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>
51 #include <sys/types.h>
53 #ifdef HAVE_ARPA_INET_H
54 # include <arpa/inet.h>
56 #ifdef HAVE_LINUX_DCCP_H
57 # include <linux/dccp.h>
60 # define IPPROTO_DCCP 33
62 #ifndef IPPROTO_UDPLITE
63 # define IPPROTO_UDPLITE 136
70 /*****************************************************************************
72 *****************************************************************************/
74 #define DEST_TEXT N_("Destination")
75 #define DEST_LONGTEXT N_( \
76 "This is the output URL that will be used." )
77 #define SDP_TEXT N_("SDP")
78 #define SDP_LONGTEXT N_( \
79 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
80 "session will be made available. You must use a url: http://location to " \
81 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
82 "for the SDP to be announced via SAP." )
83 #define SAP_TEXT N_("SAP announcing")
84 #define SAP_LONGTEXT N_("Announce this session with SAP.")
85 #define MUX_TEXT N_("Muxer")
86 #define MUX_LONGTEXT N_( \
87 "This allows you to specify the muxer used for the streaming output. " \
88 "Default is to use no muxer (standard RTP stream)." )
90 #define NAME_TEXT N_("Session name")
91 #define NAME_LONGTEXT N_( \
92 "This is the name of the session that will be announced in the SDP " \
93 "(Session Descriptor)." )
94 #define CAT_TEXT N_("Session category")
95 #define CAT_LONGTEXT N_( \
96 "This allows you to specify a category for the session, " \
97 "that will be announced if you choose to use SAP." )
98 #define DESC_TEXT N_("Session description")
99 #define DESC_LONGTEXT N_( \
100 "This allows you to give a short description with details about the stream, " \
101 "that will be announced in the SDP (Session Descriptor)." )
102 #define URL_TEXT N_("Session URL")
103 #define URL_LONGTEXT N_( \
104 "This allows you to give a URL with more details about the stream " \
105 "(often the website of the streaming organization), that will " \
106 "be announced in the SDP (Session Descriptor)." )
107 #define EMAIL_TEXT N_("Session email")
108 #define EMAIL_LONGTEXT N_( \
109 "This allows you to give a contact mail address for the stream, that will " \
110 "be announced in the SDP (Session Descriptor)." )
111 #define PHONE_TEXT N_("Session phone number")
112 #define PHONE_LONGTEXT N_( \
113 "This allows you to give a contact telephone number for the stream, that will " \
114 "be announced in the SDP (Session Descriptor)." )
116 #define PORT_TEXT N_("Port")
117 #define PORT_LONGTEXT N_( \
118 "This allows you to specify the base port for the RTP streaming." )
119 #define PORT_AUDIO_TEXT N_("Audio port")
120 #define PORT_AUDIO_LONGTEXT N_( \
121 "This allows you to specify the default audio port for the RTP streaming." )
122 #define PORT_VIDEO_TEXT N_("Video port")
123 #define PORT_VIDEO_LONGTEXT N_( \
124 "This allows you to specify the default video port for the RTP streaming." )
126 #define TTL_TEXT N_("Hop limit (TTL)")
127 #define TTL_LONGTEXT N_( \
128 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
129 "the multicast packets sent by the stream output (-1 = use operating " \
130 "system built-in default).")
132 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
133 #define RTCP_MUX_LONGTEXT N_( \
134 "This sends and receives RTCP packet multiplexed over the same port " \
137 #define CACHING_TEXT N_("Caching value (ms)")
138 #define CACHING_LONGTEXT N_( \
139 "Default caching value for outbound RTP streams. This " \
140 "value should be set in milliseconds." )
142 #define PROTO_TEXT N_("Transport protocol")
143 #define PROTO_LONGTEXT N_( \
144 "This selects which transport protocol to use for RTP." )
146 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
147 #define SRTP_KEY_LONGTEXT N_( \
148 "RTP packets will be integrity-protected and ciphered "\
149 "with this Secure RTP master shared secret key. "\
150 "This must be a 32-character-long hexadecimal string.")
152 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
153 #define SRTP_SALT_LONGTEXT N_( \
154 "Secure RTP requires a (non-secret) master salt value. " \
155 "This must be a 28-character-long hexadecimal string.")
