i18: Translate one more string
[vlc/vlc-acra.git] / modules / stream_out / rtp.c
blob72f684e616f79924073a6b809a63e443bd8bd42d
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 /*****************************************************************************
25 * Preamble
26 *****************************************************************************/
28 #ifdef HAVE_CONFIG_H
29 # include "config.h"
30 #endif
32 #include <vlc_common.h>
33 #include <vlc_plugin.h>
34 #include <vlc_sout.h>
35 #include <vlc_block.h>
37 #include <vlc_httpd.h>
38 #include <vlc_url.h>
39 #include <vlc_network.h>
40 #include <vlc_charset.h>
41 #include <vlc_strings.h>
42 #include <srtp.h>
44 #include "rtp.h"
46 #ifdef HAVE_UNISTD_H
47 # include <sys/types.h>
48 # include <unistd.h>
49 # include <fcntl.h>
50 # include <sys/stat.h>
51 #endif
52 #ifdef HAVE_LINUX_DCCP_H
53 # include <linux/dccp.h>
54 #endif
55 #ifndef IPPROTO_DCCP
56 # define IPPROTO_DCCP 33
57 #endif
58 #ifndef IPPROTO_UDPLITE
59 # define IPPROTO_UDPLITE 136
60 #endif
62 #include <errno.h>
64 #include <assert.h>
66 /*****************************************************************************
67 * Module descriptor
68 *****************************************************************************/
70 #define DEST_TEXT N_("Destination")
71 #define DEST_LONGTEXT N_( \
72 "This is the output URL that will be used." )
73 #define SDP_TEXT N_("SDP")
74 #define SDP_LONGTEXT N_( \
75 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
76 "session will be made available. You must use an url: http://location to " \
77 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
78 "for the SDP to be announced via SAP." )
79 #define SAP_TEXT N_("SAP announcing")
80 #define SAP_LONGTEXT N_("Announce this session with SAP.")
81 #define MUX_TEXT N_("Muxer")
82 #define MUX_LONGTEXT N_( \
83 "This allows you to specify the muxer used for the streaming output. " \
84 "Default is to use no muxer (standard RTP stream)." )
86 #define NAME_TEXT N_("Session name")
87 #define NAME_LONGTEXT N_( \
88 "This is the name of the session that will be announced in the SDP " \
89 "(Session Descriptor)." )
90 #define DESC_TEXT N_("Session description")
91 #define DESC_LONGTEXT N_( \
92 "This allows you to give a short description with details about the stream, " \
93 "that will be announced in the SDP (Session Descriptor)." )
94 #define URL_TEXT N_("Session URL")
95 #define URL_LONGTEXT N_( \
96 "This allows you to give an URL with more details about the stream " \
97 "(often the website of the streaming organization), that will " \
98 "be announced in the SDP (Session Descriptor)." )
99 #define EMAIL_TEXT N_("Session email")
100 #define EMAIL_LONGTEXT N_( \
101 "This allows you to give a contact mail address for the stream, that will " \
102 "be announced in the SDP (Session Descriptor)." )
103 #define PHONE_TEXT N_("Session phone number")
104 #define PHONE_LONGTEXT N_( \
105 "This allows you to give a contact telephone number for the stream, that will " \
106 "be announced in the SDP (Session Descriptor)." )
108 #define PORT_TEXT N_("Port")
109 #define PORT_LONGTEXT N_( \
110 "This allows you to specify the base port for the RTP streaming." )
111 #define PORT_AUDIO_TEXT N_("Audio port")
112 #define PORT_AUDIO_LONGTEXT N_( \
113 "This allows you to specify the default audio port for the RTP streaming." )
114 #define PORT_VIDEO_TEXT N_("Video port")
115 #define PORT_VIDEO_LONGTEXT N_( \
116 "This allows you to specify the default video port for the RTP streaming." )
118 #define TTL_TEXT N_("Hop limit (TTL)")
119 #define TTL_LONGTEXT N_( \
120 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
121 "the multicast packets sent by the stream output (-1 = use operating " \
122 "system built-in default).")
124 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
125 #define RTCP_MUX_LONGTEXT N_( \
126 "This sends and receives RTCP packet multiplexed over the same port " \
127 "as RTP packets." )
129 #define PROTO_TEXT N_("Transport protocol")
130 #define PROTO_LONGTEXT N_( \
131 "This selects which transport protocol to use for RTP." )
133 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
134 #define SRTP_KEY_LONGTEXT N_( \
135 "RTP packets will be integrity-protected and ciphered "\
136 "with this Secure RTP master shared secret key.")
138 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
139 #define SRTP_SALT_LONGTEXT N_( \
140 "Secure RTP requires a (non-secret) master salt value.")
