1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
32 #include <vlc_common.h>
33 #include <vlc_plugin.h>
35 #include <vlc_block.h>
37 #include <vlc_httpd.h>
39 #include <vlc_network.h>
41 #include <vlc_strings.h>
50 # 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 an 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 DESC_TEXT N_("Session description")
95 #define DESC_LONGTEXT N_( \
96 "This allows you to give a short description with details about the stream, " \
97 "that will be announced in the SDP (Session Descriptor)." )
98 #define URL_TEXT N_("Session URL")
99 #define URL_LONGTEXT N_( \
100 "This allows you to give an URL with more details about the stream " \
101 "(often the website of the streaming organization), that will " \
102 "be announced in the SDP (Session Descriptor)." )
103 #define EMAIL_TEXT N_("Session email")
104 #define EMAIL_LONGTEXT N_( \
105 "This allows you to give a contact mail address for the stream, that will " \
106 "be announced in the SDP (Session Descriptor)." )
107 #define PHONE_TEXT N_("Session phone number")
108 #define PHONE_LONGTEXT N_( \
109 "This allows you to give a contact telephone number for the stream, that will " \
110 "be announced in the SDP (Session Descriptor)." )
112 #define PORT_TEXT N_("Port")
113 #define PORT_LONGTEXT N_( \
114 "This allows you to specify the base port for the RTP streaming." )
115 #define PORT_AUDIO_TEXT N_("Audio port")
116 #define PORT_AUDIO_LONGTEXT N_( \
117 "This allows you to specify the default audio port for the RTP streaming." )
118 #define PORT_VIDEO_TEXT N_("Video port")
119 #define PORT_VIDEO_LONGTEXT N_( \
120 "This allows you to specify the default video port for the RTP streaming." )
122 #define TTL_TEXT N_("Hop limit (TTL)")
123 #define TTL_LONGTEXT N_( \
124 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
125 "the multicast packets sent by the stream output (-1 = use operating " \
126 "system built-in default).")
128 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
129 #define RTCP_MUX_LONGTEXT N_( \
130 "This sends and receives RTCP packet multiplexed over the same port " \
133 #define CACHING_TEXT N_("Caching value (ms)")
134 #define CACHING_LONGTEXT N_( \
135 "Default caching value for outbound RTP streams. This " \
136 "value should be set in milliseconds." )
138 #define PROTO_TEXT N_("Transport protocol")
139 #define PROTO_LONGTEXT N_( \
140 "This selects which transport protocol to use for RTP." )
142 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
143 #define SRTP_KEY_LONGTEXT N_( \
144 "RTP packets will be integrity-protected and ciphered "\
145 "with this Secure RTP master shared secret key.")
147 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
148 #define SRTP_SALT_LONGTEXT N_( \
149 "Secure RTP requires a (non-secret) master salt value.")
151 static const char *const ppsz_protos[] = {
152 "dccp", "sctp", "tcp", "udp", "udplite",
155 static const char *const ppsz_protocols[] = {
156 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
159 #define RFC3016_TEXT N_("MP4A LATM")
160 #define RFC3016_LONGTEXT N_( \
161 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
163 static int Open ( vlc_object_t * );
164 static void Close( vlc_object_t * );
166 #define SOUT_CFG_PREFIX "sout-rtp-"
167 #define MAX_EMPTY_BLOCKS 200
170 set_shortname( N_("RTP"))
171 set_description( N_("RTP stream output") )
172 set_capability( "sout stream", 0 )
173 add_shortcut( "rtp" )
174 set_category( CAT_SOUT )
175 set_subcategory( SUBCAT_SOUT_STREAM )
177 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
178 DEST_LONGTEXT, true )
179 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
181 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
183 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
186 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
187 NAME_LONGTEXT, true )
188 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
189 DESC_LONGTEXT, true )
190 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
192 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
193 EMAIL_LONGTEXT, true )
194 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
195 PHONE_LONGTEXT, true )
197 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
198 PROTO_LONGTEXT, false )
199 change_string_list( ppsz_protos, ppsz_protocols, NULL )
200 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
201 PORT_LONGTEXT, true )
202 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
203 PORT_AUDIO_LONGTEXT, true )
204 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
205 PORT_VIDEO_LONGTEXT, true )
207 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
209 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
210 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
211 add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
212 CACHING_TEXT, CACHING_LONGTEXT, true )
215 add_string( SOUT_CFG_PREFIX "key", "", NULL,
216 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
217 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
218 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
221 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT,
222 RFC3016_LONGTEXT, false )
224 set_callbacks( Open, Close )
227 /*****************************************************************************
228 * Exported prototypes
