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 #define RTSP_USER_TEXT N_("Username")
179 #define RTSP_USER_LONGTEXT N_("User name that will be " \
180 "requested to access the stream." )
181 #define RTSP_PASS_TEXT N_("Password")
182 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
183 "requested to access the stream." )
185 static int Open ( vlc_object_t * );
186 static void Close( vlc_object_t * );
188 #define SOUT_CFG_PREFIX "sout-rtp-"
189 #define MAX_EMPTY_BLOCKS 200
192 set_shortname( N_("RTP"))
193 set_description( N_("RTP stream output") )
194 set_capability( "sout stream", 0 )
195 add_shortcut( "rtp", "vod" )
196 set_category( CAT_SOUT )
197 set_subcategory( SUBCAT_SOUT_STREAM )
199 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
200 DEST_LONGTEXT, true )
201 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
203 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
205 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
208 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
209 NAME_LONGTEXT, true )
210 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
211 DESC_LONGTEXT, true )
212 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
214 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
215 EMAIL_LONGTEXT, true )
216 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
217 PHONE_LONGTEXT, true )
219 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
220 PROTO_LONGTEXT, false )
221 change_string_list( ppsz_protos, ppsz_protocols, NULL )
222 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
223 PORT_LONGTEXT, true )
224 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
225 PORT_AUDIO_LONGTEXT, true )
226 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
227 PORT_VIDEO_LONGTEXT, true )
229 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
231 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
232 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
233 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
234 CACHING_TEXT, CACHING_LONGTEXT, true )
237 add_string( SOUT_CFG_PREFIX "key", "",
238 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
239 add_string( SOUT_CFG_PREFIX "salt", "",
240 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
243 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
244 RFC3016_LONGTEXT, false )
246 set_callbacks( Open, Close )
249 set_shortname( N_("RTSP VoD" ) )
250 set_description( N_("RTSP VoD server") )
251 set_category( CAT_SOUT )
252 set_subcategory( SUBCAT_SOUT_VOD )
253 set_capability( "vod server", 10 )
254 set_callbacks( OpenVoD, CloseVoD )
255 add_shortcut( "rtsp" )
256 add_string ( "rtsp-host", NULL, RTSP_HOST_TEXT,
257 RTSP_HOST_LONGTEXT, true )
258 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
259 RTSP_TIMEOUT_LONGTEXT, true )
260 add_string( "sout-rtsp-user", "",
261 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
262 add_password( "sout-rtsp-pwd", "",
263 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
267 /*****************************************************************************
268 * Exported prototypes
269 *****************************************************************************/
270 static const char *const ppsz_sout_options[] = {
271 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
272 "sap", "description", "url", "email", "phone",
273 "proto", "rtcp-mux", "caching", "key", "salt",
277 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
278 static int Del ( sout_stream_t *, sout_stream_id_t * );
279 static int Send( sout_stream_t *, sout_stream_id_t *,
281 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
282 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
283 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
286 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
287 static void* ThreadSend( void * );
288 static void *rtp_listen_thread( void * );
290 static void SDPHandleUrl( sout_stream_t *, const char * );
292 static int SapSetup( sout_stream_t *p_stream );
293 static int FileSetup( sout_stream_t *p_stream );
294 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
296 static int64_t rtp_init_ts( const vod_media_t *p_media,
297 const char *psz_vod_session );
299 struct sout_stream_sys_t
303 vlc_mutex_t lock_sdp;
310 session_descriptor_t *p_session;
313 httpd_host_t *p_httpd_host;
314 httpd_file_t *p_httpd_file;
319 /* RTSP NPT and timestamp computations */
320 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
321 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
322 int64_t i_pts_offset; /* matches actual PTS to prediction */
326 char *psz_destination;
328 uint16_t i_port_audio;
329 uint16_t i_port_video;
335 vod_media_t *p_vod_media;
336 char *psz_vod_session;
338 /* in case we do TS/PS over rtp */
340 sout_access_out_t *p_grab;
346 sout_stream_id_t **es;
