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_("Username 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_sys_t *Add( sout_stream_t *, const es_format_t * );
277 static void Del ( sout_stream_t *, sout_stream_id_sys_t * );
278 static int Send( sout_stream_t *, sout_stream_id_sys_t *,
280 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *, const es_format_t * );
281 static void MuxDel ( sout_stream_t *, sout_stream_id_sys_t * );
282 static int MuxSend( sout_stream_t *, sout_stream_id_sys_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_sys_t **es;
348 typedef struct rtp_sink_t
354 struct sout_stream_id_sys_t
356 sout_stream_t *p_stream;
358 /* For RFC 4175, seqnum is extended to 32-bits */
362 uint32_t i_ts_offset;
366 uint16_t i_seq_sent_next;
369 rtp_format_t rtp_fmt;
372 /* Packetizer specific fields */
375 srtp_session_t *srtp;
380 vlc_mutex_t lock_sink;
383 rtsp_stream_id_t *rtsp_id;
389 block_fifo_t *p_fifo;
393 /*****************************************************************************
395 *****************************************************************************/
396 static int Open( vlc_object_t *p_this )
398 sout_stream_t *p_stream = (sout_stream_t*)p_this;
399 sout_stream_sys_t *p_sys = NULL;
400 config_chain_t *p_cfg = NULL;
404 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
405 ppsz_sout_options, p_stream->p_cfg );
407 p_sys = malloc( sizeof( sout_stream_sys_t ) );
411 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
413 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
414 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
415 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
416 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
418 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
420 msg_Err( p_stream, "audio and video RTP port must be distinct" );
421 free( p_sys->psz_destination );
426 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
428 if( !strcmp( p_cfg->psz_name, "sdp" )
429 && ( p_cfg->psz_value != NULL )
430 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
438 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
441 if( !strncasecmp( psz, "rtsp:", 5 ) )
447 /* Transport protocol */
448 p_sys->proto = IPPROTO_UDP;
449 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
451 if ((psz == NULL) || !strcasecmp (psz, "udp"))
452 (void)0; /* default */
454 if (!strcasecmp (psz, "dccp"))
456 p_sys->proto = IPPROTO_DCCP;
457 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
461 if (!strcasecmp (psz, "sctp"))
463 p_sys->proto = IPPROTO_TCP;
464 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
469 if (!strcasecmp (psz, "tcp"))
471 p_sys->proto = IPPROTO_TCP;
472 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
476 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
477 p_sys->proto = IPPROTO_UDPLITE;
479 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
482 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
484 p_sys->p_vod_media = NULL;
485 p_sys->psz_vod_session = NULL;
487 if (! strcmp(p_stream->psz_name, "vod"))
489 /* The VLM stops all instances before deleting a media, so this
490 * reference will remain valid during the lifetime of the rtp
492 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
494 if (p_sys->p_vod_media != NULL)
496 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
497 if (p_sys->psz_vod_session == NULL)
499 msg_Err(p_stream, "missing VoD session");
504 const char *mux = vod_get_mux(p_sys->p_vod_media);
505 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
509 if( p_sys->psz_destination == NULL && !b_rtsp
510 && p_sys->p_vod_media == NULL )
512 msg_Err( p_stream, "missing destination and not in RTSP mode" );
517 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
520 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
521 var_SetInteger( p_stream, "ttl", i_ttl );
524 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
526 /* NPT=0 time will be determined when we packetize the first packet
527 * (of any ES). But we want to be able to report rtptime in RTSP
528 * without waiting (and already did in the VoD case). So until then,
529 * we use an arbitrary reference PTS for timestamp computations, and
530 * then actual PTS will catch up using offsets. */
531 p_sys->i_npt_zero = VLC_TS_INVALID;
532 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
533 p_sys->psz_vod_session);
537 p_sys->psz_sdp = NULL;
539 p_sys->b_export_sap = false;
540 p_sys->p_session = NULL;
541 p_sys->psz_sdp_file = NULL;
543 p_sys->p_httpd_host = NULL;
544 p_sys->p_httpd_file = NULL;
546 p_stream->p_sys = p_sys;
548 vlc_mutex_init( &p_sys->lock_sdp );
549 vlc_mutex_init( &p_sys->lock_ts );
550 vlc_mutex_init( &p_sys->lock_es );
552 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
555 /* Check muxer type */
556 if( strncasecmp( psz, "ps", 2 )
557 && strncasecmp( psz, "mpeg1", 5 )
558 && strncasecmp( psz, "ts", 2 ) )
560 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
562 vlc_mutex_destroy( &p_sys->lock_sdp );