229 *****************************************************************************/
230 static const char *const ppsz_sout_options[] = {
231 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
232 "sap", "description", "url", "email", "phone",
233 "proto", "rtcp-mux", "caching", "key", "salt",
237 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
238 static int Del ( sout_stream_t *, sout_stream_id_t * );
239 static int Send( sout_stream_t *, sout_stream_id_t *,
241 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
242 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
243 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
246 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
247 static void* ThreadSend( vlc_object_t *p_this );
248 static void *rtp_listen_thread( void * );
250 static void SDPHandleUrl( sout_stream_t *, const char * );
252 static int SapSetup( sout_stream_t *p_stream );
253 static int FileSetup( sout_stream_t *p_stream );
254 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
256 struct sout_stream_sys_t
260 vlc_mutex_t lock_sdp;
267 session_descriptor_t *p_session;
270 httpd_host_t *p_httpd_host;
271 httpd_file_t *p_httpd_file;
276 /* RTSP NPT and timestamp computations */
277 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
278 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
279 int64_t i_pts_offset; /* matches actual PTS to prediction */
283 char *psz_destination;
284 uint32_t payload_bitmap;
286 uint16_t i_port_audio;
287 uint16_t i_port_video;
293 /* in case we do TS/PS over rtp */
295 sout_access_out_t *p_grab;
301 sout_stream_id_t **es;
304 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
306 typedef struct rtp_sink_t
312 struct sout_stream_id_t
316 sout_stream_t *p_stream;
319 uint8_t i_payload_type;
321 uint32_t i_ts_offset;
325 uint16_t i_seq_sent_next;
336 /* Packetizer specific fields */
339 srtp_session_t *srtp;
341 pf_rtp_packetizer_t pf_packetize;
344 vlc_mutex_t lock_sink;
347 rtsp_stream_id_t *rtsp_id;
353 block_fifo_t *p_fifo;
357 /*****************************************************************************
359 *****************************************************************************/
360 static int Open( vlc_object_t *p_this )
362 sout_stream_t *p_stream = (sout_stream_t*)p_this;
363 sout_instance_t *p_sout = p_stream->p_sout;
364 sout_stream_sys_t *p_sys = NULL;
365 config_chain_t *p_cfg = NULL;
369 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
370 ppsz_sout_options, p_stream->p_cfg );
372 p_sys = malloc( sizeof( sout_stream_sys_t ) );
376 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
378 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
379 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
380 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
381 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
383 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
385 msg_Err( p_stream, "audio and video RTP port must be distinct" );
386 free( p_sys->psz_destination );
391 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
393 if( !strcmp( p_cfg->psz_name, "sdp" )
394 && ( p_cfg->psz_value != NULL )
395 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
403 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
406 if( !strncasecmp( psz, "rtsp:", 5 ) )
412 /* Transport protocol */
413 p_sys->proto = IPPROTO_UDP;
414 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
416 if ((psz == NULL) || !strcasecmp (psz, "udp"))
417 (void)0; /* default */
419 if (!strcasecmp (psz, "dccp"))
421 p_sys->proto = IPPROTO_DCCP;
422 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
426 if (!strcasecmp (psz, "sctp"))
428 p_sys->proto = IPPROTO_TCP;
429 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
434 if (!strcasecmp (psz, "tcp"))
436 p_sys->proto = IPPROTO_TCP;
437 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
441 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
442 p_sys->proto = IPPROTO_UDPLITE;
444 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
447 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
449 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
451 msg_Err( p_stream, "missing destination and not in RTSP mode" );
456 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
457 if( p_sys->i_ttl == -1 )
459 /* Normally, we should let the default hop limit up to the core,
460 * but we have to know it to write our RTSP headers properly,
461 * which is why we ask the core. FIXME: broken when neither
462 * sout-rtp-ttl nor ttl are set. */
463 p_sys->i_ttl = var_InheritInteger( p_stream, "ttl" );
466 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
468 /* NPT=0 time will be determined when we packetize the first packet
469 * (of any ES). But we want to be able to report rtptime in RTSP
470 * without waiting. So until then, we use an arbitrary reference
471 * PTS for timestamp computations, and then actual PTS will catch
472 * up using offsets. */
473 p_sys->i_npt_zero = VLC_TS_INVALID;
474 p_sys->i_pts_zero = mdate(); /* arbitrary value, could probably be
476 p_sys->payload_bitmap = 0xFFFFFFFF;
480 p_sys->psz_sdp = NULL;
482 p_sys->b_export_sap = false;
483 p_sys->p_session = NULL;
484 p_sys->psz_sdp_file = NULL;
486 p_sys->p_httpd_host = NULL;
487 p_sys->p_httpd_file = NULL;
489 p_stream->p_sys = p_sys;
