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. "\
148 "This must be a 32-character-long hexadecimal string.")
150 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
151 #define SRTP_SALT_LONGTEXT N_( \
152 "Secure RTP requires a (non-secret) master salt value. " \
153 "This must be a 28-character-long hexadecimal string.")
155 static const char *const ppsz_protos[] = {
156 "dccp", "sctp", "tcp", "udp", "udplite",
159 static const char *const ppsz_protocols[] = {
160 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
163 #define RFC3016_TEXT N_("MP4A LATM")
164 #define RFC3016_LONGTEXT N_( \
165 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
167 #define RTSP_HOST_TEXT N_( "RTSP host address" )
168 #define RTSP_HOST_LONGTEXT N_( \
169 "This defines the address, port and path the RTSP VOD server will listen " \
170 "on.\nSyntax is address:port/path. The default is to listen on all "\
171 "interfaces (address 0.0.0.0), on port 554, with no path.\nTo listen " \
172 "only on the local interface, use \"localhost\" as address." )
174 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
175 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
176 "not receiving any RTSP request for this long. Setting it to a " \
177 "negative value or zero disables timeouts. The default is 60 (one " \
180 #define RTSP_USER_TEXT N_("Username")
181 #define RTSP_USER_LONGTEXT N_("User name that will be " \
182 "requested to access the stream." )
183 #define RTSP_PASS_TEXT N_("Password")
184 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
185 "requested to access the stream." )
187 static int Open ( vlc_object_t * );
188 static void Close( vlc_object_t * );
190 #define SOUT_CFG_PREFIX "sout-rtp-"
191 #define MAX_EMPTY_BLOCKS 200
194 set_shortname( N_("RTP"))
195 set_description( N_("RTP stream output") )
196 set_capability( "sout stream", 0 )
197 add_shortcut( "rtp", "vod" )
198 set_category( CAT_SOUT )
199 set_subcategory( SUBCAT_SOUT_STREAM )
201 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
202 DEST_LONGTEXT, true )
203 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
205 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
207 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
210 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
211 NAME_LONGTEXT, true )
212 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
213 DESC_LONGTEXT, true )
214 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
216 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
217 EMAIL_LONGTEXT, true )
218 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
219 PHONE_LONGTEXT, true )
221 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
222 PROTO_LONGTEXT, false )
223 change_string_list( ppsz_protos, ppsz_protocols, NULL )
224 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
225 PORT_LONGTEXT, true )
226 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
227 PORT_AUDIO_LONGTEXT, true )
228 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
229 PORT_VIDEO_LONGTEXT, true )
231 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
233 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
234 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
235 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
236 CACHING_TEXT, CACHING_LONGTEXT, true )
239 add_string( SOUT_CFG_PREFIX "key", "",
240 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
241 add_string( SOUT_CFG_PREFIX "salt", "",
242 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
245 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
246 RFC3016_LONGTEXT, false )
248 set_callbacks( Open, Close )
251 set_shortname( N_("RTSP VoD" ) )
252 set_description( N_("RTSP VoD server") )
253 set_category( CAT_SOUT )
254 set_subcategory( SUBCAT_SOUT_VOD )
255 set_capability( "vod server", 10 )
256 set_callbacks( OpenVoD, CloseVoD )
257 add_shortcut( "rtsp" )
258 add_string ( "rtsp-host", NULL, RTSP_HOST_TEXT,
259 RTSP_HOST_LONGTEXT, true )
260 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
261 RTSP_TIMEOUT_LONGTEXT, true )
262 add_string( "sout-rtsp-user", "",
263 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
264 add_password( "sout-rtsp-pwd", "",
265 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
269 /*****************************************************************************
270 * Exported prototypes
271 *****************************************************************************/
272 static const char *const ppsz_sout_options[] = {
273 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
274 "sap", "description", "url", "email", "phone",
275 "proto", "rtcp-mux", "caching", "key", "salt",
279 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
280 static int Del ( sout_stream_t *, sout_stream_id_t * );
281 static int Send( sout_stream_t *, sout_stream_id_t *,
283 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
284 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
285 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
288 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
289 static void* ThreadSend( void * );
290 static void *rtp_listen_thread( void * );
292 static void SDPHandleUrl( sout_stream_t *, const char * );