349 typedef struct rtp_sink_t
355 struct sout_stream_id_t
357 sout_stream_t *p_stream;
362 uint32_t i_ts_offset;
366 uint16_t i_seq_sent_next;
369 rtp_format_t rtp_fmt;
372 /* Packetizer specific fields */
375 srtp_session_t *srtp;
380 vlc_mutex_t lock_sink;
383 rtsp_stream_id_t *rtsp_id;
389 block_fifo_t *p_fifo;
393 /*****************************************************************************
395 *****************************************************************************/
396 static int Open( vlc_object_t *p_this )
398 sout_stream_t *p_stream = (sout_stream_t*)p_this;
399 sout_instance_t *p_sout = p_stream->p_sout;
400 sout_stream_sys_t *p_sys = NULL;
401 config_chain_t *p_cfg = NULL;
405 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
406 ppsz_sout_options, p_stream->p_cfg );
408 p_sys = malloc( sizeof( sout_stream_sys_t ) );
412 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
414 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
415 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
416 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
417 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
419 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
421 msg_Err( p_stream, "audio and video RTP port must be distinct" );
422 free( p_sys->psz_destination );
427 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
429 if( !strcmp( p_cfg->psz_name, "sdp" )
430 && ( p_cfg->psz_value != NULL )
431 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
439 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
442 if( !strncasecmp( psz, "rtsp:", 5 ) )
448 /* Transport protocol */
449 p_sys->proto = IPPROTO_UDP;
450 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
452 if ((psz == NULL) || !strcasecmp (psz, "udp"))
453 (void)0; /* default */
455 if (!strcasecmp (psz, "dccp"))
457 p_sys->proto = IPPROTO_DCCP;
458 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
462 if (!strcasecmp (psz, "sctp"))
464 p_sys->proto = IPPROTO_TCP;
465 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
470 if (!strcasecmp (psz, "tcp"))
472 p_sys->proto = IPPROTO_TCP;
473 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
477 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
478 p_sys->proto = IPPROTO_UDPLITE;
480 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
483 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
485 p_sys->p_vod_media = NULL;
486 p_sys->psz_vod_session = NULL;
488 if (! strcmp(p_stream->psz_name, "vod"))
490 /* The VLM stops all instances before deleting a media, so this
491 * reference will remain valid during the lifetime of the rtp
493 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
495 if (p_sys->p_vod_media != NULL)
497 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
498 if (p_sys->psz_vod_session == NULL)
500 msg_Err(p_stream, "missing VoD session");
505 const char *mux = vod_get_mux(p_sys->p_vod_media);
506 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
510 if( p_sys->psz_destination == NULL && !b_rtsp
511 && p_sys->p_vod_media == NULL )
513 msg_Err( p_stream, "missing destination and not in RTSP mode" );
518 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
521 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
522 var_SetInteger( p_stream, "ttl", i_ttl );
525 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
527 /* NPT=0 time will be determined when we packetize the first packet
528 * (of any ES). But we want to be able to report rtptime in RTSP
529 * without waiting (and already did in the VoD case). So until then,
530 * we use an arbitrary reference PTS for timestamp computations, and
531 * then actual PTS will catch up using offsets. */
532 p_sys->i_npt_zero = VLC_TS_INVALID;
533 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
534 p_sys->psz_vod_session);
538 p_sys->psz_sdp = NULL;
540 p_sys->b_export_sap = false;
541 p_sys->p_session = NULL;
542 p_sys->psz_sdp_file = NULL;
544 p_sys->p_httpd_host = NULL;
545 p_sys->p_httpd_file = NULL;
547 p_stream->p_sys = p_sys;
549 vlc_mutex_init( &p_sys->lock_sdp );
550 vlc_mutex_init( &p_sys->lock_ts );
551 vlc_mutex_init( &p_sys->lock_es );
553 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
556 /* Check muxer type */
557 if( strncasecmp( psz, "ps", 2 )
558 && strncasecmp( psz, "mpeg1", 5 )
559 && strncasecmp( psz, "ts", 2 ) )
561 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
563 vlc_mutex_destroy( &p_sys->lock_sdp );
564 vlc_mutex_destroy( &p_sys->lock_ts );
565 vlc_mutex_destroy( &p_sys->lock_es );
566 free( p_sys->psz_vod_session );
567 free( p_sys->psz_destination );
572 p_sys->p_grab = GrabberCreate( p_stream );
573 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
576 if( p_sys->p_mux == NULL )
578 msg_Err( p_stream, "cannot create muxer" );
579 sout_AccessOutDelete( p_sys->p_grab );