157 static const char *const ppsz_protos[] = {
158 "dccp", "sctp", "tcp", "udp", "udplite",
161 static const char *const ppsz_protocols[] = {
162 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
165 #define RFC3016_TEXT N_("MP4A LATM")
166 #define RFC3016_LONGTEXT N_( \
167 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
169 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
170 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
171 "not receiving any RTSP request for this long. Setting it to a " \
172 "negative value or zero disables timeouts. The default is 60 (one " \
175 #define RTSP_USER_TEXT N_("Username")
176 #define RTSP_USER_LONGTEXT N_("User name that will be " \
177 "requested to access the stream." )
178 #define RTSP_PASS_TEXT N_("Password")
179 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
180 "requested to access the stream." )
182 static int Open ( vlc_object_t * );
183 static void Close( vlc_object_t * );
185 #define SOUT_CFG_PREFIX "sout-rtp-"
186 #define MAX_EMPTY_BLOCKS 200
189 set_shortname( N_("RTP"))
190 set_description( N_("RTP stream output") )
191 set_capability( "sout stream", 0 )
192 add_shortcut( "rtp", "vod" )
193 set_category( CAT_SOUT )
194 set_subcategory( SUBCAT_SOUT_STREAM )
196 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
197 DEST_LONGTEXT, true )
198 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
200 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
202 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
205 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
206 NAME_LONGTEXT, true )
207 add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_LONGTEXT, true )
208 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
209 DESC_LONGTEXT, true )
210 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
212 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
213 EMAIL_LONGTEXT, true )
214 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
215 PHONE_LONGTEXT, true )
217 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
218 PROTO_LONGTEXT, false )
219 change_string_list( ppsz_protos, ppsz_protocols )
220 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
221 PORT_LONGTEXT, true )
222 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
223 PORT_AUDIO_LONGTEXT, true )
224 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
225 PORT_VIDEO_LONGTEXT, true )
227 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
229 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
230 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
231 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
232 CACHING_TEXT, CACHING_LONGTEXT, true )
235 add_string( SOUT_CFG_PREFIX "key", "",
236 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
237 add_string( SOUT_CFG_PREFIX "salt", "",
238 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
241 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
242 RFC3016_LONGTEXT, false )
244 set_callbacks( Open, Close )
247 set_shortname( N_("RTSP VoD" ) )
248 set_description( N_("RTSP VoD server") )
249 set_category( CAT_SOUT )
250 set_subcategory( SUBCAT_SOUT_VOD )
251 set_capability( "vod server", 10 )
252 set_callbacks( OpenVoD, CloseVoD )
253 add_shortcut( "rtsp" )
254 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
255 RTSP_TIMEOUT_LONGTEXT, true )
256 add_string( "sout-rtsp-user", "",
257 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
258 add_password( "sout-rtsp-pwd", "",
259 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
263 /*****************************************************************************
264 * Exported prototypes
265 *****************************************************************************/
266 static const char *const ppsz_sout_options[] = {
267 "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
268 "mux", "sap", "description", "url", "email", "phone",
269 "proto", "rtcp-mux", "caching",
276 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
277 static int Del ( sout_stream_t *, sout_stream_id_t * );
278 static int Send( sout_stream_t *, sout_stream_id_t *,
280 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
281 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
282 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
285 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
286 static void* ThreadSend( void * );
287 static void *rtp_listen_thread( void * );
289 static void SDPHandleUrl( sout_stream_t *, const char * );
291 static int SapSetup( sout_stream_t *p_stream );
292 static int FileSetup( sout_stream_t *p_stream );
293 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
295 static int64_t rtp_init_ts( const vod_media_t *p_media,
296 const char *psz_vod_session );
298 struct sout_stream_sys_t
302 vlc_mutex_t lock_sdp;
309 session_descriptor_t *p_session;
312 httpd_host_t *p_httpd_host;
313 httpd_file_t *p_httpd_file;
318 /* RTSP NPT and timestamp computations */
319 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
320 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
321 int64_t i_pts_offset; /* matches actual PTS to prediction */
325 char *psz_destination;
327 uint16_t i_port_audio;
328 uint16_t i_port_video;
334 vod_media_t *p_vod_media;
335 char *psz_vod_session;
337 /* in case we do TS/PS over rtp */
339 sout_access_out_t *p_grab;
345 sout_stream_id_t **es;
348 typedef struct rtp_sink_t
354 struct sout_stream_id_t
356 sout_stream_t *p_stream;
361 uint32_t i_ts_offset;
365 uint16_t i_seq_sent_next;
368 rtp_format_t rtp_fmt;
371 /* Packetizer specific fields */
374 srtp_session_t *srtp;
379 vlc_mutex_t lock_sink;
382 rtsp_stream_id_t *rtsp_id;
388 block_fifo_t *p_fifo;
392 /*****************************************************************************
394 *****************************************************************************/
395 static int Open( vlc_object_t *p_this )