294 static int SapSetup( sout_stream_t *p_stream );
295 static int FileSetup( sout_stream_t *p_stream );
296 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
298 static int64_t rtp_init_ts( const vod_media_t *p_media,
299 const char *psz_vod_session );
301 struct sout_stream_sys_t
305 vlc_mutex_t lock_sdp;
312 session_descriptor_t *p_session;
315 httpd_host_t *p_httpd_host;
316 httpd_file_t *p_httpd_file;
321 /* RTSP NPT and timestamp computations */
322 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
323 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
324 int64_t i_pts_offset; /* matches actual PTS to prediction */
328 char *psz_destination;
330 uint16_t i_port_audio;
331 uint16_t i_port_video;
337 vod_media_t *p_vod_media;
338 char *psz_vod_session;
340 /* in case we do TS/PS over rtp */
342 sout_access_out_t *p_grab;
348 sout_stream_id_t **es;
351 typedef struct rtp_sink_t
357 struct sout_stream_id_t
359 sout_stream_t *p_stream;
364 uint32_t i_ts_offset;
368 uint16_t i_seq_sent_next;
371 rtp_format_t rtp_fmt;
374 /* Packetizer specific fields */
377 srtp_session_t *srtp;
382 vlc_mutex_t lock_sink;
385 rtsp_stream_id_t *rtsp_id;
391 block_fifo_t *p_fifo;
395 /*****************************************************************************
397 *****************************************************************************/
398 static int Open( vlc_object_t *p_this )
400 sout_stream_t *p_stream = (sout_stream_t*)p_this;
401 sout_instance_t *p_sout = p_stream->p_sout;
402 sout_stream_sys_t *p_sys = NULL;
403 config_chain_t *p_cfg = NULL;
407 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
408 ppsz_sout_options, p_stream->p_cfg );
410 p_sys = malloc( sizeof( sout_stream_sys_t ) );
414 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
416 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
417 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
418 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
419 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
421 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
423 msg_Err( p_stream, "audio and video RTP port must be distinct" );
424 free( p_sys->psz_destination );
429 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
431 if( !strcmp( p_cfg->psz_name, "sdp" )
432 && ( p_cfg->psz_value != NULL )
433 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
441 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
444 if( !strncasecmp( psz, "rtsp:", 5 ) )
450 /* Transport protocol */
451 p_sys->proto = IPPROTO_UDP;
452 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
454 if ((psz == NULL) || !strcasecmp (psz, "udp"))
455 (void)0; /* default */
457 if (!strcasecmp (psz, "dccp"))
459 p_sys->proto = IPPROTO_DCCP;
460 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
464 if (!strcasecmp (psz, "sctp"))
466 p_sys->proto = IPPROTO_TCP;
467 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
472 if (!strcasecmp (psz, "tcp"))
474 p_sys->proto = IPPROTO_TCP;
475 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
479 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
480 p_sys->proto = IPPROTO_UDPLITE;
482 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
485 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
487 p_sys->p_vod_media = NULL;
488 p_sys->psz_vod_session = NULL;
490 if (! strcmp(p_stream->psz_name, "vod"))
492 /* The VLM stops all instances before deleting a media, so this
493 * reference will remain valid during the lifetime of the rtp
495 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
497 if (p_sys->p_vod_media != NULL)
499 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
500 if (p_sys->psz_vod_session == NULL)
502 msg_Err(p_stream, "missing VoD session");
507 const char *mux = vod_get_mux(p_sys->p_vod_media);
508 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
512 if( p_sys->psz_destination == NULL && !b_rtsp
513 && p_sys->p_vod_media == NULL )
515 msg_Err( p_stream, "missing destination and not in RTSP mode" );
520 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
523 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
524 var_SetInteger( p_stream, "ttl", i_ttl );
527 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
529 /* NPT=0 time will be determined when we packetize the first packet
530 * (of any ES). But we want to be able to report rtptime in RTSP
531 * without waiting (and already did in the VoD case). So until then,
532 * we use an arbitrary reference PTS for timestamp computations, and
533 * then actual PTS will catch up using offsets. */
534 p_sys->i_npt_zero = VLC_TS_INVALID;
535 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
536 p_sys->psz_vod_session);
540 p_sys->psz_sdp = NULL;
542 p_sys->b_export_sap = false;
543 p_sys->p_session = NULL;
544 p_sys->psz_sdp_file = NULL;
546 p_sys->p_httpd_host = NULL;
547 p_sys->p_httpd_file = NULL;
549 p_stream->p_sys = p_sys;
551 vlc_mutex_init( &p_sys->lock_sdp );
552 vlc_mutex_init( &p_sys->lock_ts );
553 vlc_mutex_init( &p_sys->lock_es );
555 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
558 /* Check muxer type */
559 if( strncasecmp( psz, "ps", 2 )
560 && strncasecmp( psz, "mpeg1", 5 )
561 && strncasecmp( psz, "ts", 2 ) )
563 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
