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_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
168 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
169 "not receiving any RTSP request for this long. Setting it to a " \
170 "negative value or zero disables timeouts. The default is 60 (one " \
173 #define RTSP_USER_TEXT N_("Username")
174 #define RTSP_USER_LONGTEXT N_("User name that will be " \
175 "requested to access the stream." )
176 #define RTSP_PASS_TEXT N_("Password")
177 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
178 "requested to access the stream." )
180 static int Open ( vlc_object_t * );
181 static void Close( vlc_object_t * );
183 #define SOUT_CFG_PREFIX "sout-rtp-"
184 #define MAX_EMPTY_BLOCKS 200
187 set_shortname( N_("RTP"))
188 set_description( N_("RTP stream output") )
189 set_capability( "sout stream", 0 )
190 add_shortcut( "rtp", "vod" )
191 set_category( CAT_SOUT )
192 set_subcategory( SUBCAT_SOUT_STREAM )
194 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
195 DEST_LONGTEXT, true )
196 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
198 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
200 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
203 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
204 NAME_LONGTEXT, true )
205 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
206 DESC_LONGTEXT, true )
207 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
209 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
210 EMAIL_LONGTEXT, true )
211 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
212 PHONE_LONGTEXT, true )
214 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
215 PROTO_LONGTEXT, false )
216 change_string_list( ppsz_protos, ppsz_protocols, NULL )
217 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
218 PORT_LONGTEXT, true )
219 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
220 PORT_AUDIO_LONGTEXT, true )
221 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
222 PORT_VIDEO_LONGTEXT, true )
224 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
226 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
227 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
228 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
229 CACHING_TEXT, CACHING_LONGTEXT, true )
232 add_string( SOUT_CFG_PREFIX "key", "",
233 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
234 add_string( SOUT_CFG_PREFIX "salt", "",
235 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
238 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
239 RFC3016_LONGTEXT, false )
241 set_callbacks( Open, Close )
244 set_shortname( N_("RTSP VoD" ) )
245 set_description( N_("RTSP VoD server") )
246 set_category( CAT_SOUT )
247 set_subcategory( SUBCAT_SOUT_VOD )
248 set_capability( "vod server", 10 )
249 set_callbacks( OpenVoD, CloseVoD )
250 add_shortcut( "rtsp" )
251 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
252 RTSP_TIMEOUT_LONGTEXT, true )
253 add_string( "sout-rtsp-user", "",
254 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
255 add_password( "sout-rtsp-pwd", "",
256 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
260 /*****************************************************************************
261 * Exported prototypes
262 *****************************************************************************/
263 static const char *const ppsz_sout_options[] = {
264 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
265 "sap", "description", "url", "email", "phone",
266 "proto", "rtcp-mux", "caching",
273 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
274 static int Del ( sout_stream_t *, sout_stream_id_t * );
275 static int Send( sout_stream_t *, sout_stream_id_t *,
277 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
278 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
279 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
282 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
283 static void* ThreadSend( void * );
284 static void *rtp_listen_thread( void * );
286 static void SDPHandleUrl( sout_stream_t *, const char * );
288 static int SapSetup( sout_stream_t *p_stream );
289 static int FileSetup( sout_stream_t *p_stream );
290 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
292 static int64_t rtp_init_ts( const vod_media_t *p_media,
293 const char *psz_vod_session );
295 struct sout_stream_sys_t
299 vlc_mutex_t lock_sdp;
306 session_descriptor_t *p_session;
309 httpd_host_t *p_httpd_host;
310 httpd_file_t *p_httpd_file;
315 /* RTSP NPT and timestamp computations */
316 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
317 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
318 int64_t i_pts_offset; /* matches actual PTS to prediction */
322 char *psz_destination;
324 uint16_t i_port_audio;
325 uint16_t i_port_video;
