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>
40 #include <vlc_charset.h>
41 #include <vlc_strings.h>
48 # include <sys/types.h>
51 # include <sys/stat.h>
53 #ifdef HAVE_LINUX_DCCP_H
54 # include <linux/dccp.h>
57 # define IPPROTO_DCCP 33
59 #ifndef IPPROTO_UDPLITE
60 # define IPPROTO_UDPLITE 136
67 /*****************************************************************************
69 *****************************************************************************/
71 #define DEST_TEXT N_("Destination")
72 #define DEST_LONGTEXT N_( \
73 "This is the output URL that will be used." )
74 #define SDP_TEXT N_("SDP")
75 #define SDP_LONGTEXT N_( \
76 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
77 "session will be made available. You must use an url: http://location to " \
78 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
79 "for the SDP to be announced via SAP." )
80 #define SAP_TEXT N_("SAP announcing")
81 #define SAP_LONGTEXT N_("Announce this session with SAP.")
82 #define MUX_TEXT N_("Muxer")
83 #define MUX_LONGTEXT N_( \
84 "This allows you to specify the muxer used for the streaming output. " \
85 "Default is to use no muxer (standard RTP stream)." )
87 #define NAME_TEXT N_("Session name")
88 #define NAME_LONGTEXT N_( \
89 "This is the name of the session that will be announced in the SDP " \
90 "(Session Descriptor)." )
91 #define DESC_TEXT N_("Session description")
92 #define DESC_LONGTEXT N_( \
93 "This allows you to give a short description with details about the stream, " \
94 "that will be announced in the SDP (Session Descriptor)." )
95 #define URL_TEXT N_("Session URL")
96 #define URL_LONGTEXT N_( \
97 "This allows you to give an URL with more details about the stream " \
98 "(often the website of the streaming organization), that will " \
99 "be announced in the SDP (Session Descriptor)." )
100 #define EMAIL_TEXT N_("Session email")
101 #define EMAIL_LONGTEXT N_( \
102 "This allows you to give a contact mail address for the stream, that will " \
103 "be announced in the SDP (Session Descriptor)." )
104 #define PHONE_TEXT N_("Session phone number")
105 #define PHONE_LONGTEXT N_( \
106 "This allows you to give a contact telephone number for the stream, that will " \
107 "be announced in the SDP (Session Descriptor)." )
109 #define PORT_TEXT N_("Port")
110 #define PORT_LONGTEXT N_( \
111 "This allows you to specify the base port for the RTP streaming." )
112 #define PORT_AUDIO_TEXT N_("Audio port")
113 #define PORT_AUDIO_LONGTEXT N_( \
114 "This allows you to specify the default audio port for the RTP streaming." )
115 #define PORT_VIDEO_TEXT N_("Video port")
116 #define PORT_VIDEO_LONGTEXT N_( \
117 "This allows you to specify the default video port for the RTP streaming." )
119 #define TTL_TEXT N_("Hop limit (TTL)")
120 #define TTL_LONGTEXT N_( \
121 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
122 "the multicast packets sent by the stream output (-1 = use operating " \
123 "system built-in default).")
125 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
126 #define RTCP_MUX_LONGTEXT N_( \
127 "This sends and receives RTCP packet multiplexed over the same port " \
130 #define PROTO_TEXT N_("Transport protocol")
131 #define PROTO_LONGTEXT N_( \
132 "This selects which transport protocol to use for RTP." )
134 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
135 #define SRTP_KEY_LONGTEXT N_( \
136 "RTP packets will be integrity-protected and ciphered "\
137 "with this Secure RTP master shared secret key.")
139 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
140 #define SRTP_SALT_LONGTEXT N_( \
141 "Secure RTP requires a (non-secret) master salt value.")