565 vlc_mutex_destroy( &p_sys->lock_sdp );
566 vlc_mutex_destroy( &p_sys->lock_ts );
567 vlc_mutex_destroy( &p_sys->lock_es );
568 free( p_sys->psz_vod_session );
569 free( p_sys->psz_destination );
574 p_sys->p_grab = GrabberCreate( p_stream );
575 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
578 if( p_sys->p_mux == NULL )
580 msg_Err( p_stream, "cannot create muxer" );
581 sout_AccessOutDelete( p_sys->p_grab );
582 vlc_mutex_destroy( &p_sys->lock_sdp );
583 vlc_mutex_destroy( &p_sys->lock_ts );
584 vlc_mutex_destroy( &p_sys->lock_es );
585 free( p_sys->psz_vod_session );
586 free( p_sys->psz_destination );
591 p_sys->packet = NULL;
593 p_stream->pf_add = MuxAdd;
594 p_stream->pf_del = MuxDel;
595 p_stream->pf_send = MuxSend;
600 p_sys->p_grab = NULL;
602 p_stream->pf_add = Add;
603 p_stream->pf_del = Del;
604 p_stream->pf_send = Send;
607 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
608 SDPHandleUrl( p_stream, "sap" );
610 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
613 config_chain_t *p_cfg;
615 SDPHandleUrl( p_stream, psz );
617 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
619 if( !strcmp( p_cfg->psz_name, "sdp" ) )
621 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
624 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
625 if( !strcmp( p_cfg->psz_value, psz ) )
628 SDPHandleUrl( p_stream, p_cfg->psz_value );
634 /* update p_sout->i_out_pace_nocontrol */
635 p_stream->p_sout->i_out_pace_nocontrol++;
637 if( p_sys->p_mux != NULL )
639 sout_stream_id_t *id = Add( p_stream, NULL );
650 /*****************************************************************************
652 *****************************************************************************/
653 static void Close( vlc_object_t * p_this )
655 sout_stream_t *p_stream = (sout_stream_t*)p_this;
656 sout_stream_sys_t *p_sys = p_stream->p_sys;
658 /* update p_sout->i_out_pace_nocontrol */
659 p_stream->p_sout->i_out_pace_nocontrol--;
663 assert( p_sys->i_es <= 1 );
665 sout_MuxDelete( p_sys->p_mux );
666 if ( p_sys->i_es > 0 )
667 Del( p_stream, p_sys->es[0] );
668 sout_AccessOutDelete( p_sys->p_grab );
672 block_Release( p_sys->packet );
676 if( p_sys->rtsp != NULL )
677 RtspUnsetup( p_sys->rtsp );
679 vlc_mutex_destroy( &p_sys->lock_sdp );
680 vlc_mutex_destroy( &p_sys->lock_ts );
681 vlc_mutex_destroy( &p_sys->lock_es );
683 if( p_sys->p_httpd_file )
684 httpd_FileDelete( p_sys->p_httpd_file );
686 if( p_sys->p_httpd_host )
687 httpd_HostDelete( p_sys->p_httpd_host );
689 free( p_sys->psz_sdp );
691 if( p_sys->psz_sdp_file != NULL )
694 unlink( p_sys->psz_sdp_file );
696 free( p_sys->psz_sdp_file );
698 free( p_sys->psz_vod_session );
699 free( p_sys->psz_destination );
703 /*****************************************************************************
705 *****************************************************************************/
706 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
708 sout_stream_sys_t *p_sys = p_stream->p_sys;
711 vlc_UrlParse( &url, psz_url, 0 );
712 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
714 if( p_sys->p_httpd_file )
716 msg_Err( p_stream, "you can use sdp=http:// only once" );
720 if( HttpSetup( p_stream, &url ) )
722 msg_Err( p_stream, "cannot export SDP as HTTP" );
725 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
727 if( p_sys->rtsp != NULL )
729 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
733 if( url.psz_host != NULL && *url.psz_host )
735 /* msg_Err( p_stream, "\"%s\" RTSP host ignored", url.psz_host );
736 msg_Info( p_stream, "Pass --rtsp-host=%s on the command line "
737 "instead.", url.psz_host ); */
739 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
740 var_SetString( p_stream, "rtsp-host", url.psz_host );
742 /* if( url.i_port != 0 )
744 msg_Err( p_stream, "\"%u\" RTSP port ignored", url.i_port );
745 msg_Info( p_stream, "Pass --rtsp-port=%u on the command line "
746 "instead.", url.i_port );
749 if( url.i_port <= 0 ) url.i_port = 554;
750 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
751 var_SetInteger( p_stream, "rtsp-port", url.i_port );
753 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
754 if( p_sys->rtsp == NULL )
755 msg_Err( p_stream, "cannot export SDP as RTSP" );
757 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
758 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
760 p_sys->b_export_sap = true;
761 SapSetup( p_stream );
763 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
765 if( p_sys->psz_sdp_file != NULL )
767 msg_Err( p_stream, "you can use sdp=file:// only once" );
770 p_sys->psz_sdp_file = make_path( psz_url );
771 if( p_sys->psz_sdp_file == NULL )
773 FileSetup( p_stream );
777 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
782 vlc_UrlClean( &url );
785 /*****************************************************************************
787 *****************************************************************************/
789 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
791 sout_stream_sys_t *p_sys = p_stream->p_sys;
792 char *psz_sdp = NULL;
793 struct sockaddr_storage dst;
797 * When we have a fixed destination (typically when we do multicast),
798 * we need to put the actual port numbers in the SDP.