397 sout_stream_t *p_stream = (sout_stream_t*)p_this;
398 sout_stream_sys_t *p_sys = NULL;
399 config_chain_t *p_cfg = NULL;
403 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
404 ppsz_sout_options, p_stream->p_cfg );
406 p_sys = malloc( sizeof( sout_stream_sys_t ) );
410 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
412 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
413 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
414 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
415 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
417 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
419 msg_Err( p_stream, "audio and video RTP port must be distinct" );
420 free( p_sys->psz_destination );
425 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
427 if( !strcmp( p_cfg->psz_name, "sdp" )
428 && ( p_cfg->psz_value != NULL )
429 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
437 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
440 if( !strncasecmp( psz, "rtsp:", 5 ) )
446 /* Transport protocol */
447 p_sys->proto = IPPROTO_UDP;
448 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
450 if ((psz == NULL) || !strcasecmp (psz, "udp"))
451 (void)0; /* default */
453 if (!strcasecmp (psz, "dccp"))
455 p_sys->proto = IPPROTO_DCCP;
456 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
460 if (!strcasecmp (psz, "sctp"))
462 p_sys->proto = IPPROTO_TCP;
463 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
468 if (!strcasecmp (psz, "tcp"))
470 p_sys->proto = IPPROTO_TCP;
471 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
475 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
476 p_sys->proto = IPPROTO_UDPLITE;
478 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
481 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
483 p_sys->p_vod_media = NULL;
484 p_sys->psz_vod_session = NULL;
486 if (! strcmp(p_stream->psz_name, "vod"))
488 /* The VLM stops all instances before deleting a media, so this
489 * reference will remain valid during the lifetime of the rtp
491 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
493 if (p_sys->p_vod_media != NULL)
495 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
496 if (p_sys->psz_vod_session == NULL)
498 msg_Err(p_stream, "missing VoD session");
503 const char *mux = vod_get_mux(p_sys->p_vod_media);
504 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
508 if( p_sys->psz_destination == NULL && !b_rtsp
509 && p_sys->p_vod_media == NULL )
511 msg_Err( p_stream, "missing destination and not in RTSP mode" );
516 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
519 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
520 var_SetInteger( p_stream, "ttl", i_ttl );
523 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
525 /* NPT=0 time will be determined when we packetize the first packet
526 * (of any ES). But we want to be able to report rtptime in RTSP
527 * without waiting (and already did in the VoD case). So until then,
528 * we use an arbitrary reference PTS for timestamp computations, and
529 * then actual PTS will catch up using offsets. */
530 p_sys->i_npt_zero = VLC_TS_INVALID;
531 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
532 p_sys->psz_vod_session);
536 p_sys->psz_sdp = NULL;
538 p_sys->b_export_sap = false;
539 p_sys->p_session = NULL;
540 p_sys->psz_sdp_file = NULL;
542 p_sys->p_httpd_host = NULL;
543 p_sys->p_httpd_file = NULL;
545 p_stream->p_sys = p_sys;
547 vlc_mutex_init( &p_sys->lock_sdp );
548 vlc_mutex_init( &p_sys->lock_ts );
549 vlc_mutex_init( &p_sys->lock_es );
551 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
554 /* Check muxer type */
555 if( strncasecmp( psz, "ps", 2 )
556 && strncasecmp( psz, "mpeg1", 5 )
557 && strncasecmp( psz, "ts", 2 ) )
559 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
561 vlc_mutex_destroy( &p_sys->lock_sdp );
562 vlc_mutex_destroy( &p_sys->lock_ts );
563 vlc_mutex_destroy( &p_sys->lock_es );
564 free( p_sys->psz_vod_session );
565 free( p_sys->psz_destination );
570 p_sys->p_grab = GrabberCreate( p_stream );
571 p_sys->p_mux = sout_MuxNew( p_stream->p_sout, psz, p_sys->p_grab );
574 if( p_sys->p_mux == NULL )
576 msg_Err( p_stream, "cannot create muxer" );
577 sout_AccessOutDelete( p_sys->p_grab );
578 vlc_mutex_destroy( &p_sys->lock_sdp );
579 vlc_mutex_destroy( &p_sys->lock_ts );
580 vlc_mutex_destroy( &p_sys->lock_es );
581 free( p_sys->psz_vod_session );
582 free( p_sys->psz_destination );
587 p_sys->packet = NULL;
589 p_stream->pf_add = MuxAdd;
590 p_stream->pf_del = MuxDel;
591 p_stream->pf_send = MuxSend;
596 p_sys->p_grab = NULL;
598 p_stream->pf_add = Add;
599 p_stream->pf_del = Del;
600 p_stream->pf_send = Send;
602 p_stream->pace_nocontrol = true;
604 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
605 SDPHandleUrl( p_stream, "sap" );
607 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
610 config_chain_t *p_cfg;
612 SDPHandleUrl( p_stream, psz );
614 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
616 if( !strcmp( p_cfg->psz_name, "sdp" ) )
618 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
621 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
622 if( !strcmp( p_cfg->psz_value, psz ) )
625 SDPHandleUrl( p_stream, p_cfg->psz_value );
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;
654 assert( p_sys->i_es <= 1 );
656 sout_MuxDelete( p_sys->p_mux );
657 if ( p_sys->i_es > 0 )
658 Del( p_stream, p_sys->es[0] );
659 sout_AccessOutDelete( p_sys->p_grab );
663 block_Release( p_sys->packet );
667 if( p_sys->rtsp != NULL )
668 RtspUnsetup( p_sys->rtsp );
670 vlc_mutex_destroy( &p_sys->lock_sdp );
671 vlc_mutex_destroy( &p_sys->lock_ts );
672 vlc_mutex_destroy( &p_sys->lock_es );
674 if( p_sys->p_httpd_file )
675 httpd_FileDelete( p_sys->p_httpd_file );
677 if( p_sys->p_httpd_host )
678 httpd_HostDelete( p_sys->p_httpd_host );
680 free( p_sys->psz_sdp );
