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 #define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS
34 #include <vlc_common.h>
35 #include <vlc_plugin.h>
37 #include <vlc_block.h>
39 #include <vlc_httpd.h>
41 #include <vlc_network.h>
47 # include <vlc_gcrypt.h>
52 #include <sys/types.h>
54 #ifdef HAVE_ARPA_INET_H
55 # include <arpa/inet.h>
57 #ifdef HAVE_LINUX_DCCP_H
58 # include <linux/dccp.h>
61 # define IPPROTO_DCCP 33
63 #ifndef IPPROTO_UDPLITE
64 # define IPPROTO_UDPLITE 136
71 /*****************************************************************************
73 *****************************************************************************/
75 #define DEST_TEXT N_("Destination")
76 #define DEST_LONGTEXT N_( \
77 "This is the output URL that will be used." )
78 #define SDP_TEXT N_("SDP")
79 #define SDP_LONGTEXT N_( \
80 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
81 "session will be made available. You must use a url: http://location to " \
82 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
83 "for the SDP to be announced via SAP." )
84 #define SAP_TEXT N_("SAP announcing")
85 #define SAP_LONGTEXT N_("Announce this session with SAP.")
86 #define MUX_TEXT N_("Muxer")
87 #define MUX_LONGTEXT N_( \
88 "This allows you to specify the muxer used for the streaming output. " \
89 "Default is to use no muxer (standard RTP stream)." )
91 #define NAME_TEXT N_("Session name")
92 #define NAME_LONGTEXT N_( \
93 "This is the name of the session that will be announced in the SDP " \
94 "(Session Descriptor)." )
95 #define CAT_TEXT N_("Session category")
96 #define CAT_LONGTEXT N_( \
97 "This allows you to specify a category for the session, " \
98 "that will be announced if you choose to use SAP." )
99 #define DESC_TEXT N_("Session description")
100 #define DESC_LONGTEXT N_( \
101 "This allows you to give a short description with details about the stream, " \
102 "that will be announced in the SDP (Session Descriptor)." )
103 #define URL_TEXT N_("Session URL")
104 #define URL_LONGTEXT N_( \
105 "This allows you to give a URL with more details about the stream " \
106 "(often the website of the streaming organization), that will " \
107 "be announced in the SDP (Session Descriptor)." )
108 #define EMAIL_TEXT N_("Session email")
109 #define EMAIL_LONGTEXT N_( \
110 "This allows you to give a contact mail address for the stream, that will " \
111 "be announced in the SDP (Session Descriptor)." )
112 #define PHONE_TEXT N_("Session phone number")
113 #define PHONE_LONGTEXT N_( \
114 "This allows you to give a contact telephone number for the stream, that will " \
115 "be announced in the SDP (Session Descriptor)." )
117 #define PORT_TEXT N_("Port")
118 #define PORT_LONGTEXT N_( \
119 "This allows you to specify the base port for the RTP streaming." )
120 #define PORT_AUDIO_TEXT N_("Audio port")
121 #define PORT_AUDIO_LONGTEXT N_( \
122 "This allows you to specify the default audio port for the RTP streaming." )
123 #define PORT_VIDEO_TEXT N_("Video port")
124 #define PORT_VIDEO_LONGTEXT N_( \
125 "This allows you to specify the default video port for the RTP streaming." )
127 #define TTL_TEXT N_("Hop limit (TTL)")
128 #define TTL_LONGTEXT N_( \
129 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
130 "the multicast packets sent by the stream output (-1 = use operating " \
131 "system built-in default).")
