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 CAT_TEXT N_("Session category")
97 #define CAT_LONGTEXT N_( \
98 "This allows you to specify a category for the session, " \
99 "that will be announced if you choose to use SAP." )
100 #define DESC_TEXT N_("Session description")
101 #define DESC_LONGTEXT N_( \
102 "This allows you to give a short description with details about the stream, " \
103 "that will be announced in the SDP (Session Descriptor)." )
104 #define URL_TEXT N_("Session URL")
105 #define URL_LONGTEXT N_( \
106 "This allows you to give a URL with more details about the stream " \
107 "(often the website of the streaming organization), that will " \
108 "be announced in the SDP (Session Descriptor)." )
109 #define EMAIL_TEXT N_("Session email")
110 #define EMAIL_LONGTEXT N_( \
111 "This allows you to give a contact mail address for the stream, that will " \
112 "be announced in the SDP (Session Descriptor)." )
113 #define PHONE_TEXT N_("Session phone number")
114 #define PHONE_LONGTEXT N_( \
115 "This allows you to give a contact telephone number for the stream, that will " \
116 "be announced in the SDP (Session Descriptor)." )
118 #define PORT_TEXT N_("Port")
119 #define PORT_LONGTEXT N_( \
120 "This allows you to specify the base port for the RTP streaming." )
121 #define PORT_AUDIO_TEXT N_("Audio port")
122 #define PORT_AUDIO_LONGTEXT N_( \
123 "This allows you to specify the default audio port for the RTP streaming." )
124 #define PORT_VIDEO_TEXT N_("Video port")
125 #define PORT_VIDEO_LONGTEXT N_( \
126 "This allows you to specify the default video port for the RTP streaming." )
128 #define TTL_TEXT N_("Hop limit (TTL)")
129 #define TTL_LONGTEXT N_( \
130 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
131 "the multicast packets sent by the stream output (-1 = use operating " \
132 "system built-in default).")
134 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
135 #define RTCP_MUX_LONGTEXT N_( \
136 "This sends and receives RTCP packet multiplexed over the same port " \
139 #define CACHING_TEXT N_("Caching value (ms)")
140 #define CACHING_LONGTEXT N_( \
141 "Default caching value for outbound RTP streams. This " \
142 "value should be set in milliseconds." )
144 #define PROTO_TEXT N_("Transport protocol")
145 #define PROTO_LONGTEXT N_( \
146 "This selects which transport protocol to use for RTP." )
148 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
149 #define SRTP_KEY_LONGTEXT N_( \
150 "RTP packets will be integrity-protected and ciphered "\
151 "with this Secure RTP master shared secret key. "\
152 "This must be a 32-character-long hexadecimal string.")
154 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
155 #define SRTP_SALT_LONGTEXT N_( \
156 "Secure RTP requires a (non-secret) master salt value. " \
157 "This must be a 28-character-long hexadecimal string.")
159 static const char *const ppsz_protos[] = {
160 "dccp", "sctp", "tcp", "udp", "udplite",
163 static const char *const ppsz_protocols[] = {
164 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
167 #define RFC3016_TEXT N_("MP4A LATM")
168 #define RFC3016_LONGTEXT N_( \
169 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
171 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
172 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
173 "not receiving any RTSP request for this long. Setting it to a " \
174 "negative value or zero disables timeouts. The default is 60 (one " \
177 #define RTSP_USER_TEXT N_("Username")
178 #define RTSP_USER_LONGTEXT N_("User name that will be " \
179 "requested to access the stream." )
180 #define RTSP_PASS_TEXT N_("Password")
181 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
182 "requested to access the stream." )
184 static int Open ( vlc_object_t * );
185 static void Close( vlc_object_t * );
187 #define SOUT_CFG_PREFIX "sout-rtp-"
188 #define MAX_EMPTY_BLOCKS 200
191 set_shortname( N_("RTP"))
192 set_description( N_("RTP stream output") )
193 set_capability( "sout stream", 0 )
194 add_shortcut( "rtp", "vod" )
195 set_category( CAT_SOUT )
196 set_subcategory( SUBCAT_SOUT_STREAM )
198 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
199 DEST_LONGTEXT, true )
200 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
202 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
204 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
207 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
208 NAME_LONGTEXT, true )
209 add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_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 )
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_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
257 RTSP_TIMEOUT_LONGTEXT, true )
258 add_string( "sout-rtsp-user", "",
259 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
260 add_password( "sout-rtsp-pwd", "",
261 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