682 if( p_sys->psz_sdp_file != NULL )
684 unlink( p_sys->psz_sdp_file );
685 free( p_sys->psz_sdp_file );
687 free( p_sys->psz_vod_session );
688 free( p_sys->psz_destination );
692 /*****************************************************************************
694 *****************************************************************************/
695 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
697 sout_stream_sys_t *p_sys = p_stream->p_sys;
700 vlc_UrlParse( &url, psz_url, 0 );
701 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
703 if( p_sys->p_httpd_file )
705 msg_Err( p_stream, "you can use sdp=http:// only once" );
709 if( HttpSetup( p_stream, &url ) )
711 msg_Err( p_stream, "cannot export SDP as HTTP" );
714 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
716 if( p_sys->rtsp != NULL )
718 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
722 if( url.psz_host != NULL && *url.psz_host )
724 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
725 "multiple-host configurations, use at your own risks.",
727 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
728 "command line instead." );
730 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
731 var_SetString( p_stream, "rtsp-host", url.psz_host );
733 if( url.i_port != 0 )
735 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
736 "the command line instead.", url.i_port ); */
738 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
739 var_SetInteger( p_stream, "rtsp-port", url.i_port );
742 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
743 if( p_sys->rtsp == NULL )
744 msg_Err( p_stream, "cannot export SDP as RTSP" );
746 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
747 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
749 p_sys->b_export_sap = true;
750 SapSetup( p_stream );
752 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
754 if( p_sys->psz_sdp_file != NULL )
756 msg_Err( p_stream, "you can use sdp=file:// only once" );
759 p_sys->psz_sdp_file = make_path( psz_url );
760 if( p_sys->psz_sdp_file == NULL )
762 FileSetup( p_stream );
766 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
771 vlc_UrlClean( &url );
774 /*****************************************************************************
776 *****************************************************************************/
778 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
780 sout_stream_sys_t *p_sys = p_stream->p_sys;
781 char *psz_sdp = NULL;
782 struct sockaddr_storage dst;
786 * When we have a fixed destination (typically when we do multicast),
787 * we need to put the actual port numbers in the SDP.
788 * When there is no fixed destination, we only support RTSP unicast
789 * on-demand setup, so we should rather let the clients decide which ports
791 * When there is both a fixed destination and RTSP unicast, we need to
792 * put port numbers used by the fixed destination, otherwise the SDP would
793 * become totally incorrect for multicast use. It should be noted that
794 * port numbers from SDP with RTSP are only "recommendation" from the
795 * server to the clients (per RFC2326), so only broken clients will fail
796 * to handle this properly. There is no solution but to use two differents
797 * output chain with two different RTSP URLs if you need to handle this
802 vlc_mutex_lock( &p_sys->lock_es );
803 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
804 goto out; /* hmm... */
806 if( p_sys->psz_destination != NULL )
810 /* Oh boy, this is really ugly! */
811 dstlen = sizeof( dst );
812 if( p_sys->es[0]->listen.fd != NULL )
813 getsockname( p_sys->es[0]->listen.fd[0],
814 (struct sockaddr *)&dst, &dstlen );
816 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
817 (struct sockaddr *)&dst, &dstlen );
823 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
824 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
825 && rtsp_url[7] == '[';
827 /* Dummy destination address for RTSP */
828 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
829 : sizeof( struct sockaddr_in );
830 memset (&dst, 0, dstlen);
831 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
837 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
838 NULL, 0, (struct sockaddr *)&dst, dstlen );
839 if( psz_sdp == NULL )
842 /* TODO: a=source-filter */
843 if( p_sys->rtcp_mux )
844 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
846 if( rtsp_url != NULL )
847 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
849 const char *proto = "RTP/AVP"; /* protocol */
850 if( rtsp_url == NULL )
852 switch( p_sys->proto )
857 proto = "TCP/RTP/AVP";
860 proto = "DCCP/RTP/AVP";
862 case IPPROTO_UDPLITE:
867 for( i = 0; i < p_sys->i_es; i++ )
869 sout_stream_id_t *id = p_sys->es[i];
870 rtp_format_t *rtp_fmt = &id->rtp_fmt;
871 const char *mime_major; /* major MIME type */
873 switch( rtp_fmt->cat )
876 mime_major = "video";
879 mime_major = "audio";
888 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
889 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
890 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
893 /* cf RFC4566 §5.14 */
894 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
895 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
897 if( rtsp_url != NULL )
899 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
900 if( track_url != NULL )
902 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
908 if( id->listen.fd != NULL )
909 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
910 if( p_sys->proto == IPPROTO_DCCP )
911 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
913 toupper( (unsigned char)mime_major[0] ) );
917 vlc_mutex_unlock( &p_sys->lock_es );
921 /*****************************************************************************
923 *****************************************************************************/
926 * Shrink the MTU down to a fixed packetization time (for audio).