133 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
134 #define RTCP_MUX_LONGTEXT N_( \
135 "This sends and receives RTCP packet multiplexed over the same port " \
138 #define CACHING_TEXT N_("Caching value (ms)")
139 #define CACHING_LONGTEXT N_( \
140 "Default caching value for outbound RTP streams. This " \
141 "value should be set in milliseconds." )
143 #define PROTO_TEXT N_("Transport protocol")
144 #define PROTO_LONGTEXT N_( \
145 "This selects which transport protocol to use for RTP." )
147 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
148 #define SRTP_KEY_LONGTEXT N_( \
149 "RTP packets will be integrity-protected and ciphered "\
150 "with this Secure RTP master shared secret key. "\
151 "This must be a 32-character-long hexadecimal string.")
153 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
154 #define SRTP_SALT_LONGTEXT N_( \
155 "Secure RTP requires a (non-secret) master salt value. " \
156 "This must be a 28-character-long hexadecimal string.")
158 static const char *const ppsz_protos[] = {
159 "dccp", "sctp", "tcp", "udp", "udplite",
162 static const char *const ppsz_protocols[] = {
163 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
166 #define RFC3016_TEXT N_("MP4A LATM")
167 #define RFC3016_LONGTEXT N_( \
168 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
170 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
171 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
172 "not receiving any RTSP request for this long. Setting it to a " \
173 "negative value or zero disables timeouts. The default is 60 (one " \
176 #define RTSP_USER_TEXT N_("Username")
177 #define RTSP_USER_LONGTEXT N_("Username that will be " \
178 "requested to access the stream." )
179 #define RTSP_PASS_TEXT N_("Password")
180 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
181 "requested to access the stream." )
183 static int Open ( vlc_object_t * );
184 static void Close( vlc_object_t * );
186 #define SOUT_CFG_PREFIX "sout-rtp-"
187 #define MAX_EMPTY_BLOCKS 200
190 set_shortname( N_("RTP"))
191 set_description( N_("RTP stream output") )
192 set_capability( "sout stream", 0 )
193 add_shortcut( "rtp", "vod" )
194 set_category( CAT_SOUT )
195 set_subcategory( SUBCAT_SOUT_STREAM )
197 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
198 DEST_LONGTEXT, true )
199 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
201 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
203 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
206 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
207 NAME_LONGTEXT, true )
208 add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_LONGTEXT, true )
209 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
210 DESC_LONGTEXT, true )
211 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
213 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
214 EMAIL_LONGTEXT, true )
215 add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
216 PHONE_LONGTEXT, true )
218 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
219 PROTO_LONGTEXT, false )
220 change_string_list( ppsz_protos, ppsz_protocols )
221 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
222 PORT_LONGTEXT, true )
223 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
224 PORT_AUDIO_LONGTEXT, true )
225 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
226 PORT_VIDEO_LONGTEXT, true )
228 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
230 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
231 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
232 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
233 CACHING_TEXT, CACHING_LONGTEXT, true )
236 add_string( SOUT_CFG_PREFIX "key", "",
237 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
238 add_string( SOUT_CFG_PREFIX "salt", "",
239 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
242 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
243 RFC3016_LONGTEXT, false )
245 set_callbacks( Open, Close )
248 set_shortname( N_("RTSP VoD" ) )
249 set_description( N_("RTSP VoD server") )
250 set_category( CAT_SOUT )
251 set_subcategory( SUBCAT_SOUT_VOD )
252 set_capability( "vod server", 10 )
253 set_callbacks( OpenVoD, CloseVoD )
254 add_shortcut( "rtsp" )
255 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
256 RTSP_TIMEOUT_LONGTEXT, true )
257 add_string( "sout-rtsp-user", "",
258 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
259 add_password( "sout-rtsp-pwd", "",
260 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