265 /*****************************************************************************
266 * Exported prototypes
267 *****************************************************************************/
268 static const char *const ppsz_sout_options[] = {
269 "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
270 "mux", "sap", "description", "url", "email", "phone",
271 "proto", "rtcp-mux", "caching",
278 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
279 static int Del ( sout_stream_t *, sout_stream_id_t * );
280 static int Send( sout_stream_t *, sout_stream_id_t *,
282 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
283 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
284 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
287 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
288 static void* ThreadSend( void * );
289 static void *rtp_listen_thread( void * );
291 static void SDPHandleUrl( sout_stream_t *, const char * );
293 static int SapSetup( sout_stream_t *p_stream );
294 static int FileSetup( sout_stream_t *p_stream );
295 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
297 static int64_t rtp_init_ts( const vod_media_t *p_media,
298 const char *psz_vod_session );
300 struct sout_stream_sys_t
304 vlc_mutex_t lock_sdp;
311 session_descriptor_t *p_session;
314 httpd_host_t *p_httpd_host;
315 httpd_file_t *p_httpd_file;
320 /* RTSP NPT and timestamp computations */
321 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
322 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
323 int64_t i_pts_offset; /* matches actual PTS to prediction */
327 char *psz_destination;
329 uint16_t i_port_audio;
330 uint16_t i_port_video;
336 vod_media_t *p_vod_media;
337 char *psz_vod_session;
339 /* in case we do TS/PS over rtp */
341 sout_access_out_t *p_grab;
347 sout_stream_id_t **es;
350 typedef struct rtp_sink_t
356 struct sout_stream_id_t
358 sout_stream_t *p_stream;
363 uint32_t i_ts_offset;
367 uint16_t i_seq_sent_next;
370 rtp_format_t rtp_fmt;
373 /* Packetizer specific fields */
376 srtp_session_t *srtp;
381 vlc_mutex_t lock_sink;
384 rtsp_stream_id_t *rtsp_id;
390 block_fifo_t *p_fifo;
394 /*****************************************************************************
396 *****************************************************************************/
397 static int Open( vlc_object_t *p_this )
399 sout_stream_t *p_stream = (sout_stream_t*)p_this;
400 sout_instance_t *p_sout = p_stream->p_sout;
401 sout_stream_sys_t *p_sys = NULL;
402 config_chain_t *p_cfg = NULL;
406 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
407 ppsz_sout_options, p_stream->p_cfg );
409 p_sys = malloc( sizeof( sout_stream_sys_t ) );
413 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
415 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
416 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
417 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
418 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
420 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
422 msg_Err( p_stream, "audio and video RTP port must be distinct" );
423 free( p_sys->psz_destination );
428 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
430 if( !strcmp( p_cfg->psz_name, "sdp" )
431 && ( p_cfg->psz_value != NULL )
432 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
440 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
443 if( !strncasecmp( psz, "rtsp:", 5 ) )
449 /* Transport protocol */
450 p_sys->proto = IPPROTO_UDP;
451 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
453 if ((psz == NULL) || !strcasecmp (psz, "udp"))
454 (void)0; /* default */
456 if (!strcasecmp (psz, "dccp"))
458 p_sys->proto = IPPROTO_DCCP;
459 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
463 if (!strcasecmp (psz, "sctp"))
465 p_sys->proto = IPPROTO_TCP;
466 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
471 if (!strcasecmp (psz, "tcp"))
473 p_sys->proto = IPPROTO_TCP;
474 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
478 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
479 p_sys->proto = IPPROTO_UDPLITE;
481 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
484 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
486 p_sys->p_vod_media = NULL;
487 p_sys->psz_vod_session = NULL;
489 if (! strcmp(p_stream->psz_name, "vod"))
491 /* The VLM stops all instances before deleting a media, so this
492 * reference will remain valid during the lifetime of the rtp
494 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
496 if (p_sys->p_vod_media != NULL)
498 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
499 if (p_sys->psz_vod_session == NULL)
501 msg_Err(p_stream, "missing VoD session");
506 const char *mux = vod_get_mux(p_sys->p_vod_media);
507 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
511 if( p_sys->psz_destination == NULL && !b_rtsp
512 && p_sys->p_vod_media == NULL )