331 vod_media_t *p_vod_media;
332 char *psz_vod_session;
334 /* in case we do TS/PS over rtp */
336 sout_access_out_t *p_grab;
342 sout_stream_id_t **es;
345 typedef struct rtp_sink_t
351 struct sout_stream_id_t
353 sout_stream_t *p_stream;
358 uint32_t i_ts_offset;
362 uint16_t i_seq_sent_next;
365 rtp_format_t rtp_fmt;
368 /* Packetizer specific fields */
371 srtp_session_t *srtp;
376 vlc_mutex_t lock_sink;
379 rtsp_stream_id_t *rtsp_id;
385 block_fifo_t *p_fifo;
389 /*****************************************************************************
391 *****************************************************************************/
392 static int Open( vlc_object_t *p_this )
394 sout_stream_t *p_stream = (sout_stream_t*)p_this;
395 sout_instance_t *p_sout = p_stream->p_sout;
396 sout_stream_sys_t *p_sys = NULL;
397 config_chain_t *p_cfg = NULL;
401 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
402 ppsz_sout_options, p_stream->p_cfg );
404 p_sys = malloc( sizeof( sout_stream_sys_t ) );
408 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
410 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
411 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
412 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
413 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
415 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
417 msg_Err( p_stream, "audio and video RTP port must be distinct" );
418 free( p_sys->psz_destination );
423 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
425 if( !strcmp( p_cfg->psz_name, "sdp" )
426 && ( p_cfg->psz_value != NULL )
427 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
435 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
438 if( !strncasecmp( psz, "rtsp:", 5 ) )
444 /* Transport protocol */
445 p_sys->proto = IPPROTO_UDP;
446 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
448 if ((psz == NULL) || !strcasecmp (psz, "udp"))
449 (void)0; /* default */
451 if (!strcasecmp (psz, "dccp"))
453 p_sys->proto = IPPROTO_DCCP;
454 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
458 if (!strcasecmp (psz, "sctp"))
460 p_sys->proto = IPPROTO_TCP;
461 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
466 if (!strcasecmp (psz, "tcp"))
468 p_sys->proto = IPPROTO_TCP;
469 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
473 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
474 p_sys->proto = IPPROTO_UDPLITE;
476 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
479 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
481 p_sys->p_vod_media = NULL;
482 p_sys->psz_vod_session = NULL;
484 if (! strcmp(p_stream->psz_name, "vod"))
486 /* The VLM stops all instances before deleting a media, so this
487 * reference will remain valid during the lifetime of the rtp
489 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
491 if (p_sys->p_vod_media != NULL)
493 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
494 if (p_sys->psz_vod_session == NULL)
496 msg_Err(p_stream, "missing VoD session");
501 const char *mux = vod_get_mux(p_sys->p_vod_media);
502 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
506 if( p_sys->psz_destination == NULL && !b_rtsp
507 && p_sys->p_vod_media == NULL )
509 msg_Err( p_stream, "missing destination and not in RTSP mode" );
514 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
517 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
518 var_SetInteger( p_stream, "ttl", i_ttl );
521 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
523 /* NPT=0 time will be determined when we packetize the first packet
524 * (of any ES). But we want to be able to report rtptime in RTSP
525 * without waiting (and already did in the VoD case). So until then,
526 * we use an arbitrary reference PTS for timestamp computations, and
527 * then actual PTS will catch up using offsets. */
528 p_sys->i_npt_zero = VLC_TS_INVALID;
529 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
530 p_sys->psz_vod_session);
534 p_sys->psz_sdp = NULL;
536 p_sys->b_export_sap = false;
537 p_sys->p_session = NULL;
538 p_sys->psz_sdp_file = NULL;
540 p_sys->p_httpd_host = NULL;
541 p_sys->p_httpd_file = NULL;
543 p_stream->p_sys = p_sys;
545 vlc_mutex_init( &p_sys->lock_sdp );
546 vlc_mutex_init( &p_sys->lock_ts );
547 vlc_mutex_init( &p_sys->lock_es );
549 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
552 /* Check muxer type */
553 if( strncasecmp( psz, "ps", 2 )
554 && strncasecmp( psz, "mpeg1", 5 )
555 && strncasecmp( psz, "ts", 2 ) )
557 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
559 vlc_mutex_destroy( &p_sys->lock_sdp );
560 vlc_mutex_destroy( &p_sys->lock_ts );
561 vlc_mutex_destroy( &p_sys->lock_es );
562 free( p_sys->psz_vod_session );
563 free( p_sys->psz_destination );
568 p_sys->p_grab = GrabberCreate( p_stream );
569 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