264 /*****************************************************************************
265 * Exported prototypes
266 *****************************************************************************/
267 static const char *const ppsz_sout_options[] = {
268 "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
269 "mux", "sap", "description", "url", "email", "phone",
270 "proto", "rtcp-mux", "caching",
277 static sout_stream_id_sys_t *Add( sout_stream_t *, const es_format_t * );
278 static void Del ( sout_stream_t *, sout_stream_id_sys_t * );
279 static int Send( sout_stream_t *, sout_stream_id_sys_t *,
281 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *, const es_format_t * );
282 static void MuxDel ( sout_stream_t *, sout_stream_id_sys_t * );
283 static int MuxSend( sout_stream_t *, sout_stream_id_sys_t *,
286 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
287 static void* ThreadSend( void * );
288 static void *rtp_listen_thread( void * );
290 static void SDPHandleUrl( sout_stream_t *, const char * );
292 static int SapSetup( sout_stream_t *p_stream );
293 static int FileSetup( sout_stream_t *p_stream );
294 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
296 static int64_t rtp_init_ts( const vod_media_t *p_media,
297 const char *psz_vod_session );
299 struct sout_stream_sys_t
303 vlc_mutex_t lock_sdp;
310 session_descriptor_t *p_session;
313 httpd_host_t *p_httpd_host;
314 httpd_file_t *p_httpd_file;
319 /* RTSP NPT and timestamp computations */
320 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
321 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
322 int64_t i_pts_offset; /* matches actual PTS to prediction */
326 char *psz_destination;
328 uint16_t i_port_audio;
329 uint16_t i_port_video;
335 vod_media_t *p_vod_media;
336 char *psz_vod_session;
338 /* in case we do TS/PS over rtp */
340 sout_access_out_t *p_grab;
346 sout_stream_id_sys_t **es;
349 typedef struct rtp_sink_t
355 struct sout_stream_id_sys_t
357 sout_stream_t *p_stream;
359 /* For RFC 4175, seqnum is extended to 32-bits */
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_stream_sys_t *p_sys = NULL;
401 config_chain_t *p_cfg = NULL;
405 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
406 ppsz_sout_options, p_stream->p_cfg );
408 p_sys = malloc( sizeof( sout_stream_sys_t ) );
412 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
414 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
415 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
416 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
417 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
419 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
421 msg_Err( p_stream, "audio and video RTP port must be distinct" );
422 free( p_sys->psz_destination );
427 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
429 if( !strcmp( p_cfg->psz_name, "sdp" )
430 && ( p_cfg->psz_value != NULL )
431 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
439 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
442 if( !strncasecmp( psz, "rtsp:", 5 ) )
448 /* Transport protocol */
449 p_sys->proto = IPPROTO_UDP;
450 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
452 if ((psz == NULL) || !strcasecmp (psz, "udp"))
453 (void)0; /* default */
455 if (!strcasecmp (psz, "dccp"))
457 p_sys->proto = IPPROTO_DCCP;
458 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
462 if (!strcasecmp (psz, "sctp"))
464 p_sys->proto = IPPROTO_TCP;
465 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
470 if (!strcasecmp (psz, "tcp"))
472 p_sys->proto = IPPROTO_TCP;
473 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
477 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
478 p_sys->proto = IPPROTO_UDPLITE;
480 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
483 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
485 p_sys->p_vod_media = NULL;
486 p_sys->psz_vod_session = NULL;
488 if (! strcmp(p_stream->psz_name, "vod"))
490 /* The VLM stops all instances before deleting a media, so this
491 * reference will remain valid during the lifetime of the rtp
493 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
495 if (p_sys->p_vod_media != NULL)
497 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
498 if (p_sys->psz_vod_session == NULL)
500 msg_Err(p_stream, "missing VoD session");
505 const char *mux = vod_get_mux(p_sys->p_vod_media);
506 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
510 if( p_sys->psz_destination == NULL && !b_rtsp
511 && p_sys->p_vod_media == NULL )
513 msg_Err( p_stream, "missing destination and not in RTSP mode" );