514 msg_Err( p_stream, "missing destination and not in RTSP mode" );
519 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
522 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
523 var_SetInteger( p_stream, "ttl", i_ttl );
526 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
528 /* NPT=0 time will be determined when we packetize the first packet
529 * (of any ES). But we want to be able to report rtptime in RTSP
530 * without waiting (and already did in the VoD case). So until then,
531 * we use an arbitrary reference PTS for timestamp computations, and
532 * then actual PTS will catch up using offsets. */
533 p_sys->i_npt_zero = VLC_TS_INVALID;
534 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
535 p_sys->psz_vod_session);
539 p_sys->psz_sdp = NULL;
541 p_sys->b_export_sap = false;
542 p_sys->p_session = NULL;
543 p_sys->psz_sdp_file = NULL;
545 p_sys->p_httpd_host = NULL;
546 p_sys->p_httpd_file = NULL;
548 p_stream->p_sys = p_sys;
550 vlc_mutex_init( &p_sys->lock_sdp );
551 vlc_mutex_init( &p_sys->lock_ts );
552 vlc_mutex_init( &p_sys->lock_es );
554 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
557 /* Check muxer type */
558 if( strncasecmp( psz, "ps", 2 )
559 && strncasecmp( psz, "mpeg1", 5 )
560 && strncasecmp( psz, "ts", 2 ) )
562 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
564 vlc_mutex_destroy( &p_sys->lock_sdp );
565 vlc_mutex_destroy( &p_sys->lock_ts );
566 vlc_mutex_destroy( &p_sys->lock_es );
567 free( p_sys->psz_vod_session );
568 free( p_sys->psz_destination );
573 p_sys->p_grab = GrabberCreate( p_stream );
574 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
577 if( p_sys->p_mux == NULL )
579 msg_Err( p_stream, "cannot create muxer" );
580 sout_AccessOutDelete( p_sys->p_grab );
581 vlc_mutex_destroy( &p_sys->lock_sdp );
582 vlc_mutex_destroy( &p_sys->lock_ts );
583 vlc_mutex_destroy( &p_sys->lock_es );
584 free( p_sys->psz_vod_session );
585 free( p_sys->psz_destination );
590 p_sys->packet = NULL;
592 p_stream->pf_add = MuxAdd;
593 p_stream->pf_del = MuxDel;
594 p_stream->pf_send = MuxSend;
599 p_sys->p_grab = NULL;
601 p_stream->pf_add = Add;
602 p_stream->pf_del = Del;
603 p_stream->pf_send = Send;
606 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
607 SDPHandleUrl( p_stream, "sap" );
609 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
612 config_chain_t *p_cfg;
614 SDPHandleUrl( p_stream, psz );
616 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
618 if( !strcmp( p_cfg->psz_name, "sdp" ) )
620 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
623 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
624 if( !strcmp( p_cfg->psz_value, psz ) )
627 SDPHandleUrl( p_stream, p_cfg->psz_value );
633 /* update p_sout->i_out_pace_nocontrol */
634 p_stream->p_sout->i_out_pace_nocontrol++;
636 if( p_sys->p_mux != NULL )
638 sout_stream_id_t *id = Add( p_stream, NULL );
649 /*****************************************************************************
651 *****************************************************************************/
652 static void Close( vlc_object_t * p_this )
654 sout_stream_t *p_stream = (sout_stream_t*)p_this;
655 sout_stream_sys_t *p_sys = p_stream->p_sys;
657 /* update p_sout->i_out_pace_nocontrol */
658 p_stream->p_sout->i_out_pace_nocontrol--;
662 assert( p_sys->i_es <= 1 );
664 sout_MuxDelete( p_sys->p_mux );
665 if ( p_sys->i_es > 0 )
666 Del( p_stream, p_sys->es[0] );
667 sout_AccessOutDelete( p_sys->p_grab );
671 block_Release( p_sys->packet );
675 if( p_sys->rtsp != NULL )
676 RtspUnsetup( p_sys->rtsp );
678 vlc_mutex_destroy( &p_sys->lock_sdp );
679 vlc_mutex_destroy( &p_sys->lock_ts );
680 vlc_mutex_destroy( &p_sys->lock_es );
682 if( p_sys->p_httpd_file )
683 httpd_FileDelete( p_sys->p_httpd_file );
685 if( p_sys->p_httpd_host )
686 httpd_HostDelete( p_sys->p_httpd_host );
688 free( p_sys->psz_sdp );
690 if( p_sys->psz_sdp_file != NULL )
693 unlink( p_sys->psz_sdp_file );
695 free( p_sys->psz_sdp_file );
697 free( p_sys->psz_vod_session );
698 free( p_sys->psz_destination );
702 /*****************************************************************************
704 *****************************************************************************/
705 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
707 sout_stream_sys_t *p_sys = p_stream->p_sys;
710 vlc_UrlParse( &url, psz_url, 0 );
711 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
713 if( p_sys->p_httpd_file )
715 msg_Err( p_stream, "you can use sdp=http:// only once" );
719 if( HttpSetup( p_stream, &url ) )
721 msg_Err( p_stream, "cannot export SDP as HTTP" );
724 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
726 if( p_sys->rtsp != NULL )
728 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
732 if( url.psz_host != NULL && *url.psz_host )
734 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