580 vlc_mutex_destroy( &p_sys->lock_sdp );
581 vlc_mutex_destroy( &p_sys->lock_ts );
582 vlc_mutex_destroy( &p_sys->lock_es );
583 free( p_sys->psz_vod_session );
584 free( p_sys->psz_destination );
589 p_sys->packet = NULL;
591 p_stream->pf_add = MuxAdd;
592 p_stream->pf_del = MuxDel;
593 p_stream->pf_send = MuxSend;
598 p_sys->p_grab = NULL;
600 p_stream->pf_add = Add;
601 p_stream->pf_del = Del;
602 p_stream->pf_send = Send;
605 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
606 SDPHandleUrl( p_stream, "sap" );
608 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
611 config_chain_t *p_cfg;
613 SDPHandleUrl( p_stream, psz );
615 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
617 if( !strcmp( p_cfg->psz_name, "sdp" ) )
619 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
622 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
623 if( !strcmp( p_cfg->psz_value, psz ) )
626 SDPHandleUrl( p_stream, p_cfg->psz_value );
632 /* update p_sout->i_out_pace_nocontrol */
633 p_stream->p_sout->i_out_pace_nocontrol++;
635 if( p_sys->p_mux != NULL )
637 sout_stream_id_t *id = Add( p_stream, NULL );
648 /*****************************************************************************
650 *****************************************************************************/
651 static void Close( vlc_object_t * p_this )
653 sout_stream_t *p_stream = (sout_stream_t*)p_this;
654 sout_stream_sys_t *p_sys = p_stream->p_sys;
656 /* update p_sout->i_out_pace_nocontrol */
657 p_stream->p_sout->i_out_pace_nocontrol--;
661 assert( p_sys->i_es <= 1 );
663 sout_MuxDelete( p_sys->p_mux );
664 if ( p_sys->i_es > 0 )
665 Del( p_stream, p_sys->es[0] );
666 sout_AccessOutDelete( p_sys->p_grab );
670 block_Release( p_sys->packet );
674 if( p_sys->rtsp != NULL )
675 RtspUnsetup( p_sys->rtsp );
677 vlc_mutex_destroy( &p_sys->lock_sdp );
678 vlc_mutex_destroy( &p_sys->lock_ts );
679 vlc_mutex_destroy( &p_sys->lock_es );
681 if( p_sys->p_httpd_file )
682 httpd_FileDelete( p_sys->p_httpd_file );
684 if( p_sys->p_httpd_host )
685 httpd_HostDelete( p_sys->p_httpd_host );
687 free( p_sys->psz_sdp );
689 if( p_sys->psz_sdp_file != NULL )
692 unlink( p_sys->psz_sdp_file );
694 free( p_sys->psz_sdp_file );
696 free( p_sys->psz_vod_session );
697 free( p_sys->psz_destination );
701 /*****************************************************************************
703 *****************************************************************************/
704 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
706 sout_stream_sys_t *p_sys = p_stream->p_sys;
709 vlc_UrlParse( &url, psz_url, 0 );
710 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
712 if( p_sys->p_httpd_file )
714 msg_Err( p_stream, "you can use sdp=http:// only once" );
718 if( HttpSetup( p_stream, &url ) )
720 msg_Err( p_stream, "cannot export SDP as HTTP" );
723 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
725 if( p_sys->rtsp != NULL )
727 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
731 if( url.psz_host != NULL && *url.psz_host )
733 /* msg_Err( p_stream, "\"%s\" RTSP host ignored", url.psz_host );
734 msg_Info( p_stream, "Pass --rtsp-host=%s on the command line "
735 "instead.", url.psz_host ); */
737 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
738 var_SetString( p_stream, "rtsp-host", url.psz_host );
740 /* if( url.i_port != 0 )
742 msg_Err( p_stream, "\"%u\" RTSP port ignored", url.i_port );
743 msg_Info( p_stream, "Pass --rtsp-port=%u on the command line "
744 "instead.", url.i_port );
747 if( url.i_port <= 0 ) url.i_port = 554;
748 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
749 var_SetInteger( p_stream, "rtsp-port", url.i_port );
751 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
752 if( p_sys->rtsp == NULL )
753 msg_Err( p_stream, "cannot export SDP as RTSP" );
755 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
756 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
758 p_sys->b_export_sap = true;
759 SapSetup( p_stream );
761 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
763 if( p_sys->psz_sdp_file != NULL )
765 msg_Err( p_stream, "you can use sdp=file:// only once" );
768 p_sys->psz_sdp_file = make_path( psz_url );
769 if( p_sys->psz_sdp_file == NULL )
771 FileSetup( p_stream );
775 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
780 vlc_UrlClean( &url );
783 /*****************************************************************************
785 *****************************************************************************/
787 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
789 sout_stream_sys_t *p_sys = p_stream->p_sys;
790 char *psz_sdp = NULL;
791 struct sockaddr_storage dst;
795 * When we have a fixed destination (typically when we do multicast),
796 * we need to put the actual port numbers in the SDP.