142 static const char *const ppsz_protos[] = {
143 "dccp", "sctp", "tcp", "udp", "udplite",
146 static const char *const ppsz_protocols[] = {
147 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
150 #define RFC3016_TEXT N_("MP4A LATM")
151 #define RFC3016_LONGTEXT N_( \
152 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
154 static int Open ( vlc_object_t * );
155 static void Close( vlc_object_t * );
157 #define SOUT_CFG_PREFIX "sout-rtp-"
158 #define MAX_EMPTY_BLOCKS 200
160 vlc_module_begin ()
161 set_shortname( N_("RTP"))
162 set_description( N_("RTP stream output") )
163 set_capability( "sout stream", 0 )
164 add_shortcut( "rtp" )
165 set_category( CAT_SOUT )
166 set_subcategory( SUBCAT_SOUT_STREAM )
168 add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
169 DEST_LONGTEXT, true );
170 add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
171 SDP_LONGTEXT, true );
172 add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
173 MUX_LONGTEXT, true );
174 add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
175 true );
177 add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
178 NAME_LONGTEXT, true );
179 add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
180 DESC_LONGTEXT, true );
181 add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
182 URL_LONGTEXT, true );
183 add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
184 EMAIL_LONGTEXT, true );
185 add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
186 PHONE_LONGTEXT, true );
188 add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
189 PROTO_LONGTEXT, false );
190 change_string_list( ppsz_protos, ppsz_protocols, NULL );
191 add_integer( SOUT_CFG_PREFIX "port", 50004, NULL, PORT_TEXT,
192 PORT_LONGTEXT, true );
193 add_integer( SOUT_CFG_PREFIX "port-audio", 50000, NULL, PORT_AUDIO_TEXT,
194 PORT_AUDIO_LONGTEXT, true );
195 add_integer( SOUT_CFG_PREFIX "port-video", 50002, NULL, PORT_VIDEO_TEXT,
196 PORT_VIDEO_LONGTEXT, true );
198 add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
199 TTL_LONGTEXT, true );
200 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
201 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );
203 add_string( SOUT_CFG_PREFIX "key", "", NULL,
204 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
205 add_string( SOUT_CFG_PREFIX "salt", "", NULL,
206 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );
208 add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
209 RFC3016_LONGTEXT, false );
211 set_callbacks( Open, Close )
212 vlc_module_end ()
214 /*****************************************************************************
215 * Exported prototypes
216 *****************************************************************************/
217 static const char *const ppsz_sout_options[] = {
218 "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
219 "sap", "description", "url", "email", "phone",
220 "proto", "rtcp-mux", "key", "salt",
221 "mp4a-latm", NULL
224 static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
225 static int Del ( sout_stream_t *, sout_stream_id_t * );
226 static int Send( sout_stream_t *, sout_stream_id_t *,
227 block_t* );
228 static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
229 static int MuxDel ( sout_stream_t *, sout_stream_id_t * );
230 static int MuxSend( sout_stream_t *, sout_stream_id_t *,
231 block_t* );
233 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
234 static void* ThreadSend( vlc_object_t *p_this );
236 static void SDPHandleUrl( sout_stream_t *, const char * );
238 static int SapSetup( sout_stream_t *p_stream );
239 static int FileSetup( sout_stream_t *p_stream );
240 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
242 struct sout_stream_sys_t
244 /* SDP */
245 char *psz_sdp;
246 vlc_mutex_t lock_sdp;
248 /* SDP to disk */
249 bool b_export_sdp_file;
250 char *psz_sdp_file;
252 /* SDP via SAP */
253 bool b_export_sap;
254 session_descriptor_t *p_session;
256 /* SDP via HTTP */
257 httpd_host_t *p_httpd_host;
258 httpd_file_t *p_httpd_file;
260 /* RTSP */
261 rtsp_stream_t *rtsp;
263 /* */
264 char *psz_destination;
265 uint32_t payload_bitmap;
266 uint16_t i_port;
267 uint16_t i_port_audio;
268 uint16_t i_port_video;
269 uint8_t proto;
270 bool rtcp_mux;
271 int i_ttl:9;
272 bool b_latm;
274 /* in case we do TS/PS over rtp */
275 sout_mux_t *p_mux;
276 sout_access_out_t *p_grab;
277 block_t *packet;
279 /* */
280 vlc_mutex_t lock_es;
281 int i_es;
282 sout_stream_id_t **es;
285 typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );
287 typedef struct rtp_sink_t
289 int rtp_fd;
290 rtcp_sender_t *rtcp;
291 } rtp_sink_t;
293 struct sout_stream_id_t
295 VLC_COMMON_MEMBERS
297 sout_stream_t *p_stream;
298 /* rtp field */
299 uint16_t i_sequence;
300 uint8_t i_payload_type;
301 uint8_t ssrc[4];
303 /* for sdp */
304 const char *psz_enc;
305 char *psz_fmtp;
306 int i_clock_rate;
307 int i_port;
308 int i_cat;
309 int i_channels;
310 int i_bitrate;
312 /* Packetizer specific fields */
313 int i_mtu;
314 srtp_session_t *srtp;
315 pf_rtp_packetizer_t pf_packetize;
317 /* Packets sinks */
318 vlc_mutex_t lock_sink;
319 int sinkc;
320 rtp_sink_t *sinkv;
321 rtsp_stream_id_t *rtsp_id;
322 int *listen_fd;
324 block_fifo_t *p_fifo;
325 int64_t i_caching;
328 /*****************************************************************************
329 * Open:
330 *****************************************************************************/
331 static int Open( vlc_object_t *p_this )
333 sout_stream_t *p_stream = (sout_stream_t*)p_this;
334 sout_instance_t *p_sout = p_stream->p_sout;
335 sout_stream_sys_t *p_sys = NULL;
336 config_chain_t *p_cfg = NULL;
337 char *psz;
338 bool b_rtsp = false;
340 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
341 ppsz_sout_options, p_stream->p_cfg );
343 p_sys = malloc( sizeof( sout_stream_sys_t ) );
344 if( p_sys == NULL )
345 return VLC_ENOMEM;
347 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
349 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
350 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
351 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
352 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
354 p_sys->psz_sdp_file = NULL;
356 if( p_sys->i_port_audio == p_sys->i_port_video )
358 msg_Err( p_stream, "audio and video port cannot be the same" );
359 p_sys->i_port_audio = 0;
360 p_sys->i_port_video = 0;
363 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
365 if( !strcmp( p_cfg->psz_name, "sdp" )
366 && ( p_cfg->psz_value != NULL )
367 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
369 b_rtsp = true;
370 break;
373 if( !b_rtsp )
375 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
376 if( psz != NULL )
378 if( !strncasecmp( psz, "rtsp:", 5 ) )
379 b_rtsp = true;
380 free( psz );
384 /* Transport protocol */
385 p_sys->proto = IPPROTO_UDP;
386 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
388 if ((psz == NULL) || !strcasecmp (psz, "udp"))
389 (void)0; /* default */
390 else
391 if (!strcasecmp (psz, "dccp"))
393 p_sys->proto = IPPROTO_DCCP;
394 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
396 #if 0
397 else
398 if (!strcasecmp (psz, "sctp"))