929 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
931 /* Samples per second */
932 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
933 bytes *= id->rtp_fmt.channels;
936 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
937 id->i_mtu = 12 + spl;
938 else /* MTU is too small for ptime, align to a sample boundary */
939 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
942 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
944 /* This is an overflow-proof way of doing:
945 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
947 * NOTE: this plays nice with offsets because the (equivalent)
948 * calculations are linear. */
949 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
950 return q.quot * (int64_t)i_clock_rate
951 + q.rem * (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 int val = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1043 msg_Err (p_stream, "bad SRTP key/salt combination (%s)",
1044 vlc_strerror_c(val));
1047 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1051 id->i_seq_sent_next = id->i_sequence;
1054 if( p_sys->psz_destination != NULL )
1056 /* Choose the port */
1057 uint16_t i_port = 0;
1061 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1062 i_port = p_sys->i_port_audio;
1064 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1065 i_port = p_sys->i_port_video;
1067 /* We do not need the ES lock (p_sys->lock_es) here, because
1068 * this is the only one thread that can *modify* the ES table.
1069 * The ES lock protects the other threads from our modifications
1070 * (TAB_APPEND, TAB_REMOVE). */
1071 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1072 if (i_port == p_sys->es[i]->i_port)
1073 i_port = 0; /* Port already in use! */
1074 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1078 msg_Err (p_stream, "too many RTP elementary streams");
1082 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1083 if (p == p_sys->es[i]->i_port)
1087 id->i_port = i_port;
1089 int type = SOCK_STREAM;
1091 switch( p_sys->proto )
1097 switch (id->rtp_fmt.cat)
1099 case VIDEO_ES: code = "RTPV"; break;
1100 case AUDIO_ES: code = "RTPARTPV"; break;
1101 case SPU_ES: code = "RTPTRTPV"; break;
1102 default: code = "RTPORTPV"; break;
1104 var_SetString (p_stream, "dccp-service", code);
1106 } /* fall through */
1109 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1110 p_sys->psz_destination, i_port,
1111 type, p_sys->proto );
1112 if( id->listen.fd == NULL )
1114 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1117 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1118 VLC_THREAD_PRIORITY_LOW ) )
1120 net_ListenClose( id->listen.fd );
1121 id->listen.fd = NULL;
1128 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1129 i_port, -1, p_sys->proto );
1132 msg_Err( p_stream, "cannot create RTP socket" );
1135 /* Ignore any unexpected incoming packet (including RTCP-RR
1136 * packets in case of rtcp-mux) */
1137 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1139 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1140 /* FIXME: test if this is multicast */
1147 switch( p_fmt->i_codec )
1149 case VLC_CODEC_MULAW:
1150 case VLC_CODEC_ALAW:
1152 rtp_set_ptime (id, 20, 1);
1154 case VLC_CODEC_S16B:
1155 case VLC_CODEC_S16L:
1156 rtp_set_ptime (id, 20, 2);
1158 case VLC_CODEC_S24B:
1159 rtp_set_ptime (id, 20, 3);
1165 #if 0 /* No payload formats sets this at the moment */
1168 cscov += 8 /* UDP */ + 12 /* RTP */;
1170 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1173 vlc_mutex_lock( &p_sys->lock_ts );
1174 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1175 vlc_mutex_unlock( &p_sys->lock_ts );
1177 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1178 p_sys->i_pts_offset );
1180 if( p_sys->rtsp != NULL )
1181 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1182 id->rtp_fmt.clock_rate, mcast_fd );
1184 id->p_fifo = block_FifoNew();
1185 if( unlikely(id->p_fifo == NULL) )
1187 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1189 block_FifoRelease( id->p_fifo );
1194 /* Update p_sys context */
1195 vlc_mutex_lock( &p_sys->lock_es );
1196 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1197 vlc_mutex_unlock( &p_sys->lock_es );
1199 psz_sdp = SDPGenerate( p_stream, NULL );
1201 vlc_mutex_lock( &p_sys->lock_sdp );
1202 free( p_sys->psz_sdp );
1203 p_sys->psz_sdp = psz_sdp;
1204 vlc_mutex_unlock( &p_sys->lock_sdp );
1206 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1208 /* Update SDP (sap/file) */
1209 if( p_sys->b_export_sap ) SapSetup( p_stream );
1210 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1215 Del( p_stream, id );
