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>
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 #define RTSP_HOST_TEXT N_( "RTSP host address" )
164 #define RTSP_HOST_LONGTEXT N_( \
165 "This defines the address, port and path the RTSP VOD server will listen " \
166 "on.\nSyntax is address:port/path. The default is to listen on all "\
167 "interfaces (address 0.0.0.0), on port 554, with no path.\nTo listen " \
168 "only on the local interface, use \"localhost\" as address." )
170 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
171 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
172 "not receiving any RTSP request for this long. Setting it to a " \
173 "negative value or zero disables timeouts. The default is 60 (one " \
176 static int Open ( vlc_object_t * );
177 static void Close( vlc_object_t * );
179 #define SOUT_CFG_PREFIX "sout-rtp-"
180 #define MAX_EMPTY_BLOCKS 200
183 set_shortname( N_("RTP"))
184 set_description( N_("RTP stream output") )
185 set_capability( "sout stream", 0 )
186 add_shortcut( "rtp", "vod" )
187 set_category( CAT_SOUT )
188 set_subcategory( SUBCAT_SOUT_STREAM )
190 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
191 DEST_LONGTEXT, true )
192 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
194 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
196 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
199 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
200 NAME_LONGTEXT, true )
201 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
202 DESC_LONGTEXT, true )
203 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
205 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
206 EMAIL_LONGTEXT, true )
207 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
208 PHONE_LONGTEXT, true )
210 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
211 PROTO_LONGTEXT, false )
212 change_string_list( ppsz_protos, ppsz_protocols, NULL )
213 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
214 PORT_LONGTEXT, true )
215 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
216 PORT_AUDIO_LONGTEXT, true )
217 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
218 PORT_VIDEO_LONGTEXT, true )
220 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
222 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
223 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
224 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
225 CACHING_TEXT, CACHING_LONGTEXT, true )
228 add_string( SOUT_CFG_PREFIX "key", "",
229 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
230 add_string( SOUT_CFG_PREFIX "salt", "",
231 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
234 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
235 RFC3016_LONGTEXT, false )
237 set_callbacks( Open, Close )
240 set_shortname( N_("RTSP VoD" ) )
241 set_description( N_("RTSP VoD server") )
242 set_category( CAT_SOUT )
243 set_subcategory( SUBCAT_SOUT_VOD )
244 set_capability( "vod server", 0 )
245 set_callbacks( OpenVoD, CloseVoD )
246 add_shortcut( "rtsp" )
247 add_string ( "rtsp-host", NULL, RTSP_HOST_TEXT,
248 RTSP_HOST_LONGTEXT, true )
249 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
250 RTSP_TIMEOUT_LONGTEXT, true )
254 /*****************************************************************************
255 * Exported prototypes
256 *****************************************************************************/
257 static const char *const ppsz_sout_options[] = {
258 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
259 "sap", "description", "url", "email", "phone",
260 "proto", "rtcp-mux", "caching", "key", "salt",
264 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
265 static int Del ( sout_stream_t *, sout_stream_id_t * );
266 static int Send( sout_stream_t *, sout_stream_id_t *,
268 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
269 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
270 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
273 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
274 static void* ThreadSend( void * );
275 static void *rtp_listen_thread( void * );
277 static void SDPHandleUrl( sout_stream_t *, const char * );
279 static int SapSetup( sout_stream_t *p_stream );
280 static int FileSetup( sout_stream_t *p_stream );
281 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
283 static int64_t rtp_init_ts( const vod_media_t *p_media,
284 const char *psz_vod_session );
286 struct sout_stream_sys_t
290 vlc_mutex_t lock_sdp;
297 session_descriptor_t *p_session;
300 httpd_host_t *p_httpd_host;
301 httpd_file_t *p_httpd_file;
306 /* RTSP NPT and timestamp computations */
307 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
308 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
309 int64_t i_pts_offset; /* matches actual PTS to prediction */
313 char *psz_destination;
315 uint16_t i_port_audio;
316 uint16_t i_port_video;
323 vod_media_t *p_vod_media;
