1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004, 2010 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
10 * This program is free software; you can redistribute it and/or modify
11 * it under the terms of the GNU General Public License as published by
12 * the Free Software Foundation; either version 2 of the License, or
13 * (at your option) any later version.
15 * This program is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
18 * GNU General Public License for more details.
20 * You should have received a copy of the GNU General Public License
21 * along with this program; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
23 *****************************************************************************/
25 /*****************************************************************************
27 *****************************************************************************/
33 #include <vlc_common.h>
34 #include <vlc_plugin.h>
36 #include <vlc_block.h>
38 #include <vlc_httpd.h>
40 #include <vlc_network.h>
46 # include <vlc_gcrypt.h>
52 # include <sys/types.h>
55 #ifdef HAVE_ARPA_INET_H
56 # include <arpa/inet.h>
58 #ifdef HAVE_LINUX_DCCP_H
59 # include <linux/dccp.h>
62 # define IPPROTO_DCCP 33
64 #ifndef IPPROTO_UDPLITE
65 # define IPPROTO_UDPLITE 136
72 /*****************************************************************************
74 *****************************************************************************/
76 #define DEST_TEXT N_("Destination")
77 #define DEST_LONGTEXT N_( \
78 "This is the output URL that will be used." )
79 #define SDP_TEXT N_("SDP")
80 #define SDP_LONGTEXT N_( \
81 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
82 "session will be made available. You must use a url: http://location to " \
83 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
84 "for the SDP to be announced via SAP." )
85 #define SAP_TEXT N_("SAP announcing")
86 #define SAP_LONGTEXT N_("Announce this session with SAP.")
87 #define MUX_TEXT N_("Muxer")
88 #define MUX_LONGTEXT N_( \
89 "This allows you to specify the muxer used for the streaming output. " \
90 "Default is to use no muxer (standard RTP stream)." )
92 #define NAME_TEXT N_("Session name")
93 #define NAME_LONGTEXT N_( \
94 "This is the name of the session that will be announced in the SDP " \
95 "(Session Descriptor)." )
96 #define DESC_TEXT N_("Session description")
97 #define DESC_LONGTEXT N_( \
98 "This allows you to give a short description with details about the stream, " \
99 "that will be announced in the SDP (Session Descriptor)." )
100 #define URL_TEXT N_("Session URL")
101 #define URL_LONGTEXT N_( \
102 "This allows you to give a URL with more details about the stream " \
103 "(often the website of the streaming organization), that will " \
104 "be announced in the SDP (Session Descriptor)." )
105 #define EMAIL_TEXT N_("Session email")
106 #define EMAIL_LONGTEXT N_( \
107 "This allows you to give a contact mail address for the stream, that will " \
108 "be announced in the SDP (Session Descriptor)." )
109 #define PHONE_TEXT N_("Session phone number")
110 #define PHONE_LONGTEXT N_( \
111 "This allows you to give a contact telephone number for the stream, that will " \
112 "be announced in the SDP (Session Descriptor)." )
114 #define PORT_TEXT N_("Port")
115 #define PORT_LONGTEXT N_( \
116 "This allows you to specify the base port for the RTP streaming." )
117 #define PORT_AUDIO_TEXT N_("Audio port")
118 #define PORT_AUDIO_LONGTEXT N_( \
119 "This allows you to specify the default audio port for the RTP streaming." )
120 #define PORT_VIDEO_TEXT N_("Video port")
121 #define PORT_VIDEO_LONGTEXT N_( \
122 "This allows you to specify the default video port for the RTP streaming." )
124 #define TTL_TEXT N_("Hop limit (TTL)")
125 #define TTL_LONGTEXT N_( \
126 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
127 "the multicast packets sent by the stream output (-1 = use operating " \
128 "system built-in default).")
130 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
131 #define RTCP_MUX_LONGTEXT N_( \
132 "This sends and receives RTCP packet multiplexed over the same port " \
135 #define CACHING_TEXT N_("Caching value (ms)")
136 #define CACHING_LONGTEXT N_( \
137 "Default caching value for outbound RTP streams. This " \
138 "value should be set in milliseconds." )
140 #define PROTO_TEXT N_("Transport protocol")
141 #define PROTO_LONGTEXT N_( \
142 "This selects which transport protocol to use for RTP." )
144 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
145 #define SRTP_KEY_LONGTEXT N_( \
146 "RTP packets will be integrity-protected and ciphered "\
147 "with this Secure RTP master shared secret key.")
149 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
150 #define SRTP_SALT_LONGTEXT N_( \
151 "Secure RTP requires a (non-secret) master salt value.")