400 p_sys->proto = IPPROTO_TCP;
401 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
403 #endif
404 #if 0
405 else
406 if (!strcasecmp (psz, "tcp"))
408 p_sys->proto = IPPROTO_TCP;
409 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
411 #endif
412 else
413 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
414 p_sys->proto = IPPROTO_UDPLITE;
415 else
416 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
417 psz);
418 free (psz);
419 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
421 if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
423 msg_Err( p_stream, "missing destination and not in RTSP mode" );
424 free( p_sys );
425 return VLC_EGENERIC;
428 p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
429 if( p_sys->i_ttl == -1 )
431 /* Normally, we should let the default hop limit up to the core,
432 * but we have to know it to build our SDP properly, which is why
433 * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
434 * ttl are set. */
435 p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
438 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
440 p_sys->payload_bitmap = 0;
441 p_sys->i_es = 0;
442 p_sys->es = NULL;
443 p_sys->rtsp = NULL;
444 p_sys->psz_sdp = NULL;
446 p_sys->b_export_sap = false;
447 p_sys->b_export_sdp_file = false;
448 p_sys->p_session = NULL;
450 p_sys->p_httpd_host = NULL;
451 p_sys->p_httpd_file = NULL;
453 p_stream->p_sys = p_sys;
455 vlc_mutex_init( &p_sys->lock_sdp );
456 vlc_mutex_init( &p_sys->lock_es );
458 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
459 if( psz != NULL )
461 sout_stream_id_t *id;
463 /* Check muxer type */
464 if( strncasecmp( psz, "ps", 2 )
465 && strncasecmp( psz, "mpeg1", 5 )
466 && strncasecmp( psz, "ts", 2 ) )
468 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
469 free( psz );
470 vlc_mutex_destroy( &p_sys->lock_sdp );
471 vlc_mutex_destroy( &p_sys->lock_es );
472 free( p_sys );
473 return VLC_EGENERIC;
476 p_sys->p_grab = GrabberCreate( p_stream );
477 p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
478 free( psz );
480 if( p_sys->p_mux == NULL )
482 msg_Err( p_stream, "cannot create muxer" );
483 sout_AccessOutDelete( p_sys->p_grab );
484 vlc_mutex_destroy( &p_sys->lock_sdp );
485 vlc_mutex_destroy( &p_sys->lock_es );
486 free( p_sys );
487 return VLC_EGENERIC;
490 id = Add( p_stream, NULL );
491 if( id == NULL )
493 sout_MuxDelete( p_sys->p_mux );
494 sout_AccessOutDelete( p_sys->p_grab );
495 vlc_mutex_destroy( &p_sys->lock_sdp );
496 vlc_mutex_destroy( &p_sys->lock_es );
497 free( p_sys );
498 return VLC_EGENERIC;
501 p_sys->packet = NULL;
503 p_stream->pf_add = MuxAdd;
504 p_stream->pf_del = MuxDel;
505 p_stream->pf_send = MuxSend;
507 else
509 p_sys->p_mux = NULL;
510 p_sys->p_grab = NULL;
512 p_stream->pf_add = Add;
513 p_stream->pf_del = Del;
514 p_stream->pf_send = Send;
517 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
518 SDPHandleUrl( p_stream, "sap" );
520 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
521 if( psz != NULL )
523 config_chain_t *p_cfg;
525 SDPHandleUrl( p_stream, psz );
527 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
529 if( !strcmp( p_cfg->psz_name, "sdp" ) )
531 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
532 continue;
534 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
535 if( !strcmp( p_cfg->psz_value, psz ) )
536 continue;
538 SDPHandleUrl( p_stream, p_cfg->psz_value );
541 free( psz );
544 /* update p_sout->i_out_pace_nocontrol */
545 p_stream->p_sout->i_out_pace_nocontrol++;
547 return VLC_SUCCESS;
550 /*****************************************************************************
551 * Close:
552 *****************************************************************************/
553 static void Close( vlc_object_t * p_this )
555 sout_stream_t *p_stream = (sout_stream_t*)p_this;
556 sout_stream_sys_t *p_sys = p_stream->p_sys;
558 /* update p_sout->i_out_pace_nocontrol */
559 p_stream->p_sout->i_out_pace_nocontrol--;
561 if( p_sys->p_mux )
563 assert( p_sys->i_es == 1 );
564 Del( p_stream, p_sys->es[0] );
566 sout_MuxDelete( p_sys->p_mux );
567 sout_AccessOutDelete( p_sys->p_grab );
568 if( p_sys->packet )
570 block_Release( p_sys->packet );
572 if( p_sys->b_export_sap )
574 p_sys->p_mux = NULL;
575 SapSetup( p_stream );
579 if( p_sys->rtsp != NULL )
580 RtspUnsetup( p_sys->rtsp );
582 vlc_mutex_destroy( &p_sys->lock_sdp );
583 vlc_mutex_destroy( &p_sys->lock_es );
585 if( p_sys->p_httpd_file )
586 httpd_FileDelete( p_sys->p_httpd_file );
588 if( p_sys->p_httpd_host )
589 httpd_HostDelete( p_sys->p_httpd_host );
591 free( p_sys->psz_sdp );
593 if( p_sys->b_export_sdp_file )
595 #ifdef HAVE_UNISTD_H
596 unlink( p_sys->psz_sdp_file );
597 #endif
598 free( p_sys->psz_sdp_file );
600 free( p_sys->psz_destination );
601 free( p_sys );
604 /*****************************************************************************
605 * SDPHandleUrl:
606 *****************************************************************************/
607 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
609 sout_stream_sys_t *p_sys = p_stream->p_sys;
610 vlc_url_t url;
612 vlc_UrlParse( &url, psz_url, 0 );
613 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
615 if( p_sys->p_httpd_file )
617 msg_Err( p_stream, "you can use sdp=http:// only once" );
618 goto out;
621 if( HttpSetup( p_stream, &url ) )
623 msg_Err( p_stream, "cannot export SDP as HTTP" );
626 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
628 if( p_sys->rtsp != NULL )
630 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
631 goto out;
634 /* FIXME test if destination is multicast or no destination at all */
635 p_sys->rtsp = RtspSetup( p_stream, &url );
636 if( p_sys->rtsp == NULL )
637 msg_Err( p_stream, "cannot export SDP as RTSP" );
638 else
639 if( p_sys->p_mux != NULL )
641 sout_stream_id_t *id = p_sys->es[0];
642 id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
643 p_sys->psz_destination, p_sys->i_ttl,
644 id->i_port, id->i_port + 1 );
647 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
648 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
650 p_sys->b_export_sap = true;
651 SapSetup( p_stream );
653 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
655 if( p_sys->b_export_sdp_file )
657 msg_Err( p_stream, "you can use sdp=file:// only once" );
658 goto out;
660 p_sys->b_export_sdp_file = true;
661 psz_url = &psz_url[5];
662 if( psz_url[0] == '/' && psz_url[1] == '/' )
663 psz_url += 2;
664 p_sys->psz_sdp_file = strdup( psz_url );
666 else
668 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
669 url.psz_protocol );
672 out:
673 vlc_UrlClean( &url );
676 /*****************************************************************************
677 * SDPGenerate
678 *****************************************************************************/
679 /*static*/
680 char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
682 const sout_stream_sys_t *p_sys = p_stream->p_sys;
683 char *psz_sdp;
684 struct sockaddr_storage dst;
685 socklen_t dstlen;
686 int i;
688 * When we have a fixed destination (typically when we do multicast),
689 * we need to put the actual port numbers in the SDP.