799 * When there is no fixed destination, we only support RTSP unicast
800 * on-demand setup, so we should rather let the clients decide which ports
802 * When there is both a fixed destination and RTSP unicast, we need to
803 * put port numbers used by the fixed destination, otherwise the SDP would
804 * become totally incorrect for multicast use. It should be noted that
805 * port numbers from SDP with RTSP are only "recommendation" from the
806 * server to the clients (per RFC2326), so only broken clients will fail
807 * to handle this properly. There is no solution but to use two differents
808 * output chain with two different RTSP URLs if you need to handle this
813 vlc_mutex_lock( &p_sys->lock_es );
814 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
815 goto out; /* hmm... */
817 if( p_sys->psz_destination != NULL )
821 /* Oh boy, this is really ugly! */
822 dstlen = sizeof( dst );
823 if( p_sys->es[0]->listen.fd != NULL )
824 getsockname( p_sys->es[0]->listen.fd[0],
825 (struct sockaddr *)&dst, &dstlen );
827 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
828 (struct sockaddr *)&dst, &dstlen );
834 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
835 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
836 && rtsp_url[7] == '[';
838 /* Dummy destination address for RTSP */
839 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
840 : sizeof( struct sockaddr_in );
841 memset (&dst, 0, dstlen);
842 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
848 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
849 NULL, 0, (struct sockaddr *)&dst, dstlen );
850 if( psz_sdp == NULL )
853 /* TODO: a=source-filter */
854 if( p_sys->rtcp_mux )
855 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
857 if( rtsp_url != NULL )
858 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
860 const char *proto = "RTP/AVP"; /* protocol */
861 if( rtsp_url == NULL )
863 switch( p_sys->proto )
868 proto = "TCP/RTP/AVP";
871 proto = "DCCP/RTP/AVP";
873 case IPPROTO_UDPLITE:
878 for( i = 0; i < p_sys->i_es; i++ )
880 sout_stream_id_t *id = p_sys->es[i];
881 rtp_format_t *rtp_fmt = &id->rtp_fmt;
882 const char *mime_major; /* major MIME type */
884 switch( rtp_fmt->cat )
887 mime_major = "video";
890 mime_major = "audio";
899 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
900 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
901 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
904 /* cf RFC4566 §5.14 */
905 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
906 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
908 if( rtsp_url != NULL )
910 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
911 if( track_url != NULL )
913 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
919 if( id->listen.fd != NULL )
920 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
921 if( p_sys->proto == IPPROTO_DCCP )
922 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
924 toupper( (unsigned char)mime_major[0] ) );
928 vlc_mutex_unlock( &p_sys->lock_es );
932 /*****************************************************************************
934 *****************************************************************************/
937 * Shrink the MTU down to a fixed packetization time (for audio).