153 static const char *const ppsz_protos[] = {
154 "dccp", "sctp", "tcp", "udp", "udplite",
157 static const char *const ppsz_protocols[] = {
158 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
161 #define RFC3016_TEXT N_("MP4A LATM")
162 #define RFC3016_LONGTEXT N_( \
163 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
165 #define RTSP_HOST_TEXT N_( "RTSP host address" )
166 #define RTSP_HOST_LONGTEXT N_( \
167 "This defines the address, port and path the RTSP VOD server will listen " \
168 "on.\nSyntax is address:port/path. The default is to listen on all "\
169 "interfaces (address 0.0.0.0), on port 554, with no path.\nTo listen " \
170 "only on the local interface, use \"localhost\" as address." )
172 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
173 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
174 "not receiving any RTSP request for this long. Setting it to a " \
175 "negative value or zero disables timeouts. The default is 60 (one " \
178 static int Open ( vlc_object_t * );
179 static void Close( vlc_object_t * );
181 #define SOUT_CFG_PREFIX "sout-rtp-"
182 #define MAX_EMPTY_BLOCKS 200
185 set_shortname( N_("RTP"))
186 set_description( N_("RTP stream output") )
187 set_capability( "sout stream", 0 )
188 add_shortcut( "rtp", "vod" )
189 set_category( CAT_SOUT )
190 set_subcategory( SUBCAT_SOUT_STREAM )
192 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
193 DEST_LONGTEXT, true )
194 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
196 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
198 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
201 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
202 NAME_LONGTEXT, true )
203 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
204 DESC_LONGTEXT, true )
205 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
207 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
208 EMAIL_LONGTEXT, true )
209 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
210 PHONE_LONGTEXT, true )
212 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
213 PROTO_LONGTEXT, false )
214 change_string_list( ppsz_protos, ppsz_protocols, NULL )
215 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
216 PORT_LONGTEXT, true )
217 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
218 PORT_AUDIO_LONGTEXT, true )
219 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
220 PORT_VIDEO_LONGTEXT, true )
222 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
224 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
225 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
226 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
227 CACHING_TEXT, CACHING_LONGTEXT, true )
230 add_string( SOUT_CFG_PREFIX "key", "",
231 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
232 add_string( SOUT_CFG_PREFIX "salt", "",
233 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
236 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
237 RFC3016_LONGTEXT, false )
239 set_callbacks( Open, Close )
242 set_shortname( N_("RTSP VoD" ) )
243 set_description( N_("RTSP VoD server") )
244 set_category( CAT_SOUT )
245 set_subcategory( SUBCAT_SOUT_VOD )
246 set_capability( "vod server", 10 )
247 set_callbacks( OpenVoD, CloseVoD )
248 add_shortcut( "rtsp" )
249 add_string ( "rtsp-host", NULL, RTSP_HOST_TEXT,
250 RTSP_HOST_LONGTEXT, true )
251 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
252 RTSP_TIMEOUT_LONGTEXT, true )
256 /*****************************************************************************
257 * Exported prototypes
258 *****************************************************************************/
259 static const char *const ppsz_sout_options[] = {
260 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
261 "sap", "description", "url", "email", "phone",
262 "proto", "rtcp-mux", "caching", "key", "salt",
266 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
267 static int Del ( sout_stream_t *, sout_stream_id_t * );
268 static int Send( sout_stream_t *, sout_stream_id_t *,
270 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
271 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
272 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
275 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
276 static void* ThreadSend( void * );
277 static void *rtp_listen_thread( void * );
279 static void SDPHandleUrl( sout_stream_t *, const char * );
281 static int SapSetup( sout_stream_t *p_stream );
282 static int FileSetup( sout_stream_t *p_stream );
283 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
285 static int64_t rtp_init_ts( const vod_media_t *p_media,
286 const char *psz_vod_session );
288 struct sout_stream_sys_t
292 vlc_mutex_t lock_sdp;
299 session_descriptor_t *p_session;
302 httpd_host_t *p_httpd_host;
303 httpd_file_t *p_httpd_file;
308 /* RTSP NPT and timestamp computations */
309 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
310 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
311 int64_t i_pts_offset; /* matches actual PTS to prediction */
315 char *psz_destination;
317 uint16_t i_port_audio;
318 uint16_t i_port_video;
324 vod_media_t *p_vod_media;
325 char *psz_vod_session;
327 /* in case we do TS/PS over rtp */
329 sout_access_out_t *p_grab;
335 sout_stream_id_t **es;
338 typedef struct rtp_sink_t
344 struct sout_stream_id_t
346 sout_stream_t *p_stream;
351 uint32_t i_ts_offset;
355 uint16_t i_seq_sent_next;
358 rtp_format_t rtp_fmt;
361 /* Packetizer specific fields */
364 srtp_session_t *srtp;
369 vlc_mutex_t lock_sink;
372 rtsp_stream_id_t *rtsp_id;
378 block_fifo_t *p_fifo;
382 /*****************************************************************************
384 *****************************************************************************/
385 static int Open( vlc_object_t *p_this )