1219 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1221 sout_stream_sys_t *p_sys = p_stream->p_sys;
1223 vlc_mutex_lock( &p_sys->lock_es );
1224 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1225 vlc_mutex_unlock( &p_sys->lock_es );
1227 if( likely(id->p_fifo != NULL) )
1229 vlc_cancel( id->thread );
1230 vlc_join( id->thread, NULL );
1231 block_FifoRelease( id->p_fifo );
1234 free( id->rtp_fmt.fmtp );
1236 if (p_sys->p_vod_media != NULL)
1237 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1239 RtspDelId( p_sys->rtsp, id->rtsp_id );
1240 if( id->listen.fd != NULL )
1242 vlc_cancel( id->listen.thread );
1243 vlc_join( id->listen.thread, NULL );
1244 net_ListenClose( id->listen.fd );
1246 /* Delete remaining sinks (incoming connections or explicit
1248 while( id->sinkc > 0 )
1249 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1251 if( id->srtp != NULL )
1252 srtp_destroy( id->srtp );
1255 vlc_mutex_destroy( &id->lock_sink );
1257 /* Update SDP (sap/file) */
1258 if( p_sys->b_export_sap ) SapSetup( p_stream );
1259 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1265 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1270 assert( p_stream->p_sys->p_mux == NULL );
1273 while( p_buffer != NULL )
1275 p_next = p_buffer->p_next;
1277 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1278 * as the first packet of the stream */
1279 if (id->b_first_packet)
1281 id->b_first_packet = false;
1282 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1283 !strcmp(id->rtp_fmt.ptname, "theora"))
1284 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1288 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1291 block_Release( p_buffer );
1297 /****************************************************************************
1299 ****************************************************************************/
1300 static int SapSetup( sout_stream_t *p_stream )
1302 sout_stream_sys_t *p_sys = p_stream->p_sys;
1304 /* Remove the previous session */
1305 if( p_sys->p_session != NULL)
1307 sout_AnnounceUnRegister( p_stream, p_sys->p_session);
1308 p_sys->p_session = NULL;
1311 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1312 p_sys->p_session = sout_AnnounceRegisterSDP( p_stream,
1314 p_sys->psz_destination );
1319 /****************************************************************************
1321 ****************************************************************************/
1322 static int FileSetup( sout_stream_t *p_stream )
1324 sout_stream_sys_t *p_sys = p_stream->p_sys;
1327 if( p_sys->psz_sdp == NULL )
1328 return VLC_EGENERIC; /* too early */
1330 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1332 msg_Err( p_stream, "cannot open file '%s' (%s)",
1333 p_sys->psz_sdp_file, vlc_strerror_c(errno) );
1334 return VLC_EGENERIC;
1337 fputs( p_sys->psz_sdp, f );
1343 /****************************************************************************
1345 ****************************************************************************/
1346 static int HttpCallback( httpd_file_sys_t *p_args,
1347 httpd_file_t *, uint8_t *p_request,
1348 uint8_t **pp_data, int *pi_data );
1350 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1352 sout_stream_sys_t *p_sys = p_stream->p_sys;
1354 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1355 if( p_sys->p_httpd_host )
1357 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1358 url->psz_path ? url->psz_path : "/",
1361 HttpCallback, (void*)p_sys );
1363 if( p_sys->p_httpd_file == NULL )
1365 return VLC_EGENERIC;
1370 static int HttpCallback( httpd_file_sys_t *p_args,
1371 httpd_file_t *f, uint8_t *p_request,
1372 uint8_t **pp_data, int *pi_data )
1374 VLC_UNUSED(f); VLC_UNUSED(p_request);
1375 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1377 vlc_mutex_lock( &p_sys->lock_sdp );
1378 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1380 *pi_data = strlen( p_sys->psz_sdp );
1381 *pp_data = malloc( *pi_data );
1382 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1389 vlc_mutex_unlock( &p_sys->lock_sdp );
1394 /****************************************************************************
1396 ****************************************************************************/
1397 static void* ThreadSend( void *data )
1400 # define ENOBUFS WSAENOBUFS
1401 # define EAGAIN WSAEWOULDBLOCK
1402 # define EWOULDBLOCK WSAEWOULDBLOCK
1404 sout_stream_id_t *id = data;
1405 unsigned i_caching = id->i_caching;
1409 block_t *out = block_FifoGet( id->p_fifo );
1410 block_cleanup_push (out);
1414 { /* FIXME: this is awfully inefficient */
