?  search:
summary | log | graphiclog1 | graphiclog2 | commit | commitdiff | tree | refs | edit | fork
blame | history | raw | HEAD
opengl: factorize program creation
[vlc.git] / modules / stream_out / rtp.c
blobf4a089e79a22c9846ea4fc914c9dd1c257b43205
1 /*****************************************************************************
2 * rtp.c: rtp stream output module
3 *****************************************************************************
4 * Copyright (C) 2003-2004, 2010 the VideoLAN team
5 * Copyright © 2007-2008 Rémi Denis-Courmont
6 *
7 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
8 * Pierre Ynard
9 *
10 * This program is free software; you can redistribute it and/or modify
11 * it under the terms of the GNU General Public License as published by
12 * the Free Software Foundation; either version 2 of the License, or
13 * (at your option) any later version.
14 *
15 * This program is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
18 * GNU General Public License for more details.
19 *
20 * You should have received a copy of the GNU General Public License
21 * along with this program; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
23 *****************************************************************************/
24
25 /*****************************************************************************
26 * Preamble
27 *****************************************************************************/
28
29 #ifdef HAVE_CONFIG_H
30 # include "config.h"
31 #endif
32
33 #define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS
34 #include <vlc_common.h>
35 #include <vlc_plugin.h>
36 #include <vlc_sout.h>
37 #include <vlc_block.h>
38
39 #include <vlc_httpd.h>
40 #include <vlc_url.h>
41 #include <vlc_network.h>
42 #include <vlc_fs.h>
43 #include <vlc_rand.h>
44 #include <vlc_memstream.h>
45 #ifdef HAVE_SRTP
46 # include <srtp.h>
47 # include <gcrypt.h>
48 # include <vlc_gcrypt.h>
49 #endif
50
51 #include "rtp.h"
52
53 #include <sys/types.h>
54 #include <unistd.h>
55 #ifdef HAVE_ARPA_INET_H
56 # include <arpa/inet.h>
57 #endif
58 #ifdef HAVE_LINUX_DCCP_H
59 # include <linux/dccp.h>
60 #endif
61 #ifndef IPPROTO_DCCP
62 # define IPPROTO_DCCP 33
63 #endif
64 #ifndef IPPROTO_UDPLITE
65 # define IPPROTO_UDPLITE 136
66 #endif
67
68 #include <ctype.h>
69 #include <errno.h>
70 #include <assert.h>
71
72 /*****************************************************************************
73 * Module descriptor
74 *****************************************************************************/
75
76 #define DEST_TEXT N_("Destination")
77 #define DEST_LONGTEXT N_( \
78 "This is the output URL that will be used." )
79 #define SDP_TEXT N_("SDP")
80 #define SDP_LONGTEXT N_( \
81 "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
82 "session will be made available. You must use a url: http://location to " \
83 "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
84 "for the SDP to be announced via SAP." )
85 #define SAP_TEXT N_("SAP announcing")
86 #define SAP_LONGTEXT N_("Announce this session with SAP.")
87 #define MUX_TEXT N_("Muxer")
88 #define MUX_LONGTEXT N_( \
89 "This allows you to specify the muxer used for the streaming output. " \
90 "Default is to use no muxer (standard RTP stream)." )
91
92 #define NAME_TEXT N_("Session name")
93 #define NAME_LONGTEXT N_( \
94 "This is the name of the session that will be announced in the SDP " \
95 "(Session Descriptor)." )
96 #define CAT_TEXT N_("Session category")
97 #define CAT_LONGTEXT N_( \
98 "This allows you to specify a category for the session, " \
99 "that will be announced if you choose to use SAP." )
100 #define DESC_TEXT N_("Session description")
101 #define DESC_LONGTEXT N_( \
102 "This allows you to give a short description with details about the stream, " \
103 "that will be announced in the SDP (Session Descriptor)." )
104 #define URL_TEXT N_("Session URL")
105 #define URL_LONGTEXT N_( \
106 "This allows you to give a URL with more details about the stream " \
107 "(often the website of the streaming organization), that will " \
108 "be announced in the SDP (Session Descriptor)." )
109 #define EMAIL_TEXT N_("Session email")
110 #define EMAIL_LONGTEXT N_( \
111 "This allows you to give a contact mail address for the stream, that will " \
112 "be announced in the SDP (Session Descriptor)." )
113
114 #define PORT_TEXT N_("Port")
115 #define PORT_LONGTEXT N_( \
116 "This allows you to specify the base port for the RTP streaming." )
117 #define PORT_AUDIO_TEXT N_("Audio port")
118 #define PORT_AUDIO_LONGTEXT N_( \
119 "This allows you to specify the default audio port for the RTP streaming." )
120 #define PORT_VIDEO_TEXT N_("Video port")
121 #define PORT_VIDEO_LONGTEXT N_( \
122 "This allows you to specify the default video port for the RTP streaming." )
123
124 #define TTL_TEXT N_("Hop limit (TTL)")
125 #define TTL_LONGTEXT N_( \
126 "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
127 "the multicast packets sent by the stream output (-1 = use operating " \
128 "system built-in default).")
129
130 #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
131 #define RTCP_MUX_LONGTEXT N_( \
132 "This sends and receives RTCP packet multiplexed over the same port " \
133 "as RTP packets." )
134
135 #define CACHING_TEXT N_("Caching value (ms)")
136 #define CACHING_LONGTEXT N_( \
137 "Default caching value for outbound RTP streams. This " \
138 "value should be set in milliseconds." )
139
140 #define PROTO_TEXT N_("Transport protocol")
141 #define PROTO_LONGTEXT N_( \
142 "This selects which transport protocol to use for RTP." )
143
144 #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
145 #define SRTP_KEY_LONGTEXT N_( \
146 "RTP packets will be integrity-protected and ciphered "\
147 "with this Secure RTP master shared secret key. "\
148 "This must be a 32-character-long hexadecimal string.")
149
150 #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
151 #define SRTP_SALT_LONGTEXT N_( \
152 "Secure RTP requires a (non-secret) master salt value. " \
153 "This must be a 28-character-long hexadecimal string.")