563 vlc_mutex_destroy( &p_sys->lock_ts );
564 vlc_mutex_destroy( &p_sys->lock_es );
565 free( p_sys->psz_vod_session );
566 free( p_sys->psz_destination );
571 p_sys->p_grab = GrabberCreate( p_stream );
572 p_sys->p_mux = sout_MuxNew( p_stream->p_sout, psz, p_sys->p_grab );
575 if( p_sys->p_mux == NULL )
577 msg_Err( p_stream, "cannot create muxer" );
578 sout_AccessOutDelete( p_sys->p_grab );
579 vlc_mutex_destroy( &p_sys->lock_sdp );
580 vlc_mutex_destroy( &p_sys->lock_ts );
581 vlc_mutex_destroy( &p_sys->lock_es );
582 free( p_sys->psz_vod_session );
583 free( p_sys->psz_destination );
588 p_sys->packet = NULL;
590 p_stream->pf_add = MuxAdd;
591 p_stream->pf_del = MuxDel;
592 p_stream->pf_send = MuxSend;
597 p_sys->p_grab = NULL;
599 p_stream->pf_add = Add;
600 p_stream->pf_del = Del;
601 p_stream->pf_send = Send;
603 p_stream->pace_nocontrol = true;
605 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
606 SDPHandleUrl( p_stream, "sap" );
608 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
611 config_chain_t *p_cfg;
613 SDPHandleUrl( p_stream, psz );
615 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
617 if( !strcmp( p_cfg->psz_name, "sdp" ) )
619 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
622 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
623 if( !strcmp( p_cfg->psz_value, psz ) )
626 SDPHandleUrl( p_stream, p_cfg->psz_value );
632 if( p_sys->p_mux != NULL )
634 sout_stream_id_sys_t *id = Add( p_stream, NULL );
645 /*****************************************************************************
647 *****************************************************************************/
648 static void Close( vlc_object_t * p_this )
650 sout_stream_t *p_stream = (sout_stream_t*)p_this;
651 sout_stream_sys_t *p_sys = p_stream->p_sys;
655 assert( p_sys->i_es <= 1 );
657 sout_MuxDelete( p_sys->p_mux );
658 if ( p_sys->i_es > 0 )
659 Del( p_stream, p_sys->es[0] );
660 sout_AccessOutDelete( p_sys->p_grab );
664 block_Release( p_sys->packet );
668 if( p_sys->rtsp != NULL )
669 RtspUnsetup( p_sys->rtsp );
671 vlc_mutex_destroy( &p_sys->lock_sdp );
672 vlc_mutex_destroy( &p_sys->lock_ts );
673 vlc_mutex_destroy( &p_sys->lock_es );
675 if( p_sys->p_httpd_file )
676 httpd_FileDelete( p_sys->p_httpd_file );
678 if( p_sys->p_httpd_host )
679 httpd_HostDelete( p_sys->p_httpd_host );
681 free( p_sys->psz_sdp );
683 if( p_sys->psz_sdp_file != NULL )
685 unlink( p_sys->psz_sdp_file );
686 free( p_sys->psz_sdp_file );
688 free( p_sys->psz_vod_session );
689 free( p_sys->psz_destination );
693 /*****************************************************************************
695 *****************************************************************************/
696 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
698 sout_stream_sys_t *p_sys = p_stream->p_sys;
701 vlc_UrlParse( &url, psz_url, 0 );
702 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
704 if( p_sys->p_httpd_file )
706 msg_Err( p_stream, "you can use sdp=http:// only once" );
710 if( HttpSetup( p_stream, &url ) )
712 msg_Err( p_stream, "cannot export SDP as HTTP" );
715 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
717 if( p_sys->rtsp != NULL )
719 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
723 if( url.psz_host != NULL && *url.psz_host )
725 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
726 "multiple-host configurations, use at your own risks.",
728 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
729 "command line instead." );
731 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
732 var_SetString( p_stream, "rtsp-host", url.psz_host );
734 if( url.i_port != 0 )
736 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
737 "the command line instead.", url.i_port ); */
739 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
740 var_SetInteger( p_stream, "rtsp-port", url.i_port );
743 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
744 if( p_sys->rtsp == NULL )
745 msg_Err( p_stream, "cannot export SDP as RTSP" );
747 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
748 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
750 p_sys->b_export_sap = true;
751 SapSetup( p_stream );
753 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
755 if( p_sys->psz_sdp_file != NULL )
757 msg_Err( p_stream, "you can use sdp=file:// only once" );
760 p_sys->psz_sdp_file = make_path( psz_url );
761 if( p_sys->psz_sdp_file == NULL )
763 FileSetup( p_stream );
767 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
772 vlc_UrlClean( &url );
775 /*****************************************************************************
777 *****************************************************************************/
779 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
781 sout_stream_sys_t *p_sys = p_stream->p_sys;
782 char *psz_sdp = NULL;
783 struct sockaddr_storage dst;
787 * When we have a fixed destination (typically when we do multicast),
788 * we need to put the actual port numbers in the SDP.