735 "multiple-host configurations, use at your own risks.",
737 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
738 "command line instead." );
740 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
741 var_SetString( p_stream, "rtsp-host", url.psz_host );
743 if( url.i_port != 0 )
745 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
746 "the command line instead.", url.i_port ); */
748 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
749 var_SetInteger( p_stream, "rtsp-port", url.i_port );
752 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
753 if( p_sys->rtsp == NULL )
754 msg_Err( p_stream, "cannot export SDP as RTSP" );
756 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
757 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
759 p_sys->b_export_sap = true;
760 SapSetup( p_stream );
762 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
764 if( p_sys->psz_sdp_file != NULL )
766 msg_Err( p_stream, "you can use sdp=file:// only once" );
769 p_sys->psz_sdp_file = make_path( psz_url );
770 if( p_sys->psz_sdp_file == NULL )
772 FileSetup( p_stream );
776 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
781 vlc_UrlClean( &url );
784 /*****************************************************************************
786 *****************************************************************************/
788 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
790 sout_stream_sys_t *p_sys = p_stream->p_sys;
791 char *psz_sdp = NULL;
792 struct sockaddr_storage dst;
796 * When we have a fixed destination (typically when we do multicast),
797 * we need to put the actual port numbers in the SDP.
798 * When there is no fixed destination, we only support RTSP unicast
799 * on-demand setup, so we should rather let the clients decide which ports
801 * When there is both a fixed destination and RTSP unicast, we need to
802 * put port numbers used by the fixed destination, otherwise the SDP would
803 * become totally incorrect for multicast use. It should be noted that
804 * port numbers from SDP with RTSP are only "recommendation" from the
805 * server to the clients (per RFC2326), so only broken clients will fail
806 * to handle this properly. There is no solution but to use two differents
807 * output chain with two different RTSP URLs if you need to handle this
812 vlc_mutex_lock( &p_sys->lock_es );
813 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
814 goto out; /* hmm... */
816 if( p_sys->psz_destination != NULL )
820 /* Oh boy, this is really ugly! */
821 dstlen = sizeof( dst );
822 if( p_sys->es[0]->listen.fd != NULL )
823 getsockname( p_sys->es[0]->listen.fd[0],
824 (struct sockaddr *)&dst, &dstlen );
826 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
827 (struct sockaddr *)&dst, &dstlen );
833 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
834 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
835 && rtsp_url[7] == '[';
837 /* Dummy destination address for RTSP */
838 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
839 : sizeof( struct sockaddr_in );
840 memset (&dst, 0, dstlen);
841 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
847 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
848 NULL, 0, (struct sockaddr *)&dst, dstlen );
849 if( psz_sdp == NULL )
852 /* TODO: a=source-filter */
853 if( p_sys->rtcp_mux )
854 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
856 if( rtsp_url != NULL )
857 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
859 const char *proto = "RTP/AVP"; /* protocol */
860 if( rtsp_url == NULL )
862 switch( p_sys->proto )
867 proto = "TCP/RTP/AVP";
870 proto = "DCCP/RTP/AVP";
872 case IPPROTO_UDPLITE:
877 for( i = 0; i < p_sys->i_es; i++ )
879 sout_stream_id_t *id = p_sys->es[i];
880 rtp_format_t *rtp_fmt = &id->rtp_fmt;
881 const char *mime_major; /* major MIME type */
883 switch( rtp_fmt->cat )
886 mime_major = "video";
889 mime_major = "audio";
898 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
899 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
900 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
903 /* cf RFC4566 §5.14 */
904 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
905 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
907 if( rtsp_url != NULL )
909 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
910 if( track_url != NULL )
912 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
918 if( id->listen.fd != NULL )
919 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
920 if( p_sys->proto == IPPROTO_DCCP )
921 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
923 toupper( (unsigned char)mime_major[0] ) );
927 vlc_mutex_unlock( &p_sys->lock_es );
931 /*****************************************************************************
933 *****************************************************************************/
936 * Shrink the MTU down to a fixed packetization time (for audio).