797 * When there is no fixed destination, we only support RTSP unicast
798 * on-demand setup, so we should rather let the clients decide which ports
800 * When there is both a fixed destination and RTSP unicast, we need to
801 * put port numbers used by the fixed destination, otherwise the SDP would
802 * become totally incorrect for multicast use. It should be noted that
803 * port numbers from SDP with RTSP are only "recommendation" from the
804 * server to the clients (per RFC2326), so only broken clients will fail
805 * to handle this properly. There is no solution but to use two differents
806 * output chain with two different RTSP URLs if you need to handle this
811 vlc_mutex_lock( &p_sys->lock_es );
812 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
813 goto out; /* hmm... */
815 if( p_sys->psz_destination != NULL )
819 /* Oh boy, this is really ugly! */
820 dstlen = sizeof( dst );
821 if( p_sys->es[0]->listen.fd != NULL )
822 getsockname( p_sys->es[0]->listen.fd[0],
823 (struct sockaddr *)&dst, &dstlen );
825 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
826 (struct sockaddr *)&dst, &dstlen );
832 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
833 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
834 && rtsp_url[7] == '[';
836 /* Dummy destination address for RTSP */
837 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
838 : sizeof( struct sockaddr_in );
839 memset (&dst, 0, dstlen);
840 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
846 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
847 NULL, 0, (struct sockaddr *)&dst, dstlen );
848 if( psz_sdp == NULL )
851 /* TODO: a=source-filter */
852 if( p_sys->rtcp_mux )
853 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
855 if( rtsp_url != NULL )
856 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
858 const char *proto = "RTP/AVP"; /* protocol */
859 if( rtsp_url == NULL )
861 switch( p_sys->proto )
866 proto = "TCP/RTP/AVP";
869 proto = "DCCP/RTP/AVP";
871 case IPPROTO_UDPLITE:
876 for( i = 0; i < p_sys->i_es; i++ )
878 sout_stream_id_t *id = p_sys->es[i];
879 rtp_format_t *rtp_fmt = &id->rtp_fmt;
880 const char *mime_major; /* major MIME type */
882 switch( rtp_fmt->cat )
885 mime_major = "video";
888 mime_major = "audio";
897 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
898 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
899 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
902 /* cf RFC4566 §5.14 */
903 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
904 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
906 if( rtsp_url != NULL )
908 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
909 if( track_url != NULL )
911 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
917 if( id->listen.fd != NULL )
918 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
919 if( p_sys->proto == IPPROTO_DCCP )
920 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
922 toupper( (unsigned char)mime_major[0] ) );
926 vlc_mutex_unlock( &p_sys->lock_es );
930 /*****************************************************************************
932 *****************************************************************************/
935 * Shrink the MTU down to a fixed packetization time (for audio).