940 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
942 /* Samples per second */
943 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
944 bytes *= id->rtp_fmt.channels;
947 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
948 id->i_mtu = 12 + spl;
949 else /* MTU is too small for ptime, align to a sample boundary */
950 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
953 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
955 /* NOTE: this plays nice with offsets because the calculations are
957 return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
960 /** Add an ES as a new RTP stream */
961 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
963 /* NOTE: As a special case, if we use a non-RTP
964 * mux (TS/PS), then p_fmt is NULL. */
965 sout_stream_sys_t *p_sys = p_stream->p_sys;
968 sout_stream_id_t *id = malloc( sizeof( *id ) );
969 if( unlikely(id == NULL) )
971 id->p_stream = p_stream;
973 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
974 if( id->i_mtu <= 12 + 16 )
975 id->i_mtu = 576 - 20 - 8; /* pessimistic */
976 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
981 vlc_mutex_init( &id->lock_sink );
986 id->listen.fd = NULL;
988 id->b_first_packet = true;
990 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
992 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
993 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
997 if (p_sys->p_vod_media != NULL)
999 id->rtp_fmt.ptname = NULL;
1001 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
1002 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
1003 &ssrc, &id->i_seq_sent_next);
1004 if (val == VLC_SUCCESS)
1006 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1007 /* This is ugly, but id->i_seq_sent_next needs to be
1008 * initialized inside vod_init_id() to avoid race
1010 id->i_sequence = id->i_seq_sent_next;
1012 /* vod_init_id() may fail either because the ES wasn't found in
1013 * the VoD media, or because the RTSP session is gone. In the
1014 * former case, id->rtp_fmt was left untouched. */
1015 format = (id->rtp_fmt.ptname != NULL);
1020 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1021 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1022 if (p_fmt == NULL && psz == NULL)
1024 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1026 if (val != VLC_SUCCESS)
1031 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1035 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1036 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1037 if (id->srtp == NULL)
1043 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1044 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1049 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1052 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1056 id->i_seq_sent_next = id->i_sequence;
1059 if( p_sys->psz_destination != NULL )
1061 /* Choose the port */
1062 uint16_t i_port = 0;
1066 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1067 i_port = p_sys->i_port_audio;
1069 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1070 i_port = p_sys->i_port_video;
1072 /* We do not need the ES lock (p_sys->lock_es) here, because
1073 * this is the only one thread that can *modify* the ES table.
1074 * The ES lock protects the other threads from our modifications
1075 * (TAB_APPEND, TAB_REMOVE). */
1076 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1077 if (i_port == p_sys->es[i]->i_port)
1078 i_port = 0; /* Port already in use! */
1079 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1083 msg_Err (p_stream, "too many RTP elementary streams");
1087 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1088 if (p == p_sys->es[i]->i_port)
1092 id->i_port = i_port;
1094 int type = SOCK_STREAM;
1096 switch( p_sys->proto )
1102 switch (id->rtp_fmt.cat)
1104 case VIDEO_ES: code = "RTPV"; break;
1105 case AUDIO_ES: code = "RTPARTPV"; break;
1106 case SPU_ES: code = "RTPTRTPV"; break;
1107 default: code = "RTPORTPV"; break;
1109 var_SetString (p_stream, "dccp-service", code);
1111 } /* fall through */
1114 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1115 p_sys->psz_destination, i_port,
1116 type, p_sys->proto );
1117 if( id->listen.fd == NULL )
1119 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1122 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1123 VLC_THREAD_PRIORITY_LOW ) )
1125 net_ListenClose( id->listen.fd );
1126 id->listen.fd = NULL;
1133 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1134 i_port, -1, p_sys->proto );
1137 msg_Err( p_stream, "cannot create RTP socket" );
1140 /* Ignore any unexpected incoming packet (including RTCP-RR
1141 * packets in case of rtcp-mux) */
1142 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1144 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1145 /* FIXME: test if this is multicast */
1152 switch( p_fmt->i_codec )
1154 case VLC_CODEC_MULAW:
1155 case VLC_CODEC_ALAW:
1157 rtp_set_ptime (id, 20, 1);
1159 case VLC_CODEC_S16B:
1160 case VLC_CODEC_S16L:
1161 rtp_set_ptime (id, 20, 2);