572 if( p_sys->p_mux == NULL )
574 msg_Err( p_stream, "cannot create muxer" );
575 sout_AccessOutDelete( p_sys->p_grab );
576 vlc_mutex_destroy( &p_sys->lock_sdp );
577 vlc_mutex_destroy( &p_sys->lock_ts );
578 vlc_mutex_destroy( &p_sys->lock_es );
579 free( p_sys->psz_vod_session );
580 free( p_sys->psz_destination );
585 p_sys->packet = NULL;
587 p_stream->pf_add = MuxAdd;
588 p_stream->pf_del = MuxDel;
589 p_stream->pf_send = MuxSend;
594 p_sys->p_grab = NULL;
596 p_stream->pf_add = Add;
597 p_stream->pf_del = Del;
598 p_stream->pf_send = Send;
601 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
602 SDPHandleUrl( p_stream, "sap" );
604 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
607 config_chain_t *p_cfg;
609 SDPHandleUrl( p_stream, psz );
611 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
613 if( !strcmp( p_cfg->psz_name, "sdp" ) )
615 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
618 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
619 if( !strcmp( p_cfg->psz_value, psz ) )
622 SDPHandleUrl( p_stream, p_cfg->psz_value );
628 /* update p_sout->i_out_pace_nocontrol */
629 p_stream->p_sout->i_out_pace_nocontrol++;
631 if( p_sys->p_mux != NULL )
633 sout_stream_id_t *id = Add( p_stream, NULL );
644 /*****************************************************************************
646 *****************************************************************************/
647 static void Close( vlc_object_t * p_this )
649 sout_stream_t *p_stream = (sout_stream_t*)p_this;
650 sout_stream_sys_t *p_sys = p_stream->p_sys;
652 /* update p_sout->i_out_pace_nocontrol */
653 p_stream->p_sout->i_out_pace_nocontrol--;
657 assert( p_sys->i_es <= 1 );
659 sout_MuxDelete( p_sys->p_mux );
660 if ( p_sys->i_es > 0 )
661 Del( p_stream, p_sys->es[0] );
662 sout_AccessOutDelete( p_sys->p_grab );
666 block_Release( p_sys->packet );
670 if( p_sys->rtsp != NULL )
671 RtspUnsetup( p_sys->rtsp );
673 vlc_mutex_destroy( &p_sys->lock_sdp );
674 vlc_mutex_destroy( &p_sys->lock_ts );
675 vlc_mutex_destroy( &p_sys->lock_es );
677 if( p_sys->p_httpd_file )
678 httpd_FileDelete( p_sys->p_httpd_file );
680 if( p_sys->p_httpd_host )
681 httpd_HostDelete( p_sys->p_httpd_host );
683 free( p_sys->psz_sdp );
685 if( p_sys->psz_sdp_file != NULL )
688 unlink( p_sys->psz_sdp_file );
690 free( p_sys->psz_sdp_file );
692 free( p_sys->psz_vod_session );
693 free( p_sys->psz_destination );
697 /*****************************************************************************
699 *****************************************************************************/
700 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
702 sout_stream_sys_t *p_sys = p_stream->p_sys;
705 vlc_UrlParse( &url, psz_url, 0 );
706 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
708 if( p_sys->p_httpd_file )
710 msg_Err( p_stream, "you can use sdp=http:// only once" );
714 if( HttpSetup( p_stream, &url ) )
716 msg_Err( p_stream, "cannot export SDP as HTTP" );
719 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
721 if( p_sys->rtsp != NULL )
723 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
727 if( url.psz_host != NULL && *url.psz_host )
729 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
730 "multiple-host configurations, use at your own risks.",
732 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
733 "command line instead." );
735 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
736 var_SetString( p_stream, "rtsp-host", url.psz_host );
738 if( url.i_port != 0 )
740 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
741 "the command line instead.", url.i_port ); */
743 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
744 var_SetInteger( p_stream, "rtsp-port", url.i_port );
747 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
748 if( p_sys->rtsp == NULL )
749 msg_Err( p_stream, "cannot export SDP as RTSP" );
751 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
752 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
754 p_sys->b_export_sap = true;
755 SapSetup( p_stream );
757 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
759 if( p_sys->psz_sdp_file != NULL )
761 msg_Err( p_stream, "you can use sdp=file:// only once" );
764 p_sys->psz_sdp_file = make_path( psz_url );
765 if( p_sys->psz_sdp_file == NULL )
767 FileSetup( p_stream );
771 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
776 vlc_UrlClean( &url );
779 /*****************************************************************************
781 *****************************************************************************/
783 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
785 sout_stream_sys_t *p_sys = p_stream->p_sys;
786 char *psz_sdp = NULL;
787 struct sockaddr_storage dst;
791 * When we have a fixed destination (typically when we do multicast),
792 * we need to put the actual port numbers in the SDP.