518 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
521 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
522 var_SetInteger( p_stream, "ttl", i_ttl );
525 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
527 /* NPT=0 time will be determined when we packetize the first packet
528 * (of any ES). But we want to be able to report rtptime in RTSP
529 * without waiting (and already did in the VoD case). So until then,
530 * we use an arbitrary reference PTS for timestamp computations, and
531 * then actual PTS will catch up using offsets. */
532 p_sys->i_npt_zero = VLC_TS_INVALID;
533 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
534 p_sys->psz_vod_session);
538 p_sys->psz_sdp = NULL;
540 p_sys->b_export_sap = false;
541 p_sys->p_session = NULL;
542 p_sys->psz_sdp_file = NULL;
544 p_sys->p_httpd_host = NULL;
545 p_sys->p_httpd_file = NULL;
547 p_stream->p_sys = p_sys;
549 vlc_mutex_init( &p_sys->lock_sdp );
550 vlc_mutex_init( &p_sys->lock_ts );
551 vlc_mutex_init( &p_sys->lock_es );
553 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
556 /* Check muxer type */
557 if( strncasecmp( psz, "ps", 2 )
558 && strncasecmp( psz, "mpeg1", 5 )
559 && strncasecmp( psz, "ts", 2 ) )
561 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
563 vlc_mutex_destroy( &p_sys->lock_sdp );
564 vlc_mutex_destroy( &p_sys->lock_ts );
565 vlc_mutex_destroy( &p_sys->lock_es );
566 free( p_sys->psz_vod_session );
567 free( p_sys->psz_destination );
572 p_sys->p_grab = GrabberCreate( p_stream );
573 p_sys->p_mux = sout_MuxNew( p_stream->p_sout, psz, p_sys->p_grab );
576 if( p_sys->p_mux == NULL )
578 msg_Err( p_stream, "cannot create muxer" );
579 sout_AccessOutDelete( p_sys->p_grab );
580 vlc_mutex_destroy( &p_sys->lock_sdp );
581 vlc_mutex_destroy( &p_sys->lock_ts );
582 vlc_mutex_destroy( &p_sys->lock_es );
583 free( p_sys->psz_vod_session );
584 free( p_sys->psz_destination );
589 p_sys->packet = NULL;
591 p_stream->pf_add = MuxAdd;
592 p_stream->pf_del = MuxDel;
593 p_stream->pf_send = MuxSend;
598 p_sys->p_grab = NULL;
600 p_stream->pf_add = Add;
601 p_stream->pf_del = Del;
602 p_stream->pf_send = Send;
604 p_stream->pace_nocontrol = true;
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 if( p_sys->p_mux != NULL )
635 sout_stream_id_sys_t *id = Add( p_stream, NULL );
646 /*****************************************************************************
648 *****************************************************************************/
649 static void Close( vlc_object_t * p_this )
651 sout_stream_t *p_stream = (sout_stream_t*)p_this;
652 sout_stream_sys_t *p_sys = p_stream->p_sys;
656 assert( p_sys->i_es <= 1 );
658 sout_MuxDelete( p_sys->p_mux );
659 if ( p_sys->i_es > 0 )
660 Del( p_stream, p_sys->es[0] );
661 sout_AccessOutDelete( p_sys->p_grab );
665 block_Release( p_sys->packet );
669 if( p_sys->rtsp != NULL )
670 RtspUnsetup( p_sys->rtsp );
672 vlc_mutex_destroy( &p_sys->lock_sdp );
673 vlc_mutex_destroy( &p_sys->lock_ts );
674 vlc_mutex_destroy( &p_sys->lock_es );
676 if( p_sys->p_httpd_file )
677 httpd_FileDelete( p_sys->p_httpd_file );
679 if( p_sys->p_httpd_host )
680 httpd_HostDelete( p_sys->p_httpd_host );
682 free( p_sys->psz_sdp );
684 if( p_sys->psz_sdp_file != NULL )
686 unlink( p_sys->psz_sdp_file );
687 free( p_sys->psz_sdp_file );
689 free( p_sys->psz_vod_session );
690 free( p_sys->psz_destination );
694 /*****************************************************************************
696 *****************************************************************************/
697 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
699 sout_stream_sys_t *p_sys = p_stream->p_sys;
702 vlc_UrlParse( &url, psz_url, 0 );
703 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
705 if( p_sys->p_httpd_file )
707 msg_Err( p_stream, "you can use sdp=http:// only once" );
711 if( HttpSetup( p_stream, &url ) )
713 msg_Err( p_stream, "cannot export SDP as HTTP" );
716 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
718 if( p_sys->rtsp != NULL )
720 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
724 if( url.psz_host != NULL && *url.psz_host )
726 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
727 "multiple-host configurations, use at your own risks.",
729 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
730 "command line instead." );
732 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
733 var_SetString( p_stream, "rtsp-host", url.psz_host );
735 if( url.i_port != 0 )
737 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