154
155 static const char *const ppsz_protos[] = {
156 "dccp", "sctp", "tcp", "udp", "udplite",
157 };
158
159 static const char *const ppsz_protocols[] = {
160 "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
161 };
162
163 #define RFC3016_TEXT N_("MP4A LATM")
164 #define RFC3016_LONGTEXT N_( \
165 "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
166
167 #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
168 #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
169 "not receiving any RTSP request for this long. Setting it to a " \
170 "negative value or zero disables timeouts. The default is 60 (one " \
171 "minute)." )
172
173 #define RTSP_USER_TEXT N_("Username")
174 #define RTSP_USER_LONGTEXT N_("Username that will be " \
175 "requested to access the stream." )
176 #define RTSP_PASS_TEXT N_("Password")
177 #define RTSP_PASS_LONGTEXT N_("Password that will be " \
178 "requested to access the stream." )
179
180 static int Open ( vlc_object_t * );
181 static void Close( vlc_object_t * );
182
183 #define SOUT_CFG_PREFIX "sout-rtp-"
184 #define MAX_EMPTY_BLOCKS 200
185
186 vlc_module_begin ()
187 set_shortname( N_("RTP"))
188 set_description( N_("RTP stream output") )
189 set_capability( "sout output", 0 )
190 add_shortcut( "rtp" )
191 set_category( CAT_SOUT )
192 set_subcategory( SUBCAT_SOUT_STREAM )
193
194 add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
195 DEST_LONGTEXT, true )
196 add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
197 SDP_LONGTEXT, true )
198 add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
199 MUX_LONGTEXT, true )
200 add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
201 true )
202
203 add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
204 NAME_LONGTEXT, true )
205 add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_LONGTEXT, true )
206 add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
207 DESC_LONGTEXT, true )
208 add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
209 URL_LONGTEXT, true )
210 add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
211 EMAIL_LONGTEXT, true )
212 add_obsolete_string( SOUT_CFG_PREFIX "phone" ) /* since 3.0.0 */
213
214 add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
215 PROTO_LONGTEXT, false )
216 change_string_list( ppsz_protos, ppsz_protocols )
217 add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
218 PORT_LONGTEXT, true )
219 add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
220 PORT_AUDIO_LONGTEXT, true )
221 add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
222 PORT_VIDEO_LONGTEXT, true )
223
224 add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
225 TTL_LONGTEXT, true )
226 add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
227 RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
228 add_integer( SOUT_CFG_PREFIX "caching", MS_FROM_VLC_TICK(DEFAULT_PTS_DELAY),
229 CACHING_TEXT, CACHING_LONGTEXT, true )
230
231 #ifdef HAVE_SRTP
232 add_string( SOUT_CFG_PREFIX "key", "",
233 SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
234 add_string( SOUT_CFG_PREFIX "salt", "",
235 SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
236 #endif
237
238 add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
239 RFC3016_LONGTEXT, false )
240
241 set_callbacks( Open, Close )
242 vlc_module_end ()
243
244 /*****************************************************************************
245 * Exported prototypes
246 *****************************************************************************/
247 static const char *const ppsz_sout_options[] = {
248 "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
249 "mux", "sap", "description", "url", "email",
250 "proto", "rtcp-mux", "caching",
251 #ifdef HAVE_SRTP
252 "key", "salt",
253 #endif
254 "mp4a-latm", NULL
255 };
256
257 static void *Add( sout_stream_t *, const es_format_t * );
258 static void Del( sout_stream_t *, void * );
259 static int Send( sout_stream_t *, void *, block_t * );
260
261 static void *MuxAdd( sout_stream_t *, const es_format_t * );
262 static void MuxDel( sout_stream_t *, void * );
263 static int MuxSend( sout_stream_t *, void *, block_t * );
264
265 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
266 static void* ThreadSend( void * );
267 static void *rtp_listen_thread( void * );
268
269 static void SDPHandleUrl( sout_stream_t *, const char * );
270
271 static int SapSetup( sout_stream_t *p_stream );
272 static int FileSetup( sout_stream_t *p_stream );
273 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
274
275 typedef struct
276 {
277 /* SDP */
278 char *psz_sdp;
279 vlc_mutex_t lock_sdp;
280
281 /* SDP to disk */
282 char *psz_sdp_file;
283
284 /* SDP via SAP */
285 bool b_export_sap;
286 session_descriptor_t *p_session;
287
288 /* SDP via HTTP */
289 httpd_host_t *p_httpd_host;
290 httpd_file_t *p_httpd_file;
291
292 /* RTSP */
293 rtsp_stream_t *rtsp;
294
295 /* RTSP NPT and timestamp computations */
296 vlc_tick_t i_npt_zero; /* when NPT=0 packet is sent */
297 vlc_tick_t i_pts_zero; /* predicts PTS of NPT=0 packet */
298 vlc_tick_t i_pts_offset; /* matches actual PTS to prediction */
299 vlc_mutex_t lock_ts;
300
301 /* */
302 char *psz_destination;
303 uint16_t i_port;
304 uint16_t i_port_audio;
305 uint16_t i_port_video;
306 uint8_t proto;
307 bool rtcp_mux;
308 bool b_latm;
309
310 /* in case we do TS/PS over rtp */
311 sout_mux_t *p_mux;
312 sout_access_out_t *p_grab;
313 block_t *packet;
314
315 /* */
316 vlc_mutex_t lock_es;
317 int i_es;
318 sout_stream_id_sys_t **es;
319 } sout_stream_sys_t;
320
321 typedef struct rtp_sink_t
322 {
323 int rtp_fd;
324 rtcp_sender_t *rtcp;
325 } rtp_sink_t;
326
327 struct sout_stream_id_sys_t
328 {
329 sout_stream_t *p_stream;
330 /* rtp field */
331 /* For RFC 4175, seqnum is extended to 32-bits */
332 uint32_t i_sequence;
333 bool b_first_packet;
334 bool b_ts_init;
335 uint32_t i_ts_offset;
336 uint8_t ssrc[4];
337
338 /* for rtsp */
339 uint16_t i_seq_sent_next;
340
341 /* for sdp */
342 rtp_format_t rtp_fmt;
343 int i_port;
344
345 /* Packetizer specific fields */
346 int i_mtu;
347 #ifdef HAVE_SRTP
348 srtp_session_t *srtp;
349 #endif
350
351 /* Packets sinks */
352 vlc_thread_t thread;
353 vlc_mutex_t lock_sink;
354 int sinkc;
355 rtp_sink_t *sinkv;
356 rtsp_stream_id_t *rtsp_id;
357 struct {
358 int *fd;
359 vlc_thread_t thread;
360 } listen;
361
362 block_fifo_t *p_fifo;
363 vlc_tick_t i_caching;
364 };
365
366 /*****************************************************************************
367 * Open:
368 *****************************************************************************/
369 static int Open( vlc_object_t *p_this )
370 {
371 sout_stream_t *p_stream = (sout_stream_t*)p_this;