939 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
941 /* Samples per second */
942 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
943 bytes *= id->rtp_fmt.channels;
946 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
947 id->i_mtu = 12 + spl;
948 else /* MTU is too small for ptime, align to a sample boundary */
949 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
952 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
954 /* This is an overflow-proof way of doing:
955 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
957 * NOTE: this plays nice with offsets because the (equivalent)
958 * calculations are linear. */
959 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
960 return q.quot * (int64_t)i_clock_rate
961 + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
964 /** Add an ES as a new RTP stream */
965 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
967 /* NOTE: As a special case, if we use a non-RTP
968 * mux (TS/PS), then p_fmt is NULL. */
969 sout_stream_sys_t *p_sys = p_stream->p_sys;
972 sout_stream_id_t *id = malloc( sizeof( *id ) );
973 if( unlikely(id == NULL) )
975 id->p_stream = p_stream;
977 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
978 if( id->i_mtu <= 12 + 16 )
979 id->i_mtu = 576 - 20 - 8; /* pessimistic */
980 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
985 vlc_mutex_init( &id->lock_sink );
990 id->listen.fd = NULL;
992 id->b_first_packet = true;
994 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
996 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
997 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
1001 if (p_sys->p_vod_media != NULL)
1003 id->rtp_fmt.ptname = NULL;
1005 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
1006 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
1007 &ssrc, &id->i_seq_sent_next);
1008 if (val == VLC_SUCCESS)
1010 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1011 /* This is ugly, but id->i_seq_sent_next needs to be
1012 * initialized inside vod_init_id() to avoid race
1014 id->i_sequence = id->i_seq_sent_next;
1016 /* vod_init_id() may fail either because the ES wasn't found in
1017 * the VoD media, or because the RTSP session is gone. In the
1018 * former case, id->rtp_fmt was left untouched. */
1019 format = (id->rtp_fmt.ptname != NULL);
1024 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1025 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1026 if (p_fmt == NULL && psz == NULL)
1028 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1030 if (val != VLC_SUCCESS)
1035 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1039 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1040 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1041 if (id->srtp == NULL)
1047 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1048 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1053 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1056 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1060 id->i_seq_sent_next = id->i_sequence;
1063 if( p_sys->psz_destination != NULL )
1065 /* Choose the port */
1066 uint16_t i_port = 0;
1070 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1071 i_port = p_sys->i_port_audio;
1073 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1074 i_port = p_sys->i_port_video;
1076 /* We do not need the ES lock (p_sys->lock_es) here, because
1077 * this is the only one thread that can *modify* the ES table.
1078 * The ES lock protects the other threads from our modifications
1079 * (TAB_APPEND, TAB_REMOVE). */
1080 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1081 if (i_port == p_sys->es[i]->i_port)
1082 i_port = 0; /* Port already in use! */
1083 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1087 msg_Err (p_stream, "too many RTP elementary streams");
1091 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1092 if (p == p_sys->es[i]->i_port)
1096 id->i_port = i_port;
1098 int type = SOCK_STREAM;
1100 switch( p_sys->proto )
1106 switch (id->rtp_fmt.cat)
1108 case VIDEO_ES: code = "RTPV"; break;
1109 case AUDIO_ES: code = "RTPARTPV"; break;
1110 case SPU_ES: code = "RTPTRTPV"; break;
1111 default: code = "RTPORTPV"; break;
1113 var_SetString (p_stream, "dccp-service", code);
1115 } /* fall through */
1118 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1119 p_sys->psz_destination, i_port,
1120 type, p_sys->proto );
1121 if( id->listen.fd == NULL )
1123 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1126 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1127 VLC_THREAD_PRIORITY_LOW ) )
1129 net_ListenClose( id->listen.fd );
1130 id->listen.fd = NULL;
1137 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1138 i_port, -1, p_sys->proto );
1141 msg_Err( p_stream, "cannot create RTP socket" );
1144 /* Ignore any unexpected incoming packet (including RTCP-RR
1145 * packets in case of rtcp-mux) */
1146 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1148 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1149 /* FIXME: test if this is multicast */