738 "the command line instead.", url.i_port ); */
740 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
741 var_SetInteger( p_stream, "rtsp-port", url.i_port );
744 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
745 if( p_sys->rtsp == NULL )
746 msg_Err( p_stream, "cannot export SDP as RTSP" );
748 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
749 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
751 p_sys->b_export_sap = true;
752 SapSetup( p_stream );
754 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
756 if( p_sys->psz_sdp_file != NULL )
758 msg_Err( p_stream, "you can use sdp=file:// only once" );
761 p_sys->psz_sdp_file = make_path( psz_url );
762 if( p_sys->psz_sdp_file == NULL )
764 FileSetup( p_stream );
768 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
773 vlc_UrlClean( &url );
776 /*****************************************************************************
778 *****************************************************************************/
780 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
782 sout_stream_sys_t *p_sys = p_stream->p_sys;
783 char *psz_sdp = NULL;
784 struct sockaddr_storage dst;
788 * When we have a fixed destination (typically when we do multicast),
789 * we need to put the actual port numbers in the SDP.
790 * When there is no fixed destination, we only support RTSP unicast
791 * on-demand setup, so we should rather let the clients decide which ports
793 * When there is both a fixed destination and RTSP unicast, we need to
794 * put port numbers used by the fixed destination, otherwise the SDP would
795 * become totally incorrect for multicast use. It should be noted that
796 * port numbers from SDP with RTSP are only "recommendation" from the
797 * server to the clients (per RFC2326), so only broken clients will fail
798 * to handle this properly. There is no solution but to use two differents
799 * output chain with two different RTSP URLs if you need to handle this
804 vlc_mutex_lock( &p_sys->lock_es );
805 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
806 goto out; /* hmm... */
808 if( p_sys->psz_destination != NULL )
812 /* Oh boy, this is really ugly! */
813 dstlen = sizeof( dst );
814 if( p_sys->es[0]->listen.fd != NULL )
815 getsockname( p_sys->es[0]->listen.fd[0],
816 (struct sockaddr *)&dst, &dstlen );
818 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
819 (struct sockaddr *)&dst, &dstlen );
825 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
826 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
827 && rtsp_url[7] == '[';
829 /* Dummy destination address for RTSP */
830 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
831 : sizeof( struct sockaddr_in );
832 memset (&dst, 0, dstlen);
833 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
839 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
840 NULL, 0, (struct sockaddr *)&dst, dstlen );
841 if( psz_sdp == NULL )
844 /* TODO: a=source-filter */
845 if( p_sys->rtcp_mux )
846 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
848 if( rtsp_url != NULL )
849 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
851 const char *proto = "RTP/AVP"; /* protocol */
852 if( rtsp_url == NULL )
854 switch( p_sys->proto )
859 proto = "TCP/RTP/AVP";
862 proto = "DCCP/RTP/AVP";
864 case IPPROTO_UDPLITE:
869 for( i = 0; i < p_sys->i_es; i++ )
871 sout_stream_id_sys_t *id = p_sys->es[i];
872 rtp_format_t *rtp_fmt = &id->rtp_fmt;
873 const char *mime_major; /* major MIME type */
875 switch( rtp_fmt->cat )
878 mime_major = "video";
881 mime_major = "audio";
890 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
891 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
892 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
895 /* cf RFC4566 §5.14 */
896 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
897 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
899 if( rtsp_url != NULL )
901 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
902 if( track_url != NULL )
904 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
910 if( id->listen.fd != NULL )
911 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
912 if( p_sys->proto == IPPROTO_DCCP )
913 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
915 toupper( (unsigned char)mime_major[0] ) );
919 vlc_mutex_unlock( &p_sys->lock_es );
923 /*****************************************************************************
925 *****************************************************************************/
928 * Shrink the MTU down to a fixed packetization time (for audio).