324 char *psz_vod_session;
326 /* in case we do TS/PS over rtp */
328 sout_access_out_t *p_grab;
334 sout_stream_id_t **es;
337 typedef struct rtp_sink_t
343 struct sout_stream_id_t
345 sout_stream_t *p_stream;
349 uint32_t i_ts_offset;
353 uint16_t i_seq_sent_next;
356 rtp_format_t rtp_fmt;
359 /* Packetizer specific fields */
362 srtp_session_t *srtp;
367 vlc_mutex_t lock_sink;
370 rtsp_stream_id_t *rtsp_id;
376 block_fifo_t *p_fifo;
380 /*****************************************************************************
382 *****************************************************************************/
383 static int Open( vlc_object_t *p_this )
385 sout_stream_t *p_stream = (sout_stream_t*)p_this;
386 sout_instance_t *p_sout = p_stream->p_sout;
387 sout_stream_sys_t *p_sys = NULL;
388 config_chain_t *p_cfg = NULL;
392 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
393 ppsz_sout_options, p_stream->p_cfg );
395 p_sys = malloc( sizeof( sout_stream_sys_t ) );
399 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
401 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
402 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
403 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
404 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
406 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
408 msg_Err( p_stream, "audio and video RTP port must be distinct" );
409 free( p_sys->psz_destination );
414 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
416 if( !strcmp( p_cfg->psz_name, "sdp" )
417 && ( p_cfg->psz_value != NULL )
418 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
426 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
429 if( !strncasecmp( psz, "rtsp:", 5 ) )
435 /* Transport protocol */
436 p_sys->proto = IPPROTO_UDP;
437 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
439 if ((psz == NULL) || !strcasecmp (psz, "udp"))
440 (void)0; /* default */
442 if (!strcasecmp (psz, "dccp"))
444 p_sys->proto = IPPROTO_DCCP;
445 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
449 if (!strcasecmp (psz, "sctp"))
451 p_sys->proto = IPPROTO_TCP;
452 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
457 if (!strcasecmp (psz, "tcp"))
459 p_sys->proto = IPPROTO_TCP;
460 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
464 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
465 p_sys->proto = IPPROTO_UDPLITE;
467 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
470 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
472 p_sys->p_vod_media = NULL;
473 p_sys->psz_vod_session = NULL;
475 if (! strcmp(p_stream->psz_name, "vod"))
477 /* The VLM stops all instances before deleting a media, so this
478 * reference will remain valid during the lifetime of the rtp
480 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
482 if (p_sys->p_vod_media != NULL)
484 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
485 if (p_sys->psz_vod_session == NULL)
487 msg_Err(p_stream, "missing VoD session");
492 const char *mux = vod_get_mux(p_sys->p_vod_media);
493 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
497 if( p_sys->psz_destination == NULL && !b_rtsp
498 && p_sys->p_vod_media == NULL )
500 msg_Err( p_stream, "missing destination and not in RTSP mode" );
505 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
506 if( p_sys->i_ttl == -1 )
508 /* Normally, we should let the default hop limit up to the core,
509 * but we have to know it to write our RTSP headers properly,
510 * which is why we ask the core. FIXME: broken when neither
511 * sout-rtp-ttl nor ttl are set. */
512 p_sys->i_ttl = var_InheritInteger( p_stream, "ttl" );
515 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
517 /* NPT=0 time will be determined when we packetize the first packet
518 * (of any ES). But we want to be able to report rtptime in RTSP
519 * without waiting (and already did in the VoD case). So until then,
520 * we use an arbitrary reference PTS for timestamp computations, and
521 * then actual PTS will catch up using offsets. */
522 p_sys->i_npt_zero = VLC_TS_INVALID;
523 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
524 p_sys->psz_vod_session);
528 p_sys->psz_sdp = NULL;
530 p_sys->b_export_sap = false;
531 p_sys->p_session = NULL;
532 p_sys->psz_sdp_file = NULL;
534 p_sys->p_httpd_host = NULL;
535 p_sys->p_httpd_file = NULL;
537 p_stream->p_sys = p_sys;
539 vlc_mutex_init( &p_sys->lock_sdp );
540 vlc_mutex_init( &p_sys->lock_ts );
541 vlc_mutex_init( &p_sys->lock_es );
543 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
546 sout_stream_id_t *id;
548 /* Check muxer type */
549 if( strncasecmp( psz, "ps", 2 )
550 && strncasecmp( psz, "mpeg1", 5 )
551 && strncasecmp( psz, "ts", 2 ) )
553 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
555 vlc_mutex_destroy( &p_sys->lock_sdp );
556 vlc_mutex_destroy( &p_sys->lock_es );
557 free( p_sys->psz_vod_session );
558 free( p_sys->psz_destination );
563 p_sys->p_grab = GrabberCreate( p_stream );