789 * When there is no fixed destination, we only support RTSP unicast
790 * on-demand setup, so we should rather let the clients decide which ports
792 * When there is both a fixed destination and RTSP unicast, we need to
793 * put port numbers used by the fixed destination, otherwise the SDP would
794 * become totally incorrect for multicast use. It should be noted that
795 * port numbers from SDP with RTSP are only "recommendation" from the
796 * server to the clients (per RFC2326), so only broken clients will fail
797 * to handle this properly. There is no solution but to use two differents
798 * output chain with two different RTSP URLs if you need to handle this
803 vlc_mutex_lock( &p_sys->lock_es );
804 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
805 goto out; /* hmm... */
807 if( p_sys->psz_destination != NULL )
811 /* Oh boy, this is really ugly! */
812 dstlen = sizeof( dst );
813 if( p_sys->es[0]->listen.fd != NULL )
814 getsockname( p_sys->es[0]->listen.fd[0],
815 (struct sockaddr *)&dst, &dstlen );
817 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
818 (struct sockaddr *)&dst, &dstlen );
824 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
825 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
826 && rtsp_url[7] == '[';
828 /* Dummy destination address for RTSP */
829 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
830 : sizeof( struct sockaddr_in );
831 memset (&dst, 0, dstlen);
832 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
838 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
839 NULL, 0, (struct sockaddr *)&dst, dstlen );
840 if( psz_sdp == NULL )
843 /* TODO: a=source-filter */
844 if( p_sys->rtcp_mux )
845 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
847 if( rtsp_url != NULL )
848 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
850 const char *proto = "RTP/AVP"; /* protocol */
851 if( rtsp_url == NULL )
853 switch( p_sys->proto )
858 proto = "TCP/RTP/AVP";
861 proto = "DCCP/RTP/AVP";
863 case IPPROTO_UDPLITE:
868 for( i = 0; i < p_sys->i_es; i++ )
870 sout_stream_id_sys_t *id = p_sys->es[i];
871 rtp_format_t *rtp_fmt = &id->rtp_fmt;
872 const char *mime_major; /* major MIME type */
874 switch( rtp_fmt->cat )
877 mime_major = "video";
880 mime_major = "audio";
889 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
890 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
891 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
894 /* cf RFC4566 §5.14 */
895 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
896 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
898 if( rtsp_url != NULL )
900 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
901 if( track_url != NULL )
903 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
909 if( id->listen.fd != NULL )
910 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
911 if( p_sys->proto == IPPROTO_DCCP )
912 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
914 toupper( (unsigned char)mime_major[0] ) );
918 vlc_mutex_unlock( &p_sys->lock_es );
922 /*****************************************************************************
924 *****************************************************************************/
927 * Shrink the MTU down to a fixed packetization time (for audio).