1167 #if 0 /* No payload formats sets this at the moment */
1170 cscov += 8 /* UDP */ + 12 /* RTP */;
1172 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1175 vlc_mutex_lock( &p_sys->lock_ts );
1176 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1177 vlc_mutex_unlock( &p_sys->lock_ts );
1179 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1180 p_sys->i_pts_offset );
1182 if( p_sys->rtsp != NULL )
1183 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1184 id->rtp_fmt.clock_rate, mcast_fd );
1186 id->p_fifo = block_FifoNew();
1187 if( unlikely(id->p_fifo == NULL) )
1189 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1191 block_FifoRelease( id->p_fifo );
1196 /* Update p_sys context */
1197 vlc_mutex_lock( &p_sys->lock_es );
1198 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1199 vlc_mutex_unlock( &p_sys->lock_es );
1201 psz_sdp = SDPGenerate( p_stream, NULL );
1203 vlc_mutex_lock( &p_sys->lock_sdp );
1204 free( p_sys->psz_sdp );
1205 p_sys->psz_sdp = psz_sdp;
1206 vlc_mutex_unlock( &p_sys->lock_sdp );
1208 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1210 /* Update SDP (sap/file) */
1211 if( p_sys->b_export_sap ) SapSetup( p_stream );
1212 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1217 Del( p_stream, id );
1221 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1223 sout_stream_sys_t *p_sys = p_stream->p_sys;
1225 vlc_mutex_lock( &p_sys->lock_es );
1226 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1227 vlc_mutex_unlock( &p_sys->lock_es );
1229 if( likely(id->p_fifo != NULL) )
1231 vlc_cancel( id->thread );
1232 vlc_join( id->thread, NULL );
1233 block_FifoRelease( id->p_fifo );
1236 free( id->rtp_fmt.fmtp );
1238 if (p_sys->p_vod_media != NULL)
1239 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1241 RtspDelId( p_sys->rtsp, id->rtsp_id );
1242 if( id->listen.fd != NULL )
1244 vlc_cancel( id->listen.thread );
1245 vlc_join( id->listen.thread, NULL );
1246 net_ListenClose( id->listen.fd );
1248 /* Delete remaining sinks (incoming connections or explicit
1250 while( id->sinkc > 0 )
1251 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1253 if( id->srtp != NULL )
1254 srtp_destroy( id->srtp );
1257 vlc_mutex_destroy( &id->lock_sink );
1259 /* Update SDP (sap/file) */
1260 if( p_sys->b_export_sap ) SapSetup( p_stream );
1261 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1267 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1272 assert( p_stream->p_sys->p_mux == NULL );
1275 while( p_buffer != NULL )
1277 p_next = p_buffer->p_next;
1279 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1280 * as the first packet of the stream */
1281 if (id->b_first_packet)
1283 id->b_first_packet = false;
1284 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1285 !strcmp(id->rtp_fmt.ptname, "theora"))
1286 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1290 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1293 block_Release( p_buffer );
1299 /****************************************************************************
1301 ****************************************************************************/
1302 static int SapSetup( sout_stream_t *p_stream )
1304 sout_stream_sys_t *p_sys = p_stream->p_sys;
1305 sout_instance_t *p_sout = p_stream->p_sout;
1307 /* Remove the previous session */
1308 if( p_sys->p_session != NULL)
1310 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1311 p_sys->p_session = NULL;
1314 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1315 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1317 p_sys->psz_destination );
1322 /****************************************************************************
1324 ****************************************************************************/
1325 static int FileSetup( sout_stream_t *p_stream )
1327 sout_stream_sys_t *p_sys = p_stream->p_sys;
1330 if( p_sys->psz_sdp == NULL )
1331 return VLC_EGENERIC; /* too early */
1333 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1335 msg_Err( p_stream, "cannot open file '%s' (%m)",
1336 p_sys->psz_sdp_file );
1337 return VLC_EGENERIC;
1340 fputs( p_sys->psz_sdp, f );
1346 /****************************************************************************
1348 ****************************************************************************/
1349 static int HttpCallback( httpd_file_sys_t *p_args,
1350 httpd_file_t *, uint8_t *p_request,
1351 uint8_t **pp_data, int *pi_data );
1353 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1355 sout_stream_sys_t *p_sys = p_stream->p_sys;
1357 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1358 if( p_sys->p_httpd_host )
1360 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1361 url->psz_path ? url->psz_path : "/",
1364 HttpCallback, (void*)p_sys );
1366 if( p_sys->p_httpd_file == NULL )
1368 return VLC_EGENERIC;
1373 static int HttpCallback( httpd_file_sys_t *p_args,
1374 httpd_file_t *f, uint8_t *p_request,
1375 uint8_t **pp_data, int *pi_data )
1377 VLC_UNUSED(f); VLC_UNUSED(p_request);
1378 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1380 vlc_mutex_lock( &p_sys->lock_sdp );