938 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
940 /* Samples per second */
941 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
942 bytes *= id->rtp_fmt.channels;
945 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
946 id->i_mtu = 12 + spl;
947 else /* MTU is too small for ptime, align to a sample boundary */
948 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
951 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
953 /* NOTE: this plays nice with offsets because the calculations are
955 return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
958 /** Add an ES as a new RTP stream */
959 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
961 /* NOTE: As a special case, if we use a non-RTP
962 * mux (TS/PS), then p_fmt is NULL. */
963 sout_stream_sys_t *p_sys = p_stream->p_sys;
966 sout_stream_id_t *id = malloc( sizeof( *id ) );
967 if( unlikely(id == NULL) )
969 id->p_stream = p_stream;
971 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
972 if( id->i_mtu <= 12 + 16 )
973 id->i_mtu = 576 - 20 - 8; /* pessimistic */
974 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
979 vlc_mutex_init( &id->lock_sink );
984 id->listen.fd = NULL;
986 id->b_first_packet = true;
988 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
990 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
991 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
995 if (p_sys->p_vod_media != NULL)
997 id->rtp_fmt.ptname = NULL;
999 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
1000 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
1001 &ssrc, &id->i_seq_sent_next);
1002 if (val == VLC_SUCCESS)
1004 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1005 /* This is ugly, but id->i_seq_sent_next needs to be
1006 * initialized inside vod_init_id() to avoid race
1008 id->i_sequence = id->i_seq_sent_next;
1010 /* vod_init_id() may fail either because the ES wasn't found in
1011 * the VoD media, or because the RTSP session is gone. In the
1012 * former case, id->rtp_fmt was left untouched. */
1013 format = (id->rtp_fmt.ptname != NULL);
1018 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1019 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1020 if (p_fmt == NULL && psz == NULL)
1022 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1024 if (val != VLC_SUCCESS)
1029 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1033 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1034 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1035 if (id->srtp == NULL)
1041 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1042 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1047 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1050 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1054 id->i_seq_sent_next = id->i_sequence;
1057 if( p_sys->psz_destination != NULL )
1059 /* Choose the port */
1060 uint16_t i_port = 0;
1064 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1065 i_port = p_sys->i_port_audio;
1067 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1068 i_port = p_sys->i_port_video;
1070 /* We do not need the ES lock (p_sys->lock_es) here, because
1071 * this is the only one thread that can *modify* the ES table.
1072 * The ES lock protects the other threads from our modifications
1073 * (TAB_APPEND, TAB_REMOVE). */
1074 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1075 if (i_port == p_sys->es[i]->i_port)
1076 i_port = 0; /* Port already in use! */
1077 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1081 msg_Err (p_stream, "too many RTP elementary streams");
1085 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1086 if (p == p_sys->es[i]->i_port)
1090 id->i_port = i_port;
1092 int type = SOCK_STREAM;
1094 switch( p_sys->proto )
1100 switch (id->rtp_fmt.cat)
1102 case VIDEO_ES: code = "RTPV"; break;
1103 case AUDIO_ES: code = "RTPARTPV"; break;
1104 case SPU_ES: code = "RTPTRTPV"; break;
1105 default: code = "RTPORTPV"; break;
1107 var_SetString (p_stream, "dccp-service", code);
1109 } /* fall through */
1112 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1113 p_sys->psz_destination, i_port,
1114 type, p_sys->proto );
1115 if( id->listen.fd == NULL )
1117 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1120 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1121 VLC_THREAD_PRIORITY_LOW ) )
1123 net_ListenClose( id->listen.fd );
1124 id->listen.fd = NULL;
1131 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1132 i_port, -1, p_sys->proto );
1135 msg_Err( p_stream, "cannot create RTP socket" );
1138 /* Ignore any unexpected incoming packet (including RTCP-RR
1139 * packets in case of rtcp-mux) */
1140 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1142 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1143 /* FIXME: test if this is multicast */
1150 switch( p_fmt->i_codec )
1152 case VLC_CODEC_MULAW:
1153 case VLC_CODEC_ALAW:
1155 rtp_set_ptime (id, 20, 1);
1157 case VLC_CODEC_S16B:
1158 case VLC_CODEC_S16L:
1159 rtp_set_ptime (id, 20, 2);