1415 size_t len = out->i_buffer;
1416 out = block_Realloc( out, 0, len + 10 );
1417 out->i_buffer = len;
1419 int canc = vlc_savecancel ();
1420 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1421 vlc_restorecancel (canc);
1424 msg_Dbg( id->p_stream, "SRTP sending error: %s",
1425 vlc_strerror_c(val) );
1426 block_Release( out );
1430 out->i_buffer = len;
1433 mwait (out->i_dts + i_caching);
1438 mwait (out->i_dts + i_caching);
1442 ssize_t len = out->i_buffer;
1443 int canc = vlc_savecancel ();
1445 vlc_mutex_lock( &id->lock_sink );
1446 unsigned deadc = 0; /* How many dead sockets? */
1447 int deadv[id->sinkc]; /* Dead sockets list */
1449 for( int i = 0; i < id->sinkc; i++ )
1452 if( !id->srtp ) /* FIXME: SRTCP support */
1454 SendRTCP( id->sinkv[i].rtcp, out );
1456 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1457 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1458 && net_errno != ENOBUFS && net_errno != ENOMEM )
1461 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1462 &type, &(socklen_t){ sizeof(type) });
1463 if( type == SOCK_DGRAM )
1464 /* ICMP soft error: ignore and retry */
1465 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1467 /* Broken connection */
1468 deadv[deadc++] = id->sinkv[i].rtp_fd;
1471 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1472 vlc_mutex_unlock( &id->lock_sink );
1473 block_Release( out );
1475 for( unsigned i = 0; i < deadc; i++ )
1477 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1478 rtp_del_sink( id, deadv[i] );
1480 vlc_restorecancel (canc);
1486 /* This thread dequeues incoming connections (DCCP streaming) */
1487 static void *rtp_listen_thread( void *data )
1489 sout_stream_id_t *id = data;
1491 assert( id->listen.fd != NULL );
1495 int fd = net_Accept( id->p_stream, id->listen.fd );
1498 int canc = vlc_savecancel( );
1499 rtp_add_sink( id, fd, true, NULL );
1500 vlc_restorecancel( canc );
1507 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1509 rtp_sink_t sink = { fd, NULL };
1510 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1512 if( sink.rtcp == NULL )
1513 msg_Err( id->p_stream, "RTCP failed!" );
1515 vlc_mutex_lock( &id->lock_sink );
1516 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1518 *seq = id->i_seq_sent_next;
1519 vlc_mutex_unlock( &id->lock_sink );
1523 void rtp_del_sink( sout_stream_id_t *id, int fd )
1525 rtp_sink_t sink = { fd, NULL };
1527 /* NOTE: must be safe to use if fd is not included */
1528 vlc_mutex_lock( &id->lock_sink );
1529 for( int i = 0; i < id->sinkc; i++ )
1531 if (id->sinkv[i].rtp_fd == fd)
1533 sink = id->sinkv[i];
1534 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1538 vlc_mutex_unlock( &id->lock_sink );
1540 CloseRTCP( sink.rtcp );
1541 net_Close( sink.rtp_fd );
1544 uint16_t rtp_get_seq( sout_stream_id_t *id )
1546 /* This will return values for the next packet. */
1549 vlc_mutex_lock( &id->lock_sink );
1550 seq = id->i_seq_sent_next;
1551 vlc_mutex_unlock( &id->lock_sink );
1556 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1557 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1558 * random (although we use the same reference for all the ES as a
1559 * feature). In the VoD case, this function is called independently
1560 * from several parts of the code, so we need to always return the same
1562 static int64_t rtp_init_ts( const vod_media_t *p_media,
1563 const char *psz_vod_session )
1565 if (p_media == NULL || psz_vod_session == NULL)
1569 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1570 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1571 i_ts_init ^= (uintptr_t)p_media;
1572 /* Limit the timestamp to 48 bits, this is enough and allows us
1573 * to stay away from overflows */
1574 i_ts_init &= 0xFFFFFFFFFFFF;
1578 /* Return a timestamp corresponding to packets being sent now, and that
1579 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1580 * Also return the NPT corresponding to this timestamp. If the stream
1581 * output is not started, the initial timestamp that will be used with
1582 * the first packets for NPT=0 is returned instead. */
1583 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
1584 const vod_media_t *p_media, const char *psz_vod_session,
1591 p_stream = id->p_stream;
1593 if (p_stream == NULL)
1594 return rtp_init_ts(p_media, psz_vod_session);
1596 sout_stream_sys_t *p_sys = p_stream->p_sys;
1598 vlc_mutex_lock( &p_sys->lock_ts );
1599 i_npt_zero = p_sys->i_npt_zero;
1600 vlc_mutex_unlock( &p_sys->lock_ts );
1602 if( i_npt_zero == VLC_TS_INVALID )
1603 return p_sys->i_pts_zero;