387 sout_stream_t *p_stream = (sout_stream_t*)p_this;
388 sout_instance_t *p_sout = p_stream->p_sout;
389 sout_stream_sys_t *p_sys = NULL;
390 config_chain_t *p_cfg = NULL;
394 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
395 ppsz_sout_options, p_stream->p_cfg );
397 p_sys = malloc( sizeof( sout_stream_sys_t ) );
401 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
403 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
404 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
405 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
406 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
408 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
410 msg_Err( p_stream, "audio and video RTP port must be distinct" );
411 free( p_sys->psz_destination );
416 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
418 if( !strcmp( p_cfg->psz_name, "sdp" )
419 && ( p_cfg->psz_value != NULL )
420 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
428 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
431 if( !strncasecmp( psz, "rtsp:", 5 ) )
437 /* Transport protocol */
438 p_sys->proto = IPPROTO_UDP;
439 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
441 if ((psz == NULL) || !strcasecmp (psz, "udp"))
442 (void)0; /* default */
444 if (!strcasecmp (psz, "dccp"))
446 p_sys->proto = IPPROTO_DCCP;
447 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
451 if (!strcasecmp (psz, "sctp"))
453 p_sys->proto = IPPROTO_TCP;
454 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
459 if (!strcasecmp (psz, "tcp"))
461 p_sys->proto = IPPROTO_TCP;
462 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
466 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
467 p_sys->proto = IPPROTO_UDPLITE;
469 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
472 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
474 p_sys->p_vod_media = NULL;
475 p_sys->psz_vod_session = NULL;
477 if (! strcmp(p_stream->psz_name, "vod"))
479 /* The VLM stops all instances before deleting a media, so this
480 * reference will remain valid during the lifetime of the rtp
482 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
484 if (p_sys->p_vod_media != NULL)
486 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
487 if (p_sys->psz_vod_session == NULL)
489 msg_Err(p_stream, "missing VoD session");
494 const char *mux = vod_get_mux(p_sys->p_vod_media);
495 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
499 if( p_sys->psz_destination == NULL && !b_rtsp
500 && p_sys->p_vod_media == NULL )
502 msg_Err( p_stream, "missing destination and not in RTSP mode" );
507 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
510 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
511 var_SetInteger( p_stream, "ttl", i_ttl );
514 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
516 /* NPT=0 time will be determined when we packetize the first packet
517 * (of any ES). But we want to be able to report rtptime in RTSP
518 * without waiting (and already did in the VoD case). So until then,
519 * we use an arbitrary reference PTS for timestamp computations, and
520 * then actual PTS will catch up using offsets. */
521 p_sys->i_npt_zero = VLC_TS_INVALID;
522 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
523 p_sys->psz_vod_session);
527 p_sys->psz_sdp = NULL;
529 p_sys->b_export_sap = false;
530 p_sys->p_session = NULL;
531 p_sys->psz_sdp_file = NULL;
533 p_sys->p_httpd_host = NULL;
534 p_sys->p_httpd_file = NULL;
536 p_stream->p_sys = p_sys;
538 vlc_mutex_init( &p_sys->lock_sdp );
539 vlc_mutex_init( &p_sys->lock_ts );
540 vlc_mutex_init( &p_sys->lock_es );
542 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
545 /* Check muxer type */
546 if( strncasecmp( psz, "ps", 2 )
547 && strncasecmp( psz, "mpeg1", 5 )
548 && strncasecmp( psz, "ts", 2 ) )
550 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
552 vlc_mutex_destroy( &p_sys->lock_sdp );
553 vlc_mutex_destroy( &p_sys->lock_ts );
554 vlc_mutex_destroy( &p_sys->lock_es );
555 free( p_sys->psz_vod_session );
556 free( p_sys->psz_destination );
561 p_sys->p_grab = GrabberCreate( p_stream );
562 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
565 if( p_sys->p_mux == NULL )
567 msg_Err( p_stream, "cannot create muxer" );
568 sout_AccessOutDelete( p_sys->p_grab );
569 vlc_mutex_destroy( &p_sys->lock_sdp );
570 vlc_mutex_destroy( &p_sys->lock_ts );
571 vlc_mutex_destroy( &p_sys->lock_es );
572 free( p_sys->psz_vod_session );
573 free( p_sys->psz_destination );
578 p_sys->packet = NULL;
580 p_stream->pf_add = MuxAdd;
581 p_stream->pf_del = MuxDel;
582 p_stream->pf_send = MuxSend;
587 p_sys->p_grab = NULL;
589 p_stream->pf_add = Add;
590 p_stream->pf_del = Del;
591 p_stream->pf_send = Send;
594 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
595 SDPHandleUrl( p_stream, "sap" );
597 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
600 config_chain_t *p_cfg;
602 SDPHandleUrl( p_stream, psz );
604 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
606 if( !strcmp( p_cfg->psz_name, "sdp" ) )
608 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
611 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
612 if( !strcmp( p_cfg->psz_value, psz ) )
615 SDPHandleUrl( p_stream, p_cfg->psz_value );
621 /* update p_sout->i_out_pace_nocontrol */
622 p_stream->p_sout->i_out_pace_nocontrol++;
624 if( p_sys->p_mux != NULL )
626 sout_stream_id_t *id = Add( p_stream, NULL );
637 /*****************************************************************************
639 *****************************************************************************/
640 static void Close( vlc_object_t * p_this )
642 sout_stream_t *p_stream = (sout_stream_t*)p_this;
643 sout_stream_sys_t *p_sys = p_stream->p_sys;
645 /* update p_sout->i_out_pace_nocontrol */
646 p_stream->p_sout->i_out_pace_nocontrol--;
650 assert( p_sys->i_es <= 1 );