1165 #if 0 /* No payload formats sets this at the moment */
1168 cscov += 8 /* UDP */ + 12 /* RTP */;
1170 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1173 vlc_mutex_lock( &p_sys->lock_ts );
1174 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1175 vlc_mutex_unlock( &p_sys->lock_ts );
1177 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1178 p_sys->i_pts_offset );
1180 if( p_sys->rtsp != NULL )
1181 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1182 id->rtp_fmt.clock_rate, mcast_fd );
1184 id->p_fifo = block_FifoNew();
1185 if( unlikely(id->p_fifo == NULL) )
1187 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1189 block_FifoRelease( id->p_fifo );
1194 /* Update p_sys context */
1195 vlc_mutex_lock( &p_sys->lock_es );
1196 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1197 vlc_mutex_unlock( &p_sys->lock_es );
1199 psz_sdp = SDPGenerate( p_stream, NULL );
1201 vlc_mutex_lock( &p_sys->lock_sdp );
1202 free( p_sys->psz_sdp );
1203 p_sys->psz_sdp = psz_sdp;
1204 vlc_mutex_unlock( &p_sys->lock_sdp );
1206 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1208 /* Update SDP (sap/file) */
1209 if( p_sys->b_export_sap ) SapSetup( p_stream );
1210 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1215 Del( p_stream, id );
1219 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1221 sout_stream_sys_t *p_sys = p_stream->p_sys;
1223 vlc_mutex_lock( &p_sys->lock_es );
1224 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1225 vlc_mutex_unlock( &p_sys->lock_es );
1227 if( likely(id->p_fifo != NULL) )
1229 vlc_cancel( id->thread );
1230 vlc_join( id->thread, NULL );
1231 block_FifoRelease( id->p_fifo );
1234 free( id->rtp_fmt.fmtp );
1236 if (p_sys->p_vod_media != NULL)
1237 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1239 RtspDelId( p_sys->rtsp, id->rtsp_id );
1240 if( id->listen.fd != NULL )
1242 vlc_cancel( id->listen.thread );
1243 vlc_join( id->listen.thread, NULL );
1244 net_ListenClose( id->listen.fd );
1246 /* Delete remaining sinks (incoming connections or explicit
1248 while( id->sinkc > 0 )
1249 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1251 if( id->srtp != NULL )
1252 srtp_destroy( id->srtp );
1255 vlc_mutex_destroy( &id->lock_sink );
1257 /* Update SDP (sap/file) */
1258 if( p_sys->b_export_sap ) SapSetup( p_stream );
1259 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1265 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1270 assert( p_stream->p_sys->p_mux == NULL );
1273 while( p_buffer != NULL )
1275 p_next = p_buffer->p_next;
1277 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1278 * as the first packet of the stream */
1279 if (id->b_first_packet)
1281 id->b_first_packet = false;
1282 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1283 !strcmp(id->rtp_fmt.ptname, "theora"))
1284 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1288 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1291 block_Release( p_buffer );
1297 /****************************************************************************
1299 ****************************************************************************/
1300 static int SapSetup( sout_stream_t *p_stream )
1302 sout_stream_sys_t *p_sys = p_stream->p_sys;
1303 sout_instance_t *p_sout = p_stream->p_sout;
1305 /* Remove the previous session */
1306 if( p_sys->p_session != NULL)
1308 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1309 p_sys->p_session = NULL;
1312 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1313 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1315 p_sys->psz_destination );
1320 /****************************************************************************
1322 ****************************************************************************/
1323 static int FileSetup( sout_stream_t *p_stream )
1325 sout_stream_sys_t *p_sys = p_stream->p_sys;
1328 if( p_sys->psz_sdp == NULL )
1329 return VLC_EGENERIC; /* too early */
1331 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1333 msg_Err( p_stream, "cannot open file '%s' (%m)",
1334 p_sys->psz_sdp_file );
1335 return VLC_EGENERIC;
1338 fputs( p_sys->psz_sdp, f );
1344 /****************************************************************************
1346 ****************************************************************************/
1347 static int HttpCallback( httpd_file_sys_t *p_args,
1348 httpd_file_t *, uint8_t *p_request,
1349 uint8_t **pp_data, int *pi_data );
1351 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1353 sout_stream_sys_t *p_sys = p_stream->p_sys;
1355 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1356 if( p_sys->p_httpd_host )
1358 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1359 url->psz_path ? url->psz_path : "/",
1362 HttpCallback, (void*)p_sys );
1364 if( p_sys->p_httpd_file == NULL )
1366 return VLC_EGENERIC;
1371 static int HttpCallback( httpd_file_sys_t *p_args,
1372 httpd_file_t *f, uint8_t *p_request,
1373 uint8_t **pp_data, int *pi_data )
1375 VLC_UNUSED(f); VLC_UNUSED(p_request);
1376 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1378 vlc_mutex_lock( &p_sys->lock_sdp );