143 static const char *const ppsz_protos[] = {
144 "dccp", "sctp", "tcp", "udp", "udplite",
147 static const char *const ppsz_protocols[] = {
148 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
151 #define RFC3016_TEXT N_("MP4A LATM")
152 #define RFC3016_LONGTEXT N_( \
153 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
155 static int Open ( vlc_object_t * );
156 static void Close( vlc_object_t * );
158 #define SOUT_CFG_PREFIX "sout-rtp-"
159 #define MAX_EMPTY_BLOCKS 200
162 set_shortname( N_("RTP"))
163 set_description( N_("RTP stream output") )
164 set_capability( "sout stream", 0 )
165 add_shortcut( "rtp" )
166 set_category( CAT_SOUT )
167 set_subcategory( SUBCAT_SOUT_STREAM )
169 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
170 DEST_LONGTEXT, true );
171 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
172 SDP_LONGTEXT, true );
173 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
174 MUX_LONGTEXT, true );
175 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
178 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
179 NAME_LONGTEXT, true );
180 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
181 DESC_LONGTEXT, true );
182 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
183 URL_LONGTEXT, true );
184 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
185 EMAIL_LONGTEXT, true );
186 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
187 PHONE_LONGTEXT, true );
189 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
190 PROTO_LONGTEXT, false );
191 change_string_list( ppsz_protos, ppsz_protocols, NULL );
192 add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
193 PORT_LONGTEXT, true );
194 add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
195 PORT_AUDIO_LONGTEXT, true );
196 add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
197 PORT_VIDEO_LONGTEXT, true );
199 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
200 TTL_LONGTEXT, true );
201 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
202 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );
204 add_string( SOUT_CFG_PREFIX "key", "", NULL,
205 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
206 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
207 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );
209 add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
210 RFC3016_LONGTEXT, false );
212 set_callbacks( Open, Close )
215 /*****************************************************************************
216 * Exported prototypes
217 *****************************************************************************/
218 static const char *const ppsz_sout_options[] = {
219 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
220 "sap", "description", "url", "email", "phone",
221 "proto", "rtcp-mux", "key", "salt",
225 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
226 static int Del ( sout_stream_t *, sout_stream_id_t * );
227 static int Send( sout_stream_t *, sout_stream_id_t *,
229 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
230 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
231 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
234 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
235 static void* ThreadSend( vlc_object_t *p_this );
237 static void SDPHandleUrl( sout_stream_t *, const char * );
239 static int SapSetup( sout_stream_t *p_stream );
240 static int FileSetup( sout_stream_t *p_stream );
241 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
243 struct sout_stream_sys_t
247 vlc_mutex_t lock_sdp;
250 bool b_export_sdp_file;
255 session_descriptor_t *p_session;
258 httpd_host_t *p_httpd_host;
259 httpd_file_t *p_httpd_file;
265 char *psz_destination;
266 uint32_t payload_bitmap;
268 uint16_t i_port_audio;
269 uint16_t i_port_video;
275 /* in case we do TS/PS over rtp */
277 sout_access_out_t *p_grab;
283 sout_stream_id_t **es;
286 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
288 typedef struct rtp_sink_t
294 struct sout_stream_id_t
298 sout_stream_t *p_stream;
301 uint8_t i_payload_type;
313 /* Packetizer specific fields */
315 srtp_session_t *srtp;
316 pf_rtp_packetizer_t pf_packetize;
319 vlc_mutex_t lock_sink;
322 rtsp_stream_id_t *rtsp_id;
325 block_fifo_t *p_fifo;
329 /*****************************************************************************
331 *****************************************************************************/
332 static int Open( vlc_object_t *p_this )
334 sout_stream_t *p_stream = (sout_stream_t*)p_this;
335 sout_instance_t *p_sout = p_stream->p_sout;
336 sout_stream_sys_t *p_sys = NULL;
337 config_chain_t *p_cfg = NULL;
341 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
342 ppsz_sout_options, p_stream->p_cfg );
344 p_sys = malloc( sizeof( sout_stream_sys_t ) );
348 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
350 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
351 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
352 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
353 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
355 p_sys->psz_sdp_file = NULL;
357 if( p_sys->i_port_audio == p_sys->i_port_video )
359 msg_Err( p_stream, "audio and video port cannot be the same" );
360 p_sys->i_port_audio = 0;
361 p_sys->i_port_video = 0;
364 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
366 if( !strcmp( p_cfg->psz_name, "sdp" )
367 && ( p_cfg->psz_value != NULL )
368 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
376 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
379 if( !strncasecmp( psz, "rtsp:", 5 ) )
385 /* Transport protocol */
386 p_sys->proto = IPPROTO_UDP;
387 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
389 if ((psz == NULL) || !strcasecmp (psz, "udp"))
390 (void)0; /* default */
392 if (!strcasecmp (psz, "dccp"))
394 p_sys->proto = IPPROTO_DCCP;
395 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
399 if (!strcasecmp (psz, "sctp"))
401 p_sys->proto = IPPROTO_TCP;
402 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
407 if (!strcasecmp (psz, "tcp"))
409 p_sys->proto = IPPROTO_TCP;