930 rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes)
932 /* Samples per second */
933 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
934 bytes *= id->rtp_fmt.channels;
937 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
938 id->i_mtu = 12 + spl;
939 else /* MTU is too small for ptime, align to a sample boundary */
940 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
943 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
945 /* This is an overflow-proof way of doing:
946 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
948 * NOTE: this plays nice with offsets because the (equivalent)
949 * calculations are linear. */
950 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
951 return q.quot * (int64_t)i_clock_rate
952 + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
955 /** Add an ES as a new RTP stream */
956 static sout_stream_id_sys_t *Add( sout_stream_t *p_stream,
957 const es_format_t *p_fmt )
959 /* NOTE: As a special case, if we use a non-RTP
960 * mux (TS/PS), then p_fmt is NULL. */
961 sout_stream_sys_t *p_sys = p_stream->p_sys;
964 sout_stream_id_sys_t *id = malloc( sizeof( *id ) );
965 if( unlikely(id == NULL) )
967 id->p_stream = p_stream;
969 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
970 if( id->i_mtu <= 12 + 16 )
971 id->i_mtu = 576 - 20 - 8; /* pessimistic */
972 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
977 vlc_mutex_init( &id->lock_sink );
982 id->listen.fd = NULL;
984 id->b_first_packet = true;
986 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
988 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
989 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
993 if (p_sys->p_vod_media != NULL)
995 id->rtp_fmt.ptname = NULL;
997 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
998 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
999 &ssrc, &id->i_seq_sent_next);
1000 if (val == VLC_SUCCESS)
1002 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1003 /* This is ugly, but id->i_seq_sent_next needs to be
1004 * initialized inside vod_init_id() to avoid race
1006 id->i_sequence = id->i_seq_sent_next;
1008 /* vod_init_id() may fail either because the ES wasn't found in
1009 * the VoD media, or because the RTSP session is gone. In the
1010 * former case, id->rtp_fmt was left untouched. */
1011 format = (id->rtp_fmt.ptname != NULL);
1016 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1017 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1018 if (p_fmt == NULL && psz == NULL)
1020 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1022 if (val != VLC_SUCCESS)
1027 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1031 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1032 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1033 if (id->srtp == NULL)
1039 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1040 int val = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1045 msg_Err (p_stream, "bad SRTP key/salt combination (%s)",
1046 vlc_strerror_c(val));
1049 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1053 id->i_seq_sent_next = id->i_sequence;
1056 if( p_sys->psz_destination != NULL )
1058 /* Choose the port */
1059 uint16_t i_port = 0;
1063 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1064 i_port = p_sys->i_port_audio;
1066 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1067 i_port = p_sys->i_port_video;
1069 /* We do not need the ES lock (p_sys->lock_es) here, because
1070 * this is the only one thread that can *modify* the ES table.
1071 * The ES lock protects the other threads from our modifications
1072 * (TAB_APPEND, TAB_REMOVE). */
1073 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1074 if (i_port == p_sys->es[i]->i_port)
1075 i_port = 0; /* Port already in use! */
1076 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1080 msg_Err (p_stream, "too many RTP elementary streams");
1084 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1085 if (p == p_sys->es[i]->i_port)
1089 id->i_port = i_port;
1091 int type = SOCK_STREAM;
1093 switch( p_sys->proto )
1099 switch (id->rtp_fmt.cat)
1101 case VIDEO_ES: code = "RTPV"; break;
1102 case AUDIO_ES: code = "RTPARTPV"; break;
1103 case SPU_ES: code = "RTPTRTPV"; break;
1104 default: code = "RTPORTPV"; break;
1106 var_SetString (p_stream, "dccp-service", code);
1108 } /* fall through */
1111 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1112 p_sys->psz_destination, i_port,
1113 type, p_sys->proto );
1114 if( id->listen.fd == NULL )
1116 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1119 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1120 VLC_THREAD_PRIORITY_LOW ) )
1122 net_ListenClose( id->listen.fd );
1123 id->listen.fd = NULL;