564 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
567 if( p_sys->p_mux == NULL )
569 msg_Err( p_stream, "cannot create muxer" );
570 sout_AccessOutDelete( p_sys->p_grab );
571 vlc_mutex_destroy( &p_sys->lock_sdp );
572 vlc_mutex_destroy( &p_sys->lock_es );
573 free( p_sys->psz_vod_session );
574 free( p_sys->psz_destination );
579 id = Add( p_stream, NULL );
582 sout_MuxDelete( p_sys->p_mux );
583 sout_AccessOutDelete( p_sys->p_grab );
584 vlc_mutex_destroy( &p_sys->lock_sdp );
585 vlc_mutex_destroy( &p_sys->lock_es );
586 free( p_sys->psz_vod_session );
587 free( p_sys->psz_destination );
592 p_sys->packet = NULL;
594 p_stream->pf_add = MuxAdd;
595 p_stream->pf_del = MuxDel;
596 p_stream->pf_send = MuxSend;
601 p_sys->p_grab = NULL;
603 p_stream->pf_add = Add;
604 p_stream->pf_del = Del;
605 p_stream->pf_send = Send;
608 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
609 SDPHandleUrl( p_stream, "sap" );
611 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
614 config_chain_t *p_cfg;
616 SDPHandleUrl( p_stream, psz );
618 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
620 if( !strcmp( p_cfg->psz_name, "sdp" ) )
622 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
625 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
626 if( !strcmp( p_cfg->psz_value, psz ) )
629 SDPHandleUrl( p_stream, p_cfg->psz_value );
635 /* update p_sout->i_out_pace_nocontrol */
636 p_stream->p_sout->i_out_pace_nocontrol++;
641 /*****************************************************************************
643 *****************************************************************************/
644 static void Close( vlc_object_t * p_this )
646 sout_stream_t *p_stream = (sout_stream_t*)p_this;
647 sout_stream_sys_t *p_sys = p_stream->p_sys;
649 /* update p_sout->i_out_pace_nocontrol */
650 p_stream->p_sout->i_out_pace_nocontrol--;
654 assert( p_sys->i_es == 1 );
656 sout_MuxDelete( p_sys->p_mux );
657 Del( p_stream, p_sys->es[0] );
658 sout_AccessOutDelete( p_sys->p_grab );
662 block_Release( p_sys->packet );
664 if( p_sys->b_export_sap )
667 SapSetup( p_stream );
671 if( p_sys->rtsp != NULL )
672 RtspUnsetup( p_sys->rtsp );
674 vlc_mutex_destroy( &p_sys->lock_sdp );
675 vlc_mutex_destroy( &p_sys->lock_ts );
676 vlc_mutex_destroy( &p_sys->lock_es );
678 if( p_sys->p_httpd_file )
679 httpd_FileDelete( p_sys->p_httpd_file );
681 if( p_sys->p_httpd_host )
682 httpd_HostDelete( p_sys->p_httpd_host );
684 free( p_sys->psz_sdp );
686 if( p_sys->psz_sdp_file != NULL )
689 unlink( p_sys->psz_sdp_file );
691 free( p_sys->psz_sdp_file );
693 free( p_sys->psz_vod_session );
694 free( p_sys->psz_destination );
698 /*****************************************************************************
700 *****************************************************************************/
701 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
703 sout_stream_sys_t *p_sys = p_stream->p_sys;
706 vlc_UrlParse( &url, psz_url, 0 );
707 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
709 if( p_sys->p_httpd_file )
711 msg_Err( p_stream, "you can use sdp=http:// only once" );
715 if( HttpSetup( p_stream, &url ) )
717 msg_Err( p_stream, "cannot export SDP as HTTP" );
720 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
722 if( p_sys->rtsp != NULL )
724 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
728 /* FIXME test if destination is multicast or no destination at all */
729 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, &url );
730 if( p_sys->rtsp == NULL )
731 msg_Err( p_stream, "cannot export SDP as RTSP" );
733 if( p_sys->p_mux != NULL )
735 sout_stream_id_t *id = p_sys->es[0];
736 rtsp_stream_id_t *rtsp_id = RtspAddId( p_sys->rtsp, id,
737 GetDWBE( id->ssrc ), id->rtp_fmt.clock_rate,
738 p_sys->psz_destination, p_sys->i_ttl,
739 id->i_port, id->i_port + 1 );
740 vlc_mutex_lock( &p_sys->lock_es );
741 id->rtsp_id = rtsp_id;
742 vlc_mutex_unlock( &p_sys->lock_es );
745 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
746 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
748 p_sys->b_export_sap = true;
749 SapSetup( p_stream );
751 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
753 if( p_sys->psz_sdp_file != NULL )
755 msg_Err( p_stream, "you can use sdp=file:// only once" );
758 p_sys->psz_sdp_file = make_path( psz_url );
759 if( p_sys->psz_sdp_file == NULL )
761 FileSetup( p_stream );
765 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
770 vlc_UrlClean( &url );
773 /*****************************************************************************
775 *****************************************************************************/
777 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
779 sout_stream_sys_t *p_sys = p_stream->p_sys;
780 char *psz_sdp = NULL;
781 struct sockaddr_storage dst;
785 * When we have a fixed destination (typically when we do multicast),
786 * we need to put the actual port numbers in the SDP.