1379 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1381 *pi_data = strlen( p_sys->psz_sdp );
1382 *pp_data = malloc( *pi_data );
1383 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1390 vlc_mutex_unlock( &p_sys->lock_sdp );
1395 /****************************************************************************
1397 ****************************************************************************/
1398 static void* ThreadSend( void *data )
1401 # define ECONNREFUSED WSAECONNREFUSED
1402 # define ENOPROTOOPT WSAENOPROTOOPT
1403 # define EHOSTUNREACH WSAEHOSTUNREACH
1404 # define ENETUNREACH WSAENETUNREACH
1405 # define ENETDOWN WSAENETDOWN
1406 # define ENOBUFS WSAENOBUFS
1407 # define EAGAIN WSAEWOULDBLOCK
1408 # define EWOULDBLOCK WSAEWOULDBLOCK
1410 sout_stream_id_t *id = data;
1411 unsigned i_caching = id->i_caching;
1415 block_t *out = block_FifoGet( id->p_fifo );
1416 block_cleanup_push (out);
1420 { /* FIXME: this is awfully inefficient */
1421 size_t len = out->i_buffer;
1422 out = block_Realloc( out, 0, len + 10 );
1423 out->i_buffer = len;
1425 int canc = vlc_savecancel ();
1426 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1427 vlc_restorecancel (canc);
1431 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1432 block_Release( out );
1436 out->i_buffer = len;
1439 mwait (out->i_dts + i_caching);
1444 mwait (out->i_dts + i_caching);
1448 ssize_t len = out->i_buffer;
1449 int canc = vlc_savecancel ();
1451 vlc_mutex_lock( &id->lock_sink );
1452 unsigned deadc = 0; /* How many dead sockets? */
1453 int deadv[id->sinkc]; /* Dead sockets list */
1455 for( int i = 0; i < id->sinkc; i++ )
1458 if( !id->srtp ) /* FIXME: SRTCP support */
1460 SendRTCP( id->sinkv[i].rtcp, out );
1462 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1463 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1464 && net_errno != ENOBUFS && net_errno != ENOMEM )
1467 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1468 &type, &(socklen_t){ sizeof(type) });
1469 if( type == SOCK_DGRAM )
1470 /* ICMP soft error: ignore and retry */
1471 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1473 /* Broken connection */
1474 deadv[deadc++] = id->sinkv[i].rtp_fd;
1477 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1478 vlc_mutex_unlock( &id->lock_sink );
1479 block_Release( out );
1481 for( unsigned i = 0; i < deadc; i++ )
1483 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1484 rtp_del_sink( id, deadv[i] );
1486 vlc_restorecancel (canc);
1492 /* This thread dequeues incoming connections (DCCP streaming) */
1493 static void *rtp_listen_thread( void *data )
1495 sout_stream_id_t *id = data;
1497 assert( id->listen.fd != NULL );
1501 int fd = net_Accept( id->p_stream, id->listen.fd );
1504 int canc = vlc_savecancel( );
1505 rtp_add_sink( id, fd, true, NULL );
1506 vlc_restorecancel( canc );
1513 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1515 rtp_sink_t sink = { fd, NULL };
1516 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1518 if( sink.rtcp == NULL )
1519 msg_Err( id->p_stream, "RTCP failed!" );
1521 vlc_mutex_lock( &id->lock_sink );
1522 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1524 *seq = id->i_seq_sent_next;
1525 vlc_mutex_unlock( &id->lock_sink );
1529 void rtp_del_sink( sout_stream_id_t *id, int fd )
1531 rtp_sink_t sink = { fd, NULL };
1533 /* NOTE: must be safe to use if fd is not included */
1534 vlc_mutex_lock( &id->lock_sink );
1535 for( int i = 0; i < id->sinkc; i++ )
1537 if (id->sinkv[i].rtp_fd == fd)
1539 sink = id->sinkv[i];
1540 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1544 vlc_mutex_unlock( &id->lock_sink );
1546 CloseRTCP( sink.rtcp );
1547 net_Close( sink.rtp_fd );
1550 uint16_t rtp_get_seq( sout_stream_id_t *id )
1552 /* This will return values for the next packet. */
1555 vlc_mutex_lock( &id->lock_sink );
1556 seq = id->i_seq_sent_next;
1557 vlc_mutex_unlock( &id->lock_sink );
1562 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1563 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1564 * random (although we use the same reference for all the ES as a
1565 * feature). In the VoD case, this function is called independently
1566 * from several parts of the code, so we need to always return the same
1568 static int64_t rtp_init_ts( const vod_media_t *p_media,
1569 const char *psz_vod_session )
1571 if (p_media == NULL || psz_vod_session == NULL)
1575 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1576 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1577 i_ts_init ^= (uintptr_t)p_media;
1578 /* Limit the timestamp to 48 bytes, this is enough and allows us
1579 * to stay away from overflows */
1580 i_ts_init &= 0xFFFFFFFFFFFF;
1584 /* Return a timestamp corresponding to packets being sent now, and that
1585 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1586 * Also return the NPT corresponding to this timestamp. If the stream
1587 * output is not started, the initial timestamp that will be used with
1588 * the first packets for NPT=0 is returned instead. */