1130 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1131 i_port, -1, p_sys->proto );
1134 msg_Err( p_stream, "cannot create RTP socket" );
1137 /* Ignore any unexpected incoming packet (including RTCP-RR
1138 * packets in case of rtcp-mux) */
1139 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1141 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1142 /* FIXME: test if this is multicast */
1149 switch( p_fmt->i_codec )
1151 case VLC_CODEC_MULAW:
1152 case VLC_CODEC_ALAW:
1154 rtp_set_ptime (id, 20, 1);
1156 case VLC_CODEC_S16B:
1157 case VLC_CODEC_S16L:
1158 rtp_set_ptime (id, 20, 2);
1160 case VLC_CODEC_S24B:
1161 rtp_set_ptime (id, 20, 3);
1167 #if 0 /* No payload formats sets this at the moment */
1170 cscov += 8 /* UDP */ + 12 /* RTP */;
1172 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1175 vlc_mutex_lock( &p_sys->lock_ts );
1176 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1177 vlc_mutex_unlock( &p_sys->lock_ts );
1179 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1180 p_sys->i_pts_offset );
1182 if( p_sys->rtsp != NULL )
1183 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1184 id->rtp_fmt.clock_rate, mcast_fd );
1186 id->p_fifo = block_FifoNew();
1187 if( unlikely(id->p_fifo == NULL) )
1189 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1191 block_FifoRelease( id->p_fifo );
1196 /* Update p_sys context */
1197 vlc_mutex_lock( &p_sys->lock_es );
1198 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1199 vlc_mutex_unlock( &p_sys->lock_es );
1201 psz_sdp = SDPGenerate( p_stream, NULL );
1203 vlc_mutex_lock( &p_sys->lock_sdp );
1204 free( p_sys->psz_sdp );
1205 p_sys->psz_sdp = psz_sdp;
1206 vlc_mutex_unlock( &p_sys->lock_sdp );
1208 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1210 /* Update SDP (sap/file) */
1211 if( p_sys->b_export_sap ) SapSetup( p_stream );
1212 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1217 Del( p_stream, id );
1221 static void Del( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1223 sout_stream_sys_t *p_sys = p_stream->p_sys;
1225 vlc_mutex_lock( &p_sys->lock_es );
1226 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1227 vlc_mutex_unlock( &p_sys->lock_es );
1229 if( likely(id->p_fifo != NULL) )
1231 vlc_cancel( id->thread );
1232 vlc_join( id->thread, NULL );
1233 block_FifoRelease( id->p_fifo );
1236 free( id->rtp_fmt.fmtp );
1238 if (p_sys->p_vod_media != NULL)
1239 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1241 RtspDelId( p_sys->rtsp, id->rtsp_id );
1242 if( id->listen.fd != NULL )
1244 vlc_cancel( id->listen.thread );
1245 vlc_join( id->listen.thread, NULL );
1246 net_ListenClose( id->listen.fd );
1248 /* Delete remaining sinks (incoming connections or explicit
1250 while( id->sinkc > 0 )
1251 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1253 if( id->srtp != NULL )
1254 srtp_destroy( id->srtp );
1257 vlc_mutex_destroy( &id->lock_sink );
1259 /* Update SDP (sap/file) */
1260 if( p_sys->b_export_sap ) SapSetup( p_stream );
1261 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1266 static int Send( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1269 assert( p_stream->p_sys->p_mux == NULL );
1272 while( p_buffer != NULL )
1274 block_t *p_next = p_buffer->p_next;
1275 p_buffer->p_next = NULL;
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 ) )
1296 /****************************************************************************
1298 ****************************************************************************/
1299 static int SapSetup( sout_stream_t *p_stream )
1301 sout_stream_sys_t *p_sys = p_stream->p_sys;
1303 /* Remove the previous session */
1304 if( p_sys->p_session != NULL)
1306 sout_AnnounceUnRegister( p_stream, p_sys->p_session);
1307 p_sys->p_session = NULL;
1310 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1311 p_sys->p_session = sout_AnnounceRegisterSDP( p_stream,
1313 p_sys->psz_destination );
1318 /****************************************************************************
1320 ****************************************************************************/
1321 static int FileSetup( sout_stream_t *p_stream )
1323 sout_stream_sys_t *p_sys = p_stream->p_sys;
1326 if( p_sys->psz_sdp == NULL )
1327 return VLC_EGENERIC; /* too early */
1329 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1331 msg_Err( p_stream, "cannot open file '%s' (%s)",
1332 p_sys->psz_sdp_file, vlc_strerror_c(errno) );
1333 return VLC_EGENERIC;
1336 fputs( p_sys->psz_sdp, f );
1342 /****************************************************************************
1344 ****************************************************************************/
1345 static int HttpCallback( httpd_file_sys_t *p_args,
1346 httpd_file_t *, uint8_t *p_request,