410 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
414 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
415 p_sys->proto = IPPROTO_UDPLITE;
417 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
420 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
422 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
424 msg_Err( p_stream, "missing destination and not in RTSP mode" );
429 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
430 if( p_sys->i_ttl == -1 )
432 /* Normally, we should let the default hop limit up to the core,
433 * but we have to know it to build our SDP properly, which is why
434 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
436 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
439 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
441 p_sys->payload_bitmap = 0;
445 p_sys->psz_sdp = NULL;
447 p_sys->b_export_sap = false;
448 p_sys->b_export_sdp_file = false;
449 p_sys->p_session = NULL;
451 p_sys->p_httpd_host = NULL;
452 p_sys->p_httpd_file = NULL;
454 p_stream->p_sys = p_sys;
456 vlc_mutex_init( &p_sys->lock_sdp );
457 vlc_mutex_init( &p_sys->lock_es );
459 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
462 sout_stream_id_t *id;
464 /* Check muxer type */
465 if( strncasecmp( psz, "ps", 2 )
466 && strncasecmp( psz, "mpeg1", 5 )
467 && strncasecmp( psz, "ts", 2 ) )
469 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
471 vlc_mutex_destroy( &p_sys->lock_sdp );
472 vlc_mutex_destroy( &p_sys->lock_es );
477 p_sys->p_grab = GrabberCreate( p_stream );
478 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
481 if( p_sys->p_mux == NULL )
483 msg_Err( p_stream, "cannot create muxer" );
484 sout_AccessOutDelete( p_sys->p_grab );
485 vlc_mutex_destroy( &p_sys->lock_sdp );
486 vlc_mutex_destroy( &p_sys->lock_es );
491 id = Add( p_stream, NULL );
494 sout_MuxDelete( p_sys->p_mux );
495 sout_AccessOutDelete( p_sys->p_grab );
496 vlc_mutex_destroy( &p_sys->lock_sdp );
497 vlc_mutex_destroy( &p_sys->lock_es );
502 p_sys->packet = NULL;
504 p_stream->pf_add = MuxAdd;
505 p_stream->pf_del = MuxDel;
506 p_stream->pf_send = MuxSend;
511 p_sys->p_grab = NULL;
513 p_stream->pf_add = Add;
514 p_stream->pf_del = Del;
515 p_stream->pf_send = Send;
518 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
519 SDPHandleUrl( p_stream, "sap" );
521 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
524 config_chain_t *p_cfg;
526 SDPHandleUrl( p_stream, psz );
528 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
530 if( !strcmp( p_cfg->psz_name, "sdp" ) )
532 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
535 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
536 if( !strcmp( p_cfg->psz_value, psz ) )
539 SDPHandleUrl( p_stream, p_cfg->psz_value );
545 /* update p_sout->i_out_pace_nocontrol */
546 p_stream->p_sout->i_out_pace_nocontrol++;
551 /*****************************************************************************
553 *****************************************************************************/
554 static void Close( vlc_object_t * p_this )
556 sout_stream_t *p_stream = (sout_stream_t*)p_this;
557 sout_stream_sys_t *p_sys = p_stream->p_sys;
559 /* update p_sout->i_out_pace_nocontrol */
560 p_stream->p_sout->i_out_pace_nocontrol--;
564 assert( p_sys->i_es == 1 );
565 Del( p_stream, p_sys->es[0] );
567 sout_MuxDelete( p_sys->p_mux );
568 sout_AccessOutDelete( p_sys->p_grab );
571 block_Release( p_sys->packet );
573 if( p_sys->b_export_sap )
576 SapSetup( p_stream );
580 if( p_sys->rtsp != NULL )
581 RtspUnsetup( p_sys->rtsp );
583 vlc_mutex_destroy( &p_sys->lock_sdp );
584 vlc_mutex_destroy( &p_sys->lock_es );
586 if( p_sys->p_httpd_file )
587 httpd_FileDelete( p_sys->p_httpd_file );
589 if( p_sys->p_httpd_host )
590 httpd_HostDelete( p_sys->p_httpd_host );
592 free( p_sys->psz_sdp );
594 if( p_sys->b_export_sdp_file )
597 unlink( p_sys->psz_sdp_file );
599 free( p_sys->psz_sdp_file );
601 free( p_sys->psz_destination );
605 /*****************************************************************************
607 *****************************************************************************/
608 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
610 sout_stream_sys_t *p_sys = p_stream->p_sys;
613 vlc_UrlParse( &url, psz_url, 0 );
614 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
616 if( p_sys->p_httpd_file )
618 msg_Err( p_stream, "you can use sdp=http:// only once" );
622 if( HttpSetup( p_stream, &url ) )
624 msg_Err( p_stream, "cannot export SDP as HTTP" );
627 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
629 if( p_sys->rtsp != NULL )
631 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
635 /* FIXME test if destination is multicast or no destination at all */
636 p_sys->rtsp = RtspSetup( p_stream, &url );
637 if( p_sys->rtsp == NULL )
638 msg_Err( p_stream, "cannot export SDP as RTSP" );
640 if( p_sys->p_mux != NULL )
642 sout_stream_id_t *id = p_sys->es[0];
643 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
644 p_sys->psz_destination, p_sys->i_ttl,
645 id->i_port, id->i_port + 1 );
648 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
649 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
651 p_sys->b_export_sap = true;
652 SapSetup( p_stream );
654 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
656 if( p_sys->b_export_sdp_file )
658 msg_Err( p_stream, "you can use sdp=file:// only once" );
661 p_sys->b_export_sdp_file = true;
662 psz_url = &psz_url[5];
663 if( psz_url[0] == '/' && psz_url[1] == '/' )
665 p_sys->psz_sdp_file = strdup( psz_url );
669 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
674 vlc_UrlClean( &url );
677 /*****************************************************************************
679 *****************************************************************************/
681 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
683 const sout_stream_sys_t *p_sys = p_stream->p_sys;
685 struct sockaddr_storage dst;
689 * When we have a fixed destination (typically when we do multicast),
690 * we need to put the actual port numbers in the SDP.