793 * When there is no fixed destination, we only support RTSP unicast
794 * on-demand setup, so we should rather let the clients decide which ports
796 * When there is both a fixed destination and RTSP unicast, we need to
797 * put port numbers used by the fixed destination, otherwise the SDP would
798 * become totally incorrect for multicast use. It should be noted that
799 * port numbers from SDP with RTSP are only "recommendation" from the
800 * server to the clients (per RFC2326), so only broken clients will fail
801 * to handle this properly. There is no solution but to use two differents
802 * output chain with two different RTSP URLs if you need to handle this
807 vlc_mutex_lock( &p_sys->lock_es );
808 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
809 goto out; /* hmm... */
811 if( p_sys->psz_destination != NULL )
815 /* Oh boy, this is really ugly! */
816 dstlen = sizeof( dst );
817 if( p_sys->es[0]->listen.fd != NULL )
818 getsockname( p_sys->es[0]->listen.fd[0],
819 (struct sockaddr *)&dst, &dstlen );
821 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
822 (struct sockaddr *)&dst, &dstlen );
828 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
829 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
830 && rtsp_url[7] == '[';
832 /* Dummy destination address for RTSP */
833 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
834 : sizeof( struct sockaddr_in );
835 memset (&dst, 0, dstlen);
836 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
842 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
843 NULL, 0, (struct sockaddr *)&dst, dstlen );
844 if( psz_sdp == NULL )
847 /* TODO: a=source-filter */
848 if( p_sys->rtcp_mux )
849 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
851 if( rtsp_url != NULL )
852 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
854 const char *proto = "RTP/AVP"; /* protocol */
855 if( rtsp_url == NULL )
857 switch( p_sys->proto )
862 proto = "TCP/RTP/AVP";
865 proto = "DCCP/RTP/AVP";
867 case IPPROTO_UDPLITE:
872 for( i = 0; i < p_sys->i_es; i++ )
874 sout_stream_id_t *id = p_sys->es[i];
875 rtp_format_t *rtp_fmt = &id->rtp_fmt;
876 const char *mime_major; /* major MIME type */
878 switch( rtp_fmt->cat )
881 mime_major = "video";
884 mime_major = "audio";
893 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
894 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
895 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
898 /* cf RFC4566 §5.14 */
899 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
900 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
902 if( rtsp_url != NULL )
904 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
905 if( track_url != NULL )
907 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
913 if( id->listen.fd != NULL )
914 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
915 if( p_sys->proto == IPPROTO_DCCP )
916 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
918 toupper( (unsigned char)mime_major[0] ) );
922 vlc_mutex_unlock( &p_sys->lock_es );
926 /*****************************************************************************
928 *****************************************************************************/
931 * Shrink the MTU down to a fixed packetization time (for audio).
934 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
936 /* Samples per second */
937 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
938 bytes *= id->rtp_fmt.channels;
941 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
942 id->i_mtu = 12 + spl;
943 else /* MTU is too small for ptime, align to a sample boundary */
944 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
947 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
949 /* NOTE: this plays nice with offsets because the calculations are
951 return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
954 /** Add an ES as a new RTP stream */
955 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
957 /* NOTE: As a special case, if we use a non-RTP
958 * mux (TS/PS), then p_fmt is NULL. */
959 sout_stream_sys_t *p_sys = p_stream->p_sys;
962 sout_stream_id_t *id = malloc( sizeof( *id ) );
963 if( unlikely(id == NULL) )
965 id->p_stream = p_stream;
967 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
968 if( id->i_mtu <= 12 + 16 )
969 id->i_mtu = 576 - 20 - 8; /* pessimistic */
970 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
975 vlc_mutex_init( &id->lock_sink );
980 id->listen.fd = NULL;
982 id->b_first_packet = true;
984 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
986 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
987 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
991 if (p_sys->p_vod_media != NULL)
993 id->rtp_fmt.ptname = NULL;
995 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
996 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
997 &ssrc, &id->i_seq_sent_next);
998 if (val == VLC_SUCCESS)
1000 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1001 /* This is ugly, but id->i_seq_sent_next needs to be
1002 * initialized inside vod_init_id() to avoid race
1004 id->i_sequence = id->i_seq_sent_next;
1006 /* vod_init_id() may fail either because the ES wasn't found in
1007 * the VoD media, or because the RTSP session is gone. In the
1008 * former case, id->rtp_fmt was left untouched. */
1009 format = (id->rtp_fmt.ptname != NULL);
1014 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1015 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1016 if (p_fmt == NULL && psz == NULL)
1018 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1020 if (val != VLC_SUCCESS)
1025 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1029 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1030 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1031 if (id->srtp == NULL)
1037 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1038 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1043 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1046 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1050 id->i_seq_sent_next = id->i_sequence;
1053 if( p_sys->psz_destination != NULL )
1055 /* Choose the port */
1056 uint16_t i_port = 0;
1060 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1061 i_port = p_sys->i_port_audio;
1063 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1064 i_port = p_sys->i_port_video;
1066 /* We do not need the ES lock (p_sys->lock_es) here, because
1067 * this is the only one thread that can *modify* the ES table.