1347 uint8_t **pp_data, int *pi_data );
1349 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1351 sout_stream_sys_t *p_sys = p_stream->p_sys;
1353 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1354 if( p_sys->p_httpd_host )
1356 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1357 url->psz_path ? url->psz_path : "/",
1360 HttpCallback, (void*)p_sys );
1362 if( p_sys->p_httpd_file == NULL )
1364 return VLC_EGENERIC;
1369 static int HttpCallback( httpd_file_sys_t *p_args,
1370 httpd_file_t *f, uint8_t *p_request,
1371 uint8_t **pp_data, int *pi_data )
1373 VLC_UNUSED(f); VLC_UNUSED(p_request);
1374 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1376 vlc_mutex_lock( &p_sys->lock_sdp );
1377 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1379 *pi_data = strlen( p_sys->psz_sdp );
1380 *pp_data = malloc( *pi_data );
1381 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1388 vlc_mutex_unlock( &p_sys->lock_sdp );
1393 /****************************************************************************
1395 ****************************************************************************/
1396 static void* ThreadSend( void *data )
1399 # define ENOBUFS WSAENOBUFS
1400 # define EAGAIN WSAEWOULDBLOCK
1401 # define EWOULDBLOCK WSAEWOULDBLOCK
1403 sout_stream_id_sys_t *id = data;
1404 unsigned i_caching = id->i_caching;
1408 block_t *out = block_FifoGet( id->p_fifo );
1409 block_cleanup_push (out);
1413 { /* FIXME: this is awfully inefficient */
1414 size_t len = out->i_buffer;
1415 out = block_Realloc( out, 0, len + 10 );
1416 out->i_buffer = len;
1418 int canc = vlc_savecancel ();
1419 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1420 vlc_restorecancel (canc);
1423 msg_Dbg( id->p_stream, "SRTP sending error: %s",
1424 vlc_strerror_c(val) );
1425 block_Release( out );
1429 out->i_buffer = len;
1432 mwait (out->i_dts + i_caching);
1437 mwait (out->i_dts + i_caching);
1441 ssize_t len = out->i_buffer;
1442 int canc = vlc_savecancel ();
1444 vlc_mutex_lock( &id->lock_sink );
1445 unsigned deadc = 0; /* How many dead sockets? */
1446 int deadv[id->sinkc]; /* Dead sockets list */
1448 for( int i = 0; i < id->sinkc; i++ )
1451 if( !id->srtp ) /* FIXME: SRTCP support */
1453 SendRTCP( id->sinkv[i].rtcp, out );
1455 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1456 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1457 && net_errno != ENOBUFS && net_errno != ENOMEM )
1460 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1461 &type, &(socklen_t){ sizeof(type) });
1462 if( type == SOCK_DGRAM )
1463 /* ICMP soft error: ignore and retry */
1464 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1466 /* Broken connection */
1467 deadv[deadc++] = id->sinkv[i].rtp_fd;
1470 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1471 vlc_mutex_unlock( &id->lock_sink );
1472 block_Release( out );
1474 for( unsigned i = 0; i < deadc; i++ )
1476 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1477 rtp_del_sink( id, deadv[i] );
1479 vlc_restorecancel (canc);
1485 /* This thread dequeues incoming connections (DCCP streaming) */
1486 static void *rtp_listen_thread( void *data )
1488 sout_stream_id_sys_t *id = data;
1490 assert( id->listen.fd != NULL );
1494 int fd = net_Accept( id->p_stream, id->listen.fd );
1497 int canc = vlc_savecancel( );
1498 rtp_add_sink( id, fd, true, NULL );
1499 vlc_restorecancel( canc );
1502 vlc_assert_unreachable();
1506 int rtp_add_sink( sout_stream_id_sys_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1508 rtp_sink_t sink = { fd, NULL };
1509 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1511 if( sink.rtcp == NULL )
1512 msg_Err( id->p_stream, "RTCP failed!" );
1514 vlc_mutex_lock( &id->lock_sink );
1515 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1517 *seq = id->i_seq_sent_next;
1518 vlc_mutex_unlock( &id->lock_sink );
1522 void rtp_del_sink( sout_stream_id_sys_t *id, int fd )
1524 rtp_sink_t sink = { fd, NULL };
1526 /* NOTE: must be safe to use if fd is not included */
1527 vlc_mutex_lock( &id->lock_sink );
1528 for( int i = 0; i < id->sinkc; i++ )
1530 if (id->sinkv[i].rtp_fd == fd)
1532 sink = id->sinkv[i];
1533 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1537 vlc_mutex_unlock( &id->lock_sink );
1539 CloseRTCP( sink.rtcp );
1540 net_Close( sink.rtp_fd );
1543 uint16_t rtp_get_seq( sout_stream_id_sys_t *id )
1545 /* This will return values for the next packet. */
1548 vlc_mutex_lock( &id->lock_sink );
1549 seq = id->i_seq_sent_next;
1550 vlc_mutex_unlock( &id->lock_sink );
1555 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1556 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1557 * random (although we use the same reference for all the ES as a