691 * When there is no fixed destination, we only support RTSP unicast
692 * on-demand setup, so we should rather let the clients decide which ports
694 * When there is both a fixed destination and RTSP unicast, we need to
695 * put port numbers used by the fixed destination, otherwise the SDP would
696 * become totally incorrect for multicast use. It should be noted that
697 * port numbers from SDP with RTSP are only "recommendation" from the
698 * server to the clients (per RFC2326), so only broken clients will fail
699 * to handle this properly. There is no solution but to use two differents
700 * output chain with two different RTSP URLs if you need to handle this
705 if( p_sys->psz_destination != NULL )
709 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
710 dstlen = sizeof( dst );
711 if( p_sys->es[0]->listen_fd != NULL )
712 getsockname( p_sys->es[0]->listen_fd[0],
713 (struct sockaddr *)&dst, &dstlen );
715 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
716 (struct sockaddr *)&dst, &dstlen );
722 /* Dummy destination address for RTSP */
723 memset (&dst, 0, sizeof( struct sockaddr_in ) );
724 dst.ss_family = AF_INET;
728 dstlen = sizeof( struct sockaddr_in );
731 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
732 NULL, 0, (struct sockaddr *)&dst, dstlen );
733 if( psz_sdp == NULL )
736 /* TODO: a=source-filter */
737 if( p_sys->rtcp_mux )
738 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
740 if( rtsp_url != NULL )
741 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
743 /* FIXME: locking?! */
744 for( i = 0; i < p_sys->i_es; i++ )
746 sout_stream_id_t *id = p_sys->es[i];
747 const char *mime_major; /* major MIME type */
748 const char *proto = "RTP/AVP"; /* protocol */
753 mime_major = "video";
756 mime_major = "audio";
765 if( rtsp_url == NULL )
767 switch( p_sys->proto )
772 proto = "TCP/RTP/AVP";
775 proto = "DCCP/RTP/AVP";
777 case IPPROTO_UDPLITE:
782 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
783 id->i_payload_type, false, id->i_bitrate,
784 id->psz_enc, id->i_clock_rate, id->i_channels,
787 if( rtsp_url != NULL )
789 assert( strlen( rtsp_url ) > 0 );
790 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
791 sdp_AddAttribute ( &psz_sdp, "control",
792 addslash ? "%s/trackID=%u" : "%strackID=%u",
797 if( id->listen_fd != NULL )
798 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
799 if( p_sys->proto == IPPROTO_DCCP )
800 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
801 "SC:RTP%c", toupper( mime_major[0] ) );
808 /*****************************************************************************
810 *****************************************************************************/
812 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
814 static const char hex[16] = "0123456789abcdef";
817 for( i = 0; i < i_data; i++ )
819 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
820 s[2*i+1] = hex[(p_data[i] )&0xf];
826 * Shrink the MTU down to a fixed packetization time (for audio).
829 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
831 /* Samples per second */
832 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
833 bytes *= id->i_channels;
836 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
837 id->i_mtu = 12 + spl;
838 else /* MTU is too small for ptime, align to a sample boundary */
839 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
842 /** Add an ES as a new RTP stream */
843 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
845 /* NOTE: As a special case, if we use a non-RTP
846 * mux (TS/PS), then p_fmt is NULL. */
847 sout_stream_sys_t *p_sys = p_stream->p_sys;
848 sout_stream_id_t *id;
849 int i_port, cscov = -1;
852 if (0xffffffff == p_sys->payload_bitmap)
854 msg_Err (p_stream, "too many RTP elementary streams");
858 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
861 vlc_object_attach( id, p_stream );
863 /* Choose the port */
868 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
870 i_port = p_sys->i_port_audio;
871 p_sys->i_port_audio = 0;
874 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
876 i_port = p_sys->i_port_video;
877 p_sys->i_port_video = 0;
882 if( p_sys->i_port != p_sys->i_port_audio
883 && p_sys->i_port != p_sys->i_port_video )
885 i_port = p_sys->i_port;
892 id->p_stream = p_stream;
894 /* Look for free dymanic payload type */
895 id->i_payload_type = 96;
896 while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
897 id->i_payload_type++;
898 assert (id->i_payload_type < 128);
900 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
901 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
905 id->i_clock_rate = 90000; /* most common case for video */
910 id->i_cat = p_fmt->i_cat;
911 if( p_fmt->i_cat == AUDIO_ES )
913 id->i_clock_rate = p_fmt->audio.i_rate;
914 id->i_channels = p_fmt->audio.i_channels;
916 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
920 id->i_cat = VIDEO_ES;
