmt->i_cat == AUDIO_ES )
939 id->i_clock_rate = p_fmt->audio.i_rate;
940 id->i_channels = p_fmt->audio.i_channels;
942 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
946 id->i_cat = VIDEO_ES;
950 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
951 if( id->i_mtu <= 12 + 16 )
952 id->i_mtu = 576 - 20 - 8; /* pessimistic */
953 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
955 id->pf_packetize = NULL;
960 vlc_mutex_init( &id->lock_sink );
965 id->listen.fd = NULL;
968 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
971 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
974 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
975 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
976 if (id->srtp == NULL)
982 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
983 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
988 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
991 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
995 id->i_seq_sent_next = id->i_sequence;
997 if( p_sys->psz_destination != NULL )
999 /* Choose the port */
1000 uint16_t i_port = 0;
1004 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1005 i_port = p_sys->i_port_audio;
1007 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1008 i_port = p_sys->i_port_video;
1010 /* We do not need the ES lock (p_sys->lock_es) here, because
1011 * this is the only one thread that can *modify* the ES table.
1012 * The ES lock protects the other threads from our modifications
1013 * (TAB_APPEND, TAB_REMOVE). */
1014 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1015 if (i_port == p_sys->es[i]->i_port)
1016 i_port = 0; /* Port already in use! */
1017 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1021 msg_Err (p_stream, "too many RTP elementary streams");
1025 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1026 if (p == p_sys->es[i]->i_port)
1030 id->i_port = i_port;
1032 int type = SOCK_STREAM;
1034 switch( p_sys->proto )
1042 case VIDEO_ES: code = "RTPV"; break;
1043 case AUDIO_ES: code = "RTPARTPV"; break;
1044 case SPU_ES: code = "RTPTRTPV"; break;
1045 default: code = "RTPORTPV"; break;
1047 var_SetString (p_stream, "dccp-service", code);
1049 } /* fall through */
1052 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1053 p_sys->psz_destination, i_port,
1054 type, p_sys->proto );
1055 if( id->listen.fd == NULL )
1057 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1060 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1061 VLC_THREAD_PRIORITY_LOW ) )
1063 net_ListenClose( id->listen.fd );
1064 id->listen.fd = NULL;
1071 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1072 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1073 i_port, ttl, p_sys->proto );
1076 msg_Err( p_stream, "cannot create RTP socket" );
1079 /* Ignore any unexpected incoming packet (including RTCP-RR
1080 * packets in case of rtcp-mux) */
1081 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1083 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1090 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1092 if( psz == NULL ) /* Uho! */
1095 if( strncmp( psz, "ts", 2 ) == 0 )
1097 id->i_payload_type = 33;
1098 id->psz_enc = "MP2T";
1102 id->psz_enc = "MP2P";
1107 switch( p_fmt->i_codec )
1109 case VLC_CODEC_MULAW:
1110 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1111 id->i_payload_type = 0;
1112 id->psz_enc = "PCMU";
1113 id->pf_packetize = rtp_packetize_split;
1114 rtp_set_ptime (id, 20, 1);
1116 case VLC_CODEC_ALAW:
1117 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1118 id->i_payload_type = 8;
1119 id->psz_enc = "PCMA";
1120 id->pf_packetize = rtp_packetize_split;
1121 rtp_set_ptime (id, 20, 1);
1123 case VLC_CODEC_S16B:
1124 case VLC_CODEC_S16L:
1125 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1127 id->i_payload_type = 11;
1129 else if( p_fmt->audio.i_channels == 2 &&
1130 p_fmt->audio.i_rate == 44100 )
1132 id->i_payload_type = 10;
1134 id->psz_enc = "L16";
1135 if( p_fmt->i_codec == VLC_CODEC_S16B )
1136 id->pf_packetize = rtp_packetize_split;
1138 id->pf_packetize = rtp_packetize_swab;
1139 rtp_set_ptime (id, 20, 2);
1143 id->pf_packetize = rtp_packetize_split;
1144 rtp_set_ptime (id, 20, 1);
1146 case VLC_CODEC_MPGA:
1147 id->i_payload_type = 14;
1148 id->psz_enc = "MPA";
1149 id->i_clock_rate = 90000; /* not 44100 */
1150 id->pf_packetize = rtp_packetize_mpa;
1152 case VLC_CODEC_MPGV:
1153 id->i_payload_type = 32;
1154 id->psz_enc = "MPV";
1155 id->pf_packetize = rtp_packetize_mpv;
1157 case VLC_CODEC_ADPCM_G726:
1158 switch( p_fmt->i_bitrate / 1000 )
1161 id->psz_enc = "G726-16";
1162 id->pf_packetize = rtp_packetize_g726_16;
1165 id->psz_enc = "G726-24";
1166 id->pf_packetize = rtp_packetize_g726_24;
1169 id->psz_enc = "G726-32";
1170 id->pf_packetize = rtp_packetize_g726_32;
1173 id->psz_enc = "G726-40";
1174 id->pf_packetize = rtp_packetize_g726_40;
1177 msg_Err( p_stream, "cannot add this stream (unsupported "
1178 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1183 id->psz_enc = "ac3";
1184 id->pf_packetize = rtp_packetize_ac3;
1186 case VLC_CODEC_H263:
1187 id->psz_enc = "H263-1998";
1188 id->pf_packetize = rtp_packetize_h263;
1190 case VLC_CODEC_H264:
1191 id->psz_enc = "H264";
1192 id->pf_packetize = rtp_packetize_h264;
1193 id->psz_fmtp = NULL;
1195 if( p_fmt->i_extra > 0 )
1197 uint8_t *p_buffer = p_fmt->p_extra;
1198 int i_buffer = p_fmt->i_extra;
1199 char *p_64_sps = NULL;
1200 char *p_64_pps = NULL;
1203 while( i_buffer > 4 )
1208 while( p_buffer[0] != 0 || p_buffer[1] != 0 ||
1213 if( i_buffer == 0 ) break;
1216 if( i_buffer < 4 || memcmp(p_buffer, "\x00\x00\x01", 3 ) )
1218 msg_Dbg( p_stream, "No startcode found..");
1224 const int i_nal_type = p_buffer[0]&0x1f;
1226 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1229 for( i_offset = 0; i_offset+2 < i_buffer ; i_offset++)
1231 if( !memcmp(p_buffer + i_offset, "\x00\x00\x01", 3 ) )
1233 /* we found another startcode */
1234 while( i_offset > 0 && 0 == p_buffer[ i_offset - 1 ] )
1243 msg_Dbg( p_stream, "No-info found in nal ");
1247 if( i_nal_type == 7 )
1250 p_64_sps = vlc_b64_encode_binary( p_buffer, i_size );
1251 /* XXX: nothing ensures that i_size >= 4 ?? */
1252 sprintf_hexa( hexa, &p_buffer[1], 3 );
1254 else if( i_nal_type == 8 )
1257 p_64_pps = vlc_b64_encode_binary( p_buffer, i_size );
1263 if( p_64_sps && p_64_pps &&
1264 ( asprintf( &id->psz_fmtp,
1265 "packetization-mode=1;profile-level-id=%s;"
1266 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1267 p_64_pps ) == -1 ) )
1268 id->psz_fmtp = NULL;
1273 id->psz_fmtp = strdup( "packetization-mode=1" );
1276 case VLC_CODEC_MP4V:
1278 id->psz_enc = "MP4V-ES";
1279 id->pf_packetize = rtp_packetize_split;
1280 if( p_fmt->i_extra > 0 )
1282 char hexa[2*p_fmt->i_extra +1];
1283 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1284 if( asprintf( &id->psz_fmtp,
1285 "profile-level-id=3; config=%s;", hexa ) == -1 )
1286 id->psz_fmtp = NULL;
1290 case VLC_CODEC_MP4A:
1294 char hexa[2*p_fmt->i_extra +1];
1296 id->psz_enc = "mpeg4-generic";
1297 id->pf_packetize = rtp_packetize_mp4a;
1298 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1299 if( asprintf( &id->psz_fmtp,
1300 "streamtype=5; profile-level-id=15; "
1301 "mode=AAC-hbr; config=%s; SizeLength=13; "
1302 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1304 id->psz_fmtp = NULL;
1310 unsigned char config[6];
1311 unsigned int aacsrates[15] = {
1312 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1313 16000, 12000, 11025, 8000, 7350, 0, 0 };
1315 for( i = 0; i < 15; i++ )