787 * When there is no fixed destination, we only support RTSP unicast
788 * on-demand setup, so we should rather let the clients decide which ports
790 * When there is both a fixed destination and RTSP unicast, we need to
791 * put port numbers used by the fixed destination, otherwise the SDP would
792 * become totally incorrect for multicast use. It should be noted that
793 * port numbers from SDP with RTSP are only "recommendation" from the
794 * server to the clients (per RFC2326), so only broken clients will fail
795 * to handle this properly. There is no solution but to use two differents
796 * output chain with two different RTSP URLs if you need to handle this
801 vlc_mutex_lock( &p_sys->lock_es );
802 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
803 goto out; /* hmm... */
805 if( p_sys->psz_destination != NULL )
809 /* Oh boy, this is really ugly! */
810 dstlen = sizeof( dst );
811 if( p_sys->es[0]->listen.fd != NULL )
812 getsockname( p_sys->es[0]->listen.fd[0],
813 (struct sockaddr *)&dst, &dstlen );
815 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
816 (struct sockaddr *)&dst, &dstlen );
822 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
823 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
824 && rtsp_url[7] == '[';
826 /* Dummy destination address for RTSP */
827 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
828 : sizeof( struct sockaddr_in );
829 memset (&dst, 0, dstlen);
830 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
836 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
837 NULL, 0, (struct sockaddr *)&dst, dstlen );
838 if( psz_sdp == NULL )
841 /* TODO: a=source-filter */
842 if( p_sys->rtcp_mux )
843 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
845 if( rtsp_url != NULL )
846 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
848 const char *proto = "RTP/AVP"; /* protocol */
849 if( rtsp_url == NULL )
851 switch( p_sys->proto )
856 proto = "TCP/RTP/AVP";
859 proto = "DCCP/RTP/AVP";
861 case IPPROTO_UDPLITE:
866 for( i = 0; i < p_sys->i_es; i++ )
868 sout_stream_id_t *id = p_sys->es[i];
869 rtp_format_t *rtp_fmt = &id->rtp_fmt;
870 const char *mime_major; /* major MIME type */
872 switch( rtp_fmt->cat )
875 mime_major = "video";
878 mime_major = "audio";
887 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
888 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
889 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
892 /* cf RFC4566 §5.14 */
893 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
894 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
896 if( rtsp_url != NULL )
898 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
899 if( track_url != NULL )
901 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
907 if( id->listen.fd != NULL )
908 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
909 if( p_sys->proto == IPPROTO_DCCP )
910 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
911 "SC:RTP%c", toupper( mime_major[0] ) );
915 vlc_mutex_unlock( &p_sys->lock_es );
919 /*****************************************************************************
921 *****************************************************************************/
924 * Shrink the MTU down to a fixed packetization time (for audio).