652 sout_MuxDelete( p_sys->p_mux );
653 if ( p_sys->i_es > 0 )
654 Del( p_stream, p_sys->es[0] );
655 sout_AccessOutDelete( p_sys->p_grab );
659 block_Release( p_sys->packet );
663 if( p_sys->rtsp != NULL )
664 RtspUnsetup( p_sys->rtsp );
666 vlc_mutex_destroy( &p_sys->lock_sdp );
667 vlc_mutex_destroy( &p_sys->lock_ts );
668 vlc_mutex_destroy( &p_sys->lock_es );
670 if( p_sys->p_httpd_file )
671 httpd_FileDelete( p_sys->p_httpd_file );
673 if( p_sys->p_httpd_host )
674 httpd_HostDelete( p_sys->p_httpd_host );
676 free( p_sys->psz_sdp );
678 if( p_sys->psz_sdp_file != NULL )
681 unlink( p_sys->psz_sdp_file );
683 free( p_sys->psz_sdp_file );
685 free( p_sys->psz_vod_session );
686 free( p_sys->psz_destination );
690 /*****************************************************************************
692 *****************************************************************************/
693 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
695 sout_stream_sys_t *p_sys = p_stream->p_sys;
698 vlc_UrlParse( &url, psz_url, 0 );
699 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
701 if( p_sys->p_httpd_file )
703 msg_Err( p_stream, "you can use sdp=http:// only once" );
707 if( HttpSetup( p_stream, &url ) )
709 msg_Err( p_stream, "cannot export SDP as HTTP" );
712 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
714 if( p_sys->rtsp != NULL )
716 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
720 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, &url );
721 if( p_sys->rtsp == NULL )
722 msg_Err( p_stream, "cannot export SDP as RTSP" );
724 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
725 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
727 p_sys->b_export_sap = true;
728 SapSetup( p_stream );
730 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
732 if( p_sys->psz_sdp_file != NULL )
734 msg_Err( p_stream, "you can use sdp=file:// only once" );
737 p_sys->psz_sdp_file = make_path( psz_url );
738 if( p_sys->psz_sdp_file == NULL )
740 FileSetup( p_stream );
744 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
749 vlc_UrlClean( &url );
752 /*****************************************************************************
754 *****************************************************************************/
756 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
758 sout_stream_sys_t *p_sys = p_stream->p_sys;
759 char *psz_sdp = NULL;
760 struct sockaddr_storage dst;
764 * When we have a fixed destination (typically when we do multicast),
765 * we need to put the actual port numbers in the SDP.
766 * When there is no fixed destination, we only support RTSP unicast
767 * on-demand setup, so we should rather let the clients decide which ports
769 * When there is both a fixed destination and RTSP unicast, we need to
770 * put port numbers used by the fixed destination, otherwise the SDP would
771 * become totally incorrect for multicast use. It should be noted that
772 * port numbers from SDP with RTSP are only "recommendation" from the
773 * server to the clients (per RFC2326), so only broken clients will fail
774 * to handle this properly. There is no solution but to use two differents
775 * output chain with two different RTSP URLs if you need to handle this
780 vlc_mutex_lock( &p_sys->lock_es );
781 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
782 goto out; /* hmm... */
784 if( p_sys->psz_destination != NULL )
788 /* Oh boy, this is really ugly! */
789 dstlen = sizeof( dst );
790 if( p_sys->es[0]->listen.fd != NULL )
791 getsockname( p_sys->es[0]->listen.fd[0],
792 (struct sockaddr *)&dst, &dstlen );
794 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
795 (struct sockaddr *)&dst, &dstlen );
801 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
802 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
803 && rtsp_url[7] == '[';
805 /* Dummy destination address for RTSP */
806 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
807 : sizeof( struct sockaddr_in );
808 memset (&dst, 0, dstlen);
809 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
815 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
816 NULL, 0, (struct sockaddr *)&dst, dstlen );
817 if( psz_sdp == NULL )
820 /* TODO: a=source-filter */
821 if( p_sys->rtcp_mux )
822 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
824 if( rtsp_url != NULL )
825 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
827 const char *proto = "RTP/AVP"; /* protocol */
828 if( rtsp_url == NULL )
830 switch( p_sys->proto )
835 proto = "TCP/RTP/AVP";
838 proto = "DCCP/RTP/AVP";
840 case IPPROTO_UDPLITE:
845 for( i = 0; i < p_sys->i_es; i++ )
847 sout_stream_id_t *id = p_sys->es[i];
848 rtp_format_t *rtp_fmt = &id->rtp_fmt;
849 const char *mime_major; /* major MIME type */
851 switch( rtp_fmt->cat )
854 mime_major = "video";
857 mime_major = "audio";
866 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
867 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
868 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
871 /* cf RFC4566 §5.14 */
872 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
873 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
875 if( rtsp_url != NULL )
877 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
878 if( track_url != NULL )
880 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
886 if( id->listen.fd != NULL )
887 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
888 if( p_sys->proto == IPPROTO_DCCP )
889 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
891 toupper( (unsigned char)mime_major[0] ) );
895 vlc_mutex_unlock( &p_sys->lock_es );
899 /*****************************************************************************
901 *****************************************************************************/
904 * Shrink the MTU down to a fixed packetization time (for audio).