372 sout_stream_sys_t *p_sys = NULL;
373 char *psz;
374 bool b_rtsp = false;
375
376 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
377 ppsz_sout_options, p_stream->p_cfg );
378
379 p_sys = malloc( sizeof( sout_stream_sys_t ) );
380 if( p_sys == NULL )
381 return VLC_ENOMEM;
382
383 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
384
385 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
386 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
387 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
388 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
389
390 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
391 {
392 msg_Err( p_stream, "audio and video RTP port must be distinct" );
393 free( p_sys->psz_destination );
394 free( p_sys );
395 return VLC_EGENERIC;
396 }
397
398 for( config_chain_t *p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
399 {
400 if( !strcmp( p_cfg->psz_name, "sdp" )
401 && ( p_cfg->psz_value != NULL )
402 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
403 {
404 b_rtsp = true;
405 break;
406 }
407 }
408 if( !b_rtsp )
409 {
410 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
411 if( psz != NULL )
412 {
413 if( !strncasecmp( psz, "rtsp:", 5 ) )
414 b_rtsp = true;
415 free( psz );
416 }
417 }
418
419 /* Transport protocol */
420 p_sys->proto = IPPROTO_UDP;
421 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
422
423 if ((psz == NULL) || !strcasecmp (psz, "udp"))
424 (void)0; /* default */
425 else
426 if (!strcasecmp (psz, "dccp"))
427 {
428 p_sys->proto = IPPROTO_DCCP;
429 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
430 }
431 #if 0
432 else
433 if (!strcasecmp (psz, "sctp"))
434 {
435 p_sys->proto = IPPROTO_TCP;
436 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
437 }
438 #endif
439 #if 0
440 else
441 if (!strcasecmp (psz, "tcp"))
442 {
443 p_sys->proto = IPPROTO_TCP;
444 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
445 }
446 #endif
447 else
448 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
449 p_sys->proto = IPPROTO_UDPLITE;
450 else
451 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
452 psz);
453 free (psz);
454 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
455
456 if( p_sys->psz_destination == NULL && !b_rtsp )
457 {
458 msg_Err( p_stream, "missing destination and not in RTSP mode" );
459 free( p_sys );
460 return VLC_EGENERIC;
461 }
462
463 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
464 if( i_ttl != -1 )
465 {
466 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
467 var_SetInteger( p_stream, "ttl", i_ttl );
468 }
469
470 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
471
472 /* NPT=0 time will be determined when we packetize the first packet
473 * (of any ES). But we want to be able to report rtptime in RTSP
474 * without waiting. So until then,
475 * we use an arbitrary reference PTS for timestamp computations, and
476 * then actual PTS will catch up using offsets. */
477 p_sys->i_npt_zero = VLC_TICK_INVALID;
478 p_sys->i_pts_zero = vlc_tick_now();
479 p_sys->i_es = 0;
480 p_sys->es = NULL;
481 p_sys->rtsp = NULL;
482 p_sys->psz_sdp = NULL;
483
484 p_sys->b_export_sap = false;
485 p_sys->p_session = NULL;
486 p_sys->psz_sdp_file = NULL;
487
488 p_sys->p_httpd_host = NULL;
489 p_sys->p_httpd_file = NULL;
490
491 p_stream->p_sys = p_sys;
492
493 vlc_mutex_init( &p_sys->lock_sdp );
494 vlc_mutex_init( &p_sys->lock_ts );
495 vlc_mutex_init( &p_sys->lock_es );
496
497 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
498 if( psz != NULL )
499 {
500 /* Check muxer type */
501 if( strncasecmp( psz, "ps", 2 )
502 && strncasecmp( psz, "mpeg1", 5 )
503 && strncasecmp( psz, "ts", 2 ) )
504 {
505 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
506 free( psz );
507 free( p_sys->psz_destination );
508 free( p_sys );
509 return VLC_EGENERIC;
510 }
511
512 p_sys->p_grab = GrabberCreate( p_stream );
513 p_sys->p_mux = sout_MuxNew( p_sys->p_grab, psz );
514 free( psz );
515
516 if( p_sys->p_mux == NULL )
517 {
518 msg_Err( p_stream, "cannot create muxer" );
519 sout_AccessOutDelete( p_sys->p_grab );
520 free( p_sys->psz_destination );
521 free( p_sys );
522 return VLC_EGENERIC;
523 }
524
525 p_sys->packet = NULL;
526
527 p_stream->pf_add = MuxAdd;
528 p_stream->pf_del = MuxDel;
529 p_stream->pf_send = MuxSend;
530 }
531 else
532 {
533 p_sys->p_mux = NULL;
534 p_sys->p_grab = NULL;
535
536 p_stream->pf_add = Add;
537 p_stream->pf_del = Del;
538 p_stream->pf_send = Send;
539 }
540 p_stream->pace_nocontrol = true;
541
542 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
543 SDPHandleUrl( p_stream, "sap://" );
544
545 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
546 if( psz != NULL )
547 {
548 config_chain_t *p_cfg;
549
550 SDPHandleUrl( p_stream, psz );
551
552 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
553 {
554 if( !strcmp( p_cfg->psz_name, "sdp" ) )
555 {
556 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
557 continue;
558
559 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
560 if( !strcmp( p_cfg->psz_value, psz ) )
561 continue;
562
563 SDPHandleUrl( p_stream, p_cfg->psz_value );
564 }
565 }
566 free( psz );
567 }
568
569 if( p_sys->p_mux != NULL )
570 {
571 sout_stream_id_sys_t *id = Add( p_stream, NULL );
572 if( id == NULL )
573 {
574 Close( p_this );
575 return VLC_EGENERIC;
576 }
577 }
578
579 return VLC_SUCCESS;
580 }
581
582 /*****************************************************************************
583 * Close:
584 *****************************************************************************/
585 static void Close( vlc_object_t * p_this )
586 {
587 sout_stream_t *p_stream = (sout_stream_t*)p_this;
588 sout_stream_sys_t *p_sys = p_stream->p_sys;
589
590 if( p_sys->p_mux )
591 {
592 assert( p_sys->i_es <= 1 );
593
594 sout_MuxDelete( p_sys->p_mux );
595 if ( p_sys->i_es > 0 )
596 Del( p_stream, p_sys->es[0] );
597 sout_AccessOutDelete( p_sys->p_grab );
598
599 if( p_sys->packet )
600 {
601 block_Release( p_sys->packet );
602 }
603 }
604
605 if( p_sys->rtsp != NULL )
606 RtspUnsetup( p_sys->rtsp );
607
608 if( p_sys->p_httpd_file )
609 httpd_FileDelete( p_sys->p_httpd_file );
610
611 if( p_sys->p_httpd_host )
612 httpd_HostDelete( p_sys->p_httpd_host );
613
614 free( p_sys->psz_sdp );
615
616 if( p_sys->psz_sdp_file != NULL )
617 {
618 unlink( p_sys->psz_sdp_file );
619 free( p_sys->psz_sdp_file );
620 }
621 free( p_sys->psz_destination );
622 free( p_sys );
623 }
624
625 /*****************************************************************************
626 * SDPHandleUrl:
627 *****************************************************************************/
628 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
629 {