1316 if( p_fmt->audio.i_rate == aacsrates[i] )
1322 config[3]=p_fmt->audio.i_channels<<4;
1326 id->psz_enc = "MP4A-LATM";
1327 id->pf_packetize = rtp_packetize_mp4a_latm;
1328 sprintf_hexa( hexa, config, 6 );
1329 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1330 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1331 id->psz_fmtp = NULL;
1335 case VLC_CODEC_AMR_NB:
1336 id->psz_enc = "AMR";
1337 id->psz_fmtp = strdup( "octet-align=1" );
1338 id->pf_packetize = rtp_packetize_amr;
1340 case VLC_CODEC_AMR_WB:
1341 id->psz_enc = "AMR-WB";
1342 id->psz_fmtp = strdup( "octet-align=1" );
1343 id->pf_packetize = rtp_packetize_amr;
1345 case VLC_CODEC_SPEEX:
1346 id->psz_enc = "SPEEX";
1347 id->pf_packetize = rtp_packetize_spx;
1349 case VLC_CODEC_ITU_T140:
1350 id->psz_enc = "t140" ;
1351 id->i_clock_rate = 1000;
1352 id->pf_packetize = rtp_packetize_t140;
1356 msg_Err( p_stream, "cannot add this stream (unsupported "
1357 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1360 if (id->i_payload_type >= 96)
1361 /* Mark dynamic payload type in use */
1362 p_sys->payload_bitmap &= ~(1 << (127 - id->i_payload_type));
1364 #if 0 /* No payload formats sets this at the moment */
1367 cscov += 8 /* UDP */ + 12 /* RTP */;
1369 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1372 vlc_mutex_lock( &p_sys->lock_ts );
1373 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1374 vlc_mutex_unlock( &p_sys->lock_ts );
1376 id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
1378 if( p_sys->rtsp != NULL )
1379 id->rtsp_id = RtspAddId( p_sys->rtsp, id,
1380 GetDWBE( id->ssrc ),
1381 p_sys->psz_destination,
1382 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1384 id->p_fifo = block_FifoNew();
1385 if( unlikely(id->p_fifo == NULL) )
1387 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1389 block_FifoRelease( id->p_fifo );
1394 /* Update p_sys context */
1395 vlc_mutex_lock( &p_sys->lock_es );
1396 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1397 vlc_mutex_unlock( &p_sys->lock_es );
1399 psz_sdp = SDPGenerate( p_stream, NULL );
1401 vlc_mutex_lock( &p_sys->lock_sdp );
1402 free( p_sys->psz_sdp );
1403 p_sys->psz_sdp = psz_sdp;
1404 vlc_mutex_unlock( &p_sys->lock_sdp );
1406 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1408 /* Update SDP (sap/file) */
1409 if( p_sys->b_export_sap ) SapSetup( p_stream );
1410 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1415 Del( p_stream, id );
1419 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1421 sout_stream_sys_t *p_sys = p_stream->p_sys;
1423 vlc_mutex_lock( &p_sys->lock_es );
1424 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1425 vlc_mutex_unlock( &p_sys->lock_es );
1427 if( likely(id->p_fifo != NULL) )
1429 vlc_cancel( id->thread );
1430 vlc_join( id->thread, NULL );
1431 block_FifoRelease( id->p_fifo );
1434 /* Release dynamic payload type */
1435 if (id->i_payload_type >= 96)
1436 p_sys->payload_bitmap |= 1 << (127 - id->i_payload_type);
1438 free( id->psz_fmtp );
1441 RtspDelId( p_sys->rtsp, id->rtsp_id );
1442 if( id->listen.fd != NULL )
1444 vlc_cancel( id->listen.thread );
1445 vlc_join( id->listen.thread, NULL );
1446 net_ListenClose( id->listen.fd );
1448 /* Delete remaining sinks (incoming connections or explicit
1450 while( id->sinkc > 0 )
1451 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1453 if( id->srtp != NULL )
1454 srtp_destroy( id->srtp );
1457 vlc_mutex_destroy( &id->lock_sink );
1459 /* Update SDP (sap/file) */
1460 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1461 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1467 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1472 assert( p_stream->p_sys->p_mux == NULL );
1475 while( p_buffer != NULL )
1477 p_next = p_buffer->p_next;