924 id->i_mtu = config_GetInt( p_stream, "mtu" );
925 if( id->i_mtu <= 12 + 16 )
926 id->i_mtu = 576 - 20 - 8; /* pessimistic */
927 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
930 id->pf_packetize = NULL;
932 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
935 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
936 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
937 if (id->srtp == NULL)
943 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
944 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
949 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
952 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
955 vlc_mutex_init( &id->lock_sink );
960 id->listen_fd = NULL;
963 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
965 if( p_sys->psz_destination != NULL )
966 switch( p_sys->proto )
973 case VIDEO_ES: code = "RTPV"; break;
974 case AUDIO_ES: code = "RTPARTPV"; break;
975 case SPU_ES: code = "RTPTRTPV"; break;
976 default: code = "RTPORTPV"; break;
978 var_SetString (p_stream, "dccp-service", code);
981 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
982 p_sys->psz_destination, i_port,
984 if( id->listen_fd == NULL )
986 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
993 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
994 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
995 i_port, ttl, p_sys->proto );
998 msg_Err( p_stream, "cannot create RTP socket" );
1001 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1007 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1009 if( psz == NULL ) /* Uho! */
1012 if( strncmp( psz, "ts", 2 ) == 0 )
1014 id->i_payload_type = 33;
1015 id->psz_enc = "MP2T";
1019 id->psz_enc = "MP2P";
1024 switch( p_fmt->i_codec )
1026 case VLC_FOURCC( 'u', 'l', 'a', 'w' ):
1027 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1028 id->i_payload_type = 0;
1029 id->psz_enc = "PCMU";
1030 id->pf_packetize = rtp_packetize_split;
1031 rtp_set_ptime (id, 20, 1);
1033 case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
1034 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1035 id->i_payload_type = 8;
1036 id->psz_enc = "PCMA";
1037 id->pf_packetize = rtp_packetize_split;
1038 rtp_set_ptime (id, 20, 1);
1040 case VLC_FOURCC( 's', '1', '6', 'b' ):
1041 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1043 id->i_payload_type = 11;
1045 else if( p_fmt->audio.i_channels == 2 &&
1046 p_fmt->audio.i_rate == 44100 )
1048 id->i_payload_type = 10;
1050 id->psz_enc = "L16";
1051 id->pf_packetize = rtp_packetize_split;
1052 rtp_set_ptime (id, 20, 2);
1054 case VLC_FOURCC( 'u', '8', ' ', ' ' ):
1056 id->pf_packetize = rtp_packetize_split;
1057 rtp_set_ptime (id, 20, 1);
1059 case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
1060 case VLC_FOURCC( 'm', 'p', '3', ' ' ):
1061 id->i_payload_type = 14;
1062 id->psz_enc = "MPA";
1063 id->i_clock_rate = 90000; /* not 44100 */
1064 id->pf_packetize = rtp_packetize_mpa;
1066 case VLC_FOURCC( 'm', 'p', 'g', 'v' ):
1067 id->i_payload_type = 32;
1068 id->psz_enc = "MPV";
1069 id->pf_packetize = rtp_packetize_mpv;
1071 case VLC_FOURCC( 'G', '7', '2', '6' ):
1072 case VLC_FOURCC( 'g', '7', '2', '6' ):
1073 switch( p_fmt->i_bitrate / 1000 )
1076 id->psz_enc = "G726-16";
1077 id->pf_packetize = rtp_packetize_g726_16;
1080 id->psz_enc = "G726-24";
1081 id->pf_packetize = rtp_packetize_g726_24;
1084 id->psz_enc = "G726-32";
1085 id->pf_packetize = rtp_packetize_g726_32;
1088 id->psz_enc = "G726-40";
1089 id->pf_packetize = rtp_packetize_g726_40;
1093 case VLC_FOURCC( 'a', '5', '2', ' ' ):
1094 id->psz_enc = "ac3";
1095 id->pf_packetize = rtp_packetize_ac3;
1097 case VLC_FOURCC( 'H', '2', '6', '3' ):
1098 id->psz_enc = "H263-1998";
1099 id->pf_packetize = rtp_packetize_h263;
1101 case VLC_FOURCC( 'h', '2', '6', '4' ):
1102 id->psz_enc = "H264";
1103 id->pf_packetize = rtp_packetize_h264;
1104 id->psz_fmtp = NULL;
1106 if( p_fmt->i_extra > 0 )
1108 uint8_t *p_buffer = p_fmt->p_extra;
1109 int i_buffer = p_fmt->i_extra;
1110 char *p_64_sps = NULL;
1111 char *p_64_pps = NULL;
1114 while( i_buffer > 4 &&
1115 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1116 p_buffer[2] == 0 && p_buffer[3] == 1 )
1118 const int i_nal_type = p_buffer[4]&0x1f;
1122 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1125 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1127 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1129 /* we found another startcode */
1134 if( i_nal_type == 7 )
1136 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1137 sprintf_hexa( hexa, &p_buffer[5], 3 );
1139 else if( i_nal_type == 8 )
1141 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1147 if( p_64_sps && p_64_pps &&
1148 ( asprintf( &id->psz_fmtp,
1149 "packetization-mode=1;profile-level-id=%s;"
1150 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1151 p_64_pps ) == -1 ) )
1152 id->psz_fmtp = NULL;
1157 id->psz_fmtp = strdup( "packetization-mode=1" );