630 sout_stream_sys_t *p_sys = p_stream->p_sys;
631 vlc_url_t url;
632
633 vlc_UrlParse( &url, psz_url );
634 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
635 {
636 if( p_sys->p_httpd_file )
637 {
638 msg_Err( p_stream, "you can use sdp=http:// only once" );
639 goto out;
640 }
641
642 if( HttpSetup( p_stream, &url ) )
643 {
644 msg_Err( p_stream, "cannot export SDP as HTTP" );
645 }
646 }
647 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
648 {
649 if( p_sys->rtsp != NULL )
650 {
651 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
652 goto out;
653 }
654
655 if( url.psz_host != NULL && *url.psz_host )
656 {
657 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
658 "multiple-host configurations, use at your own risks.",
659 url.psz_host );
660 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
661 "command line instead." );
662
663 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
664 var_SetString( p_stream, "rtsp-host", url.psz_host );
665 }
666 if( url.i_port != 0 )
667 {
668 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
669 "the command line instead.", url.i_port ); */
670
671 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
672 var_SetInteger( p_stream, "rtsp-port", url.i_port );
673 }
674
675 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), url.psz_path );
676 if( p_sys->rtsp == NULL )
677 msg_Err( p_stream, "cannot export SDP as RTSP" );
678 }
679 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
680 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
681 {
682 p_sys->b_export_sap = true;
683 SapSetup( p_stream );
684 }
685 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
686 {
687 if( p_sys->psz_sdp_file != NULL )
688 {
689 msg_Err( p_stream, "you can use sdp=file:// only once" );
690 goto out;
691 }
692 p_sys->psz_sdp_file = vlc_uri2path( psz_url );
693 if( p_sys->psz_sdp_file == NULL )
694 goto out;
695 FileSetup( p_stream );
696 }
697 else
698 {
699 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
700 url.psz_protocol );
701 }
702
703 out:
704 vlc_UrlClean( &url );
705 }
706
707 /*****************************************************************************
708 * SDPGenerate
709 *****************************************************************************/
710 /*static*/
711 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
712 {
713 sout_stream_sys_t *p_sys = p_stream->p_sys;
714 struct vlc_memstream sdp;
715 struct sockaddr_storage dst;
716 char *psz_sdp = NULL;
717 socklen_t dstlen;
718 int i;
719 /*
720 * When we have a fixed destination (typically when we do multicast),
721 * we need to put the actual port numbers in the SDP.
722 * When there is no fixed destination, we only support RTSP unicast
723 * on-demand setup, so we should rather let the clients decide which ports
724 * to use.
725 * When there is both a fixed destination and RTSP unicast, we need to
726 * put port numbers used by the fixed destination, otherwise the SDP would
727 * become totally incorrect for multicast use. It should be noted that
728 * port numbers from SDP with RTSP are only "recommendation" from the
729 * server to the clients (per RFC2326), so only broken clients will fail
730 * to handle this properly. There is no solution but to use two differents
731 * output chain with two different RTSP URLs if you need to handle this
732 * scenario.
733 */
734 int inclport;
735
736 vlc_mutex_lock( &p_sys->lock_es );
737 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
738 goto out; /* hmm... */
739
740 if( p_sys->psz_destination != NULL )
741 {
742 inclport = 1;
743
744 /* Oh boy, this is really ugly! */
745 dstlen = sizeof( dst );
746 if( p_sys->es[0]->listen.fd != NULL )
747 getsockname( p_sys->es[0]->listen.fd[0],
748 (struct sockaddr *)&dst, &dstlen );
749 else
750 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
751 (struct sockaddr *)&dst, &dstlen );
752 }
753 else
754 {
755 inclport = 0;
756
757 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
758 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
759 && rtsp_url[7] == '[';
760
761 /* Dummy destination address for RTSP */
762 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
763 : sizeof( struct sockaddr_in );
764 memset (&dst, 0, dstlen);
765 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
766 #ifdef HAVE_SA_LEN
767 dst.ss_len = dstlen;
768 #endif
769 }
770
771 if( vlc_sdp_Start( &sdp, VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
772 NULL, 0, (struct sockaddr *)&dst, dstlen ) )
773 goto out;
774
775 /* TODO: a=source-filter */
776 if( p_sys->rtcp_mux )
777 sdp_AddAttribute( &sdp, "rtcp-mux", NULL );
778
779 if( rtsp_url != NULL )
780 sdp_AddAttribute ( &sdp, "control", "%s", rtsp_url );
781
782 const char *proto = "RTP/AVP"; /* protocol */
783 if( rtsp_url == NULL )
784 {
785 switch( p_sys->proto )
786 {
787 case IPPROTO_UDP:
788 break;
789 case IPPROTO_TCP:
790 proto = "TCP/RTP/AVP";
791 break;
792 case IPPROTO_DCCP:
793 proto = "DCCP/RTP/AVP";
794 break;
795 case IPPROTO_UDPLITE:
796 return psz_sdp;
797 }
798 }
799
800 for( i = 0; i < p_sys->i_es; i++ )
801 {
802 sout_stream_id_sys_t *id = p_sys->es[i];
803 rtp_format_t *rtp_fmt = &id->rtp_fmt;
804 const char *mime_major; /* major MIME type */
805
806 switch( rtp_fmt->cat )
807 {
808 case VIDEO_ES:
809 mime_major = "video";
810 break;
811 case AUDIO_ES:
812 mime_major = "audio";
813 break;
814 case SPU_ES:
815 mime_major = "text";
816 break;
817 default:
818 continue;
819 }
820
821 sdp_AddMedia( &sdp, mime_major, proto, inclport * id->i_port,
822 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
823 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
824 rtp_fmt->fmtp);
825
826 /* cf RFC4566 §5.14 */
827 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
828 sdp_AddAttribute( &sdp, "rtcp", "%u", id->i_port + 1 );
829
830 if( rtsp_url != NULL )
831 {
832 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
833 if( track_url != NULL )
834 {
835 sdp_AddAttribute( &sdp, "control", "%s", track_url );
836 free( track_url );
837 }
838 }
839 else
840 {
841 if( id->listen.fd != NULL )
842 sdp_AddAttribute( &sdp, "setup", "passive" );
843 if( p_sys->proto == IPPROTO_DCCP )
844 sdp_AddAttribute( &sdp, "dccp-service-code", "SC:RTP%c",
845 toupper( (unsigned char)mime_major[0] ) );
846 }
847 }
848
849 if( vlc_memstream_close( &sdp ) == 0 )
850 psz_sdp = sdp.ptr;
851 out:
852 vlc_mutex_unlock( &p_sys->lock_es );
853 return psz_sdp;
854 }
855
856 /*****************************************************************************
857 * RTP mux
858 *****************************************************************************/
859
860 /**
861 * Shrink the MTU down to a fixed packetization time (for audio).