1068 * The ES lock protects the other threads from our modifications
1069 * (TAB_APPEND, TAB_REMOVE). */
1070 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1071 if (i_port == p_sys->es[i]->i_port)
1072 i_port = 0; /* Port already in use! */
1073 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1077 msg_Err (p_stream, "too many RTP elementary streams");
1081 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1082 if (p == p_sys->es[i]->i_port)
1086 id->i_port = i_port;
1088 int type = SOCK_STREAM;
1090 switch( p_sys->proto )
1096 switch (id->rtp_fmt.cat)
1098 case VIDEO_ES: code = "RTPV"; break;
1099 case AUDIO_ES: code = "RTPARTPV"; break;
1100 case SPU_ES: code = "RTPTRTPV"; break;
1101 default: code = "RTPORTPV"; break;
1103 var_SetString (p_stream, "dccp-service", code);
1105 } /* fall through */
1108 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1109 p_sys->psz_destination, i_port,
1110 type, p_sys->proto );
1111 if( id->listen.fd == NULL )
1113 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1116 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1117 VLC_THREAD_PRIORITY_LOW ) )
1119 net_ListenClose( id->listen.fd );
1120 id->listen.fd = NULL;
1127 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1128 i_port, -1, p_sys->proto );
1131 msg_Err( p_stream, "cannot create RTP socket" );
1134 /* Ignore any unexpected incoming packet (including RTCP-RR
1135 * packets in case of rtcp-mux) */
1136 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1138 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1139 /* FIXME: test if this is multicast */
1146 switch( p_fmt->i_codec )
1148 case VLC_CODEC_MULAW:
1149 case VLC_CODEC_ALAW:
1151 rtp_set_ptime (id, 20, 1);
1153 case VLC_CODEC_S16B:
1154 case VLC_CODEC_S16L:
1155 rtp_set_ptime (id, 20, 2);
1161 #if 0 /* No payload formats sets this at the moment */
1164 cscov += 8 /* UDP */ + 12 /* RTP */;
1166 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1169 vlc_mutex_lock( &p_sys->lock_ts );
1170 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1171 vlc_mutex_unlock( &p_sys->lock_ts );
1173 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1174 p_sys->i_pts_offset );
1176 if( p_sys->rtsp != NULL )
1177 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1178 id->rtp_fmt.clock_rate, mcast_fd );
1180 id->p_fifo = block_FifoNew();
1181 if( unlikely(id->p_fifo == NULL) )
1183 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1185 block_FifoRelease( id->p_fifo );
1190 /* Update p_sys context */
1191 vlc_mutex_lock( &p_sys->lock_es );
1192 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1193 vlc_mutex_unlock( &p_sys->lock_es );
1195 psz_sdp = SDPGenerate( p_stream, NULL );
1197 vlc_mutex_lock( &p_sys->lock_sdp );
1198 free( p_sys->psz_sdp );
1199 p_sys->psz_sdp = psz_sdp;
1200 vlc_mutex_unlock( &p_sys->lock_sdp );
1202 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1204 /* Update SDP (sap/file) */
1205 if( p_sys->b_export_sap ) SapSetup( p_stream );
1206 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1211 Del( p_stream, id );
1215 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1217 sout_stream_sys_t *p_sys = p_stream->p_sys;
1219 vlc_mutex_lock( &p_sys->lock_es );
1220 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1221 vlc_mutex_unlock( &p_sys->lock_es );
1223 if( likely(id->p_fifo != NULL) )
1225 vlc_cancel( id->thread );
1226 vlc_join( id->thread, NULL );
1227 block_FifoRelease( id->p_fifo );
1230 free( id->rtp_fmt.fmtp );
1232 if (p_sys->p_vod_media != NULL)
1233 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1235 RtspDelId( p_sys->rtsp, id->rtsp_id );
1236 if( id->listen.fd != NULL )
1238 vlc_cancel( id->listen.thread );
1239 vlc_join( id->listen.thread, NULL );
1240 net_ListenClose( id->listen.fd );
1242 /* Delete remaining sinks (incoming connections or explicit
1244 while( id->sinkc > 0 )
1245 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1247 if( id->srtp != NULL )
1248 srtp_destroy( id->srtp );
1251 vlc_mutex_destroy( &id->lock_sink );
1253 /* Update SDP (sap/file) */
1254 if( p_sys->b_export_sap ) SapSetup( p_stream );
1255 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1261 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1266 assert( p_stream->p_sys->p_mux == NULL );