1381 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1383 *pi_data = strlen( p_sys->psz_sdp );
1384 *pp_data = malloc( *pi_data );
1385 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1392 vlc_mutex_unlock( &p_sys->lock_sdp );
1397 /****************************************************************************
1399 ****************************************************************************/
1400 static void* ThreadSend( void *data )
1403 # define ECONNREFUSED WSAECONNREFUSED
1404 # define ENOPROTOOPT WSAENOPROTOOPT
1405 # define EHOSTUNREACH WSAEHOSTUNREACH
1406 # define ENETUNREACH WSAENETUNREACH
1407 # define ENETDOWN WSAENETDOWN
1408 # define ENOBUFS WSAENOBUFS
1409 # define EAGAIN WSAEWOULDBLOCK
1410 # define EWOULDBLOCK WSAEWOULDBLOCK
1412 sout_stream_id_t *id = data;
1413 unsigned i_caching = id->i_caching;
1417 block_t *out = block_FifoGet( id->p_fifo );
1418 block_cleanup_push (out);
1422 { /* FIXME: this is awfully inefficient */
1423 size_t len = out->i_buffer;
1424 out = block_Realloc( out, 0, len + 10 );
1425 out->i_buffer = len;
1427 int canc = vlc_savecancel ();
1428 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1429 vlc_restorecancel (canc);
1433 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1434 block_Release( out );
1438 out->i_buffer = len;
1441 mwait (out->i_dts + i_caching);
1446 mwait (out->i_dts + i_caching);
1450 ssize_t len = out->i_buffer;
1451 int canc = vlc_savecancel ();
1453 vlc_mutex_lock( &id->lock_sink );
1454 unsigned deadc = 0; /* How many dead sockets? */
1455 int deadv[id->sinkc]; /* Dead sockets list */
1457 for( int i = 0; i < id->sinkc; i++ )
1460 if( !id->srtp ) /* FIXME: SRTCP support */
1462 SendRTCP( id->sinkv[i].rtcp, out );
1464 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1465 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1466 && net_errno != ENOBUFS && net_errno != ENOMEM )
1469 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1470 &type, &(socklen_t){ sizeof(type) });
1471 if( type == SOCK_DGRAM )
1472 /* ICMP soft error: ignore and retry */
1473 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1475 /* Broken connection */
1476 deadv[deadc++] = id->sinkv[i].rtp_fd;
1479 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1480 vlc_mutex_unlock( &id->lock_sink );
1481 block_Release( out );
1483 for( unsigned i = 0; i < deadc; i++ )
1485 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1486 rtp_del_sink( id, deadv[i] );
1488 vlc_restorecancel (canc);
1494 /* This thread dequeues incoming connections (DCCP streaming) */
1495 static void *rtp_listen_thread( void *data )
1497 sout_stream_id_t *id = data;
1499 assert( id->listen.fd != NULL );
1503 int fd = net_Accept( id->p_stream, id->listen.fd );
1506 int canc = vlc_savecancel( );
1507 rtp_add_sink( id, fd, true, NULL );
1508 vlc_restorecancel( canc );
1515 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1517 rtp_sink_t sink = { fd, NULL };
1518 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1520 if( sink.rtcp == NULL )
1521 msg_Err( id->p_stream, "RTCP failed!" );
1523 vlc_mutex_lock( &id->lock_sink );
1524 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1526 *seq = id->i_seq_sent_next;
1527 vlc_mutex_unlock( &id->lock_sink );
1531 void rtp_del_sink( sout_stream_id_t *id, int fd )
1533 rtp_sink_t sink = { fd, NULL };
1535 /* NOTE: must be safe to use if fd is not included */
1536 vlc_mutex_lock( &id->lock_sink );
1537 for( int i = 0; i < id->sinkc; i++ )
1539 if (id->sinkv[i].rtp_fd == fd)
1541 sink = id->sinkv[i];
1542 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1546 vlc_mutex_unlock( &id->lock_sink );
1548 CloseRTCP( sink.rtcp );
1549 net_Close( sink.rtp_fd );
1552 uint16_t rtp_get_seq( sout_stream_id_t *id )
1554 /* This will return values for the next packet. */
1557 vlc_mutex_lock( &id->lock_sink );
1558 seq = id->i_seq_sent_next;
1559 vlc_mutex_unlock( &id->lock_sink );
1564 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1565 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1566 * random (although we use the same reference for all the ES as a
1567 * feature). In the VoD case, this function is called independently
1568 * from several parts of the code, so we need to always return the same
1570 static int64_t rtp_init_ts( const vod_media_t *p_media,
1571 const char *psz_vod_session )
1573 if (p_media == NULL || psz_vod_session == NULL)
1577 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1578 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1579 i_ts_init ^= (uintptr_t)p_media;
1580 /* Limit the timestamp to 48 bytes, this is enough and allows us
1581 * to stay away from overflows */
1582 i_ts_init &= 0xFFFFFFFFFFFF;
1586 /* Return a timestamp corresponding to packets being sent now, and that
1587 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1588 * Also return the NPT corresponding to this timestamp. If the stream
1589 * output is not started, the initial timestamp that will be used with
1590 * the first packets for NPT=0 is returned instead. */