491 vlc_mutex_init( &p_sys->lock_sdp );
492 vlc_mutex_init( &p_sys->lock_ts );
493 vlc_mutex_init( &p_sys->lock_es );
495 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
498 sout_stream_id_t *id;
500 /* Check muxer type */
501 if( strncasecmp( psz, "ps", 2 )
502 && strncasecmp( psz, "mpeg1", 5 )
503 && strncasecmp( psz, "ts", 2 ) )
505 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
507 vlc_mutex_destroy( &p_sys->lock_sdp );
508 vlc_mutex_destroy( &p_sys->lock_es );
509 free( p_sys->psz_destination );
514 p_sys->p_grab = GrabberCreate( p_stream );
515 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
518 if( p_sys->p_mux == NULL )
520 msg_Err( p_stream, "cannot create muxer" );
521 sout_AccessOutDelete( p_sys->p_grab );
522 vlc_mutex_destroy( &p_sys->lock_sdp );
523 vlc_mutex_destroy( &p_sys->lock_es );
524 free( p_sys->psz_destination );
529 id = Add( p_stream, NULL );
532 sout_MuxDelete( p_sys->p_mux );
533 sout_AccessOutDelete( p_sys->p_grab );
534 vlc_mutex_destroy( &p_sys->lock_sdp );
535 vlc_mutex_destroy( &p_sys->lock_es );
536 free( p_sys->psz_destination );
541 p_sys->packet = NULL;
543 p_stream->pf_add = MuxAdd;
544 p_stream->pf_del = MuxDel;
545 p_stream->pf_send = MuxSend;
550 p_sys->p_grab = NULL;
552 p_stream->pf_add = Add;
553 p_stream->pf_del = Del;
554 p_stream->pf_send = Send;
557 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
558 SDPHandleUrl( p_stream, "sap" );
560 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
563 config_chain_t *p_cfg;
565 SDPHandleUrl( p_stream, psz );
567 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
569 if( !strcmp( p_cfg->psz_name, "sdp" ) )
571 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
574 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
575 if( !strcmp( p_cfg->psz_value, psz ) )
578 SDPHandleUrl( p_stream, p_cfg->psz_value );
584 /* update p_sout->i_out_pace_nocontrol */
585 p_stream->p_sout->i_out_pace_nocontrol++;
590 /*****************************************************************************
592 *****************************************************************************/
593 static void Close( vlc_object_t * p_this )
595 sout_stream_t *p_stream = (sout_stream_t*)p_this;
596 sout_stream_sys_t *p_sys = p_stream->p_sys;
598 /* update p_sout->i_out_pace_nocontrol */
599 p_stream->p_sout->i_out_pace_nocontrol--;
603 assert( p_sys->i_es == 1 );
605 sout_MuxDelete( p_sys->p_mux );
606 Del( p_stream, p_sys->es[0] );
607 sout_AccessOutDelete( p_sys->p_grab );
611 block_Release( p_sys->packet );
613 if( p_sys->b_export_sap )
616 SapSetup( p_stream );
620 if( p_sys->rtsp != NULL )
621 RtspUnsetup( p_sys->rtsp );
623 vlc_mutex_destroy( &p_sys->lock_sdp );
624 vlc_mutex_destroy( &p_sys->lock_ts );
625 vlc_mutex_destroy( &p_sys->lock_es );
627 if( p_sys->p_httpd_file )
628 httpd_FileDelete( p_sys->p_httpd_file );
630 if( p_sys->p_httpd_host )
631 httpd_HostDelete( p_sys->p_httpd_host );
633 free( p_sys->psz_sdp );
635 if( p_sys->psz_sdp_file != NULL )
638 unlink( p_sys->psz_sdp_file );
640 free( p_sys->psz_sdp_file );
642 free( p_sys->psz_destination );
646 /*****************************************************************************
648 *****************************************************************************/
649 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
651 sout_stream_sys_t *p_sys = p_stream->p_sys;
654 vlc_UrlParse( &url, psz_url, 0 );
655 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
657 if( p_sys->p_httpd_file )
659 msg_Err( p_stream, "you can use sdp=http:// only once" );
663 if( HttpSetup( p_stream, &url ) )
665 msg_Err( p_stream, "cannot export SDP as HTTP" );
668 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
670 if( p_sys->rtsp != NULL )
672 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
676 /* FIXME test if destination is multicast or no destination at all */
677 p_sys->rtsp = RtspSetup( p_stream, &url );
678 if( p_sys->rtsp == NULL )
679 msg_Err( p_stream, "cannot export SDP as RTSP" );
681 if( p_sys->p_mux != NULL )
683 sout_stream_id_t *id = p_sys->es[0];
684 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
685 p_sys->psz_destination, p_sys->i_ttl,
686 id->i_port, id->i_port + 1 );
689 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
690 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
692 p_sys->b_export_sap = true;
693 SapSetup( p_stream );
695 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
697 if( p_sys->psz_sdp_file != NULL )
699 msg_Err( p_stream, "you can use sdp=file:// only once" );
702 psz_url = &psz_url[5];
703 if( psz_url[0] == '/' && psz_url[1] == '/' )
705 p_sys->psz_sdp_file = strdup( psz_url );
706 if( p_sys->psz_sdp_file == NULL )
708 decode_URI( p_sys->psz_sdp_file ); /* FIXME? */
709 FileSetup( p_stream );
713 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
718 vlc_UrlClean( &url );
721 /*****************************************************************************
723 *****************************************************************************/
725 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
727 const sout_stream_sys_t *p_sys = p_stream->p_sys;
729 struct sockaddr_storage dst;
733 * When we have a fixed destination (typically when we do multicast),
734 * we need to put the actual port numbers in the SDP.