1269 while( p_buffer != NULL )
1271 p_next = p_buffer->p_next;
1273 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1274 * as the first packet of the stream */
1275 if (id->b_first_packet)
1277 id->b_first_packet = false;
1278 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1279 !strcmp(id->rtp_fmt.ptname, "theora"))
1280 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1284 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1287 block_Release( p_buffer );
1293 /****************************************************************************
1295 ****************************************************************************/
1296 static int SapSetup( sout_stream_t *p_stream )
1298 sout_stream_sys_t *p_sys = p_stream->p_sys;
1299 sout_instance_t *p_sout = p_stream->p_sout;
1301 /* Remove the previous session */
1302 if( p_sys->p_session != NULL)
1304 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1305 p_sys->p_session = NULL;
1308 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1309 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1311 p_sys->psz_destination );
1316 /****************************************************************************
1318 ****************************************************************************/
1319 static int FileSetup( sout_stream_t *p_stream )
1321 sout_stream_sys_t *p_sys = p_stream->p_sys;
1324 if( p_sys->psz_sdp == NULL )
1325 return VLC_EGENERIC; /* too early */
1327 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1329 msg_Err( p_stream, "cannot open file '%s' (%m)",
1330 p_sys->psz_sdp_file );
1331 return VLC_EGENERIC;
1334 fputs( p_sys->psz_sdp, f );
1340 /****************************************************************************
1342 ****************************************************************************/
1343 static int HttpCallback( httpd_file_sys_t *p_args,
1344 httpd_file_t *, uint8_t *p_request,
1345 uint8_t **pp_data, int *pi_data );
1347 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1349 sout_stream_sys_t *p_sys = p_stream->p_sys;
1351 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1352 if( p_sys->p_httpd_host )
1354 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1355 url->psz_path ? url->psz_path : "/",
1358 HttpCallback, (void*)p_sys );
1360 if( p_sys->p_httpd_file == NULL )
1362 return VLC_EGENERIC;
1367 static int HttpCallback( httpd_file_sys_t *p_args,
1368 httpd_file_t *f, uint8_t *p_request,
1369 uint8_t **pp_data, int *pi_data )
1371 VLC_UNUSED(f); VLC_UNUSED(p_request);
1372 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1374 vlc_mutex_lock( &p_sys->lock_sdp );
1375 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1377 *pi_data = strlen( p_sys->psz_sdp );
1378 *pp_data = malloc( *pi_data );
1379 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1386 vlc_mutex_unlock( &p_sys->lock_sdp );
1391 /****************************************************************************
1393 ****************************************************************************/
1394 static void* ThreadSend( void *data )
1397 # define ECONNREFUSED WSAECONNREFUSED
1398 # define ENOPROTOOPT WSAENOPROTOOPT
1399 # define EHOSTUNREACH WSAEHOSTUNREACH
1400 # define ENETUNREACH WSAENETUNREACH
1401 # define ENETDOWN WSAENETDOWN
1402 # define ENOBUFS WSAENOBUFS
1403 # define EAGAIN WSAEWOULDBLOCK
1404 # define EWOULDBLOCK WSAEWOULDBLOCK
1406 sout_stream_id_t *id = data;
1407 unsigned i_caching = id->i_caching;
1411 block_t *out = block_FifoGet( id->p_fifo );
1412 block_cleanup_push (out);
1416 { /* FIXME: this is awfully inefficient */
1417 size_t len = out->i_buffer;
1418 out = block_Realloc( out, 0, len + 10 );
1419 out->i_buffer = len;
1421 int canc = vlc_savecancel ();
1422 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1423 vlc_restorecancel (canc);
1427 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1428 block_Release( out );
1432 out->i_buffer = len;
1435 mwait (out->i_dts + i_caching);
1440 mwait (out->i_dts + i_caching);
1444 ssize_t len = out->i_buffer;
1445 int canc = vlc_savecancel ();
1447 vlc_mutex_lock( &id->lock_sink );
1448 unsigned deadc = 0; /* How many dead sockets? */
1449 int deadv[id->sinkc]; /* Dead sockets list */