1478 if( id->pf_packetize( id, p_buffer ) )
1481 block_Release( p_buffer );
1487 /****************************************************************************
1489 ****************************************************************************/
1490 static int SapSetup( sout_stream_t *p_stream )
1492 sout_stream_sys_t *p_sys = p_stream->p_sys;
1493 sout_instance_t *p_sout = p_stream->p_sout;
1495 /* Remove the previous session */
1496 if( p_sys->p_session != NULL)
1498 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1499 p_sys->p_session = NULL;
1502 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1504 announce_method_t *p_method = sout_SAPMethod();
1505 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1507 p_sys->psz_destination,
1509 sout_MethodRelease( p_method );
1515 /****************************************************************************
1517 ****************************************************************************/
1518 static int FileSetup( sout_stream_t *p_stream )
1520 sout_stream_sys_t *p_sys = p_stream->p_sys;
1523 if( p_sys->psz_sdp == NULL )
1524 return VLC_EGENERIC; /* too early */
1526 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1528 msg_Err( p_stream, "cannot open file '%s' (%m)",
1529 p_sys->psz_sdp_file );
1530 return VLC_EGENERIC;
1533 fputs( p_sys->psz_sdp, f );
1539 /****************************************************************************
1541 ****************************************************************************/
1542 static int HttpCallback( httpd_file_sys_t *p_args,
1543 httpd_file_t *, uint8_t *p_request,
1544 uint8_t **pp_data, int *pi_data );
1546 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1548 sout_stream_sys_t *p_sys = p_stream->p_sys;
1550 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1551 url->i_port > 0 ? url->i_port : 80 );
1552 if( p_sys->p_httpd_host )
1554 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1555 url->psz_path ? url->psz_path : "/",
1558 HttpCallback, (void*)p_sys );
1560 if( p_sys->p_httpd_file == NULL )
1562 return VLC_EGENERIC;
1567 static int HttpCallback( httpd_file_sys_t *p_args,
1568 httpd_file_t *f, uint8_t *p_request,
1569 uint8_t **pp_data, int *pi_data )
1571 VLC_UNUSED(f); VLC_UNUSED(p_request);
1572 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1574 vlc_mutex_lock( &p_sys->lock_sdp );
1575 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1577 *pi_data = strlen( p_sys->psz_sdp );
1578 *pp_data = malloc( *pi_data );
1579 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1586 vlc_mutex_unlock( &p_sys->lock_sdp );
1591 /****************************************************************************
1593 ****************************************************************************/
1594 static void* ThreadSend( void *data )
1597 # define ECONNREFUSED WSAECONNREFUSED
1598 # define ENOPROTOOPT WSAENOPROTOOPT
1599 # define EHOSTUNREACH WSAEHOSTUNREACH
1600 # define ENETUNREACH WSAENETUNREACH
1601 # define ENETDOWN WSAENETDOWN
1602 # define ENOBUFS WSAENOBUFS
1603 # define EAGAIN WSAEWOULDBLOCK
1604 # define EWOULDBLOCK WSAEWOULDBLOCK
1606 sout_stream_id_t *id = data;
1607 unsigned i_caching = id->i_caching;
1611 block_t *out = block_FifoGet( id->p_fifo );
1612 block_cleanup_push (out);
1616 { /* FIXME: this is awfully inefficient */
1617 size_t len = out->i_buffer;
1618 out = block_Realloc( out, 0, len + 10 );
1619 out->i_buffer = len;
1621 int canc = vlc_savecancel ();
1622 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1623 vlc_restorecancel (canc);
1627 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1628 block_Release( out );
1632 out->i_buffer = len;
1636 mwait (out->i_dts + i_caching);
1641 ssize_t len = out->i_buffer;
1642 int canc = vlc_savecancel ();
1644 vlc_mutex_lock( &id->lock_sink );