907 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
909 /* Samples per second */
910 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
911 bytes *= id->rtp_fmt.channels;
914 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
915 id->i_mtu = 12 + spl;
916 else /* MTU is too small for ptime, align to a sample boundary */
917 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
920 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
922 /* NOTE: this plays nice with offsets because the calculations are
924 return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
927 /** Add an ES as a new RTP stream */
928 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
930 /* NOTE: As a special case, if we use a non-RTP
931 * mux (TS/PS), then p_fmt is NULL. */
932 sout_stream_sys_t *p_sys = p_stream->p_sys;
935 sout_stream_id_t *id = malloc( sizeof( *id ) );
936 if( unlikely(id == NULL) )
938 id->p_stream = p_stream;
940 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
941 if( id->i_mtu <= 12 + 16 )
942 id->i_mtu = 576 - 20 - 8; /* pessimistic */
943 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
948 vlc_mutex_init( &id->lock_sink );
953 id->listen.fd = NULL;
955 id->b_first_packet = true;
957 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
959 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
960 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
964 if (p_sys->p_vod_media != NULL)
966 id->rtp_fmt.ptname = NULL;
968 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
969 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
970 &ssrc, &id->i_seq_sent_next);
971 if (val == VLC_SUCCESS)
973 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
974 /* This is ugly, but id->i_seq_sent_next needs to be
975 * initialized inside vod_init_id() to avoid race
977 id->i_sequence = id->i_seq_sent_next;
979 /* vod_init_id() may fail either because the ES wasn't found in
980 * the VoD media, or because the RTSP session is gone. In the
981 * former case, id->rtp_fmt was left untouched. */
982 format = (id->rtp_fmt.ptname != NULL);
987 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
988 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
989 if (p_fmt == NULL && psz == NULL)
991 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
993 if (val != VLC_SUCCESS)
998 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1002 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1003 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1004 if (id->srtp == NULL)
1010 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1011 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1016 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1019 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1023 id->i_seq_sent_next = id->i_sequence;
1026 if( p_sys->psz_destination != NULL )
1028 /* Choose the port */
1029 uint16_t i_port = 0;
1033 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1034 i_port = p_sys->i_port_audio;
1036 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1037 i_port = p_sys->i_port_video;
1039 /* We do not need the ES lock (p_sys->lock_es) here, because
1040 * this is the only one thread that can *modify* the ES table.
1041 * The ES lock protects the other threads from our modifications
1042 * (TAB_APPEND, TAB_REMOVE). */
1043 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1044 if (i_port == p_sys->es[i]->i_port)
1045 i_port = 0; /* Port already in use! */
1046 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1050 msg_Err (p_stream, "too many RTP elementary streams");
1054 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1055 if (p == p_sys->es[i]->i_port)
1059 id->i_port = i_port;
1061 int type = SOCK_STREAM;
1063 switch( p_sys->proto )
1069 switch (id->rtp_fmt.cat)
1071 case VIDEO_ES: code = "RTPV"; break;
1072 case AUDIO_ES: code = "RTPARTPV"; break;
1073 case SPU_ES: code = "RTPTRTPV"; break;
1074 default: code = "RTPORTPV"; break;
1076 var_SetString (p_stream, "dccp-service", code);
1078 } /* fall through */
1081 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1082 p_sys->psz_destination, i_port,
1083 type, p_sys->proto );
1084 if( id->listen.fd == NULL )
1086 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1089 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1090 VLC_THREAD_PRIORITY_LOW ) )
1092 net_ListenClose( id->listen.fd );
1093 id->listen.fd = NULL;
1100 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1101 i_port, -1, p_sys->proto );
1104 msg_Err( p_stream, "cannot create RTP socket" );
1107 /* Ignore any unexpected incoming packet (including RTCP-RR
1108 * packets in case of rtcp-mux) */
1109 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1111 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1112 /* FIXME: test if this is multicast */
1119 switch( p_fmt->i_codec )
1121 case VLC_CODEC_MULAW:
1122 case VLC_CODEC_ALAW:
1124 rtp_set_ptime (id, 20, 1);
1126 case VLC_CODEC_S16B:
1127 case VLC_CODEC_S16L:
1128 rtp_set_ptime (id, 20, 2);
1134 #if 0 /* No payload formats sets this at the moment */