1451 for( int i = 0; i < id->sinkc; i++ )
1454 if( !id->srtp ) /* FIXME: SRTCP support */
1456 SendRTCP( id->sinkv[i].rtcp, out );
1458 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1459 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1460 && net_errno != ENOBUFS && net_errno != ENOMEM )
1463 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1464 &type, &(socklen_t){ sizeof(type) });
1465 if( type == SOCK_DGRAM )
1466 /* ICMP soft error: ignore and retry */
1467 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1469 /* Broken connection */
1470 deadv[deadc++] = id->sinkv[i].rtp_fd;
1473 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1474 vlc_mutex_unlock( &id->lock_sink );
1475 block_Release( out );
1477 for( unsigned i = 0; i < deadc; i++ )
1479 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1480 rtp_del_sink( id, deadv[i] );
1482 vlc_restorecancel (canc);
1488 /* This thread dequeues incoming connections (DCCP streaming) */
1489 static void *rtp_listen_thread( void *data )
1491 sout_stream_id_t *id = data;
1493 assert( id->listen.fd != NULL );
1497 int fd = net_Accept( id->p_stream, id->listen.fd );
1500 int canc = vlc_savecancel( );
1501 rtp_add_sink( id, fd, true, NULL );
1502 vlc_restorecancel( canc );
1509 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1511 rtp_sink_t sink = { fd, NULL };
1512 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1514 if( sink.rtcp == NULL )
1515 msg_Err( id->p_stream, "RTCP failed!" );
1517 vlc_mutex_lock( &id->lock_sink );
1518 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1520 *seq = id->i_seq_sent_next;
1521 vlc_mutex_unlock( &id->lock_sink );
1525 void rtp_del_sink( sout_stream_id_t *id, int fd )
1527 rtp_sink_t sink = { fd, NULL };
1529 /* NOTE: must be safe to use if fd is not included */
1530 vlc_mutex_lock( &id->lock_sink );
1531 for( int i = 0; i < id->sinkc; i++ )
1533 if (id->sinkv[i].rtp_fd == fd)
1535 sink = id->sinkv[i];
1536 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1540 vlc_mutex_unlock( &id->lock_sink );
1542 CloseRTCP( sink.rtcp );
1543 net_Close( sink.rtp_fd );
1546 uint16_t rtp_get_seq( sout_stream_id_t *id )
1548 /* This will return values for the next packet. */
1551 vlc_mutex_lock( &id->lock_sink );
1552 seq = id->i_seq_sent_next;
1553 vlc_mutex_unlock( &id->lock_sink );
1558 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1559 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1560 * random (although we use the same reference for all the ES as a
1561 * feature). In the VoD case, this function is called independently
1562 * from several parts of the code, so we need to always return the same
1564 static int64_t rtp_init_ts( const vod_media_t *p_media,
1565 const char *psz_vod_session )
1567 if (p_media == NULL || psz_vod_session == NULL)
1571 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1572 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1573 i_ts_init ^= (uintptr_t)p_media;
1574 /* Limit the timestamp to 48 bytes, this is enough and allows us
1575 * to stay away from overflows */
1576 i_ts_init &= 0xFFFFFFFFFFFF;
1580 /* Return a timestamp corresponding to packets being sent now, and that
1581 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1582 * Also return the NPT corresponding to this timestamp. If the stream
1583 * output is not started, the initial timestamp that will be used with
1584 * the first packets for NPT=0 is returned instead. */
1585 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1586 const vod_media_t *p_media, const char *psz_vod_session,
1593 p_stream = id->p_stream;
1595 if (p_stream == NULL)
1596 return rtp_init_ts(p_media, psz_vod_session);
1598 sout_stream_sys_t *p_sys = p_stream->p_sys;
1600 vlc_mutex_lock( &p_sys->lock_ts );
1601 i_npt_zero = p_sys->i_npt_zero;
1602 vlc_mutex_unlock( &p_sys->lock_ts );
1604 if( i_npt_zero == VLC_TS_INVALID )
1605 return p_sys->i_pts_zero;
1607 mtime_t now = mdate();
1608 if( now < i_npt_zero )
1609 return p_sys->i_pts_zero;
1611 int64_t npt = now - i_npt_zero;
1615 return p_sys->i_pts_zero + npt;
1618 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1619 int b_marker, int64_t i_pts )
1621 if( !id->b_ts_init )
1623 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1624 vlc_mutex_lock( &p_sys->lock_ts );