1156 switch( p_fmt->i_codec )
1158 case VLC_CODEC_MULAW:
1159 case VLC_CODEC_ALAW:
1161 rtp_set_ptime (id, 20, 1);
1163 case VLC_CODEC_S16B:
1164 case VLC_CODEC_S16L:
1165 rtp_set_ptime (id, 20, 2);
1171 #if 0 /* No payload formats sets this at the moment */
1174 cscov += 8 /* UDP */ + 12 /* RTP */;
1176 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1179 vlc_mutex_lock( &p_sys->lock_ts );
1180 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1181 vlc_mutex_unlock( &p_sys->lock_ts );
1183 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1184 p_sys->i_pts_offset );
1186 if( p_sys->rtsp != NULL )
1187 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1188 id->rtp_fmt.clock_rate, mcast_fd );
1190 id->p_fifo = block_FifoNew();
1191 if( unlikely(id->p_fifo == NULL) )
1193 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1195 block_FifoRelease( id->p_fifo );
1200 /* Update p_sys context */
1201 vlc_mutex_lock( &p_sys->lock_es );
1202 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1203 vlc_mutex_unlock( &p_sys->lock_es );
1205 psz_sdp = SDPGenerate( p_stream, NULL );
1207 vlc_mutex_lock( &p_sys->lock_sdp );
1208 free( p_sys->psz_sdp );
1209 p_sys->psz_sdp = psz_sdp;
1210 vlc_mutex_unlock( &p_sys->lock_sdp );
1212 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1214 /* Update SDP (sap/file) */
1215 if( p_sys->b_export_sap ) SapSetup( p_stream );
1216 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1221 Del( p_stream, id );
1225 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1227 sout_stream_sys_t *p_sys = p_stream->p_sys;
1229 vlc_mutex_lock( &p_sys->lock_es );
1230 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1231 vlc_mutex_unlock( &p_sys->lock_es );
1233 if( likely(id->p_fifo != NULL) )
1235 vlc_cancel( id->thread );
1236 vlc_join( id->thread, NULL );
1237 block_FifoRelease( id->p_fifo );
1240 free( id->rtp_fmt.fmtp );
1242 if (p_sys->p_vod_media != NULL)
1243 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1245 RtspDelId( p_sys->rtsp, id->rtsp_id );
1246 if( id->listen.fd != NULL )
1248 vlc_cancel( id->listen.thread );
1249 vlc_join( id->listen.thread, NULL );
1250 net_ListenClose( id->listen.fd );
1252 /* Delete remaining sinks (incoming connections or explicit
1254 while( id->sinkc > 0 )
1255 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1257 if( id->srtp != NULL )
1258 srtp_destroy( id->srtp );
1261 vlc_mutex_destroy( &id->lock_sink );
1263 /* Update SDP (sap/file) */
1264 if( p_sys->b_export_sap ) SapSetup( p_stream );
1265 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1271 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1276 assert( p_stream->p_sys->p_mux == NULL );
1279 while( p_buffer != NULL )
1281 p_next = p_buffer->p_next;
1283 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1284 * as the first packet of the stream */
1285 if (id->b_first_packet)
1287 id->b_first_packet = false;
1288 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1289 !strcmp(id->rtp_fmt.ptname, "theora"))
1290 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1294 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1297 block_Release( p_buffer );
1303 /****************************************************************************
1305 ****************************************************************************/
1306 static int SapSetup( sout_stream_t *p_stream )
1308 sout_stream_sys_t *p_sys = p_stream->p_sys;
1309 sout_instance_t *p_sout = p_stream->p_sout;
1311 /* Remove the previous session */
1312 if( p_sys->p_session != NULL)
1314 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1315 p_sys->p_session = NULL;
1318 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1319 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1321 p_sys->psz_destination );
1326 /****************************************************************************
1328 ****************************************************************************/
1329 static int FileSetup( sout_stream_t *p_stream )
1331 sout_stream_sys_t *p_sys = p_stream->p_sys;
1334 if( p_sys->psz_sdp == NULL )
1335 return VLC_EGENERIC; /* too early */
1337 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1339 msg_Err( p_stream, "cannot open file '%s' (%m)",
1340 p_sys->psz_sdp_file );
1341 return VLC_EGENERIC;
1344 fputs( p_sys->psz_sdp, f );
1350 /****************************************************************************
1352 ****************************************************************************/
1353 static int HttpCallback( httpd_file_sys_t *p_args,
1354 httpd_file_t *, uint8_t *p_request,
1355 uint8_t **pp_data, int *pi_data );
1357 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1359 sout_stream_sys_t *p_sys = p_stream->p_sys;
1361 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1362 if( p_sys->p_httpd_host )
1364 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1365 url->psz_path ? url->psz_path : "/",
1368 HttpCallback, (void*)p_sys );
1370 if( p_sys->p_httpd_file == NULL )
1372 return VLC_EGENERIC;
1377 static int HttpCallback( httpd_file_sys_t *p_args,