735 * When there is no fixed destination, we only support RTSP unicast
736 * on-demand setup, so we should rather let the clients decide which ports
738 * When there is both a fixed destination and RTSP unicast, we need to
739 * put port numbers used by the fixed destination, otherwise the SDP would
740 * become totally incorrect for multicast use. It should be noted that
741 * port numbers from SDP with RTSP are only "recommendation" from the
742 * server to the clients (per RFC2326), so only broken clients will fail
743 * to handle this properly. There is no solution but to use two differents
744 * output chain with two different RTSP URLs if you need to handle this
749 if( p_sys->psz_destination != NULL )
753 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
754 dstlen = sizeof( dst );
755 if( p_sys->es[0]->listen.fd != NULL )
756 getsockname( p_sys->es[0]->listen.fd[0],
757 (struct sockaddr *)&dst, &dstlen );
759 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
760 (struct sockaddr *)&dst, &dstlen );
766 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
767 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
768 && rtsp_url[7] == '[';
770 /* Dummy destination address for RTSP */
771 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
772 : sizeof( struct sockaddr_in );
773 memset (&dst, 0, dstlen);
774 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
780 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
781 NULL, 0, (struct sockaddr *)&dst, dstlen );
782 if( psz_sdp == NULL )
785 /* TODO: a=source-filter */
786 if( p_sys->rtcp_mux )
787 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
789 if( rtsp_url != NULL )
790 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
792 const char *proto = "RTP/AVP"; /* protocol */
793 if( rtsp_url == NULL )
795 switch( p_sys->proto )
800 proto = "TCP/RTP/AVP";
803 proto = "DCCP/RTP/AVP";
805 case IPPROTO_UDPLITE:
810 /* FIXME: locking?! */
811 for( i = 0; i < p_sys->i_es; i++ )
813 sout_stream_id_t *id = p_sys->es[i];
814 const char *mime_major; /* major MIME type */
819 mime_major = "video";
822 mime_major = "audio";
831 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
832 id->i_payload_type, false, id->i_bitrate,
833 id->psz_enc, id->i_clock_rate, id->i_channels,
836 if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
837 sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
839 if( rtsp_url != NULL )
841 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
842 if( track_url != NULL )
844 sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
850 if( id->listen.fd != NULL )
851 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
852 if( p_sys->proto == IPPROTO_DCCP )
853 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
854 "SC:RTP%c", toupper( mime_major[0] ) );
861 /*****************************************************************************
863 *****************************************************************************/
865 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
867 static const char hex[16] = "0123456789abcdef";
870 for( i = 0; i < i_data; i++ )
872 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
873 s[2*i+1] = hex[(p_data[i] )&0xf];
879 * Shrink the MTU down to a fixed packetization time (for audio).
882 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
884 /* Samples per second */
885 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
886 bytes *= id->i_channels;
889 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
890 id->i_mtu = 12 + spl;
891 else /* MTU is too small for ptime, align to a sample boundary */
892 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
895 uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts )
897 /* NOTE: this plays nice with offsets because the calculations are
899 return i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
902 /** Add an ES as a new RTP stream */
903 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
905 /* NOTE: As a special case, if we use a non-RTP
906 * mux (TS/PS), then p_fmt is NULL. */
907 sout_stream_sys_t *p_sys = p_stream->p_sys;
908 sout_stream_id_t *id;
911 if (0 == p_sys->payload_bitmap)
913 msg_Err (p_stream, "too many RTP elementary streams");
917 /* Choose the port */
922 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
923 i_port = p_sys->i_port_audio;
925 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
926 i_port = p_sys->i_port_video;
928 /* We do not need the ES lock (p_sys->lock_es) here, because this is the
929 * only one thread that can *modify* the ES table. The ES lock protects
930 * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
931 for (int i = 0; i_port && (i < p_sys->i_es); i++)
932 if (i_port == p_sys->es[i]->i_port)
933 i_port = 0; /* Port already in use! */
934 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
938 msg_Err (p_stream, "too many RTP elementary streams");
942 for (int i = 0; i_port && (i < p_sys->i_es); i++)
943 if (p == p_sys->es[i]->i_port)
947 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
950 vlc_object_attach( id, p_stream );
952 id->p_stream = p_stream;
954 /* Look for free dymanic payload type */
955 id->i_payload_type = 96 + clz32 (p_sys->payload_bitmap);
956 assert (id->i_payload_type < 128);
958 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
959 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