927 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
929 /* Samples per second */
930 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
931 bytes *= id->rtp_fmt.channels;
934 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
935 id->i_mtu = 12 + spl;
936 else /* MTU is too small for ptime, align to a sample boundary */
937 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
940 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
942 /* NOTE: this plays nice with offsets because the calculations are
944 return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
947 /** Add an ES as a new RTP stream */
948 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
950 /* NOTE: As a special case, if we use a non-RTP
951 * mux (TS/PS), then p_fmt is NULL. */
952 sout_stream_sys_t *p_sys = p_stream->p_sys;
955 sout_stream_id_t *id = malloc( sizeof( *id ) );
956 if( unlikely(id == NULL) )
958 id->p_stream = p_stream;
960 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
961 if( id->i_mtu <= 12 + 16 )
962 id->i_mtu = 576 - 20 - 8; /* pessimistic */
963 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
968 vlc_mutex_init( &id->lock_sink );
973 id->listen.fd = NULL;
976 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
978 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
979 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
983 if (p_sys->p_vod_media != NULL)
985 id->rtp_fmt.ptname = NULL;
987 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
988 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
989 &ssrc, &id->i_seq_sent_next);
990 if (val == VLC_SUCCESS)
992 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
993 /* This is ugly, but id->i_seq_sent_next needs to be
994 * initialized inside vod_init_id() to avoid race
996 id->i_sequence = id->i_seq_sent_next;
998 /* vod_init_id() may fail either because the ES wasn't found in
999 * the VoD media, or because that track wasn't SETUP. In the
1000 * former case, id->rtp_fmt was left untouched. */
1001 format = (id->rtp_fmt.ptname != NULL);
1006 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1007 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1008 if (p_fmt == NULL && psz == NULL)
1010 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1012 if (val != VLC_SUCCESS)
1017 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1020 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1021 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1022 if (id->srtp == NULL)
1028 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1029 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1034 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1037 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1041 id->i_seq_sent_next = id->i_sequence;
1043 if( p_sys->psz_destination != NULL )
1045 /* Choose the port */
1046 uint16_t i_port = 0;
1050 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1051 i_port = p_sys->i_port_audio;
1053 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1054 i_port = p_sys->i_port_video;
1056 /* We do not need the ES lock (p_sys->lock_es) here, because
1057 * this is the only one thread that can *modify* the ES table.
1058 * The ES lock protects the other threads from our modifications
1059 * (TAB_APPEND, TAB_REMOVE). */
1060 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1061 if (i_port == p_sys->es[i]->i_port)
1062 i_port = 0; /* Port already in use! */
1063 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1067 msg_Err (p_stream, "too many RTP elementary streams");
1071 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1072 if (p == p_sys->es[i]->i_port)
1076 id->i_port = i_port;
1078 int type = SOCK_STREAM;
1080 switch( p_sys->proto )
1086 switch (id->rtp_fmt.cat)
1088 case VIDEO_ES: code = "RTPV"; break;
1089 case AUDIO_ES: code = "RTPARTPV"; break;
1090 case SPU_ES: code = "RTPTRTPV"; break;
1091 default: code = "RTPORTPV"; break;
1093 var_SetString (p_stream, "dccp-service", code);
1095 } /* fall through */
1098 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1099 p_sys->psz_destination, i_port,
1100 type, p_sys->proto );
1101 if( id->listen.fd == NULL )
1103 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1106 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1107 VLC_THREAD_PRIORITY_LOW ) )
1109 net_ListenClose( id->listen.fd );
1110 id->listen.fd = NULL;
1117 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1118 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1119 i_port, ttl, p_sys->proto );
1122 msg_Err( p_stream, "cannot create RTP socket" );
1125 /* Ignore any unexpected incoming packet (including RTCP-RR
1126 * packets in case of rtcp-mux) */
1127 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1129 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1135 switch( p_fmt->i_codec )
1137 case VLC_CODEC_MULAW:
1138 case VLC_CODEC_ALAW:
1140 rtp_set_ptime (id, 20, 1);
1142 case VLC_CODEC_S16B:
1143 case VLC_CODEC_S16L:
1144 rtp_set_ptime (id, 20, 2);