1137 cscov += 8 /* UDP */ + 12 /* RTP */;
1139 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1142 vlc_mutex_lock( &p_sys->lock_ts );
1143 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1144 vlc_mutex_unlock( &p_sys->lock_ts );
1146 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1147 p_sys->i_pts_offset );
1149 if( p_sys->rtsp != NULL )
1150 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1151 id->rtp_fmt.clock_rate, mcast_fd );
1153 id->p_fifo = block_FifoNew();
1154 if( unlikely(id->p_fifo == NULL) )
1156 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1158 block_FifoRelease( id->p_fifo );
1163 /* Update p_sys context */
1164 vlc_mutex_lock( &p_sys->lock_es );
1165 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1166 vlc_mutex_unlock( &p_sys->lock_es );
1168 psz_sdp = SDPGenerate( p_stream, NULL );
1170 vlc_mutex_lock( &p_sys->lock_sdp );
1171 free( p_sys->psz_sdp );
1172 p_sys->psz_sdp = psz_sdp;
1173 vlc_mutex_unlock( &p_sys->lock_sdp );
1175 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1177 /* Update SDP (sap/file) */
1178 if( p_sys->b_export_sap ) SapSetup( p_stream );
1179 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1184 Del( p_stream, id );
1188 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1190 sout_stream_sys_t *p_sys = p_stream->p_sys;
1192 vlc_mutex_lock( &p_sys->lock_es );
1193 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1194 vlc_mutex_unlock( &p_sys->lock_es );
1196 if( likely(id->p_fifo != NULL) )
1198 vlc_cancel( id->thread );
1199 vlc_join( id->thread, NULL );
1200 block_FifoRelease( id->p_fifo );
1203 free( id->rtp_fmt.fmtp );
1205 if (p_sys->p_vod_media != NULL)
1206 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1208 RtspDelId( p_sys->rtsp, id->rtsp_id );
1209 if( id->listen.fd != NULL )
1211 vlc_cancel( id->listen.thread );
1212 vlc_join( id->listen.thread, NULL );
1213 net_ListenClose( id->listen.fd );
1215 /* Delete remaining sinks (incoming connections or explicit
1217 while( id->sinkc > 0 )
1218 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1220 if( id->srtp != NULL )
1221 srtp_destroy( id->srtp );
1224 vlc_mutex_destroy( &id->lock_sink );
1226 /* Update SDP (sap/file) */
1227 if( p_sys->b_export_sap ) SapSetup( p_stream );
1228 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1234 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1239 assert( p_stream->p_sys->p_mux == NULL );
1242 while( p_buffer != NULL )
1244 p_next = p_buffer->p_next;
1246 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1247 * as the first packet of the stream */
1248 if (id->b_first_packet)
1250 id->b_first_packet = false;
1251 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1252 !strcmp(id->rtp_fmt.ptname, "theora"))
1253 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1257 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1260 block_Release( p_buffer );
1266 /****************************************************************************
1268 ****************************************************************************/
1269 static int SapSetup( sout_stream_t *p_stream )
1271 sout_stream_sys_t *p_sys = p_stream->p_sys;
1272 sout_instance_t *p_sout = p_stream->p_sout;
1274 /* Remove the previous session */
1275 if( p_sys->p_session != NULL)
1277 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1278 p_sys->p_session = NULL;
1281 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1282 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1284 p_sys->psz_destination );
1289 /****************************************************************************
1291 ****************************************************************************/
1292 static int FileSetup( sout_stream_t *p_stream )
1294 sout_stream_sys_t *p_sys = p_stream->p_sys;
1297 if( p_sys->psz_sdp == NULL )
1298 return VLC_EGENERIC; /* too early */
1300 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1302 msg_Err( p_stream, "cannot open file '%s' (%m)",
1303 p_sys->psz_sdp_file );
1304 return VLC_EGENERIC;
1307 fputs( p_sys->psz_sdp, f );
1313 /****************************************************************************
1315 ****************************************************************************/
1316 static int HttpCallback( httpd_file_sys_t *p_args,
1317 httpd_file_t *, uint8_t *p_request,
1318 uint8_t **pp_data, int *pi_data );
1320 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1322 sout_stream_sys_t *p_sys = p_stream->p_sys;
1324 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1325 if( p_sys->p_httpd_host )
1327 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1328 url->psz_path ? url->psz_path : "/",
1331 HttpCallback, (void*)p_sys );
1333 if( p_sys->p_httpd_file == NULL )
1335 return VLC_EGENERIC;
1340 static int HttpCallback( httpd_file_sys_t *p_args,
1341 httpd_file_t *f, uint8_t *p_request,
1342 uint8_t **pp_data, int *pi_data )
1344 VLC_UNUSED(f); VLC_UNUSED(p_request);
1345 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1347 vlc_mutex_lock( &p_sys->lock_sdp );