1160 case VLC_FOURCC( 'm', 'p', '4', 'v' ):
1162 char hexa[2*p_fmt->i_extra +1];
1164 id->psz_enc = "MP4V-ES";
1165 id->pf_packetize = rtp_packetize_split;
1166 if( p_fmt->i_extra > 0 )
1168 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1169 if( asprintf( &id->psz_fmtp,
1170 "profile-level-id=3; config=%s;", hexa ) == -1 )
1171 id->psz_fmtp = NULL;
1175 case VLC_FOURCC( 'm', 'p', '4', 'a' ):
1179 char hexa[2*p_fmt->i_extra +1];
1181 id->psz_enc = "mpeg4-generic";
1182 id->pf_packetize = rtp_packetize_mp4a;
1183 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1184 if( asprintf( &id->psz_fmtp,
1185 "streamtype=5; profile-level-id=15; "
1186 "mode=AAC-hbr; config=%s; SizeLength=13; "
1187 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1189 id->psz_fmtp = NULL;
1195 unsigned char config[6];
1196 unsigned int aacsrates[15] = {
1197 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1198 16000, 12000, 11025, 8000, 7350, 0, 0 };
1200 for( i = 0; i < 15; i++ )
1201 if( p_fmt->audio.i_rate == aacsrates[i] )
1207 config[3]=p_fmt->audio.i_channels<<4;
1211 id->psz_enc = "MP4A-LATM";
1212 id->pf_packetize = rtp_packetize_mp4a_latm;
1213 sprintf_hexa( hexa, config, 6 );
1214 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1215 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1216 id->psz_fmtp = NULL;
1220 case VLC_FOURCC( 's', 'a', 'm', 'r' ):
1221 id->psz_enc = "AMR";
1222 id->psz_fmtp = strdup( "octet-align=1" );
1223 id->pf_packetize = rtp_packetize_amr;
1225 case VLC_FOURCC( 's', 'a', 'w', 'b' ):
1226 id->psz_enc = "AMR-WB";
1227 id->psz_fmtp = strdup( "octet-align=1" );
1228 id->pf_packetize = rtp_packetize_amr;
1230 case VLC_FOURCC( 's', 'p', 'x', ' ' ):
1231 id->psz_enc = "SPEEX";
1232 id->pf_packetize = rtp_packetize_spx;
1234 case VLC_FOURCC( 't', '1', '4', '0' ):
1235 id->psz_enc = "t140" ;
1236 id->i_clock_rate = 1000;
1237 id->pf_packetize = rtp_packetize_t140;
1241 msg_Err( p_stream, "cannot add this stream (unsupported "
1242 "codec:%4.4s)", (char*)&p_fmt->i_codec );
1245 if (id->i_payload_type >= 96)
1246 /* Mark dynamic payload type in use */
1247 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1249 #if 0 /* No payload formats sets this at the moment */
1251 cscov += 8 /* UDP */ + 12 /* RTP */;
1253 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1256 if( p_sys->rtsp != NULL )
1257 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1258 GetDWBE( id->ssrc ),
1259 p_sys->psz_destination,
1260 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1262 id->p_fifo = block_FifoNew();
1263 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1264 VLC_THREAD_PRIORITY_HIGHEST, false ) )
1267 /* Update p_sys context */
1268 vlc_mutex_lock( &p_sys->lock_es );
1269 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1270 vlc_mutex_unlock( &p_sys->lock_es );
1272 psz_sdp = SDPGenerate( p_stream, NULL );
1274 vlc_mutex_lock( &p_sys->lock_sdp );
1275 free( p_sys->psz_sdp );
1276 p_sys->psz_sdp = psz_sdp;
1277 vlc_mutex_unlock( &p_sys->lock_sdp );
1279 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1281 /* Update SDP (sap/file) */
1282 if( p_sys->b_export_sap ) SapSetup( p_stream );
1283 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1288 Del( p_stream, id );
1292 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1294 sout_stream_sys_t *p_sys = p_stream->p_sys;
1296 if( id->p_fifo != NULL )
1298 vlc_object_kill( id );
1299 vlc_thread_join( id );
1300 block_FifoRelease( id->p_fifo );
1303 vlc_mutex_lock( &p_sys->lock_es );
1304 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1305 vlc_mutex_unlock( &p_sys->lock_es );
1308 if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
1309 p_sys->i_port_audio = id->i_port;
1310 if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
1311 p_sys->i_port_video = id->i_port;
1312 /* Release dynamic payload type */
1313 if (id->i_payload_type >= 96)
1314 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1316 free( id->psz_fmtp );
1319 RtspDelId( p_sys->rtsp, id->rtsp_id );
1321 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1322 if( id->listen_fd != NULL )
1323 net_ListenClose( id->listen_fd );
1324 if( id->srtp != NULL )
1325 srtp_destroy( id->srtp );
1327 vlc_mutex_destroy( &id->lock_sink );
1329 /* Update SDP (sap/file) */
1330 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1331 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1333 vlc_object_detach( id );
1334 vlc_object_release( id );
1338 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1343 assert( p_stream->p_sys->p_mux == NULL );
1346 while( p_buffer != NULL )
1348 p_next = p_buffer->p_next;
1349 if( id->pf_packetize( id, p_buffer ) )
1352 block_Release( p_buffer );
1358 /****************************************************************************
1360 ****************************************************************************/
1361 static int SapSetup( sout_stream_t *p_stream )