961 id->i_seq_sent_next = id->i_sequence;
965 id->i_clock_rate = 90000; /* most common case for video */
970 id->i_cat = p_fmt->i_cat;
971 if( p_fmt->i_cat == AUDIO_ES )
973 id->i_clock_rate = p_fmt->audio.i_rate;
974 id->i_channels = p_fmt->audio.i_channels;
976 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
980 id->i_cat = VIDEO_ES;
984 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
985 if( id->i_mtu <= 12 + 16 )
986 id->i_mtu = 576 - 20 - 8; /* pessimistic */
987 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
989 id->pf_packetize = NULL;
994 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
997 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
998 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
999 if (id->srtp == NULL)
1005 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1006 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1011 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
1014 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1018 vlc_mutex_init( &id->lock_sink );
1023 id->listen.fd = NULL;
1026 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
1028 if( p_sys->psz_destination != NULL )
1029 switch( p_sys->proto )
1036 case VIDEO_ES: code = "RTPV"; break;
1037 case AUDIO_ES: code = "RTPARTPV"; break;
1038 case SPU_ES: code = "RTPTRTPV"; break;
1039 default: code = "RTPORTPV"; break;
1041 var_SetString (p_stream, "dccp-service", code);
1042 } /* fall through */
1044 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1045 p_sys->psz_destination, i_port,
1047 if( id->listen.fd == NULL )
1049 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1052 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1053 VLC_THREAD_PRIORITY_LOW ) )
1055 net_ListenClose( id->listen.fd );
1056 id->listen.fd = NULL;
1063 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
1064 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1065 i_port, ttl, p_sys->proto );
1068 msg_Err( p_stream, "cannot create RTP socket" );
1071 /* Ignore any unexpected incoming packet (including RTCP-RR
1072 * packets in case of rtcp-mux) */
1073 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1075 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1081 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1083 if( psz == NULL ) /* Uho! */
1086 if( strncmp( psz, "ts", 2 ) == 0 )
1088 id->i_payload_type = 33;
1089 id->psz_enc = "MP2T";
1093 id->psz_enc = "MP2P";
1098 switch( p_fmt->i_codec )
1100 case VLC_CODEC_MULAW:
1101 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1102 id->i_payload_type = 0;
1103 id->psz_enc = "PCMU";
1104 id->pf_packetize = rtp_packetize_split;
1105 rtp_set_ptime (id, 20, 1);
1107 case VLC_CODEC_ALAW:
1108 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1109 id->i_payload_type = 8;
1110 id->psz_enc = "PCMA";
1111 id->pf_packetize = rtp_packetize_split;
1112 rtp_set_ptime (id, 20, 1);
1114 case VLC_CODEC_S16B:
1115 case VLC_CODEC_S16L:
1116 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1118 id->i_payload_type = 11;
1120 else if( p_fmt->audio.i_channels == 2 &&
1121 p_fmt->audio.i_rate == 44100 )
1123 id->i_payload_type = 10;
1125 id->psz_enc = "L16";
1126 if( p_fmt->i_codec == VLC_CODEC_S16B )
1127 id->pf_packetize = rtp_packetize_split;
1129 id->pf_packetize = rtp_packetize_swab;
1130 rtp_set_ptime (id, 20, 2);
1134 id->pf_packetize = rtp_packetize_split;
1135 rtp_set_ptime (id, 20, 1);
1137 case VLC_CODEC_MPGA:
1138 id->i_payload_type = 14;
1139 id->psz_enc = "MPA";
1140 id->i_clock_rate = 90000; /* not 44100 */
1141 id->pf_packetize = rtp_packetize_mpa;
1143 case VLC_CODEC_MPGV:
1144 id->i_payload_type = 32;
1145 id->psz_enc = "MPV";
1146 id->pf_packetize = rtp_packetize_mpv;
1148 case VLC_CODEC_ADPCM_G726:
1149 switch( p_fmt->i_bitrate / 1000 )
1152 id->psz_enc = "G726-16";
1153 id->pf_packetize = rtp_packetize_g726_16;
1156 id->psz_enc = "G726-24";
1157 id->pf_packetize = rtp_packetize_g726_24;
1160 id->psz_enc = "G726-32";
1161 id->pf_packetize = rtp_packetize_g726_32;
1164 id->psz_enc = "G726-40";
1165 id->pf_packetize = rtp_packetize_g726_40;
1168 msg_Err( p_stream, "cannot add this stream (unsupported "
1169 "G.726 bit rate: %u)", p_fmt->i_bitrate );
1174 id->psz_enc = "ac3";
1175 id->pf_packetize = rtp_packetize_ac3;
1177 case VLC_CODEC_H263:
1178 id->psz_enc = "H263-1998";
1179 id->pf_packetize = rtp_packetize_h263;
1181 case VLC_CODEC_H264:
1182 id->psz_enc = "H264";
1183 id->pf_packetize = rtp_packetize_h264;
1184 id->psz_fmtp = NULL;
1186 if( p_fmt->i_extra > 0 )
1188 uint8_t *p_buffer = p_fmt->p_extra;
1189 int i_buffer = p_fmt->i_extra;
1190 char *p_64_sps = NULL;
1191 char *p_64_pps = NULL;
1194 while( i_buffer > 4 &&
1195 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1196 p_buffer[2] == 0 && p_buffer[3] == 1 )
1198 const int i_nal_type = p_buffer[4]&0x1f;
1202 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1205 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1207 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1209 /* we found another startcode */
1214 if( i_nal_type == 7 )
1216 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1217 sprintf_hexa( hexa, &p_buffer[5], 3 );
1219 else if( i_nal_type == 8 )
1221 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1227 if( p_64_sps && p_64_pps &&
1228 ( asprintf( &id->psz_fmtp,
1229 "packetization-mode=1;profile-level-id=%s;"