1645 unsigned deadc = 0; /* How many dead sockets? */
1646 int deadv[id->sinkc]; /* Dead sockets list */
1648 for( int i = 0; i < id->sinkc; i++ )
1651 if( !id->srtp ) /* FIXME: SRTCP support */
1653 SendRTCP( id->sinkv[i].rtcp, out );
1655 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1659 /* Soft errors (e.g. ICMP): */
1660 case ECONNREFUSED: /* Port unreachable */
1663 case EPROTO: /* Protocol unreachable */
1665 case EHOSTUNREACH: /* Host unreachable */
1666 case ENETUNREACH: /* Network unreachable */
1667 case ENETDOWN: /* Entire network down */
1668 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1669 /* Transient congestion: */
1670 case ENOMEM: /* out of socket buffers */
1673 #if (EAGAIN != EWOULDBLOCK)
1679 deadv[deadc++] = id->sinkv[i].rtp_fd;
1681 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1682 vlc_mutex_unlock( &id->lock_sink );
1683 block_Release( out );
1685 for( unsigned i = 0; i < deadc; i++ )
1687 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1688 rtp_del_sink( id, deadv[i] );
1690 vlc_restorecancel (canc);
1696 /* This thread dequeues incoming connections (DCCP streaming) */
1697 static void *rtp_listen_thread( void *data )
1699 sout_stream_id_t *id = data;
1701 assert( id->listen.fd != NULL );
1705 int fd = net_Accept( id->p_stream, id->listen.fd );
1708 int canc = vlc_savecancel( );
1709 rtp_add_sink( id, fd, true, NULL );
1710 vlc_restorecancel( canc );
1717 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1719 rtp_sink_t sink = { fd, NULL };
1720 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1722 if( sink.rtcp == NULL )
1723 msg_Err( id->p_stream, "RTCP failed!" );
1725 vlc_mutex_lock( &id->lock_sink );
1726 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1728 *seq = id->i_seq_sent_next;
1729 vlc_mutex_unlock( &id->lock_sink );
1733 void rtp_del_sink( sout_stream_id_t *id, int fd )
1735 rtp_sink_t sink = { fd, NULL };
1737 /* NOTE: must be safe to use if fd is not included */
1738 vlc_mutex_lock( &id->lock_sink );
1739 for( int i = 0; i < id->sinkc; i++ )
1741 if (id->sinkv[i].rtp_fd == fd)
1743 sink = id->sinkv[i];
1744 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1748 vlc_mutex_unlock( &id->lock_sink );
1750 CloseRTCP( sink.rtcp );
1751 net_Close( sink.rtp_fd );
1754 uint16_t rtp_get_seq( sout_stream_id_t *id )
1756 /* This will return values for the next packet. */
1759 vlc_mutex_lock( &id->lock_sink );
1760 seq = id->i_seq_sent_next;
1761 vlc_mutex_unlock( &id->lock_sink );
1766 /* Return a timestamp corresponding to packets being sent now, and that
1767 * can be passed to rtp_compute_ts() to get rtptime values for each ES. */
1768 int64_t rtp_get_ts( const sout_stream_t *p_stream )
1770 sout_stream_sys_t *p_sys = p_stream->p_sys;
1772 vlc_mutex_lock( &p_sys->lock_ts );
1773 i_npt_zero = p_sys->i_npt_zero;
1774 vlc_mutex_unlock( &p_sys->lock_ts );
1776 if( i_npt_zero == VLC_TS_INVALID )
1777 return p_sys->i_pts_zero;
1779 mtime_t now = mdate();
1780 if( now < i_npt_zero )
1781 return p_sys->i_pts_zero;
1783 return p_sys->i_pts_zero + (now - i_npt_zero);
1786 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1787 int b_marker, int64_t i_pts )
1789 if( !id->b_ts_init )
1791 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1792 vlc_mutex_lock( &p_sys->lock_ts );
1793 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1795 /* This is the first packet of any ES. We initialize the
1796 * NPT=0 time reference, and the offset to match the
1797 * arbitrary PTS reference. */
1798 p_sys->i_npt_zero = i_pts + id->i_caching;
1799 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1801 vlc_mutex_unlock( &p_sys->lock_ts );
1803 /* And in any case this is the first packet of this ES, so we
1804 * initialize the offset for this ES. */