690 * When there is no fixed destination, we only support RTSP unicast
691 * on-demand setup, so we should rather let the clients decide which ports
692 * to use.
693 * When there is both a fixed destination and RTSP unicast, we need to
694 * put port numbers used by the fixed destination, otherwise the SDP would
695 * become totally incorrect for multicast use. It should be noted that
696 * port numbers from SDP with RTSP are only "recommendation" from the
697 * server to the clients (per RFC2326), so only broken clients will fail
698 * to handle this properly. There is no solution but to use two differents
699 * output chain with two different RTSP URLs if you need to handle this
700 * scenario.
702 int inclport;
704 if( p_sys->psz_destination != NULL )
706 inclport = 1;
708 /* Oh boy, this is really ugly! (+ race condition on lock_es) */
709 dstlen = sizeof( dst );
710 if( p_sys->es[0]->listen_fd != NULL )
711 getsockname( p_sys->es[0]->listen_fd[0],
712 (struct sockaddr *)&dst, &dstlen );
713 else
714 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
715 (struct sockaddr *)&dst, &dstlen );
717 else
719 inclport = 0;
721 /* Dummy destination address for RTSP */
722 memset (&dst, 0, sizeof( struct sockaddr_in ) );
723 dst.ss_family = AF_INET;
724 #ifdef HAVE_SA_LEN
725 dst.ss_len =
726 #endif
727 dstlen = sizeof( struct sockaddr_in );
730 psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
731 NULL, 0, (struct sockaddr *)&dst, dstlen );
732 if( psz_sdp == NULL )
733 return NULL;
735 /* TODO: a=source-filter */
736 if( p_sys->rtcp_mux )
737 sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
739 if( rtsp_url != NULL )
740 sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
742 /* FIXME: locking?! */
743 for( i = 0; i < p_sys->i_es; i++ )
745 sout_stream_id_t *id = p_sys->es[i];
746 const char *mime_major; /* major MIME type */
747 const char *proto = "RTP/AVP"; /* protocol */
749 switch( id->i_cat )
751 case VIDEO_ES:
752 mime_major = "video";
753 break;
754 case AUDIO_ES:
755 mime_major = "audio";
756 break;
757 case SPU_ES:
758 mime_major = "text";
759 break;
760 default:
761 continue;
764 if( rtsp_url == NULL )
766 switch( p_sys->proto )
768 case IPPROTO_UDP:
769 break;
770 case IPPROTO_TCP:
771 proto = "TCP/RTP/AVP";
772 break;
773 case IPPROTO_DCCP:
774 proto = "DCCP/RTP/AVP";
775 break;
776 case IPPROTO_UDPLITE:
777 continue;
781 sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
782 id->i_payload_type, false, id->i_bitrate,
783 id->psz_enc, id->i_clock_rate, id->i_channels,
784 id->psz_fmtp);
786 if( rtsp_url != NULL )
788 assert( strlen( rtsp_url ) > 0 );
789 bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
790 sdp_AddAttribute ( &psz_sdp, "control",
791 addslash ? "%s/trackID=%u" : "%strackID=%u",
792 rtsp_url, i );
794 else
796 if( id->listen_fd != NULL )
797 sdp_AddAttribute( &psz_sdp, "setup", "passive" );
798 if( p_sys->proto == IPPROTO_DCCP )
799 sdp_AddAttribute( &psz_sdp, "dccp-service-code",
800 "SC:RTP%c", toupper( mime_major[0] ) );
804 return psz_sdp;
807 /*****************************************************************************
808 * RTP mux
809 *****************************************************************************/
811 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
813 static const char hex[16] = "0123456789abcdef";
814 int i;
816 for( i = 0; i < i_data; i++ )
818 s[2*i+0] = hex[(p_data[i]>>4)&0xf];
819 s[2*i+1] = hex[(p_data[i] )&0xf];
821 s[2*i_data] = '\0';
825 * Shrink the MTU down to a fixed packetization time (for audio).