1378 httpd_file_t *f, uint8_t *p_request,
1379 uint8_t **pp_data, int *pi_data )
1381 VLC_UNUSED(f); VLC_UNUSED(p_request);
1382 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1384 vlc_mutex_lock( &p_sys->lock_sdp );
1385 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1387 *pi_data = strlen( p_sys->psz_sdp );
1388 *pp_data = malloc( *pi_data );
1389 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1396 vlc_mutex_unlock( &p_sys->lock_sdp );
1401 /****************************************************************************
1403 ****************************************************************************/
1404 static void* ThreadSend( void *data )
1407 # define ENOBUFS WSAENOBUFS
1408 # define EAGAIN WSAEWOULDBLOCK
1409 # define EWOULDBLOCK WSAEWOULDBLOCK
1411 sout_stream_id_t *id = data;
1412 unsigned i_caching = id->i_caching;
1416 block_t *out = block_FifoGet( id->p_fifo );
1417 block_cleanup_push (out);
1421 { /* FIXME: this is awfully inefficient */
1422 size_t len = out->i_buffer;
1423 out = block_Realloc( out, 0, len + 10 );
1424 out->i_buffer = len;
1426 int canc = vlc_savecancel ();
1427 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1428 vlc_restorecancel (canc);
1432 msg_Dbg( id->p_stream, "SRTP sending error: %m" );
1433 block_Release( out );
1437 out->i_buffer = len;
1440 mwait (out->i_dts + i_caching);
1445 mwait (out->i_dts + i_caching);
1449 ssize_t len = out->i_buffer;
1450 int canc = vlc_savecancel ();
1452 vlc_mutex_lock( &id->lock_sink );
1453 unsigned deadc = 0; /* How many dead sockets? */
1454 int deadv[id->sinkc]; /* Dead sockets list */
1456 for( int i = 0; i < id->sinkc; i++ )
1459 if( !id->srtp ) /* FIXME: SRTCP support */
1461 SendRTCP( id->sinkv[i].rtcp, out );
1463 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1464 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1465 && net_errno != ENOBUFS && net_errno != ENOMEM )
1468 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1469 &type, &(socklen_t){ sizeof(type) });
1470 if( type == SOCK_DGRAM )
1471 /* ICMP soft error: ignore and retry */
1472 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1474 /* Broken connection */
1475 deadv[deadc++] = id->sinkv[i].rtp_fd;
1478 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1479 vlc_mutex_unlock( &id->lock_sink );
1480 block_Release( out );
1482 for( unsigned i = 0; i < deadc; i++ )
1484 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1485 rtp_del_sink( id, deadv[i] );
1487 vlc_restorecancel (canc);
1493 /* This thread dequeues incoming connections (DCCP streaming) */
1494 static void *rtp_listen_thread( void *data )
1496 sout_stream_id_t *id = data;
1498 assert( id->listen.fd != NULL );
1502 int fd = net_Accept( id->p_stream, id->listen.fd );
1505 int canc = vlc_savecancel( );
1506 rtp_add_sink( id, fd, true, NULL );
1507 vlc_restorecancel( canc );
1514 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1516 rtp_sink_t sink = { fd, NULL };
1517 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1519 if( sink.rtcp == NULL )
1520 msg_Err( id->p_stream, "RTCP failed!" );
1522 vlc_mutex_lock( &id->lock_sink );
1523 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1525 *seq = id->i_seq_sent_next;
1526 vlc_mutex_unlock( &id->lock_sink );
1530 void rtp_del_sink( sout_stream_id_t *id, int fd )
1532 rtp_sink_t sink = { fd, NULL };
1534 /* NOTE: must be safe to use if fd is not included */
1535 vlc_mutex_lock( &id->lock_sink );
1536 for( int i = 0; i < id->sinkc; i++ )
1538 if (id->sinkv[i].rtp_fd == fd)
1540 sink = id->sinkv[i];
1541 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1545 vlc_mutex_unlock( &id->lock_sink );
1547 CloseRTCP( sink.rtcp );
1548 net_Close( sink.rtp_fd );
1551 uint16_t rtp_get_seq( sout_stream_id_t *id )
1553 /* This will return values for the next packet. */
1556 vlc_mutex_lock( &id->lock_sink );
1557 seq = id->i_seq_sent_next;
1558 vlc_mutex_unlock( &id->lock_sink );
1563 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1564 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1565 * random (although we use the same reference for all the ES as a
1566 * feature). In the VoD case, this function is called independently
1567 * from several parts of the code, so we need to always return the same
1569 static int64_t rtp_init_ts( const vod_media_t *p_media,
1570 const char *psz_vod_session )
1572 if (p_media == NULL || psz_vod_session == NULL)
1576 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1577 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1578 i_ts_init ^= (uintptr_t)p_media;
1579 /* Limit the timestamp to 48 bits, this is enough and allows us
1580 * to stay away from overflows */
1581 i_ts_init &= 0xFFFFFFFFFFFF;
1585 /* Return a timestamp corresponding to packets being sent now, and that
1586 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1587 * Also return the NPT corresponding to this timestamp. If the stream
1588 * output is not started, the initial timestamp that will be used with
1589 * the first packets for NPT=0 is returned instead. */