1230 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1231 p_64_pps ) == -1 ) )
1232 id->psz_fmtp = NULL;
1237 id->psz_fmtp = strdup( "packetization-mode=1" );
1240 case VLC_CODEC_MP4V:
1242 id->psz_enc = "MP4V-ES";
1243 id->pf_packetize = rtp_packetize_split;
1244 if( p_fmt->i_extra > 0 )
1246 char hexa[2*p_fmt->i_extra +1];
1247 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1248 if( asprintf( &id->psz_fmtp,
1249 "profile-level-id=3; config=%s;", hexa ) == -1 )
1250 id->psz_fmtp = NULL;
1254 case VLC_CODEC_MP4A:
1258 char hexa[2*p_fmt->i_extra +1];
1260 id->psz_enc = "mpeg4-generic";
1261 id->pf_packetize = rtp_packetize_mp4a;
1262 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1263 if( asprintf( &id->psz_fmtp,
1264 "streamtype=5; profile-level-id=15; "
1265 "mode=AAC-hbr; config=%s; SizeLength=13; "
1266 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1268 id->psz_fmtp = NULL;
1274 unsigned char config[6];
1275 unsigned int aacsrates[15] = {
1276 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1277 16000, 12000, 11025, 8000, 7350, 0, 0 };
1279 for( i = 0; i < 15; i++ )
1280 if( p_fmt->audio.i_rate == aacsrates[i] )
1286 config[3]=p_fmt->audio.i_channels<<4;
1290 id->psz_enc = "MP4A-LATM";
1291 id->pf_packetize = rtp_packetize_mp4a_latm;
1292 sprintf_hexa( hexa, config, 6 );
1293 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1294 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1295 id->psz_fmtp = NULL;
1299 case VLC_CODEC_AMR_NB:
1300 id->psz_enc = "AMR";
1301 id->psz_fmtp = strdup( "octet-align=1" );
1302 id->pf_packetize = rtp_packetize_amr;
1304 case VLC_CODEC_AMR_WB:
1305 id->psz_enc = "AMR-WB";
1306 id->psz_fmtp = strdup( "octet-align=1" );
1307 id->pf_packetize = rtp_packetize_amr;
1309 case VLC_CODEC_SPEEX:
1310 id->psz_enc = "SPEEX";
1311 id->pf_packetize = rtp_packetize_spx;
1313 case VLC_CODEC_ITU_T140:
1314 id->psz_enc = "t140" ;
1315 id->i_clock_rate = 1000;
1316 id->pf_packetize = rtp_packetize_t140;
1320 msg_Err( p_stream, "cannot add this stream (unsupported "
1321 "codec: %4.4s)", (char*)&p_fmt->i_codec );
1324 if (id->i_payload_type >= 96)
1325 /* Mark dynamic payload type in use */
1326 p_sys->payload_bitmap &= ~(1 << (127 - id->i_payload_type));
1328 #if 0 /* No payload formats sets this at the moment */
1331 cscov += 8 /* UDP */ + 12 /* RTP */;
1333 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1336 vlc_mutex_lock( &p_sys->lock_ts );
1337 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1338 vlc_mutex_unlock( &p_sys->lock_ts );
1340 id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
1342 if( p_sys->rtsp != NULL )
1343 id->rtsp_id = RtspAddId( p_sys->rtsp, id,
1344 GetDWBE( id->ssrc ),
1345 p_sys->psz_destination,
1346 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1348 id->p_fifo = block_FifoNew();
1349 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1350 VLC_THREAD_PRIORITY_HIGHEST ) )
1353 /* Update p_sys context */
1354 vlc_mutex_lock( &p_sys->lock_es );
1355 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1356 vlc_mutex_unlock( &p_sys->lock_es );
1358 psz_sdp = SDPGenerate( p_stream, NULL );
1360 vlc_mutex_lock( &p_sys->lock_sdp );
1361 free( p_sys->psz_sdp );
1362 p_sys->psz_sdp = psz_sdp;
1363 vlc_mutex_unlock( &p_sys->lock_sdp );
1365 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1367 /* Update SDP (sap/file) */
1368 if( p_sys->b_export_sap ) SapSetup( p_stream );
1369 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1374 Del( p_stream, id );
1378 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1380 sout_stream_sys_t *p_sys = p_stream->p_sys;
1382 if( id->p_fifo != NULL )
1384 vlc_object_kill( id );
1385 vlc_thread_join( id );
1386 block_FifoRelease( id->p_fifo );
1389 vlc_mutex_lock( &p_sys->lock_es );
1390 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1391 vlc_mutex_unlock( &p_sys->lock_es );
1393 /* Release dynamic payload type */
1394 if (id->i_payload_type >= 96)
1395 p_sys->payload_bitmap |= 1 << (127 - id->i_payload_type);
1397 free( id->psz_fmtp );
1400 RtspDelId( p_sys->rtsp, id->rtsp_id );
1401 if( id->listen.fd != NULL )
1403 vlc_cancel( id->listen.thread );
1404 vlc_join( id->listen.thread, NULL );
1405 net_ListenClose( id->listen.fd );
1407 /* Delete remaining sinks (incoming connections or explicit
1409 while( id->sinkc > 0 )
1410 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1412 if( id->srtp != NULL )
1413 srtp_destroy( id->srtp );
1416 vlc_mutex_destroy( &id->lock_sink );
1418 /* Update SDP (sap/file) */
1419 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1420 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1422 vlc_object_release( id );
1426 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1431 assert( p_stream->p_sys->p_mux == NULL );
1434 while( p_buffer != NULL )
1436 p_next = p_buffer->p_next;
1437 if( id->pf_packetize( id, p_buffer ) )
1440 block_Release( p_buffer );
1446 /****************************************************************************
1448 ****************************************************************************/
1449 static int SapSetup( sout_stream_t *p_stream )
1451 sout_stream_sys_t *p_sys = p_stream->p_sys;
1452 sout_instance_t *p_sout = p_stream->p_sout;
1454 /* Remove the previous session */
1455 if( p_sys->p_session != NULL)
1457 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1458 p_sys->p_session = NULL;