931 rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes)
933 /* Samples per second */
934 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
935 bytes *= id->rtp_fmt.channels;
938 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
939 id->i_mtu = 12 + spl;
940 else /* MTU is too small for ptime, align to a sample boundary */
941 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
944 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
946 /* This is an overflow-proof way of doing:
947 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
949 * NOTE: this plays nice with offsets because the (equivalent)
950 * calculations are linear. */
951 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
952 return q.quot * (int64_t)i_clock_rate
953 + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
956 /** Add an ES as a new RTP stream */
957 static sout_stream_id_sys_t *Add( sout_stream_t *p_stream,
958 const es_format_t *p_fmt )
960 /* NOTE: As a special case, if we use a non-RTP
961 * mux (TS/PS), then p_fmt is NULL. */
962 sout_stream_sys_t *p_sys = p_stream->p_sys;
965 sout_stream_id_sys_t *id = malloc( sizeof( *id ) );
966 if( unlikely(id == NULL) )
968 id->p_stream = p_stream;
970 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
971 if( id->i_mtu <= 12 + 16 )
972 id->i_mtu = 576 - 20 - 8; /* pessimistic */
973 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
978 vlc_mutex_init( &id->lock_sink );
983 id->listen.fd = NULL;
985 id->b_first_packet = true;
987 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
989 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
990 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
994 if (p_sys->p_vod_media != NULL)
996 id->rtp_fmt.ptname = NULL;
998 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
999 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
1000 &ssrc, &id->i_seq_sent_next);
1001 if (val == VLC_SUCCESS)
1003 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1004 /* This is ugly, but id->i_seq_sent_next needs to be
1005 * initialized inside vod_init_id() to avoid race
1007 id->i_sequence = id->i_seq_sent_next;
1009 /* vod_init_id() may fail either because the ES wasn't found in
1010 * the VoD media, or because the RTSP session is gone. In the
1011 * former case, id->rtp_fmt was left untouched. */
1012 format = (id->rtp_fmt.ptname != NULL);
1017 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1018 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1019 if (p_fmt == NULL && psz == NULL)
1021 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1023 if (val != VLC_SUCCESS)
1028 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1032 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1033 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1034 if (id->srtp == NULL)
1040 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1041 int val = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1046 msg_Err (p_stream, "bad SRTP key/salt combination (%s)",
1047 vlc_strerror_c(val));
1050 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1054 id->i_seq_sent_next = id->i_sequence;
1057 if( p_sys->psz_destination != NULL )
1059 /* Choose the port */
1060 uint16_t i_port = 0;
1064 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1065 i_port = p_sys->i_port_audio;
1067 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1068 i_port = p_sys->i_port_video;
1070 /* We do not need the ES lock (p_sys->lock_es) here, because
1071 * this is the only one thread that can *modify* the ES table.