1589 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1590 const vod_media_t *p_media, const char *psz_vod_session,
1597 p_stream = id->p_stream;
1599 if (p_stream == NULL)
1600 return rtp_init_ts(p_media, psz_vod_session);
1602 sout_stream_sys_t *p_sys = p_stream->p_sys;
1604 vlc_mutex_lock( &p_sys->lock_ts );
1605 i_npt_zero = p_sys->i_npt_zero;
1606 vlc_mutex_unlock( &p_sys->lock_ts );
1608 if( i_npt_zero == VLC_TS_INVALID )
1609 return p_sys->i_pts_zero;
1611 mtime_t now = mdate();
1612 if( now < i_npt_zero )
1613 return p_sys->i_pts_zero;
1615 int64_t npt = now - i_npt_zero;
1619 return p_sys->i_pts_zero + npt;
1622 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1623 int b_marker, int64_t i_pts )
1625 if( !id->b_ts_init )
1627 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1628 vlc_mutex_lock( &p_sys->lock_ts );
1629 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1631 /* This is the first packet of any ES. We initialize the
1632 * NPT=0 time reference, and the offset to match the
1633 * arbitrary PTS reference. */
1634 p_sys->i_npt_zero = i_pts + id->i_caching;
1635 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1637 vlc_mutex_unlock( &p_sys->lock_ts );
1639 /* And in any case this is the first packet of this ES, so we
1640 * initialize the offset for this ES. */
1641 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1642 p_sys->i_pts_offset );
1643 id->b_ts_init = true;
1646 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1649 out->p_buffer[0] = 0x80;
1650 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1651 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1652 out->p_buffer[3] = ( id->i_sequence )&0xff;
1653 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1654 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1655 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1656 out->p_buffer[7] = ( i_timestamp )&0xff;
1658 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1664 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1666 block_FifoPut( id->p_fifo, out );
1670 * @return configured max RTP payload size (including payload type-specific
1671 * headers, excluding RTP and transport headers)
1673 size_t rtp_mtu (const sout_stream_id_t *id)
1675 return id->i_mtu - 12;
1678 /*****************************************************************************
1680 *****************************************************************************/
1682 /** Add an ES to a non-RTP muxed stream */
1683 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1685 sout_input_t *p_input;
1686 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1687 assert( p_mux != NULL );
1689 p_input = sout_MuxAddStream( p_mux, p_fmt );
1690 if( p_input == NULL )
1692 msg_Err( p_stream, "cannot add this stream to the muxer" );
1696 return (sout_stream_id_t *)p_input;
1700 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1703 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1704 assert( p_mux != NULL );
1706 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1711 /** Remove an ES from a non-RTP muxed stream */
1712 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1714 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1715 assert( p_mux != NULL );
1717 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1722 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1723 const block_t *p_buffer )
1725 sout_stream_sys_t *p_sys = p_stream->p_sys;
1726 sout_stream_id_t *id = p_sys->es[0];
1728 int64_t i_dts = p_buffer->i_dts;
1730 uint8_t *p_data = p_buffer->p_buffer;
1731 size_t i_data = p_buffer->i_buffer;
1732 size_t i_max = id->i_mtu - 12;
1734 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1740 /* output complete packet */
1741 if( p_sys->packet &&
1742 p_sys->packet->i_buffer + i_data > i_max )
1744 rtp_packetize_send( id, p_sys->packet );
1745 p_sys->packet = NULL;
1748 if( p_sys->packet == NULL )
1750 /* allocate a new packet */
1751 p_sys->packet = block_New( p_stream, id->i_mtu );
1752 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1753 p_sys->packet->i_dts = i_dts;
1754 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1755 i_dts += p_sys->packet->i_length;
1758 i_size = __MIN( i_data,
1759 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1761 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1764 p_sys->packet->i_buffer += i_size;
1773 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1776 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1782 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1784 p_next = p_buffer->p_next;
1785 block_Release( p_buffer );
1793 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1795 sout_access_out_t *p_grab;
1797 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1798 if( p_grab == NULL )
1801 p_grab->p_module = NULL;
1802 p_grab->psz_access = strdup( "grab" );
1803 p_grab->p_cfg = NULL;
1804 p_grab->psz_path = strdup( "" );
1805 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1806 p_grab->pf_seek = NULL;
1807 p_grab->pf_write = AccessOutGrabberWrite;