1461 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1463 announce_method_t *p_method = sout_SAPMethod();
1464 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1466 p_sys->psz_destination,
1468 sout_MethodRelease( p_method );
1474 /****************************************************************************
1476 ****************************************************************************/
1477 static int FileSetup( sout_stream_t *p_stream )
1479 sout_stream_sys_t *p_sys = p_stream->p_sys;
1482 if( p_sys->psz_sdp == NULL )
1483 return VLC_EGENERIC; /* too early */
1485 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1487 msg_Err( p_stream, "cannot open file '%s' (%m)",
1488 p_sys->psz_sdp_file );
1489 return VLC_EGENERIC;
1492 fputs( p_sys->psz_sdp, f );
1498 /****************************************************************************
1500 ****************************************************************************/
1501 static int HttpCallback( httpd_file_sys_t *p_args,
1502 httpd_file_t *, uint8_t *p_request,
1503 uint8_t **pp_data, int *pi_data );
1505 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1507 sout_stream_sys_t *p_sys = p_stream->p_sys;
1509 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1510 url->i_port > 0 ? url->i_port : 80 );
1511 if( p_sys->p_httpd_host )
1513 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1514 url->psz_path ? url->psz_path : "/",
1517 HttpCallback, (void*)p_sys );
1519 if( p_sys->p_httpd_file == NULL )
1521 return VLC_EGENERIC;
1526 static int HttpCallback( httpd_file_sys_t *p_args,
1527 httpd_file_t *f, uint8_t *p_request,
1528 uint8_t **pp_data, int *pi_data )
1530 VLC_UNUSED(f); VLC_UNUSED(p_request);
1531 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1533 vlc_mutex_lock( &p_sys->lock_sdp );
1534 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1536 *pi_data = strlen( p_sys->psz_sdp );
1537 *pp_data = malloc( *pi_data );
1538 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1545 vlc_mutex_unlock( &p_sys->lock_sdp );
1550 /****************************************************************************
1552 ****************************************************************************/
1553 static void* ThreadSend( vlc_object_t *p_this )
1556 # define ECONNREFUSED WSAECONNREFUSED
1557 # define ENOPROTOOPT WSAENOPROTOOPT
1558 # define EHOSTUNREACH WSAEHOSTUNREACH
1559 # define ENETUNREACH WSAENETUNREACH
1560 # define ENETDOWN WSAENETDOWN
1561 # define ENOBUFS WSAENOBUFS
1562 # define EAGAIN WSAEWOULDBLOCK
1563 # define EWOULDBLOCK WSAEWOULDBLOCK
1565 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1566 unsigned i_caching = id->i_caching;
1570 block_t *out = block_FifoGet( id->p_fifo );
1571 block_cleanup_push (out);
1575 { /* FIXME: this is awfully inefficient */
1576 size_t len = out->i_buffer;
1577 out = block_Realloc( out, 0, len + 10 );
1578 out->i_buffer = len;
1580 int canc = vlc_savecancel ();
1581 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1582 vlc_restorecancel (canc);
1586 msg_Dbg( id, "SRTP sending error: %m" );
1587 block_Release( out );
1591 out->i_buffer = len;
1595 mwait (out->i_dts + i_caching);
1600 ssize_t len = out->i_buffer;
1601 int canc = vlc_savecancel ();
1603 vlc_mutex_lock( &id->lock_sink );
1604 unsigned deadc = 0; /* How many dead sockets? */
1605 int deadv[id->sinkc]; /* Dead sockets list */
1607 for( int i = 0; i < id->sinkc; i++ )
1610 if( !id->srtp ) /* FIXME: SRTCP support */
1612 SendRTCP( id->sinkv[i].rtcp, out );
1614 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1618 /* Soft errors (e.g. ICMP): */
1619 case ECONNREFUSED: /* Port unreachable */
1622 case EPROTO: /* Protocol unreachable */
1624 case EHOSTUNREACH: /* Host unreachable */
1625 case ENETUNREACH: /* Network unreachable */
1626 case ENETDOWN: /* Entire network down */
1627 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1628 /* Transient congestion: */
1629 case ENOMEM: /* out of socket buffers */
1632 #if (EAGAIN != EWOULDBLOCK)
1638 deadv[deadc++] = id->sinkv[i].rtp_fd;
1640 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1641 vlc_mutex_unlock( &id->lock_sink );
1642 block_Release( out );
1644 for( unsigned i = 0; i < deadc; i++ )
1646 msg_Dbg( id, "removing socket %d", deadv[i] );
1647 rtp_del_sink( id, deadv[i] );
1649 vlc_restorecancel (canc);
1655 /* This thread dequeues incoming connections (DCCP streaming) */
1656 static void *rtp_listen_thread( void *data )
1658 sout_stream_id_t *id = data;
1660 assert( id->listen.fd != NULL );
1664 int fd = net_Accept( id, id->listen.fd );
1667 int canc = vlc_savecancel( );
1668 rtp_add_sink( id, fd, true, NULL );
1669 vlc_restorecancel( canc );
1676 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1678 rtp_sink_t sink = { fd, NULL };
1679 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1681 if( sink.rtcp == NULL )
1682 msg_Err( id, "RTCP failed!" );
1684 vlc_mutex_lock( &id->lock_sink );
1685 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1687 *seq = id->i_seq_sent_next;
1688 vlc_mutex_unlock( &id->lock_sink );
1692 void rtp_del_sink( sout_stream_id_t *id, int fd )
1694 rtp_sink_t sink = { fd, NULL };
1696 /* NOTE: must be safe to use if fd is not included */
1697 vlc_mutex_lock( &id->lock_sink );
1698 for( int i = 0; i < id->sinkc; i++ )
1700 if (id->sinkv[i].rtp_fd == fd)
1702 sink = id->sinkv[i];