1348 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1350 *pi_data = strlen( p_sys->psz_sdp );
1351 *pp_data = malloc( *pi_data );
1352 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1359 vlc_mutex_unlock( &p_sys->lock_sdp );
1364 /****************************************************************************
1366 ****************************************************************************/
1367 static void* ThreadSend( void *data )
1370 # define ECONNREFUSED WSAECONNREFUSED
1371 # define ENOPROTOOPT WSAENOPROTOOPT
1372 # define EHOSTUNREACH WSAEHOSTUNREACH
1373 # define ENETUNREACH WSAENETUNREACH
1374 # define ENETDOWN WSAENETDOWN
1375 # define ENOBUFS WSAENOBUFS
1376 # define EAGAIN WSAEWOULDBLOCK
1377 # define EWOULDBLOCK WSAEWOULDBLOCK
1379 sout_stream_id_t *id = data;
1380 unsigned i_caching = id->i_caching;
1384 block_t *out = block_FifoGet( id->p_fifo );
1385 block_cleanup_push (out);
1389 { /* FIXME: this is awfully inefficient */
1390 size_t len = out->i_buffer;
1391 out = block_Realloc( out, 0, len + 10 );
1392 out->i_buffer = len;
1394 int canc = vlc_savecancel ();
1395 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1396 vlc_restorecancel (canc);
1400 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1401 block_Release( out );
1405 out->i_buffer = len;
1408 mwait (out->i_dts + i_caching);
1413 mwait (out->i_dts + i_caching);
1417 ssize_t len = out->i_buffer;
1418 int canc = vlc_savecancel ();
1420 vlc_mutex_lock( &id->lock_sink );
1421 unsigned deadc = 0; /* How many dead sockets? */
1422 int deadv[id->sinkc]; /* Dead sockets list */
1424 for( int i = 0; i < id->sinkc; i++ )
1427 if( !id->srtp ) /* FIXME: SRTCP support */
1429 SendRTCP( id->sinkv[i].rtcp, out );
1431 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1432 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1433 && net_errno != ENOBUFS && net_errno != ENOMEM )
1436 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1437 &type, &(socklen_t){ sizeof(type) });
1438 if( type == SOCK_DGRAM )
1439 /* ICMP soft error: ignore and retry */
1440 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1442 /* Broken connection */
1443 deadv[deadc++] = id->sinkv[i].rtp_fd;
1446 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1447 vlc_mutex_unlock( &id->lock_sink );
1448 block_Release( out );
1450 for( unsigned i = 0; i < deadc; i++ )
1452 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1453 rtp_del_sink( id, deadv[i] );
1455 vlc_restorecancel (canc);
1461 /* This thread dequeues incoming connections (DCCP streaming) */
1462 static void *rtp_listen_thread( void *data )
1464 sout_stream_id_t *id = data;
1466 assert( id->listen.fd != NULL );
1470 int fd = net_Accept( id->p_stream, id->listen.fd );
1473 int canc = vlc_savecancel( );
1474 rtp_add_sink( id, fd, true, NULL );
1475 vlc_restorecancel( canc );
1482 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1484 rtp_sink_t sink = { fd, NULL };
1485 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1487 if( sink.rtcp == NULL )
1488 msg_Err( id->p_stream, "RTCP failed!" );
1490 vlc_mutex_lock( &id->lock_sink );
1491 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1493 *seq = id->i_seq_sent_next;
1494 vlc_mutex_unlock( &id->lock_sink );
1498 void rtp_del_sink( sout_stream_id_t *id, int fd )
1500 rtp_sink_t sink = { fd, NULL };
1502 /* NOTE: must be safe to use if fd is not included */
1503 vlc_mutex_lock( &id->lock_sink );
1504 for( int i = 0; i < id->sinkc; i++ )
1506 if (id->sinkv[i].rtp_fd == fd)
1508 sink = id->sinkv[i];
1509 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1513 vlc_mutex_unlock( &id->lock_sink );
1515 CloseRTCP( sink.rtcp );
1516 net_Close( sink.rtp_fd );
1519 uint16_t rtp_get_seq( sout_stream_id_t *id )
1521 /* This will return values for the next packet. */
1524 vlc_mutex_lock( &id->lock_sink );
1525 seq = id->i_seq_sent_next;
1526 vlc_mutex_unlock( &id->lock_sink );
1531 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1532 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1533 * random (although we use the same reference for all the ES as a
1534 * feature). In the VoD case, this function is called independently
1535 * from several parts of the code, so we need to always return the same
1537 static int64_t rtp_init_ts( const vod_media_t *p_media,
1538 const char *psz_vod_session )
1540 if (p_media == NULL || psz_vod_session == NULL)
1544 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1545 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1546 i_ts_init ^= (uintptr_t)p_media;
1547 /* Limit the timestamp to 48 bytes, this is enough and allows us
1548 * to stay away from overflows */
1549 i_ts_init &= 0xFFFFFFFFFFFF;
1553 /* Return a timestamp corresponding to packets being sent now, and that
1554 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1555 * Also return the NPT corresponding to this timestamp. If the stream
1556 * output is not started, the initial timestamp that will be used with
1557 * the first packets for NPT=0 is returned instead. */