862 */
863 static void
864 rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes)
865 {
866 /* Samples per second */
867 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
868 bytes *= id->rtp_fmt.channels;
869 spl *= bytes;
870
871 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
872 id->i_mtu = 12 + spl;
873 else /* MTU is too small for ptime, align to a sample boundary */
874 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
875 }
876
877 uint32_t rtp_compute_ts( unsigned i_clock_rate, vlc_tick_t i_pts )
878 {
879 /* This is an overflow-proof way of doing:
880 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
881 *
882 * NOTE: this plays nice with offsets because the (equivalent)
883 * calculations are linear. */
884 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
885 return q.quot * (int64_t)i_clock_rate
886 + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
887 }
888
889 /** Add an ES as a new RTP stream */
890 static void *Add( sout_stream_t *p_stream, const es_format_t *p_fmt )
891 {
892 /* NOTE: As a special case, if we use a non-RTP
893 * mux (TS/PS), then p_fmt is NULL. */
894 sout_stream_sys_t *p_sys = p_stream->p_sys;
895 char *psz_sdp;
896
897 sout_stream_id_sys_t *id = malloc( sizeof( *id ) );
898 if( unlikely(id == NULL) )
899 return NULL;
900 id->p_stream = p_stream;
901
902 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
903 if( id->i_mtu <= 12 + 16 )
904 id->i_mtu = 576 - 20 - 8; /* pessimistic */
905 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
906
907 #ifdef HAVE_SRTP
908 id->srtp = NULL;
909 #endif
910 vlc_mutex_init( &id->lock_sink );
911 id->sinkc = 0;
912 id->sinkv = NULL;
913 id->rtsp_id = NULL;
914 id->p_fifo = NULL;
915 id->listen.fd = NULL;
916
917 id->b_first_packet = true;
918 id->i_caching =
919 VLC_TICK_FROM_MS(var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching"));
920
921 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
922 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
923
924 bool format = false;
925
926 if (!format)
927 {
928 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
929 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
930 if (p_fmt == NULL && psz == NULL)
931 goto error;
932 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
933 free( psz );
934 if (val != VLC_SUCCESS)
935 goto error;
936 }
937
938 #ifdef HAVE_SRTP
939 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
940 if (key)
941 {
942 vlc_gcrypt_init ();
943 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
944 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
945 if (id->srtp == NULL)
946 {
947 free (key);
948 goto error;
949 }
950
951 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
952 int val = srtp_setkeystring (id->srtp, key, salt ? salt : "");
953 free (salt);
954 free (key);
955 if (val)
956 {
957 msg_Err (p_stream, "bad SRTP key/salt combination (%s)",
958 vlc_strerror_c(val));
959 goto error;
960 }
961 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
962 }
963 #endif
964
965 id->i_seq_sent_next = id->i_sequence;
966
967 int mcast_fd = -1;
968 if( p_sys->psz_destination != NULL )
969 {
970 /* Choose the port */
971 uint16_t i_port = 0;
972 if( p_fmt == NULL )
973 ;
974 else
975 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
976 i_port = p_sys->i_port_audio;
977 else
978 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
979 i_port = p_sys->i_port_video;
980
981 /* We do not need the ES lock (p_sys->lock_es) here, because
982 * this is the only one thread that can *modify* the ES table.