1805 id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
1806 id->b_ts_init = true;
1809 uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset;
1811 out->p_buffer[0] = 0x80;
1812 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1813 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1814 out->p_buffer[3] = ( id->i_sequence )&0xff;
1815 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1816 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1817 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1818 out->p_buffer[7] = ( i_timestamp )&0xff;
1820 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1826 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1828 block_FifoPut( id->p_fifo, out );
1832 * @return configured max RTP payload size (including payload type-specific
1833 * headers, excluding RTP and transport headers)
1835 size_t rtp_mtu (const sout_stream_id_t *id)
1837 return id->i_mtu - 12;
1840 /*****************************************************************************
1842 *****************************************************************************/
1844 /** Add an ES to a non-RTP muxed stream */
1845 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1847 sout_input_t *p_input;
1848 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1849 assert( p_mux != NULL );
1851 p_input = sout_MuxAddStream( p_mux, p_fmt );
1852 if( p_input == NULL )
1854 msg_Err( p_stream, "cannot add this stream to the muxer" );
1858 return (sout_stream_id_t *)p_input;
1862 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1865 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1866 assert( p_mux != NULL );
1868 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1873 /** Remove an ES from a non-RTP muxed stream */
1874 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1876 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1877 assert( p_mux != NULL );
1879 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1884 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1885 const block_t *p_buffer )
1887 sout_stream_sys_t *p_sys = p_stream->p_sys;
1888 sout_stream_id_t *id = p_sys->es[0];
1890 int64_t i_dts = p_buffer->i_dts;
1892 uint8_t *p_data = p_buffer->p_buffer;
1893 size_t i_data = p_buffer->i_buffer;
1894 size_t i_max = id->i_mtu - 12;
1896 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1902 /* output complete packet */
1903 if( p_sys->packet &&
1904 p_sys->packet->i_buffer + i_data > i_max )
1906 rtp_packetize_send( id, p_sys->packet );
1907 p_sys->packet = NULL;
1910 if( p_sys->packet == NULL )
1912 /* allocate a new packet */
1913 p_sys->packet = block_New( p_stream, id->i_mtu );
1914 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1915 p_sys->packet->i_dts = i_dts;
1916 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1917 i_dts += p_sys->packet->i_length;
1920 i_size = __MIN( i_data,
1921 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1923 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1926 p_sys->packet->i_buffer += i_size;
1935 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1938 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1944 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1946 p_next = p_buffer->p_next;
1947 block_Release( p_buffer );
1955 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1957 sout_access_out_t *p_grab;
1959 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1960 if( p_grab == NULL )
1963 p_grab->p_module = NULL;
1964 p_grab->psz_access = strdup( "grab" );
1965 p_grab->p_cfg = NULL;
1966 p_grab->psz_path = strdup( "" );
1967 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1968 p_grab->pf_seek = NULL;
1969 p_grab->pf_write = AccessOutGrabberWrite;
1970 vlc_object_attach( p_grab, p_stream );