1150 #if 0 /* No payload formats sets this at the moment */
1153 cscov += 8 /* UDP */ + 12 /* RTP */;
1155 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1158 vlc_mutex_lock( &p_sys->lock_ts );
1159 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1160 vlc_mutex_unlock( &p_sys->lock_ts );
1162 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1163 p_sys->i_pts_offset );
1165 if( p_sys->rtsp != NULL )
1166 id->rtsp_id = RtspAddId( p_sys->rtsp, id,
1167 GetDWBE( id->ssrc ),
1168 id->rtp_fmt.clock_rate,
1169 p_sys->psz_destination,
1170 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1172 id->p_fifo = block_FifoNew();
1173 if( unlikely(id->p_fifo == NULL) )
1175 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1177 block_FifoRelease( id->p_fifo );
1182 /* Update p_sys context */
1183 vlc_mutex_lock( &p_sys->lock_es );
1184 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1185 vlc_mutex_unlock( &p_sys->lock_es );
1187 psz_sdp = SDPGenerate( p_stream, NULL );
1189 vlc_mutex_lock( &p_sys->lock_sdp );
1190 free( p_sys->psz_sdp );
1191 p_sys->psz_sdp = psz_sdp;
1192 vlc_mutex_unlock( &p_sys->lock_sdp );
1194 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1196 /* Update SDP (sap/file) */
1197 if( p_sys->b_export_sap ) SapSetup( p_stream );
1198 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1203 Del( p_stream, id );
1207 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1209 sout_stream_sys_t *p_sys = p_stream->p_sys;
1211 vlc_mutex_lock( &p_sys->lock_es );
1212 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1213 vlc_mutex_unlock( &p_sys->lock_es );
1215 if( likely(id->p_fifo != NULL) )
1217 vlc_cancel( id->thread );
1218 vlc_join( id->thread, NULL );
1219 block_FifoRelease( id->p_fifo );
1222 free( id->rtp_fmt.fmtp );
1224 if (p_sys->p_vod_media != NULL)
1225 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1227 RtspDelId( p_sys->rtsp, id->rtsp_id );
1228 if( id->listen.fd != NULL )
1230 vlc_cancel( id->listen.thread );
1231 vlc_join( id->listen.thread, NULL );
1232 net_ListenClose( id->listen.fd );
1234 /* Delete remaining sinks (incoming connections or explicit
1236 while( id->sinkc > 0 )
1237 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1239 if( id->srtp != NULL )
1240 srtp_destroy( id->srtp );
1243 vlc_mutex_destroy( &id->lock_sink );
1245 /* Update SDP (sap/file) */
1246 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1247 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1253 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1258 assert( p_stream->p_sys->p_mux == NULL );
1261 while( p_buffer != NULL )
1263 p_next = p_buffer->p_next;
1264 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1267 block_Release( p_buffer );
1273 /****************************************************************************
1275 ****************************************************************************/
1276 static int SapSetup( sout_stream_t *p_stream )
1278 sout_stream_sys_t *p_sys = p_stream->p_sys;
1279 sout_instance_t *p_sout = p_stream->p_sout;
1281 /* Remove the previous session */
1282 if( p_sys->p_session != NULL)
1284 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1285 p_sys->p_session = NULL;
1288 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1290 announce_method_t *p_method = sout_SAPMethod();
1291 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1293 p_sys->psz_destination,
1295 sout_MethodRelease( p_method );
1301 /****************************************************************************
1303 ****************************************************************************/
1304 static int FileSetup( sout_stream_t *p_stream )
1306 sout_stream_sys_t *p_sys = p_stream->p_sys;
1309 if( p_sys->psz_sdp == NULL )
1310 return VLC_EGENERIC; /* too early */
1312 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1314 msg_Err( p_stream, "cannot open file '%s' (%m)",
1315 p_sys->psz_sdp_file );
1316 return VLC_EGENERIC;
1319 fputs( p_sys->psz_sdp, f );
1325 /****************************************************************************
1327 ****************************************************************************/
1328 static int HttpCallback( httpd_file_sys_t *p_args,
1329 httpd_file_t *, uint8_t *p_request,
1330 uint8_t **pp_data, int *pi_data );
1332 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1334 sout_stream_sys_t *p_sys = p_stream->p_sys;
1336 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1337 url->i_port > 0 ? url->i_port : 80 );
1338 if( p_sys->p_httpd_host )
1340 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1341 url->psz_path ? url->psz_path : "/",
1344 HttpCallback, (void*)p_sys );
1346 if( p_sys->p_httpd_file == NULL )
1348 return VLC_EGENERIC;
1353 static int HttpCallback( httpd_file_sys_t *p_args,
1354 httpd_file_t *f, uint8_t *p_request,
1355 uint8_t **pp_data, int *pi_data )
1357 VLC_UNUSED(f); VLC_UNUSED(p_request);
1358 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1360 vlc_mutex_lock( &p_sys->lock_sdp );