1363 sout_stream_sys_t *p_sys = p_stream->p_sys;
1364 sout_instance_t *p_sout = p_stream->p_sout;
1366 /* Remove the previous session */
1367 if( p_sys->p_session != NULL)
1369 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1370 p_sys->p_session = NULL;
1373 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1375 announce_method_t *p_method = sout_SAPMethod();
1376 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1378 p_sys->psz_destination,
1380 sout_MethodRelease( p_method );
1386 /****************************************************************************
1388 ****************************************************************************/
1389 static int FileSetup( sout_stream_t *p_stream )
1391 sout_stream_sys_t *p_sys = p_stream->p_sys;
1394 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1396 msg_Err( p_stream, "cannot open file '%s' (%m)",
1397 p_sys->psz_sdp_file );
1398 return VLC_EGENERIC;
1401 fputs( p_sys->psz_sdp, f );
1407 /****************************************************************************
1409 ****************************************************************************/
1410 static int HttpCallback( httpd_file_sys_t *p_args,
1411 httpd_file_t *, uint8_t *p_request,
1412 uint8_t **pp_data, int *pi_data );
1414 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1416 sout_stream_sys_t *p_sys = p_stream->p_sys;
1418 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1419 url->i_port > 0 ? url->i_port : 80 );
1420 if( p_sys->p_httpd_host )
1422 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1423 url->psz_path ? url->psz_path : "/",
1426 HttpCallback, (void*)p_sys );
1428 if( p_sys->p_httpd_file == NULL )
1430 return VLC_EGENERIC;
1435 static int HttpCallback( httpd_file_sys_t *p_args,
1436 httpd_file_t *f, uint8_t *p_request,
1437 uint8_t **pp_data, int *pi_data )
1439 VLC_UNUSED(f); VLC_UNUSED(p_request);
1440 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1442 vlc_mutex_lock( &p_sys->lock_sdp );
1443 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1445 *pi_data = strlen( p_sys->psz_sdp );
1446 *pp_data = malloc( *pi_data );
1447 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1454 vlc_mutex_unlock( &p_sys->lock_sdp );
1459 /****************************************************************************
1461 ****************************************************************************/
1462 static void* ThreadSend( vlc_object_t *p_this )
1464 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1465 unsigned i_caching = id->i_caching;
1469 block_t *out = block_FifoGet( id->p_fifo );
1470 block_cleanup_push (out);
1473 { /* FIXME: this is awfully inefficient */
1474 size_t len = out->i_buffer;
1475 out = block_Realloc( out, 0, len + 10 );
1476 out->i_buffer = len;
1478 int canc = vlc_savecancel ();
1479 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1480 vlc_restorecancel (canc);
1484 msg_Dbg( id, "SRTP sending error: %m" );
1485 block_Release( out );
1489 out->i_buffer = len;
1493 mwait (out->i_dts + i_caching);
1498 ssize_t len = out->i_buffer;
1499 int canc = vlc_savecancel ();
1501 vlc_mutex_lock( &id->lock_sink );
1502 unsigned deadc = 0; /* How many dead sockets? */
1503 int deadv[id->sinkc]; /* Dead sockets list */
1505 for( int i = 0; i < id->sinkc; i++ )
1507 if( !id->srtp ) /* FIXME: SRTCP support */
1508 SendRTCP( id->sinkv[i].rtcp, out );
1510 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1512 /* Retry sending to root out soft-errors */
1513 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1516 deadv[deadc++] = id->sinkv[i].rtp_fd;
1518 vlc_mutex_unlock( &id->lock_sink );
1519 block_Release( out );
1521 for( unsigned i = 0; i < deadc; i++ )
1523 msg_Dbg( id, "removing socket %d", deadv[i] );
1524 rtp_del_sink( id, deadv[i] );
1527 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1528 while( id->listen_fd != NULL )
1530 int fd = net_Accept( id, id->listen_fd, 0 );
1533 msg_Dbg( id, "adding socket %d", fd );
1534 rtp_add_sink( id, fd, true );
1536 vlc_restorecancel (canc);
1541 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1543 rtp_sink_t sink = { fd, NULL };
1544 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1546 if( sink.rtcp == NULL )
1547 msg_Err( id, "RTCP failed!" );
1549 vlc_mutex_lock( &id->lock_sink );
1550 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1551 vlc_mutex_unlock( &id->lock_sink );
1555 void rtp_del_sink( sout_stream_id_t *id, int fd )
1557 rtp_sink_t sink = { fd, NULL };
1559 /* NOTE: must be safe to use if fd is not included */
1560 vlc_mutex_lock( &id->lock_sink );
1561 for( int i = 0; i < id->sinkc; i++ )
1563 if (id->sinkv[i].rtp_fd == fd)
1565 sink = id->sinkv[i];
1566 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1570 vlc_mutex_unlock( &id->lock_sink );
1572 CloseRTCP( sink.rtcp );
1573 net_Close( sink.rtp_fd );
1576 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1578 /* This will return values for the next packet.