1590 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1591 const vod_media_t *p_media, const char *psz_vod_session,
1598 p_stream = id->p_stream;
1600 if (p_stream == NULL)
1601 return rtp_init_ts(p_media, psz_vod_session);
1603 sout_stream_sys_t *p_sys = p_stream->p_sys;
1605 vlc_mutex_lock( &p_sys->lock_ts );
1606 i_npt_zero = p_sys->i_npt_zero;
1607 vlc_mutex_unlock( &p_sys->lock_ts );
1609 if( i_npt_zero == VLC_TS_INVALID )
1610 return p_sys->i_pts_zero;
1612 mtime_t now = mdate();
1613 if( now < i_npt_zero )
1614 return p_sys->i_pts_zero;
1616 int64_t npt = now - i_npt_zero;
1620 return p_sys->i_pts_zero + npt;
1623 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1624 int b_marker, int64_t i_pts )
1626 if( !id->b_ts_init )
1628 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1629 vlc_mutex_lock( &p_sys->lock_ts );
1630 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1632 /* This is the first packet of any ES. We initialize the
1633 * NPT=0 time reference, and the offset to match the
1634 * arbitrary PTS reference. */
1635 p_sys->i_npt_zero = i_pts + id->i_caching;
1636 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1638 vlc_mutex_unlock( &p_sys->lock_ts );
1640 /* And in any case this is the first packet of this ES, so we
1641 * initialize the offset for this ES. */
1642 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1643 p_sys->i_pts_offset );
1644 id->b_ts_init = true;
1647 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1650 out->p_buffer[0] = 0x80;
1651 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1652 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1653 out->p_buffer[3] = ( id->i_sequence )&0xff;
1654 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1655 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1656 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1657 out->p_buffer[7] = ( i_timestamp )&0xff;
1659 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1665 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1667 block_FifoPut( id->p_fifo, out );
1671 * @return configured max RTP payload size (including payload type-specific
1672 * headers, excluding RTP and transport headers)
1674 size_t rtp_mtu (const sout_stream_id_t *id)
1676 return id->i_mtu - 12;
1679 /*****************************************************************************
1681 *****************************************************************************/
1683 /** Add an ES to a non-RTP muxed stream */
1684 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1686 sout_input_t *p_input;
1687 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1688 assert( p_mux != NULL );
1690 p_input = sout_MuxAddStream( p_mux, p_fmt );
1691 if( p_input == NULL )
1693 msg_Err( p_stream, "cannot add this stream to the muxer" );
1697 return (sout_stream_id_t *)p_input;
1701 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1704 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1705 assert( p_mux != NULL );
1707 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1712 /** Remove an ES from a non-RTP muxed stream */
1713 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1715 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1716 assert( p_mux != NULL );
1718 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1723 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1724 const block_t *p_buffer )
1726 sout_stream_sys_t *p_sys = p_stream->p_sys;
1727 sout_stream_id_t *id = p_sys->es[0];
1729 int64_t i_dts = p_buffer->i_dts;
1731 uint8_t *p_data = p_buffer->p_buffer;
1732 size_t i_data = p_buffer->i_buffer;
1733 size_t i_max = id->i_mtu - 12;
1735 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1741 /* output complete packet */
1742 if( p_sys->packet &&
1743 p_sys->packet->i_buffer + i_data > i_max )
1745 rtp_packetize_send( id, p_sys->packet );
1746 p_sys->packet = NULL;
1749 if( p_sys->packet == NULL )
1751 /* allocate a new packet */
1752 p_sys->packet = block_New( p_stream, id->i_mtu );
1753 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1754 p_sys->packet->i_dts = i_dts;
1755 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1756 i_dts += p_sys->packet->i_length;
1759 i_size = __MIN( i_data,
1760 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1762 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1765 p_sys->packet->i_buffer += i_size;
1774 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1777 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1783 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1785 p_next = p_buffer->p_next;
1786 block_Release( p_buffer );
1794 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1796 sout_access_out_t *p_grab;
1798 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1799 if( p_grab == NULL )
1802 p_grab->p_module = NULL;
1803 p_grab->psz_access = strdup( "grab" );
1804 p_grab->p_cfg = NULL;
1805 p_grab->psz_path = strdup( "" );
1806 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1807 p_grab->pf_seek = NULL;
1808 p_grab->pf_write = AccessOutGrabberWrite;