1703 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1707 vlc_mutex_unlock( &id->lock_sink );
1709 CloseRTCP( sink.rtcp );
1710 net_Close( sink.rtp_fd );
1713 uint16_t rtp_get_seq( sout_stream_id_t *id )
1715 /* This will return values for the next packet. */
1718 vlc_mutex_lock( &id->lock_sink );
1719 seq = id->i_seq_sent_next;
1720 vlc_mutex_unlock( &id->lock_sink );
1725 /* Return a timestamp corresponding to packets being sent now, and that
1726 * can be passed to rtp_compute_ts() to get rtptime values for each ES. */
1727 int64_t rtp_get_ts( const sout_stream_t *p_stream )
1729 sout_stream_sys_t *p_sys = p_stream->p_sys;
1731 vlc_mutex_lock( &p_sys->lock_ts );
1732 i_npt_zero = p_sys->i_npt_zero;
1733 vlc_mutex_unlock( &p_sys->lock_ts );
1735 if( i_npt_zero == VLC_TS_INVALID )
1736 return p_sys->i_pts_zero;
1738 mtime_t now = mdate();
1739 if( now < i_npt_zero )
1740 return p_sys->i_pts_zero;
1742 return p_sys->i_pts_zero + (now - i_npt_zero);
1745 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1746 int b_marker, int64_t i_pts )
1748 if( !id->b_ts_init )
1750 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1751 vlc_mutex_lock( &p_sys->lock_ts );
1752 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1754 /* This is the first packet of any ES. We initialize the
1755 * NPT=0 time reference, and the offset to match the
1756 * arbitrary PTS reference. */
1757 p_sys->i_npt_zero = i_pts + id->i_caching;
1758 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1760 vlc_mutex_unlock( &p_sys->lock_ts );
1762 /* And in any case this is the first packet of this ES, so we
1763 * initialize the offset for this ES. */
1764 id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
1765 id->b_ts_init = true;
1768 uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset;
1770 out->p_buffer[0] = 0x80;
1771 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1772 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1773 out->p_buffer[3] = ( id->i_sequence )&0xff;
1774 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1775 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1776 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1777 out->p_buffer[7] = ( i_timestamp )&0xff;
1779 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1785 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1787 block_FifoPut( id->p_fifo, out );
1791 * @return configured max RTP payload size (including payload type-specific
1792 * headers, excluding RTP and transport headers)
1794 size_t rtp_mtu (const sout_stream_id_t *id)
1796 return id->i_mtu - 12;
1799 /*****************************************************************************
1801 *****************************************************************************/
1803 /** Add an ES to a non-RTP muxed stream */
1804 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1806 sout_input_t *p_input;
1807 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1808 assert( p_mux != NULL );
1810 p_input = sout_MuxAddStream( p_mux, p_fmt );
1811 if( p_input == NULL )
1813 msg_Err( p_stream, "cannot add this stream to the muxer" );
1817 return (sout_stream_id_t *)p_input;
1821 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1824 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1825 assert( p_mux != NULL );
1827 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1832 /** Remove an ES from a non-RTP muxed stream */
1833 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1835 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1836 assert( p_mux != NULL );
1838 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1843 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1844 const block_t *p_buffer )
1846 sout_stream_sys_t *p_sys = p_stream->p_sys;
1847 sout_stream_id_t *id = p_sys->es[0];
1849 int64_t i_dts = p_buffer->i_dts;
1851 uint8_t *p_data = p_buffer->p_buffer;
1852 size_t i_data = p_buffer->i_buffer;
1853 size_t i_max = id->i_mtu - 12;
1855 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1861 /* output complete packet */
1862 if( p_sys->packet &&
1863 p_sys->packet->i_buffer + i_data > i_max )
1865 rtp_packetize_send( id, p_sys->packet );
1866 p_sys->packet = NULL;
1869 if( p_sys->packet == NULL )
1871 /* allocate a new packet */
1872 p_sys->packet = block_New( p_stream, id->i_mtu );
1873 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1874 p_sys->packet->i_dts = i_dts;
1875 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1876 i_dts += p_sys->packet->i_length;
1879 i_size = __MIN( i_data,
1880 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1882 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1885 p_sys->packet->i_buffer += i_size;
1894 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1897 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1903 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1905 p_next = p_buffer->p_next;
1906 block_Release( p_buffer );
1914 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1916 sout_access_out_t *p_grab;
1918 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1919 if( p_grab == NULL )
1922 p_grab->p_module = NULL;
1923 p_grab->psz_access = strdup( "grab" );
1924 p_grab->p_cfg = NULL;
1925 p_grab->psz_path = strdup( "" );
1926 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1927 p_grab->pf_seek = NULL;
1928 p_grab->pf_write = AccessOutGrabberWrite;
1929 vlc_object_attach( p_grab, p_stream );