983 * The ES lock protects the other threads from our modifications
984 * (TAB_APPEND, TAB_REMOVE). */
985 for (int i = 0; i_port && (i < p_sys->i_es); i++)
986 if (i_port == p_sys->es[i]->i_port)
987 i_port = 0; /* Port already in use! */
988 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
989 {
990 if (p == 0)
991 {
992 msg_Err (p_stream, "too many RTP elementary streams");
993 goto error;
994 }
995 i_port = p;
996 for (int i = 0; i_port && (i < p_sys->i_es); i++)
997 if (p == p_sys->es[i]->i_port)
998 i_port = 0;
999 }
1000
1001 id->i_port = i_port;
1002
1003 int type = SOCK_STREAM;
1004
1005 switch( p_sys->proto )
1006 {
1007 #ifdef SOCK_DCCP
1008 case IPPROTO_DCCP:
1009 {
1010 const char *code;
1011 switch (id->rtp_fmt.cat)
1012 {
1013 case VIDEO_ES: code = "RTPV"; break;
1014 case AUDIO_ES: code = "RTPARTPV"; break;
1015 case SPU_ES: code = "RTPTRTPV"; break;
1016 default: code = "RTPORTPV"; break;
1017 }
1018 var_SetString (p_stream, "dccp-service", code);
1019 type = SOCK_DCCP;
1020 }
1021 #endif
1022 /* fall through */
1023 case IPPROTO_TCP:
1024 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1025 p_sys->psz_destination, i_port,
1026 type, p_sys->proto );
1027 if( id->listen.fd == NULL )
1028 {
1029 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1030 goto error;
1031 }
1032 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1033 VLC_THREAD_PRIORITY_LOW ) )
1034 {
1035 net_ListenClose( id->listen.fd );
1036 id->listen.fd = NULL;
1037 goto error;
1038 }
1039 break;
1040
1041 default:
1042 {
1043 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1044 i_port, -1, p_sys->proto );
1045 if( fd == -1 )
1046 {
1047 msg_Err( p_stream, "cannot create RTP socket" );
1048 goto error;
1049 }
1050 /* Ignore any unexpected incoming packet (including RTCP-RR
1051 * packets in case of rtcp-mux) */
1052 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1053 sizeof (int));
1054 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1055 /* FIXME: test if this is multicast */
1056 mcast_fd = fd;
1057 }
1058 }
1059 }
1060
1061 if( p_fmt != NULL )
1062 switch( p_fmt->i_codec )
1063 {
1064 case VLC_CODEC_MULAW:
1065 case VLC_CODEC_ALAW:
1066 case VLC_CODEC_U8:
1067 rtp_set_ptime (id, 20, 1);
1068 break;
1069 case VLC_CODEC_S16B:
1070 case VLC_CODEC_S16L:
1071 rtp_set_ptime (id, 20, 2);
1072 break;
1073 case VLC_CODEC_S24B:
1074 rtp_set_ptime (id, 20, 3);
1075 break;
1076 default:
1077 break;
1078 }
1079
1080 #if 0 /* No payload formats sets this at the moment */
1081 int cscov = -1;
1082 if( cscov != -1 )
1083 cscov += 8 /* UDP */ + 12 /* RTP */;
1084 if( id->sinkc > 0 )
1085 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1086 #endif
1087
1088 vlc_mutex_lock( &p_sys->lock_ts );
1089 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TICK_INVALID );
1090 vlc_mutex_unlock( &p_sys->lock_ts );
1091 if( id->b_ts_init )
1092 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1093 p_sys->i_pts_offset );
1094
1095 if( p_sys->rtsp != NULL )
1096 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1097 id->rtp_fmt.clock_rate, mcast_fd );
1098
1099 id->p_fifo = block_FifoNew();
1100 if( unlikely(id->p_fifo == NULL) )
1101 goto error;
1102 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1103 {
1104 block_FifoRelease( id->p_fifo );
1105 id->p_fifo = NULL;
1106 goto error;
1107 }
1108
1109 /* Update p_sys context */
1110 vlc_mutex_lock( &p_sys->lock_es );
1111 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1112 vlc_mutex_unlock( &p_sys->lock_es );
1113
1114 psz_sdp = SDPGenerate( p_stream, NULL );
1115
1116 vlc_mutex_lock( &p_sys->lock_sdp );
1117 free( p_sys->psz_sdp );
1118 p_sys->psz_sdp = psz_sdp;
1119 vlc_mutex_unlock( &p_sys->lock_sdp );
1120
1121 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1122
1123 /* Update SDP (sap/file) */
1124 if( p_sys->b_export_sap ) SapSetup( p_stream );
1125 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1126
1127 return id;
1128
1129 error:
1130 Del( p_stream, id );
1131 return NULL;
1132 }
1133
1134 static void Del( sout_stream_t *p_stream, void *_id )
1135 {
1136 sout_stream_sys_t *p_sys = p_stream->p_sys;
1137 sout_stream_id_sys_t *id = (sout_stream_id_sys_t *)_id;
1138
1139 vlc_mutex_lock( &p_sys->lock_es );
1140 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1141 vlc_mutex_unlock( &p_sys->lock_es );
1142
1143 if( likely(id->p_fifo != NULL) )
1144 {
1145 vlc_cancel( id->thread );
1146 vlc_join( id->thread, NULL );
1147 block_FifoRelease( id->p_fifo );
1148 }
1149
1150 free( id->rtp_fmt.fmtp );
1151
1152 if( id->rtsp_id )
1153 RtspDelId( p_sys->rtsp, id->rtsp_id );
1154 if( id->listen.fd != NULL )
1155 {
1156 vlc_cancel( id->listen.thread );
1157 vlc_join( id->listen.thread, NULL );
1158 net_ListenClose( id->listen.fd );
1159 }
1160 /* Delete remaining sinks (incoming connections or explicit
1161 * outgoing dst=) */
1162 while( id->sinkc > 0 )
1163 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1164 #ifdef HAVE_SRTP
1165 if( id->srtp != NULL )
1166 srtp_destroy( id->srtp );
1167 #endif
1168
1169 /* Update SDP (sap/file) */
1170 if( p_sys->b_export_sap ) SapSetup( p_stream );
1171 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1172
1173 free( id );
1174 }
1175
1176 static int Send( sout_stream_t *p_stream, void *_id, block_t *p_buffer )
1177 {
1178 sout_stream_id_sys_t *id = (sout_stream_id_sys_t *)_id;
1179 assert( ((sout_stream_sys_t *)p_stream->p_sys)->p_mux == NULL );
1180
1181 while( p_buffer != NULL )
1182 {
1183 block_t *p_next = p_buffer->p_next;
1184 p_buffer->p_next = NULL;
1185
1186 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1187 * as the first packet of the stream */
1188 if (id->b_first_packet)
1189 {
1190 id->b_first_packet = false;
1191 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1192 !strcmp(id->rtp_fmt.ptname, "theora"))
1193 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1194 p_buffer->i_pts);
1195 }
1196
1197 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1198 break;