827 static void
828 rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
830 /* Samples per second */
831 size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
832 bytes *= id->i_channels;
833 spl *= bytes;
835 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
836 id->i_mtu = 12 + spl;
837 else /* MTU is too small for ptime, align to a sample boundary */
838 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
841 /** Add an ES as a new RTP stream */
842 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
844 /* NOTE: As a special case, if we use a non-RTP
845 * mux (TS/PS), then p_fmt is NULL. */
846 sout_stream_sys_t *p_sys = p_stream->p_sys;
847 sout_stream_id_t *id;
848 int i_port, cscov = -1;
849 char *psz_sdp;
851 if (0xffffffff == p_sys->payload_bitmap)
853 msg_Err (p_stream, "too many RTP elementary streams");
854 return NULL;
857 id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
858 if( id == NULL )
859 return NULL;
860 vlc_object_attach( id, p_stream );
862 /* Choose the port */
863 i_port = 0;
864 if( p_fmt == NULL )
866 else
867 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
869 i_port = p_sys->i_port_audio;
870 p_sys->i_port_audio = 0;
872 else
873 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
875 i_port = p_sys->i_port_video;
876 p_sys->i_port_video = 0;
879 while( i_port == 0 )
881 if( p_sys->i_port != p_sys->i_port_audio
882 && p_sys->i_port != p_sys->i_port_video )
884 i_port = p_sys->i_port;
885 p_sys->i_port += 2;
886 break;
888 p_sys->i_port += 2;
891 id->p_stream = p_stream;
893 id->i_sequence = rand()&0xffff;
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 id->ssrc[0] = rand()&0xff;
901 id->ssrc[1] = rand()&0xff;
902 id->ssrc[2] = rand()&0xff;
903 id->ssrc[3] = rand()&0xff;
905 id->psz_enc = NULL;
906 id->psz_fmtp = NULL;
907 id->i_clock_rate = 90000; /* most common case for video */
908 id->i_channels = 0;
909 id->i_port = i_port;
910 if( p_fmt != NULL )
912 id->i_cat = p_fmt->i_cat;
913 if( p_fmt->i_cat == AUDIO_ES )
915 id->i_clock_rate = p_fmt->audio.i_rate;
916 id->i_channels = p_fmt->audio.i_channels;
918 id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
920 else
922 id->i_cat = VIDEO_ES;
923 id->i_bitrate = 0;
926 id->i_mtu = config_GetInt( p_stream, "mtu" );
927 if( id->i_mtu <= 12 + 16 )
928 id->i_mtu = 576 - 20 - 8; /* pessimistic */
929 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
931 id->srtp = NULL;
932 id->pf_packetize = NULL;
934 char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
935 if (key)
937 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
938 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
939 if (id->srtp == NULL)
941 free (key);
942 goto error;
945 char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
946 errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
947 free (salt);
948 free (key);
949 if (errno)
951 msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
952 goto error;
954 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
957 vlc_mutex_init( &id->lock_sink );
958 id->sinkc = 0;
959 id->sinkv = NULL;
960 id->rtsp_id = NULL;
961 id->p_fifo = NULL;
962 id->listen_fd = NULL;
964 id->i_caching =
965 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
967 if( p_sys->psz_destination != NULL )
968 switch( p_sys->proto )
970 case IPPROTO_DCCP:
972 const char *code;
973 switch (id->i_cat)
975 case VIDEO_ES: code = "RTPV"; break;
976 case AUDIO_ES: code = "RTPARTPV"; break;
977 case SPU_ES: code = "RTPTRPTV"; break;
978 default: code = "RTPORTPV"; break;
980 var_SetString (p_stream, "dccp-service", code);
981 } /* fall through */
982 case IPPROTO_TCP:
983 id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
984 p_sys->psz_destination, i_port,
985 p_sys->proto );
986 if( id->listen_fd == NULL )
988 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
989 goto error;
991 break;
993 default:
995 int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
996 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
997 i_port, ttl, p_sys->proto );
998 if( fd == -1 )
1000 msg_Err( p_stream, "cannot create RTP socket" );
1001 goto error;
1003 rtp_add_sink( id, fd, p_sys->rtcp_mux );
1007 if( p_fmt == NULL )
1009 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1011 if( psz == NULL ) /* Uho! */
1013 else
1014 if( strncmp( psz, "ts", 2 ) == 0 )
1016 id->i_payload_type = 33;
1017 id->psz_enc = "MP2T";
1019 else
1021 id->psz_enc = "MP2P";
1023 free( psz );
1025 else
1026 switch( p_fmt->i_codec )
1028 case VLC_FOURCC( 'u', 'l', 'a', 'w' ):
1029 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1030 id->i_payload_type = 0;
1031 id->psz_enc = "PCMU";
1032 id->pf_packetize = rtp_packetize_split;
1033 rtp_set_ptime (id, 20, 1);
1034 break;
1035 case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
1036 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
1037 id->i_payload_type = 8;
1038 id->psz_enc = "PCMA";
1039 id->pf_packetize = rtp_packetize_split;
1040 rtp_set_ptime (id, 20, 1);
1041 break;
1042 case VLC_FOURCC( 's', '1', '6', 'b' ):
1043 if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
1045 id->i_payload_type = 11;
1047 else if( p_fmt->audio.i_channels == 2 &&
1048 p_fmt->audio.i_rate == 44100 )
1050 id->i_payload_type = 10;
1052 id->psz_enc = "L16";
1053 id->pf_packetize = rtp_packetize_split;
1054 rtp_set_ptime (id, 20, 2);
1055 break;
1056 case VLC_FOURCC( 'u', '8', ' ', ' ' ):
1057 id->psz_enc = "L8";
1058 id->pf_packetize = rtp_packetize_split;
1059 rtp_set_ptime (id, 20, 1);
1060 break;
1061 case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
1062 case VLC_FOURCC( 'm', 'p', '3', ' ' ):
1063 id->i_payload_type = 14;
1064 id->psz_enc = "MPA";
1065 id->i_clock_rate = 90000; /* not 44100 */
1066 id->pf_packetize = rtp_packetize_mpa;
1067 break;
1068 case VLC_FOURCC( 'm', 'p', 'g', 'v' ):
1069 id->i_payload_type = 32;
1070 id->psz_enc = "MPV";
1071 id->pf_packetize = rtp_packetize_mpv;
1072 break;
1073 case VLC_FOURCC( 'G', '7', '2', '6' ):
1074 case VLC_FOURCC( 'g', '7', '2', '6' ):