1605 mtime_t now = mdate();
1606 if( now < i_npt_zero )
1607 return p_sys->i_pts_zero;
1609 int64_t npt = now - i_npt_zero;
1613 return p_sys->i_pts_zero + npt;
1616 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1617 int b_marker, int64_t i_pts )
1619 if( !id->b_ts_init )
1621 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1622 vlc_mutex_lock( &p_sys->lock_ts );
1623 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1625 /* This is the first packet of any ES. We initialize the
1626 * NPT=0 time reference, and the offset to match the
1627 * arbitrary PTS reference. */
1628 p_sys->i_npt_zero = i_pts + id->i_caching;
1629 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1631 vlc_mutex_unlock( &p_sys->lock_ts );
1633 /* And in any case this is the first packet of this ES, so we
1634 * initialize the offset for this ES. */
1635 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1636 p_sys->i_pts_offset );
1637 id->b_ts_init = true;
1640 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1643 out->p_buffer[0] = 0x80;
1644 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1645 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1646 out->p_buffer[3] = ( id->i_sequence )&0xff;
1647 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1648 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1649 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1650 out->p_buffer[7] = ( i_timestamp )&0xff;
1652 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1658 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1660 block_FifoPut( id->p_fifo, out );
1664 * @return configured max RTP payload size (including payload type-specific
1665 * headers, excluding RTP and transport headers)
1667 size_t rtp_mtu (const sout_stream_id_t *id)
1669 return id->i_mtu - 12;
1672 /*****************************************************************************
1674 *****************************************************************************/
1676 /** Add an ES to a non-RTP muxed stream */
1677 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1679 sout_input_t *p_input;
1680 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1681 assert( p_mux != NULL );
1683 p_input = sout_MuxAddStream( p_mux, p_fmt );
1684 if( p_input == NULL )
1686 msg_Err( p_stream, "cannot add this stream to the muxer" );
1690 return (sout_stream_id_t *)p_input;
1694 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1697 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1698 assert( p_mux != NULL );
1700 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1705 /** Remove an ES from a non-RTP muxed stream */
1706 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1708 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1709 assert( p_mux != NULL );
1711 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1716 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1717 const block_t *p_buffer )
1719 sout_stream_sys_t *p_sys = p_stream->p_sys;
1720 sout_stream_id_t *id = p_sys->es[0];
1722 int64_t i_dts = p_buffer->i_dts;
1724 uint8_t *p_data = p_buffer->p_buffer;
1725 size_t i_data = p_buffer->i_buffer;
1726 size_t i_max = id->i_mtu - 12;
1728 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1734 /* output complete packet */
1735 if( p_sys->packet &&
1736 p_sys->packet->i_buffer + i_data > i_max )
1738 rtp_packetize_send( id, p_sys->packet );
1739 p_sys->packet = NULL;
1742 if( p_sys->packet == NULL )
1744 /* allocate a new packet */
1745 p_sys->packet = block_Alloc( id->i_mtu );
1746 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1747 p_sys->packet->i_dts = i_dts;
1748 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1749 i_dts += p_sys->packet->i_length;
1752 i_size = __MIN( i_data,
1753 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1755 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1758 p_sys->packet->i_buffer += i_size;
1767 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1770 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1776 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1778 p_next = p_buffer->p_next;
1779 block_Release( p_buffer );
1787 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1789 sout_access_out_t *p_grab;
1791 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1792 if( p_grab == NULL )
1795 p_grab->p_module = NULL;
1796 p_grab->psz_access = strdup( "grab" );
1797 p_grab->p_cfg = NULL;
1798 p_grab->psz_path = strdup( "" );
1799 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1800 p_grab->pf_seek = NULL;
1801 p_grab->pf_write = AccessOutGrabberWrite;