1558 * feature). In the VoD case, this function is called independently
1559 * from several parts of the code, so we need to always return the same
1561 static int64_t rtp_init_ts( const vod_media_t *p_media,
1562 const char *psz_vod_session )
1564 if (p_media == NULL || psz_vod_session == NULL)
1568 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1569 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1570 i_ts_init ^= (uintptr_t)p_media;
1571 /* Limit the timestamp to 48 bits, this is enough and allows us
1572 * to stay away from overflows */
1573 i_ts_init &= 0xFFFFFFFFFFFF;
1577 /* Return a timestamp corresponding to packets being sent now, and that
1578 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1579 * Also return the NPT corresponding to this timestamp. If the stream
1580 * output is not started, the initial timestamp that will be used with
1581 * the first packets for NPT=0 is returned instead. */
1582 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_sys_t *id,
1583 const vod_media_t *p_media, const char *psz_vod_session,
1590 p_stream = id->p_stream;
1592 if (p_stream == NULL)
1593 return rtp_init_ts(p_media, psz_vod_session);
1595 sout_stream_sys_t *p_sys = p_stream->p_sys;
1597 vlc_mutex_lock( &p_sys->lock_ts );
1598 i_npt_zero = p_sys->i_npt_zero;
1599 vlc_mutex_unlock( &p_sys->lock_ts );
1601 if( i_npt_zero == VLC_TS_INVALID )
1602 return p_sys->i_pts_zero;
1604 mtime_t now = mdate();
1605 if( now < i_npt_zero )
1606 return p_sys->i_pts_zero;
1608 int64_t npt = now - i_npt_zero;
1612 return p_sys->i_pts_zero + npt;
1615 void rtp_packetize_common( sout_stream_id_sys_t *id, block_t *out,
1616 int b_marker, int64_t i_pts )
1618 if( !id->b_ts_init )
1620 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1621 vlc_mutex_lock( &p_sys->lock_ts );
1622 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1624 /* This is the first packet of any ES. We initialize the
1625 * NPT=0 time reference, and the offset to match the
1626 * arbitrary PTS reference. */
1627 p_sys->i_npt_zero = i_pts + id->i_caching;
1628 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1630 vlc_mutex_unlock( &p_sys->lock_ts );
1632 /* And in any case this is the first packet of this ES, so we
1633 * initialize the offset for this ES. */
1634 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1635 p_sys->i_pts_offset );
1636 id->b_ts_init = true;
1639 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1642 out->p_buffer[0] = 0x80;
1643 out->p_buffer[1] = (b_marker?0x80:0x00)|id->rtp_fmt.payload_type;
1644 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1645 out->p_buffer[3] = ( id->i_sequence )&0xff;
1646 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1647 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1648 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1649 out->p_buffer[7] = ( i_timestamp )&0xff;
1651 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1656 uint16_t rtp_get_extended_sequence( sout_stream_id_sys_t *id )
1658 return id->i_sequence >> 16;
1661 void rtp_packetize_send( sout_stream_id_sys_t *id, block_t *out )
1663 block_FifoPut( id->p_fifo, out );
1667 * @return configured max RTP payload size (including payload type-specific
1668 * headers, excluding RTP and transport headers)
1670 size_t rtp_mtu (const sout_stream_id_sys_t *id)
1672 return id->i_mtu - 12;
1675 /*****************************************************************************
1677 *****************************************************************************/
1679 /** Add an ES to a non-RTP muxed stream */
1680 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *p_stream,
1681 const es_format_t *p_fmt )
1683 sout_input_t *p_input;
1684 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1685 assert( p_mux != NULL );
1687 p_input = sout_MuxAddStream( p_mux, p_fmt );
1688 if( p_input == NULL )
1690 msg_Err( p_stream, "cannot add this stream to the muxer" );
1694 return (sout_stream_id_sys_t *)p_input;
1698 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1701 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1702 assert( p_mux != NULL );
1704 return sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1708 /** Remove an ES from a non-RTP muxed stream */
1709 static void MuxDel( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1711 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1712 assert( p_mux != NULL );
1714 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_sys_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_Alloc( 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;
1807 void rtp_get_video_geometry( sout_stream_id_sys_t *id, int *width, int *height )
1809 int ret = sscanf( id->rtp_fmt.fmtp, "%*s width=%d; height=%d; ", width, height );