1361 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1363 *pi_data = strlen( p_sys->psz_sdp );
1364 *pp_data = malloc( *pi_data );
1365 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1372 vlc_mutex_unlock( &p_sys->lock_sdp );
1377 /****************************************************************************
1379 ****************************************************************************/
1380 static void* ThreadSend( void *data )
1383 # define ECONNREFUSED WSAECONNREFUSED
1384 # define ENOPROTOOPT WSAENOPROTOOPT
1385 # define EHOSTUNREACH WSAEHOSTUNREACH
1386 # define ENETUNREACH WSAENETUNREACH
1387 # define ENETDOWN WSAENETDOWN
1388 # define ENOBUFS WSAENOBUFS
1389 # define EAGAIN WSAEWOULDBLOCK
1390 # define EWOULDBLOCK WSAEWOULDBLOCK
1392 sout_stream_id_t *id = data;
1393 unsigned i_caching = id->i_caching;
1397 block_t *out = block_FifoGet( id->p_fifo );
1398 block_cleanup_push (out);
1402 { /* FIXME: this is awfully inefficient */
1403 size_t len = out->i_buffer;
1404 out = block_Realloc( out, 0, len + 10 );
1405 out->i_buffer = len;
1407 int canc = vlc_savecancel ();
1408 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1409 vlc_restorecancel (canc);
1413 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1414 block_Release( out );
1418 out->i_buffer = len;
1422 mwait (out->i_dts + i_caching);
1427 ssize_t len = out->i_buffer;
1428 int canc = vlc_savecancel ();
1430 vlc_mutex_lock( &id->lock_sink );
1431 unsigned deadc = 0; /* How many dead sockets? */
1432 int deadv[id->sinkc]; /* Dead sockets list */
1434 for( int i = 0; i < id->sinkc; i++ )
1437 if( !id->srtp ) /* FIXME: SRTCP support */
1439 SendRTCP( id->sinkv[i].rtcp, out );
1441 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1445 /* Soft errors (e.g. ICMP): */
1446 case ECONNREFUSED: /* Port unreachable */
1449 case EPROTO: /* Protocol unreachable */
1451 case EHOSTUNREACH: /* Host unreachable */
1452 case ENETUNREACH: /* Network unreachable */
1453 case ENETDOWN: /* Entire network down */
1454 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1455 /* Transient congestion: */
1456 case ENOMEM: /* out of socket buffers */
1459 #if (EAGAIN != EWOULDBLOCK)
1465 deadv[deadc++] = id->sinkv[i].rtp_fd;
1467 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1468 vlc_mutex_unlock( &id->lock_sink );
1469 block_Release( out );
1471 for( unsigned i = 0; i < deadc; i++ )
1473 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1474 rtp_del_sink( id, deadv[i] );
1476 vlc_restorecancel (canc);
1482 /* This thread dequeues incoming connections (DCCP streaming) */
1483 static void *rtp_listen_thread( void *data )
1485 sout_stream_id_t *id = data;
1487 assert( id->listen.fd != NULL );
1491 int fd = net_Accept( id->p_stream, id->listen.fd );
1494 int canc = vlc_savecancel( );
1495 rtp_add_sink( id, fd, true, NULL );
1496 vlc_restorecancel( canc );
1503 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1505 rtp_sink_t sink = { fd, NULL };
1506 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1508 if( sink.rtcp == NULL )
1509 msg_Err( id->p_stream, "RTCP failed!" );
1511 vlc_mutex_lock( &id->lock_sink );
1512 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1514 *seq = id->i_seq_sent_next;
1515 vlc_mutex_unlock( &id->lock_sink );
1519 void rtp_del_sink( sout_stream_id_t *id, int fd )
1521 rtp_sink_t sink = { fd, NULL };
1523 /* NOTE: must be safe to use if fd is not included */
1524 vlc_mutex_lock( &id->lock_sink );
1525 for( int i = 0; i < id->sinkc; i++ )
1527 if (id->sinkv[i].rtp_fd == fd)
1529 sink = id->sinkv[i];
1530 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1534 vlc_mutex_unlock( &id->lock_sink );
1536 CloseRTCP( sink.rtcp );
1537 net_Close( sink.rtp_fd );
1540 uint16_t rtp_get_seq( sout_stream_id_t *id )
1542 /* This will return values for the next packet. */
1545 vlc_mutex_lock( &id->lock_sink );
1546 seq = id->i_seq_sent_next;
1547 vlc_mutex_unlock( &id->lock_sink );
1552 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1553 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1554 * random (although we use the same reference for all the ES as a
1555 * feature). In the VoD case, this function is called independently
1556 * from several parts of the code, so we need to always return the same
1558 static int64_t rtp_init_ts( const vod_media_t *p_media,
1559 const char *psz_vod_session )
1561 if (p_media == NULL || psz_vod_session == NULL)
1565 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1566 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1567 i_ts_init ^= (uint64_t) p_media;
1568 /* Limit the timestamp to 48 bytes, this is enough and allows us
1569 * to stay away from overflows */
1570 i_ts_init &= 0xFFFFFFFFFFFF;
1574 /* Return a timestamp corresponding to packets being sent now, and that
1575 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1576 * If the stream output is not started, the initial timestamp that will
1577 * be used with the first packets is returned instead. */