1625 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1627 /* This is the first packet of any ES. We initialize the
1628 * NPT=0 time reference, and the offset to match the
1629 * arbitrary PTS reference. */
1630 p_sys->i_npt_zero = i_pts + id->i_caching;
1631 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1633 vlc_mutex_unlock( &p_sys->lock_ts );
1635 /* And in any case this is the first packet of this ES, so we
1636 * initialize the offset for this ES. */
1637 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1638 p_sys->i_pts_offset );
1639 id->b_ts_init = true;
1642 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1645 out->p_buffer[0] = 0x80;
1646 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1647 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1648 out->p_buffer[3] = ( id->i_sequence )&0xff;
1649 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1650 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1651 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1652 out->p_buffer[7] = ( i_timestamp )&0xff;
1654 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1660 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1662 block_FifoPut( id->p_fifo, out );
1666 * @return configured max RTP payload size (including payload type-specific
1667 * headers, excluding RTP and transport headers)
1669 size_t rtp_mtu (const sout_stream_id_t *id)
1671 return id->i_mtu - 12;
1674 /*****************************************************************************
1676 *****************************************************************************/
1678 /** Add an ES to a non-RTP muxed stream */
1679 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1681 sout_input_t *p_input;
1682 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1683 assert( p_mux != NULL );
1685 p_input = sout_MuxAddStream( p_mux, p_fmt );
1686 if( p_input == NULL )
1688 msg_Err( p_stream, "cannot add this stream to the muxer" );
1692 return (sout_stream_id_t *)p_input;
1696 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1699 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1700 assert( p_mux != NULL );
1702 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1707 /** Remove an ES from a non-RTP muxed stream */
1708 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1710 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1711 assert( p_mux != NULL );
1713 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1718 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1719 const block_t *p_buffer )
1721 sout_stream_sys_t *p_sys = p_stream->p_sys;
1722 sout_stream_id_t *id = p_sys->es[0];
1724 int64_t i_dts = p_buffer->i_dts;
1726 uint8_t *p_data = p_buffer->p_buffer;
1727 size_t i_data = p_buffer->i_buffer;
1728 size_t i_max = id->i_mtu - 12;
1730 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1736 /* output complete packet */
1737 if( p_sys->packet &&
1738 p_sys->packet->i_buffer + i_data > i_max )
1740 rtp_packetize_send( id, p_sys->packet );
1741 p_sys->packet = NULL;
1744 if( p_sys->packet == NULL )
1746 /* allocate a new packet */
1747 p_sys->packet = block_New( p_stream, id->i_mtu );
1748 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1749 p_sys->packet->i_dts = i_dts;
1750 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1751 i_dts += p_sys->packet->i_length;
1754 i_size = __MIN( i_data,
1755 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1757 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1760 p_sys->packet->i_buffer += i_size;
1769 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1772 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1778 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1780 p_next = p_buffer->p_next;
1781 block_Release( p_buffer );
1789 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1791 sout_access_out_t *p_grab;
1793 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1794 if( p_grab == NULL )
1797 p_grab->p_module = NULL;
1798 p_grab->psz_access = strdup( "grab" );
1799 p_grab->p_cfg = NULL;
1800 p_grab->psz_path = strdup( "" );
1801 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1802 p_grab->pf_seek = NULL;
1803 p_grab->pf_write = AccessOutGrabberWrite;