1072 * The ES lock protects the other threads from our modifications
1073 * (TAB_APPEND, TAB_REMOVE). */
1074 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1075 if (i_port == p_sys->es[i]->i_port)
1076 i_port = 0; /* Port already in use! */
1077 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1081 msg_Err (p_stream, "too many RTP elementary streams");
1085 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1086 if (p == p_sys->es[i]->i_port)
1090 id->i_port = i_port;
1092 int type = SOCK_STREAM;
1094 switch( p_sys->proto )
1100 switch (id->rtp_fmt.cat)
1102 case VIDEO_ES: code = "RTPV"; break;
1103 case AUDIO_ES: code = "RTPARTPV"; break;
1104 case SPU_ES: code = "RTPTRTPV"; break;
1105 default: code = "RTPORTPV"; break;
1107 var_SetString (p_stream, "dccp-service", code);
1109 } /* fall through */
1112 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1113 p_sys->psz_destination, i_port,
1114 type, p_sys->proto );
1115 if( id->listen.fd == NULL )
1117 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1120 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1121 VLC_THREAD_PRIORITY_LOW ) )
1123 net_ListenClose( id->listen.fd );
1124 id->listen.fd = NULL;
1131 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1132 i_port, -1, p_sys->proto );
1135 msg_Err( p_stream, "cannot create RTP socket" );
1138 /* Ignore any unexpected incoming packet (including RTCP-RR
1139 * packets in case of rtcp-mux) */
1140 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1142 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1143 /* FIXME: test if this is multicast */
1150 switch( p_fmt->i_codec )
1152 case VLC_CODEC_MULAW:
1153 case VLC_CODEC_ALAW:
1155 rtp_set_ptime (id, 20, 1);
1157 case VLC_CODEC_S16B:
1158 case VLC_CODEC_S16L:
1159 rtp_set_ptime (id, 20, 2);
1161 case VLC_CODEC_S24B:
1162 rtp_set_ptime (id, 20, 3);
1168 #if 0 /* No payload formats sets this at the moment */
1171 cscov += 8 /* UDP */ + 12 /* RTP */;
1173 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1176 vlc_mutex_lock( &p_sys->lock_ts );
1177 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1178 vlc_mutex_unlock( &p_sys->lock_ts );
1180 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1181 p_sys->i_pts_offset );
1183 if( p_sys->rtsp != NULL )
1184 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1185 id->rtp_fmt.clock_rate, mcast_fd );
1187 id->p_fifo = block_FifoNew();
1188 if( unlikely(id->p_fifo == NULL) )
1190 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1192 block_FifoRelease( id->p_fifo );
1197 /* Update p_sys context */
1198 vlc_mutex_lock( &p_sys->lock_es );
1199 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1200 vlc_mutex_unlock( &p_sys->lock_es );
1202 psz_sdp = SDPGenerate( p_stream, NULL );
1204 vlc_mutex_lock( &p_sys->lock_sdp );
1205 free( p_sys->psz_sdp );
1206 p_sys->psz_sdp = psz_sdp;
1207 vlc_mutex_unlock( &p_sys->lock_sdp );
1209 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1211 /* Update SDP (sap/file) */
1212 if( p_sys->b_export_sap ) SapSetup( p_stream );
1213 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1218 Del( p_stream, id );
1222 static void Del( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1224 sout_stream_sys_t *p_sys = p_stream->p_sys;
1226 vlc_mutex_lock( &p_sys->lock_es );
1227 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1228 vlc_mutex_unlock( &p_sys->lock_es );
1230 if( likely(id->p_fifo != NULL) )
1232 vlc_cancel( id->thread );
1233 vlc_join( id->thread, NULL );
1234 block_FifoRelease( id->p_fifo );
1237 free( id->rtp_fmt.fmtp );
1239 if (p_sys->p_vod_media != NULL)
1240 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1242 RtspDelId( p_sys->rtsp, id->rtsp_id );
1243 if( id->listen.fd != NULL )
1245 vlc_cancel( id->listen.thread );
1246 vlc_join( id->listen.thread, NULL );
1247 net_ListenClose( id->listen.fd );
1249 /* Delete remaining sinks (incoming connections or explicit
1251 while( id->sinkc > 0 )
1252 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1254 if( id->srtp != NULL )
1255 srtp_destroy( id->srtp );
1258 vlc_mutex_destroy( &id->lock_sink );
1260 /* Update SDP (sap/file) */
1261 if( p_sys->b_export_sap ) SapSetup( p_stream );
1262 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1267 static int Send( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1270 assert( p_stream->p_sys->p_mux == NULL );
1273 while( p_buffer != NULL )