1558 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1559 const vod_media_t *p_media, const char *psz_vod_session,
1566 p_stream = id->p_stream;
1568 if (p_stream == NULL)
1569 return rtp_init_ts(p_media, psz_vod_session);
1571 sout_stream_sys_t *p_sys = p_stream->p_sys;
1573 vlc_mutex_lock( &p_sys->lock_ts );
1574 i_npt_zero = p_sys->i_npt_zero;
1575 vlc_mutex_unlock( &p_sys->lock_ts );
1577 if( i_npt_zero == VLC_TS_INVALID )
1578 return p_sys->i_pts_zero;
1580 mtime_t now = mdate();
1581 if( now < i_npt_zero )
1582 return p_sys->i_pts_zero;
1584 int64_t npt = now - i_npt_zero;
1588 return p_sys->i_pts_zero + npt;
1591 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1592 int b_marker, int64_t i_pts )
1594 if( !id->b_ts_init )
1596 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1597 vlc_mutex_lock( &p_sys->lock_ts );
1598 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1600 /* This is the first packet of any ES. We initialize the
1601 * NPT=0 time reference, and the offset to match the
1602 * arbitrary PTS reference. */
1603 p_sys->i_npt_zero = i_pts + id->i_caching;
1604 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1606 vlc_mutex_unlock( &p_sys->lock_ts );
1608 /* And in any case this is the first packet of this ES, so we
1609 * initialize the offset for this ES. */
1610 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1611 p_sys->i_pts_offset );
1612 id->b_ts_init = true;
1615 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1618 out->p_buffer[0] = 0x80;
1619 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1620 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1621 out->p_buffer[3] = ( id->i_sequence )&0xff;
1622 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1623 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1624 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1625 out->p_buffer[7] = ( i_timestamp )&0xff;
1627 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1633 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1635 block_FifoPut( id->p_fifo, out );
1639 * @return configured max RTP payload size (including payload type-specific
1640 * headers, excluding RTP and transport headers)
1642 size_t rtp_mtu (const sout_stream_id_t *id)
1644 return id->i_mtu - 12;
1647 /*****************************************************************************
1649 *****************************************************************************/
1651 /** Add an ES to a non-RTP muxed stream */
1652 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1654 sout_input_t *p_input;
1655 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1656 assert( p_mux != NULL );
1658 p_input = sout_MuxAddStream( p_mux, p_fmt );
1659 if( p_input == NULL )
1661 msg_Err( p_stream, "cannot add this stream to the muxer" );
1665 return (sout_stream_id_t *)p_input;
1669 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1672 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1673 assert( p_mux != NULL );
1675 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1680 /** Remove an ES from a non-RTP muxed stream */
1681 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1683 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1684 assert( p_mux != NULL );
1686 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1691 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1692 const block_t *p_buffer )
1694 sout_stream_sys_t *p_sys = p_stream->p_sys;
1695 sout_stream_id_t *id = p_sys->es[0];
1697 int64_t i_dts = p_buffer->i_dts;
1699 uint8_t *p_data = p_buffer->p_buffer;
1700 size_t i_data = p_buffer->i_buffer;
1701 size_t i_max = id->i_mtu - 12;
1703 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1709 /* output complete packet */
1710 if( p_sys->packet &&
1711 p_sys->packet->i_buffer + i_data > i_max )
1713 rtp_packetize_send( id, p_sys->packet );
1714 p_sys->packet = NULL;
1717 if( p_sys->packet == NULL )
1719 /* allocate a new packet */
1720 p_sys->packet = block_New( p_stream, id->i_mtu );
1721 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1722 p_sys->packet->i_dts = i_dts;
1723 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1724 i_dts += p_sys->packet->i_length;
1727 i_size = __MIN( i_data,
1728 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1730 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1733 p_sys->packet->i_buffer += i_size;
1742 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1745 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1751 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1753 p_next = p_buffer->p_next;
1754 block_Release( p_buffer );
1762 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1764 sout_access_out_t *p_grab;
1766 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1767 if( p_grab == NULL )
1770 p_grab->p_module = NULL;
1771 p_grab->psz_access = strdup( "grab" );
1772 p_grab->p_cfg = NULL;
1773 p_grab->psz_path = strdup( "" );
1774 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1775 p_grab->pf_seek = NULL;
1776 p_grab->pf_write = AccessOutGrabberWrite;