1075 switch( p_fmt->i_bitrate / 1000 )
1077 case 16:
1078 id->psz_enc = "G726-16";
1079 id->pf_packetize = rtp_packetize_g726_16;
1080 break;
1081 case 24:
1082 id->psz_enc = "G726-24";
1083 id->pf_packetize = rtp_packetize_g726_24;
1084 break;
1085 case 32:
1086 id->psz_enc = "G726-32";
1087 id->pf_packetize = rtp_packetize_g726_32;
1088 break;
1089 case 40:
1090 id->psz_enc = "G726-40";
1091 id->pf_packetize = rtp_packetize_g726_40;
1092 break;
1094 break;
1095 case VLC_FOURCC( 'a', '5', '2', ' ' ):
1096 id->psz_enc = "ac3";
1097 id->pf_packetize = rtp_packetize_ac3;
1098 break;
1099 case VLC_FOURCC( 'H', '2', '6', '3' ):
1100 id->psz_enc = "H263-1998";
1101 id->pf_packetize = rtp_packetize_h263;
1102 break;
1103 case VLC_FOURCC( 'h', '2', '6', '4' ):
1104 id->psz_enc = "H264";
1105 id->pf_packetize = rtp_packetize_h264;
1106 id->psz_fmtp = NULL;
1108 if( p_fmt->i_extra > 0 )
1110 uint8_t *p_buffer = p_fmt->p_extra;
1111 int i_buffer = p_fmt->i_extra;
1112 char *p_64_sps = NULL;
1113 char *p_64_pps = NULL;
1114 char hexa[6+1];
1116 while( i_buffer > 4 &&
1117 p_buffer[0] == 0 && p_buffer[1] == 0 &&
1118 p_buffer[2] == 0 && p_buffer[3] == 1 )
1120 const int i_nal_type = p_buffer[4]&0x1f;
1121 int i_offset;
1122 int i_size = 0;
1124 msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
1126 i_size = i_buffer;
1127 for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
1129 if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
1131 /* we found another startcode */
1132 i_size = i_offset;
1133 break;
1136 if( i_nal_type == 7 )
1138 p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1139 sprintf_hexa( hexa, &p_buffer[5], 3 );
1141 else if( i_nal_type == 8 )
1143 p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
1145 i_buffer -= i_size;
1146 p_buffer += i_size;
1148 /* */
1149 if( p_64_sps && p_64_pps &&
1150 ( asprintf( &id->psz_fmtp,
1151 "packetization-mode=1;profile-level-id=%s;"
1152 "sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
1153 p_64_pps ) == -1 ) )
1154 id->psz_fmtp = NULL;
1155 free( p_64_sps );
1156 free( p_64_pps );
1158 if( !id->psz_fmtp )
1159 id->psz_fmtp = strdup( "packetization-mode=1" );
1160 break;
1162 case VLC_FOURCC( 'm', 'p', '4', 'v' ):
1164 char hexa[2*p_fmt->i_extra +1];
1166 id->psz_enc = "MP4V-ES";
1167 id->pf_packetize = rtp_packetize_split;
1168 if( p_fmt->i_extra > 0 )
1170 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1171 if( asprintf( &id->psz_fmtp,
1172 "profile-level-id=3; config=%s;", hexa ) == -1 )
1173 id->psz_fmtp = NULL;
1175 break;
1177 case VLC_FOURCC( 'm', 'p', '4', 'a' ):
1179 if(!p_sys->b_latm)
1181 char hexa[2*p_fmt->i_extra +1];
1183 id->psz_enc = "mpeg4-generic";
1184 id->pf_packetize = rtp_packetize_mp4a;
1185 sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
1186 if( asprintf( &id->psz_fmtp,
1187 "streamtype=5; profile-level-id=15; "
1188 "mode=AAC-hbr; config=%s; SizeLength=13; "
1189 "IndexLength=3; IndexDeltaLength=3; Profile=1;",
1190 hexa ) == -1 )
1191 id->psz_fmtp = NULL;
1193 else
1195 char hexa[13];
1196 int i;
1197 unsigned char config[6];
1198 unsigned int aacsrates[15] = {
1199 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
1200 16000, 12000, 11025, 8000, 7350, 0, 0 };
1202 for( i = 0; i < 15; i++ )
1203 if( p_fmt->audio.i_rate == aacsrates[i] )
1204 break;
1206 config[0]=0x40;
1207 config[1]=0;
1208 config[2]=0x20|i;
1209 config[3]=p_fmt->audio.i_channels<<4;
1210 config[4]=0x3f;
1211 config[5]=0xc0;
1213 id->psz_enc = "MP4A-LATM";
1214 id->pf_packetize = rtp_packetize_mp4a_latm;
1215 sprintf_hexa( hexa, config, 6 );
1216 if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
1217 "object=2; cpresent=0; config=%s", hexa ) == -1 )
1218 id->psz_fmtp = NULL;
1220 break;
1222 case VLC_FOURCC( 's', 'a', 'm', 'r' ):
1223 id->psz_enc = "AMR";
1224 id->psz_fmtp = strdup( "octet-align=1" );
1225 id->pf_packetize = rtp_packetize_amr;
1226 break;
1227 case VLC_FOURCC( 's', 'a', 'w', 'b' ):
1228 id->psz_enc = "AMR-WB";
1229 id->psz_fmtp = strdup( "octet-align=1" );
1230 id->pf_packetize = rtp_packetize_amr;
1231 break;
1232 case VLC_FOURCC( 's', 'p', 'x', ' ' ):
1233 id->psz_enc = "SPEEX";
1234 id->pf_packetize = rtp_packetize_spx;
1235 break;
1236 case VLC_FOURCC( 't', '1', '4', '0' ):
1237 id->psz_enc = "t140" ;
1238 id->i_clock_rate = 1000;
1239 id->pf_packetize = rtp_packetize_t140;
1240 break;
1242 default:
1243 msg_Err( p_stream, "cannot add this stream (unsupported "
1244 "codec:%4.4s)", (char*)&p_fmt->i_codec );
1245 goto error;
1247 if (id->i_payload_type >= 96)
1248 /* Mark dynamic payload type in use */
1249 p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
1251 #if 0 /* No payload formats sets this at the moment */
1252 if( cscov != -1 )
1253 cscov += 8 /* UDP */ + 12 /* RTP */;
1254 if( id->sinkc > 0 )
1255 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1256 #endif
1258 if( p_sys->rtsp != NULL )
1259 id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
1260 GetDWBE( id->ssrc ),
1261 p_sys->psz_destination,
1262 p_sys->i_ttl, id->i_port, id->i_port + 1 );
1264 id->p_fifo = block_FifoNew();
1265 if( vlc_thread_create( id, "RTP send thread", ThreadSend,
1266 VLC_THREAD_PRIORITY_HIGHEST, false ) )
1267 goto error;
1269 /* Update p_sys context */
1270 vlc_mutex_lock( &p_sys->lock_es );
1271 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1272 vlc_mutex_unlock( &p_sys->lock_es );
1274 psz_sdp = SDPGenerate( p_stream, NULL );
1276 vlc_mutex_lock( &p_sys->lock_sdp );
1277 free( p_sys->psz_sdp );
1278 p_sys->psz_sdp = psz_sdp;
1279 vlc_mutex_unlock( &p_sys->lock_sdp );
1281 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1283 /* Update SDP (sap/file) */
1284 if( p_sys->b_export_sap ) SapSetup( p_stream );
1285 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1287 return id;
1289 error:
1290 Del( p_stream, id );
1291 return NULL;
1294 static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
1296 sout_stream_sys_t *p_sys = p_stream->p_sys;
1298 if( id->p_fifo != NULL )
1300 vlc_object_kill( id );
1301 vlc_thread_join( id );