1275 block_t *p_next = p_buffer->p_next;
1276 p_buffer->p_next = NULL;
1278 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1279 * as the first packet of the stream */
1280 if (id->b_first_packet)
1282 id->b_first_packet = false;
1283 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1284 !strcmp(id->rtp_fmt.ptname, "theora"))
1285 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1289 if( id->rtp_fmt.pf_packetize( id, 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_sys_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_sys_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 );
1503 vlc_assert_unreachable();
1507 int rtp_add_sink( sout_stream_id_sys_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_sys_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_sys_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_sys_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_sys_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 );
1657 uint16_t rtp_get_extended_sequence( sout_stream_id_sys_t *id )
1659 return id->i_sequence >> 16;
1662 void rtp_packetize_send( sout_stream_id_sys_t *id, block_t *out )
1664 block_FifoPut( id->p_fifo, out );
1668 * @return configured max RTP payload size (including payload type-specific
1669 * headers, excluding RTP and transport headers)
1671 size_t rtp_mtu (const sout_stream_id_sys_t *id)
1673 return id->i_mtu - 12;
1676 /*****************************************************************************
1678 *****************************************************************************/
1680 /** Add an ES to a non-RTP muxed stream */
1681 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *p_stream,
1682 const es_format_t *p_fmt )
1684 sout_input_t *p_input;
1685 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1686 assert( p_mux != NULL );
1688 p_input = sout_MuxAddStream( p_mux, p_fmt );
1689 if( p_input == NULL )
1691 msg_Err( p_stream, "cannot add this stream to the muxer" );
1695 return (sout_stream_id_sys_t *)p_input;
1699 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1702 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1703 assert( p_mux != NULL );
1705 return sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1709 /** Remove an ES from a non-RTP muxed stream */
1710 static void MuxDel( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1712 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1713 assert( p_mux != NULL );
1715 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1719 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1720 const block_t *p_buffer )
1722 sout_stream_sys_t *p_sys = p_stream->p_sys;
1723 sout_stream_id_sys_t *id = p_sys->es[0];
1725 int64_t i_dts = p_buffer->i_dts;
1727 uint8_t *p_data = p_buffer->p_buffer;
1728 size_t i_data = p_buffer->i_buffer;
1729 size_t i_max = id->i_mtu - 12;
1731 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1737 /* output complete packet */
1738 if( p_sys->packet &&
1739 p_sys->packet->i_buffer + i_data > i_max )
1741 rtp_packetize_send( id, p_sys->packet );
1742 p_sys->packet = NULL;
1745 if( p_sys->packet == NULL )
1747 /* allocate a new packet */
1748 p_sys->packet = block_Alloc( id->i_mtu );
1749 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1750 p_sys->packet->i_dts = i_dts;
1751 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1752 i_dts += p_sys->packet->i_length;
1755 i_size = __MIN( i_data,
1756 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1758 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1761 p_sys->packet->i_buffer += i_size;
1770 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1773 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1779 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1781 p_next = p_buffer->p_next;
1782 block_Release( p_buffer );
1790 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1792 sout_access_out_t *p_grab;
1794 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1795 if( p_grab == NULL )
1798 p_grab->p_module = NULL;
1799 p_grab->psz_access = strdup( "grab" );
1800 p_grab->p_cfg = NULL;
1801 p_grab->psz_path = strdup( "" );
1802 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1803 p_grab->pf_seek = NULL;
1804 p_grab->pf_write = AccessOutGrabberWrite;
1808 void rtp_get_video_geometry( sout_stream_id_sys_t *id, int *width, int *height )
1810 int ret = sscanf( id->rtp_fmt.fmtp, "%*s width=%d; height=%d; ", width, height );