1199
1200 p_buffer = p_next;
1201 }
1202 return VLC_SUCCESS;
1203 }
1204
1205 /****************************************************************************
1206 * SAP:
1207 ****************************************************************************/
1208 static int SapSetup( sout_stream_t *p_stream )
1209 {
1210 sout_stream_sys_t *p_sys = p_stream->p_sys;
1211
1212 /* Remove the previous session */
1213 if( p_sys->p_session != NULL)
1214 {
1215 sout_AnnounceUnRegister( p_stream, p_sys->p_session);
1216 p_sys->p_session = NULL;
1217 }
1218
1219 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1220 p_sys->p_session = sout_AnnounceRegisterSDP( p_stream,
1221 p_sys->psz_sdp,
1222 p_sys->psz_destination );
1223
1224 return VLC_SUCCESS;
1225 }
1226
1227 /****************************************************************************
1228 * File:
1229 ****************************************************************************/
1230 static int FileSetup( sout_stream_t *p_stream )
1231 {
1232 sout_stream_sys_t *p_sys = p_stream->p_sys;
1233 FILE *f;
1234
1235 if( p_sys->psz_sdp == NULL )
1236 return VLC_EGENERIC; /* too early */
1237
1238 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1239 {
1240 msg_Err( p_stream, "cannot open file '%s' (%s)",
1241 p_sys->psz_sdp_file, vlc_strerror_c(errno) );
1242 return VLC_EGENERIC;
1243 }
1244
1245 fputs( p_sys->psz_sdp, f );
1246 fclose( f );
1247
1248 return VLC_SUCCESS;
1249 }
1250
1251 /****************************************************************************
1252 * HTTP:
1253 ****************************************************************************/
1254 static int HttpCallback( httpd_file_sys_t *p_args,
1255 httpd_file_t *, uint8_t *p_request,
1256 uint8_t **pp_data, int *pi_data );
1257
1258 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1259 {
1260 sout_stream_sys_t *p_sys = p_stream->p_sys;
1261
1262 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1263 if( p_sys->p_httpd_host )
1264 {
1265 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1266 url->psz_path ? url->psz_path : "/",
1267 "application/sdp",
1268 NULL, NULL,
1269 HttpCallback, (void*)p_sys );
1270 }
1271 if( p_sys->p_httpd_file == NULL )
1272 {
1273 return VLC_EGENERIC;
1274 }
1275 return VLC_SUCCESS;
1276 }
1277
1278 static int HttpCallback( httpd_file_sys_t *p_args,
1279 httpd_file_t *f, uint8_t *p_request,
1280 uint8_t **pp_data, int *pi_data )
1281 {
1282 VLC_UNUSED(f); VLC_UNUSED(p_request);
1283 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1284
1285 vlc_mutex_lock( &p_sys->lock_sdp );
1286 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1287 {
1288 *pi_data = strlen( p_sys->psz_sdp );
1289 *pp_data = malloc( *pi_data );
1290 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1291 }
1292 else
1293 {
1294 *pp_data = NULL;
1295 *pi_data = 0;
1296 }
1297 vlc_mutex_unlock( &p_sys->lock_sdp );
1298
1299 return VLC_SUCCESS;
1300 }
1301
1302 /****************************************************************************
1303 * RTP send
1304 ****************************************************************************/
1305 static void* ThreadSend( void *data )
1306 {
1307 #ifdef _WIN32
1308 # define ENOBUFS WSAENOBUFS
1309 # define EAGAIN WSAEWOULDBLOCK
1310 # define EWOULDBLOCK WSAEWOULDBLOCK
1311 #endif
1312 sout_stream_id_sys_t *id = data;
1313 vlc_tick_t i_caching = id->i_caching;
1314
1315 for (;;)
1316 {
1317 block_t *out = block_FifoGet( id->p_fifo );
1318 block_cleanup_push (out);
1319
1320 #ifdef HAVE_SRTP
1321 if( id->srtp )
1322 { /* FIXME: this is awfully inefficient */
1323 size_t len = out->i_buffer;
1324 out = block_Realloc( out, 0, len + 10 );
1325 out->i_buffer = len;
1326
1327 int canc = vlc_savecancel ();
1328 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1329 vlc_restorecancel (canc);
1330 if( val )
1331 {
1332 msg_Dbg( id->p_stream, "SRTP sending error: %s",
1333 vlc_strerror_c(val) );
1334 block_Release( out );
1335 out = NULL;
1336 }
1337 else
1338 out->i_buffer = len;
1339 }
1340 if (out)
1341 vlc_tick_wait (out->i_dts + i_caching);
1342 vlc_cleanup_pop ();
1343 if (out == NULL)
1344 continue;
1345 #else
1346 vlc_tick_wait (out->i_dts + i_caching);
1347 vlc_cleanup_pop ();
1348 #endif
1349
1350 ssize_t len = out->i_buffer;
1351 int canc = vlc_savecancel ();
1352
1353 vlc_mutex_lock( &id->lock_sink );
1354 unsigned deadc = 0; /* How many dead sockets? */
1355 int deadv[id->sinkc ? id->sinkc : 1]; /* Dead sockets list */
1356
1357 for( int i = 0; i < id->sinkc; i++ )
1358 {
1359 #ifdef HAVE_SRTP
1360 if( !id->srtp ) /* FIXME: SRTCP support */
1361 #endif
1362 SendRTCP( id->sinkv[i].rtcp, out );
1363
1364 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1365 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1366 && net_errno != ENOBUFS && net_errno != ENOMEM )
1367 {
1368 int type;
1369 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1370 &type, &(socklen_t){ sizeof(type) });
1371 if( type == SOCK_DGRAM )
1372 /* ICMP soft error: ignore and retry */
1373 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1374 else
1375 /* Broken connection */
1376 deadv[deadc++] = id->sinkv[i].rtp_fd;
1377 }
1378 }
1379 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1380 vlc_mutex_unlock( &id->lock_sink );
1381 block_Release( out );
1382
1383 for( unsigned i = 0; i < deadc; i++ )
1384 {
1385 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1386 rtp_del_sink( id, deadv[i] );
1387 }
1388 vlc_restorecancel (canc);
1389 }
1390 return NULL;
1391 }
1392
1393
1394 /* This thread dequeues incoming connections (DCCP streaming) */
1395 static void *rtp_listen_thread( void *data )
1396 {
1397 sout_stream_id_sys_t *id = data;
1398
1399 assert( id->listen.fd != NULL );
1400
1401 for( ;; )
1402 {
1403 int fd = net_Accept( id->p_stream, id->listen.fd );
1404 if( fd == -1 )
1405 continue;
1406 int canc = vlc_savecancel( );
1407 rtp_add_sink( id, fd, true, NULL );
1408 vlc_restorecancel( canc );