1302 block_FifoRelease( id->p_fifo );
1305 vlc_mutex_lock( &p_sys->lock_es );
1306 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1307 vlc_mutex_unlock( &p_sys->lock_es );
1309 /* Release port */
1310 if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
1311 p_sys->i_port_audio = id->i_port;
1312 if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
1313 p_sys->i_port_video = id->i_port;
1314 /* Release dynamic payload type */
1315 if (id->i_payload_type >= 96)
1316 p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
1318 free( id->psz_fmtp );
1320 if( id->rtsp_id )
1321 RtspDelId( p_sys->rtsp, id->rtsp_id );
1322 if( id->sinkc > 0 )
1323 rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
1324 if( id->listen_fd != NULL )
1325 net_ListenClose( id->listen_fd );
1326 if( id->srtp != NULL )
1327 srtp_destroy( id->srtp );
1329 vlc_mutex_destroy( &id->lock_sink );
1331 /* Update SDP (sap/file) */
1332 if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
1333 if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
1335 vlc_object_detach( id );
1336 vlc_object_release( id );
1337 return VLC_SUCCESS;
1340 static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
1341 block_t *p_buffer )
1343 block_t *p_next;
1345 assert( p_stream->p_sys->p_mux == NULL );
1346 (void)p_stream;
1348 while( p_buffer != NULL )
1350 p_next = p_buffer->p_next;
1351 if( id->pf_packetize( id, p_buffer ) )
1352 break;
1354 block_Release( p_buffer );
1355 p_buffer = p_next;
1357 return VLC_SUCCESS;
1360 /****************************************************************************
1361 * SAP:
1362 ****************************************************************************/
1363 static int SapSetup( sout_stream_t *p_stream )
1365 sout_stream_sys_t *p_sys = p_stream->p_sys;
1366 sout_instance_t *p_sout = p_stream->p_sout;
1368 /* Remove the previous session */
1369 if( p_sys->p_session != NULL)
1371 sout_AnnounceUnRegister( p_sout, p_sys->p_session);
1372 p_sys->p_session = NULL;
1375 if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
1377 announce_method_t *p_method = sout_SAPMethod();
1378 p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
1379 p_sys->psz_sdp,
1380 p_sys->psz_destination,
1381 p_method );
1382 sout_MethodRelease( p_method );
1385 return VLC_SUCCESS;
1388 /****************************************************************************
1389 * File:
1390 ****************************************************************************/
1391 static int FileSetup( sout_stream_t *p_stream )
1393 sout_stream_sys_t *p_sys = p_stream->p_sys;
1394 FILE *f;
1396 if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1398 msg_Err( p_stream, "cannot open file '%s' (%m)",
1399 p_sys->psz_sdp_file );
1400 return VLC_EGENERIC;
1403 fputs( p_sys->psz_sdp, f );
1404 fclose( f );
1406 return VLC_SUCCESS;
1409 /****************************************************************************
1410 * HTTP:
1411 ****************************************************************************/
1412 static int HttpCallback( httpd_file_sys_t *p_args,
1413 httpd_file_t *, uint8_t *p_request,
1414 uint8_t **pp_data, int *pi_data );
1416 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1418 sout_stream_sys_t *p_sys = p_stream->p_sys;
1420 p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
1421 url->i_port > 0 ? url->i_port : 80 );
1422 if( p_sys->p_httpd_host )
1424 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1425 url->psz_path ? url->psz_path : "/",
1426 "application/sdp",
1427 NULL, NULL, NULL,
1428 HttpCallback, (void*)p_sys );
1430 if( p_sys->p_httpd_file == NULL )
1432 return VLC_EGENERIC;
1434 return VLC_SUCCESS;
1437 static int HttpCallback( httpd_file_sys_t *p_args,
1438 httpd_file_t *f, uint8_t *p_request,
1439 uint8_t **pp_data, int *pi_data )
1441 VLC_UNUSED(f); VLC_UNUSED(p_request);
1442 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1444 vlc_mutex_lock( &p_sys->lock_sdp );
1445 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1447 *pi_data = strlen( p_sys->psz_sdp );
1448 *pp_data = malloc( *pi_data );
1449 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1451 else
1453 *pp_data = NULL;
1454 *pi_data = 0;
1456 vlc_mutex_unlock( &p_sys->lock_sdp );
1458 return VLC_SUCCESS;
1461 /****************************************************************************
1462 * RTP send
1463 ****************************************************************************/
1464 static void* ThreadSend( vlc_object_t *p_this )
1466 sout_stream_id_t *id = (sout_stream_id_t *)p_this;
1467 unsigned i_caching = id->i_caching;
1469 for (;;)
1471 block_t *out = block_FifoGet( id->p_fifo );
1472 block_cleanup_push (out);
1474 if( id->srtp )
1475 { /* FIXME: this is awfully inefficient */
1476 size_t len = out->i_buffer;
1477 out = block_Realloc( out, 0, len + 10 );
1478 out->i_buffer = len;
1480 int canc = vlc_savecancel ();
1481 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1482 vlc_restorecancel (canc);
1483 if( val )
1485 errno = val;
1486 msg_Dbg( id, "SRTP sending error: %m" );
1487 block_Release( out );
1488 out = NULL;
1490 else
1491 out->i_buffer = len;
1494 if (out)
1495 mwait (out->i_dts + i_caching);
1496 vlc_cleanup_pop ();
1497 if (out == NULL)
1498 continue;
1500 ssize_t len = out->i_buffer;
1501 int canc = vlc_savecancel ();
1503 vlc_mutex_lock( &id->lock_sink );
1504 unsigned deadc = 0; /* How many dead sockets? */
1505 int deadv[id->sinkc]; /* Dead sockets list */
1507 for( int i = 0; i < id->sinkc; i++ )
1509 if( !id->srtp ) /* FIXME: SRTCP support */
1510 SendRTCP( id->sinkv[i].rtcp, out );
1512 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1513 continue;
1514 /* Retry sending to root out soft-errors */
1515 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
1516 continue;
1518 deadv[deadc++] = id->sinkv[i].rtp_fd;
1520 vlc_mutex_unlock( &id->lock_sink );
1521 block_Release( out );
1523 for( unsigned i = 0; i < deadc; i++ )
1525 msg_Dbg( id, "removing socket %d", deadv[i] );
1526 rtp_del_sink( id, deadv[i] );