1591 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1592 const vod_media_t *p_media, const char *psz_vod_session,
1599 p_stream = id->p_stream;
1601 if (p_stream == NULL)
1602 return rtp_init_ts(p_media, psz_vod_session);
1604 sout_stream_sys_t *p_sys = p_stream->p_sys;
1606 vlc_mutex_lock( &p_sys->lock_ts );
1607 i_npt_zero = p_sys->i_npt_zero;
1608 vlc_mutex_unlock( &p_sys->lock_ts );
1610 if( i_npt_zero == VLC_TS_INVALID )
1611 return p_sys->i_pts_zero;
1613 mtime_t now = mdate();
1614 if( now < i_npt_zero )
1615 return p_sys->i_pts_zero;
1617 int64_t npt = now - i_npt_zero;
1621 return p_sys->i_pts_zero + npt;
1624 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1625 int b_marker, int64_t i_pts )
1627 if( !id->b_ts_init )
1629 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1630 vlc_mutex_lock( &p_sys->lock_ts );
1631 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1633 /* This is the first packet of any ES. We initialize the
1634 * NPT=0 time reference, and the offset to match the
1635 * arbitrary PTS reference. */
1636 p_sys->i_npt_zero = i_pts + id->i_caching;
1637 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1639 vlc_mutex_unlock( &p_sys->lock_ts );
1641 /* And in any case this is the first packet of this ES, so we
1642 * initialize the offset for this ES. */
1643 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1644 p_sys->i_pts_offset );
1645 id->b_ts_init = true;
1648 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1651 out->p_buffer[0] = 0x80;
1652 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1653 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1654 out->p_buffer[3] = ( id->i_sequence )&0xff;
1655 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1656 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1657 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1658 out->p_buffer[7] = ( i_timestamp )&0xff;
1660 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1666 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1668 block_FifoPut( id->p_fifo, out );
1672 * @return configured max RTP payload size (including payload type-specific
1673 * headers, excluding RTP and transport headers)
1675 size_t rtp_mtu (const sout_stream_id_t *id)
1677 return id->i_mtu - 12;
1680 /*****************************************************************************
1682 *****************************************************************************/
1684 /** Add an ES to a non-RTP muxed stream */
1685 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1687 sout_input_t *p_input;
1688 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1689 assert( p_mux != NULL );
1691 p_input = sout_MuxAddStream( p_mux, p_fmt );
1692 if( p_input == NULL )
1694 msg_Err( p_stream, "cannot add this stream to the muxer" );
1698 return (sout_stream_id_t *)p_input;
1702 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1705 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1706 assert( p_mux != NULL );
1708 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1713 /** Remove an ES from a non-RTP muxed stream */
1714 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1716 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1717 assert( p_mux != NULL );
1719 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1724 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1725 const block_t *p_buffer )
1727 sout_stream_sys_t *p_sys = p_stream->p_sys;
1728 sout_stream_id_t *id = p_sys->es[0];
1730 int64_t i_dts = p_buffer->i_dts;
1732 uint8_t *p_data = p_buffer->p_buffer;
1733 size_t i_data = p_buffer->i_buffer;
1734 size_t i_max = id->i_mtu - 12;
1736 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1742 /* output complete packet */
1743 if( p_sys->packet &&
1744 p_sys->packet->i_buffer + i_data > i_max )
1746 rtp_packetize_send( id, p_sys->packet );
1747 p_sys->packet = NULL;
1750 if( p_sys->packet == NULL )
1752 /* allocate a new packet */
1753 p_sys->packet = block_New( p_stream, id->i_mtu );
1754 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1755 p_sys->packet->i_dts = i_dts;
1756 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1757 i_dts += p_sys->packet->i_length;
1760 i_size = __MIN( i_data,
1761 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1763 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1766 p_sys->packet->i_buffer += i_size;
1775 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1778 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1784 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1786 p_next = p_buffer->p_next;
1787 block_Release( p_buffer );
1795 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1797 sout_access_out_t *p_grab;
1799 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1800 if( p_grab == NULL )
1803 p_grab->p_module = NULL;
1804 p_grab->psz_access = strdup( "grab" );
1805 p_grab->p_cfg = NULL;
1806 p_grab->psz_path = strdup( "" );
1807 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1808 p_grab->pf_seek = NULL;
1809 p_grab->pf_write = AccessOutGrabberWrite;