1579 * Accounting for caching would not be totally trivial. */
1580 return id->i_sequence;
1583 /* FIXME: this is pretty bad - if we remove and then insert an ES
1584 * the number will get unsynched from inside RTSP */
1585 unsigned rtp_get_num( const sout_stream_id_t *id )
1587 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1590 vlc_mutex_lock( &p_sys->lock_es );
1591 for( i = 0; i < p_sys->i_es; i++ )
1593 if( id == p_sys->es[i] )
1596 vlc_mutex_unlock( &p_sys->lock_es );
1602 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1603 int b_marker, int64_t i_pts )
1605 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
1607 out->p_buffer[0] = 0x80;
1608 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1609 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1610 out->p_buffer[3] = ( id->i_sequence )&0xff;
1611 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1612 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1613 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1614 out->p_buffer[7] = ( i_timestamp )&0xff;
1616 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1622 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1624 block_FifoPut( id->p_fifo, out );
1628 * @return configured max RTP payload size (including payload type-specific
1629 * headers, excluding RTP and transport headers)
1631 size_t rtp_mtu (const sout_stream_id_t *id)
1633 return id->i_mtu - 12;
1636 /*****************************************************************************
1638 *****************************************************************************/
1640 /** Add an ES to a non-RTP muxed stream */
1641 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1643 sout_input_t *p_input;
1644 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1645 assert( p_mux != NULL );
1647 p_input = sout_MuxAddStream( p_mux, p_fmt );
1648 if( p_input == NULL )
1650 msg_Err( p_stream, "cannot add this stream to the muxer" );
1654 return (sout_stream_id_t *)p_input;
1658 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1661 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1662 assert( p_mux != NULL );
1664 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1669 /** Remove an ES from a non-RTP muxed stream */
1670 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1672 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1673 assert( p_mux != NULL );
1675 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1680 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1681 const block_t *p_buffer )
1683 sout_stream_sys_t *p_sys = p_stream->p_sys;
1684 sout_stream_id_t *id = p_sys->es[0];
1686 int64_t i_dts = p_buffer->i_dts;
1688 uint8_t *p_data = p_buffer->p_buffer;
1689 size_t i_data = p_buffer->i_buffer;
1690 size_t i_max = id->i_mtu - 12;
1692 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1698 /* output complete packet */
1699 if( p_sys->packet &&
1700 p_sys->packet->i_buffer + i_data > i_max )
1702 rtp_packetize_send( id, p_sys->packet );
1703 p_sys->packet = NULL;
1706 if( p_sys->packet == NULL )
1708 /* allocate a new packet */
1709 p_sys->packet = block_New( p_stream, id->i_mtu );
1710 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1711 p_sys->packet->i_dts = i_dts;
1712 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1713 i_dts += p_sys->packet->i_length;
1716 i_size = __MIN( i_data,
1717 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1719 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1722 p_sys->packet->i_buffer += i_size;
1731 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1734 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1740 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1742 p_next = p_buffer->p_next;
1743 block_Release( p_buffer );
1751 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1753 sout_access_out_t *p_grab;
1755 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1756 if( p_grab == NULL )
1759 p_grab->p_module = NULL;
1760 p_grab->psz_access = strdup( "grab" );
1761 p_grab->p_cfg = NULL;
1762 p_grab->psz_path = strdup( "" );
1763 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1764 p_grab->pf_seek = NULL;
1765 p_grab->pf_write = AccessOutGrabberWrite;
1766 vlc_object_attach( p_grab, p_stream );