1578 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1579 const vod_media_t *p_media, const char *psz_vod_session )
1582 p_stream = id->p_stream;
1584 if (p_stream == NULL)
1585 return rtp_init_ts(p_media, psz_vod_session);
1587 sout_stream_sys_t *p_sys = p_stream->p_sys;
1589 vlc_mutex_lock( &p_sys->lock_ts );
1590 i_npt_zero = p_sys->i_npt_zero;
1591 vlc_mutex_unlock( &p_sys->lock_ts );
1593 if( i_npt_zero == VLC_TS_INVALID )
1594 return p_sys->i_pts_zero;
1596 mtime_t now = mdate();
1597 if( now < i_npt_zero )
1598 return p_sys->i_pts_zero;
1600 return p_sys->i_pts_zero + (now - i_npt_zero);
1603 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1604 int b_marker, int64_t i_pts )
1606 if( !id->b_ts_init )
1608 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1609 vlc_mutex_lock( &p_sys->lock_ts );
1610 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1612 /* This is the first packet of any ES. We initialize the
1613 * NPT=0 time reference, and the offset to match the
1614 * arbitrary PTS reference. */
1615 p_sys->i_npt_zero = i_pts + id->i_caching;
1616 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1618 vlc_mutex_unlock( &p_sys->lock_ts );
1620 /* And in any case this is the first packet of this ES, so we
1621 * initialize the offset for this ES. */
1622 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1623 p_sys->i_pts_offset );
1624 id->b_ts_init = true;
1627 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1630 out->p_buffer[0] = 0x80;
1631 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1632 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1633 out->p_buffer[3] = ( id->i_sequence )&0xff;
1634 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1635 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1636 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1637 out->p_buffer[7] = ( i_timestamp )&0xff;
1639 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1645 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1647 block_FifoPut( id->p_fifo, out );
1651 * @return configured max RTP payload size (including payload type-specific
1652 * headers, excluding RTP and transport headers)
1654 size_t rtp_mtu (const sout_stream_id_t *id)
1656 return id->i_mtu - 12;
1659 /*****************************************************************************
1661 *****************************************************************************/
1663 /** Add an ES to a non-RTP muxed stream */
1664 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1666 sout_input_t *p_input;
1667 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1668 assert( p_mux != NULL );
1670 p_input = sout_MuxAddStream( p_mux, p_fmt );
1671 if( p_input == NULL )
1673 msg_Err( p_stream, "cannot add this stream to the muxer" );
1677 return (sout_stream_id_t *)p_input;
1681 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1684 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1685 assert( p_mux != NULL );
1687 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1692 /** Remove an ES from a non-RTP muxed stream */
1693 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1695 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1696 assert( p_mux != NULL );
1698 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1703 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1704 const block_t *p_buffer )
1706 sout_stream_sys_t *p_sys = p_stream->p_sys;
1707 sout_stream_id_t *id = p_sys->es[0];
1709 int64_t i_dts = p_buffer->i_dts;
1711 uint8_t *p_data = p_buffer->p_buffer;
1712 size_t i_data = p_buffer->i_buffer;
1713 size_t i_max = id->i_mtu - 12;
1715 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1721 /* output complete packet */
1722 if( p_sys->packet &&
1723 p_sys->packet->i_buffer + i_data > i_max )
1725 rtp_packetize_send( id, p_sys->packet );
1726 p_sys->packet = NULL;
1729 if( p_sys->packet == NULL )
1731 /* allocate a new packet */
1732 p_sys->packet = block_New( p_stream, id->i_mtu );
1733 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1734 p_sys->packet->i_dts = i_dts;
1735 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1736 i_dts += p_sys->packet->i_length;
1739 i_size = __MIN( i_data,
1740 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1742 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1745 p_sys->packet->i_buffer += i_size;
1754 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1757 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1763 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1765 p_next = p_buffer->p_next;
1766 block_Release( p_buffer );
1774 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1776 sout_access_out_t *p_grab;
1778 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1779 if( p_grab == NULL )
1782 p_grab->p_module = NULL;
1783 p_grab->psz_access = strdup( "grab" );
1784 p_grab->p_cfg = NULL;
1785 p_grab->psz_path = strdup( "" );
1786 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1787 p_grab->pf_seek = NULL;
1788 p_grab->pf_write = AccessOutGrabberWrite;
1789 vlc_object_attach( p_grab, p_stream );