1529 /* Hopefully we won't overflow the SO_MAXCONN accept queue */
1530 while( id->listen_fd != NULL )
1532 int fd = net_Accept( id, id->listen_fd, 0 );
1533 if( fd == -1 )
1534 break;
1535 msg_Dbg( id, "adding socket %d", fd );
1536 rtp_add_sink( id, fd, true );
1538 vlc_restorecancel (canc);
1540 return NULL;
1543 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
1545 rtp_sink_t sink = { fd, NULL };
1546 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1547 rtcp_mux );
1548 if( sink.rtcp == NULL )
1549 msg_Err( id, "RTCP failed!" );
1551 vlc_mutex_lock( &id->lock_sink );
1552 INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
1553 vlc_mutex_unlock( &id->lock_sink );
1554 return VLC_SUCCESS;
1557 void rtp_del_sink( sout_stream_id_t *id, int fd )
1559 rtp_sink_t sink = { fd, NULL };
1561 /* NOTE: must be safe to use if fd is not included */
1562 vlc_mutex_lock( &id->lock_sink );
1563 for( int i = 0; i < id->sinkc; i++ )
1565 if (id->sinkv[i].rtp_fd == fd)
1567 sink = id->sinkv[i];
1568 REMOVE_ELEM( id->sinkv, id->sinkc, i );
1569 break;
1572 vlc_mutex_unlock( &id->lock_sink );
1574 CloseRTCP( sink.rtcp );
1575 net_Close( sink.rtp_fd );
1578 uint16_t rtp_get_seq( const sout_stream_id_t *id )
1580 /* This will return values for the next packet.
1581 * Accounting for caching would not be totally trivial. */
1582 return id->i_sequence;
1585 /* FIXME: this is pretty bad - if we remove and then insert an ES
1586 * the number will get unsynched from inside RTSP */
1587 unsigned rtp_get_num( const sout_stream_id_t *id )
1589 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1590 int i;
1592 vlc_mutex_lock( &p_sys->lock_es );
1593 for( i = 0; i < p_sys->i_es; i++ )
1595 if( id == p_sys->es[i] )
1596 break;
1598 vlc_mutex_unlock( &p_sys->lock_es );
1600 return i;
1604 void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
1605 int b_marker, int64_t i_pts )
1607 uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
1609 out->p_buffer[0] = 0x80;
1610 out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
1611 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1612 out->p_buffer[3] = ( id->i_sequence )&0xff;
1613 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1614 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1615 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1616 out->p_buffer[7] = ( i_timestamp )&0xff;
1618 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1620 out->i_buffer = 12;
1621 id->i_sequence++;
1624 void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
1626 block_FifoPut( id->p_fifo, out );
1630 * @return configured max RTP payload size (including payload type-specific
1631 * headers, excluding RTP and transport headers)
1633 size_t rtp_mtu (const sout_stream_id_t *id)
1635 return id->i_mtu - 12;
1638 /*****************************************************************************
1639 * Non-RTP mux
1640 *****************************************************************************/
1642 /** Add an ES to a non-RTP muxed stream */
1643 static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
1645 sout_input_t *p_input;
1646 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1647 assert( p_mux != NULL );
1649 p_input = sout_MuxAddStream( p_mux, p_fmt );
1650 if( p_input == NULL )
1652 msg_Err( p_stream, "cannot add this stream to the muxer" );
1653 return NULL;
1656 return (sout_stream_id_t *)p_input;
1660 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
1661 block_t *p_buffer )
1663 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1664 assert( p_mux != NULL );
1666 sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1667 return VLC_SUCCESS;
1671 /** Remove an ES from a non-RTP muxed stream */
1672 static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
1674 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1675 assert( p_mux != NULL );
1677 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1678 return VLC_SUCCESS;
1682 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1683 const block_t *p_buffer )
1685 sout_stream_sys_t *p_sys = p_stream->p_sys;
1686 sout_stream_id_t *id = p_sys->es[0];
1688 int64_t i_dts = p_buffer->i_dts;
1690 uint8_t *p_data = p_buffer->p_buffer;
1691 size_t i_data = p_buffer->i_buffer;
1692 size_t i_max = id->i_mtu - 12;
1694 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1696 while( i_data > 0 )
1698 size_t i_size;
1700 /* output complete packet */
1701 if( p_sys->packet &&
1702 p_sys->packet->i_buffer + i_data > i_max )
1704 rtp_packetize_send( id, p_sys->packet );
1705 p_sys->packet = NULL;
1708 if( p_sys->packet == NULL )
1710 /* allocate a new packet */
1711 p_sys->packet = block_New( p_stream, id->i_mtu );
1712 rtp_packetize_common( id, p_sys->packet, 1, i_dts );
1713 p_sys->packet->i_dts = i_dts;
1714 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1715 i_dts += p_sys->packet->i_length;
1718 i_size = __MIN( i_data,
1719 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1721 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1722 p_data, i_size );
1724 p_sys->packet->i_buffer += i_size;
1725 p_data += i_size;
1726 i_data -= i_size;
1729 return VLC_SUCCESS;
1733 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1734 block_t *p_buffer )
1736 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1738 while( p_buffer )
1740 block_t *p_next;
1742 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1744 p_next = p_buffer->p_next;
1745 block_Release( p_buffer );
1746 p_buffer = p_next;
1749 return VLC_SUCCESS;
1753 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1755 sout_access_out_t *p_grab;
1757 p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
1758 if( p_grab == NULL )
1759 return NULL;
1761 p_grab->p_module = NULL;
1762 p_grab->psz_access = strdup( "grab" );
1763 p_grab->p_cfg = NULL;
1764 p_grab->psz_path = strdup( "" );
1765 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1766 p_grab->pf_seek = NULL;
1767 p_grab->pf_write = AccessOutGrabberWrite;
1768 vlc_object_attach( p_grab, p_stream );
1769 return p_grab;