1409 }
1410
1411 vlc_assert_unreachable();
1412 }
1413
1414
1415 int rtp_add_sink( sout_stream_id_sys_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1416 {
1417 rtp_sink_t sink = { fd, NULL };
1418 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1419 rtcp_mux );
1420 if( sink.rtcp == NULL )
1421 msg_Err( id->p_stream, "RTCP failed!" );
1422
1423 vlc_mutex_lock( &id->lock_sink );
1424 TAB_APPEND(id->sinkc, id->sinkv, sink);
1425 if( seq != NULL )
1426 *seq = id->i_seq_sent_next;
1427 vlc_mutex_unlock( &id->lock_sink );
1428 return VLC_SUCCESS;
1429 }
1430
1431 void rtp_del_sink( sout_stream_id_sys_t *id, int fd )
1432 {
1433 rtp_sink_t sink = { fd, NULL };
1434
1435 /* NOTE: must be safe to use if fd is not included */
1436 vlc_mutex_lock( &id->lock_sink );
1437 for( int i = 0; i < id->sinkc; i++ )
1438 {
1439 if (id->sinkv[i].rtp_fd == fd)
1440 {
1441 sink = id->sinkv[i];
1442 TAB_ERASE(id->sinkc, id->sinkv, i);
1443 break;
1444 }
1445 }
1446 vlc_mutex_unlock( &id->lock_sink );
1447
1448 CloseRTCP( sink.rtcp );
1449 net_Close( sink.rtp_fd );
1450 }
1451
1452 uint16_t rtp_get_seq( sout_stream_id_sys_t *id )
1453 {
1454 /* This will return values for the next packet. */
1455 uint16_t seq;
1456
1457 vlc_mutex_lock( &id->lock_sink );
1458 seq = id->i_seq_sent_next;
1459 vlc_mutex_unlock( &id->lock_sink );
1460
1461 return seq;
1462 }
1463
1464 /* Return a timestamp corresponding to packets being sent now, and that
1465 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1466 * Also return the NPT corresponding to this timestamp. If the stream
1467 * output is not started, the initial timestamp that will be used with
1468 * the first packets for NPT=0 is returned instead. */
1469 vlc_tick_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_sys_t *id,
1470 vlc_tick_t *p_npt )
1471 {
1472 if (p_npt != NULL)
1473 *p_npt = 0;
1474
1475 if (id != NULL)
1476 p_stream = id->p_stream;
1477
1478 if (p_stream == NULL)
1479 return vlc_tick_now();
1480
1481 sout_stream_sys_t *p_sys = p_stream->p_sys;
1482 vlc_tick_t i_npt_zero;
1483 vlc_mutex_lock( &p_sys->lock_ts );
1484 i_npt_zero = p_sys->i_npt_zero;
1485 vlc_mutex_unlock( &p_sys->lock_ts );
1486
1487 if( i_npt_zero == VLC_TICK_INVALID )
1488 return p_sys->i_pts_zero;
1489
1490 vlc_tick_t now = vlc_tick_now();
1491 if( now < i_npt_zero )
1492 return p_sys->i_pts_zero;
1493
1494 vlc_tick_t npt = now - i_npt_zero;
1495 if (p_npt != NULL)
1496 *p_npt = npt;
1497
1498 return p_sys->i_pts_zero + npt;
1499 }
1500
1501 void rtp_packetize_common( sout_stream_id_sys_t *id, block_t *out,
1502 bool b_m_bit, vlc_tick_t i_pts )
1503 {
1504 if( !id->b_ts_init )
1505 {
1506 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1507 vlc_mutex_lock( &p_sys->lock_ts );
1508 if( p_sys->i_npt_zero == VLC_TICK_INVALID )
1509 {
1510 /* This is the first packet of any ES. We initialize the
1511 * NPT=0 time reference, and the offset to match the
1512 * arbitrary PTS reference. */
1513 p_sys->i_npt_zero = i_pts + id->i_caching;
1514 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1515 }
1516 vlc_mutex_unlock( &p_sys->lock_ts );
1517
1518 /* And in any case this is the first packet of this ES, so we
1519 * initialize the offset for this ES. */
1520 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1521 p_sys->i_pts_offset );
1522 id->b_ts_init = true;
1523 }
1524
1525 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1526 + id->i_ts_offset;
1527
1528 out->p_buffer[0] = 0x80;
1529 out->p_buffer[1] = (b_m_bit?0x80:0x00)|id->rtp_fmt.payload_type;
1530 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1531 out->p_buffer[3] = ( id->i_sequence )&0xff;
1532 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1533 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1534 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1535 out->p_buffer[7] = ( i_timestamp )&0xff;
1536
1537 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1538
1539 id->i_sequence++;
1540 }
1541
1542 uint16_t rtp_get_extended_sequence( sout_stream_id_sys_t *id )
1543 {
1544 return id->i_sequence >> 16;
1545 }
1546
1547 void rtp_packetize_send( sout_stream_id_sys_t *id, block_t *out )
1548 {
1549 block_FifoPut( id->p_fifo, out );
1550 }
1551
1552 /**
1553 * @return configured max RTP payload size (including payload type-specific
1554 * headers, excluding RTP and transport headers)
1555 */
1556 size_t rtp_mtu (const sout_stream_id_sys_t *id)
1557 {
1558 return id->i_mtu - 12;
1559 }
1560
1561 /*****************************************************************************
1562 * Non-RTP mux
1563 *****************************************************************************/
1564
1565 /** Add an ES to a non-RTP muxed stream */
1566 static void *MuxAdd( sout_stream_t *p_stream, const es_format_t *p_fmt )
1567 {
1568 sout_input_t *p_input;
1569 sout_stream_sys_t *p_sys = p_stream->p_sys;
1570 sout_mux_t *p_mux = p_sys->p_mux;
1571 assert( p_mux != NULL );
1572
1573 p_input = sout_MuxAddStream( p_mux, p_fmt );
1574 if( p_input == NULL )
1575 {
1576 msg_Err( p_stream, "cannot add this stream to the muxer" );
1577 return NULL;
1578 }
1579
1580 return (sout_stream_id_sys_t *)p_input;
1581 }
1582
1583
1584 static int MuxSend( sout_stream_t *p_stream, void *id, block_t *p_buffer )
1585 {
1586 sout_stream_sys_t *p_sys = p_stream->p_sys;
1587 sout_mux_t *p_mux = p_sys->p_mux;
1588 assert( p_mux != NULL );
1589
1590 return sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1591 }
1592
1593
1594 /** Remove an ES from a non-RTP muxed stream */
1595 static void MuxDel( sout_stream_t *p_stream, void *id )
1596 {
1597 sout_stream_sys_t *p_sys = p_stream->p_sys;
1598 sout_mux_t *p_mux = p_sys->p_mux;
1599 assert( p_mux != NULL );
1600
1601 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1602 }
1603
1604
1605 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1606 const block_t *p_buffer )
1607 {
1608 sout_stream_sys_t *p_sys = p_stream->p_sys;
1609 sout_stream_id_sys_t *id = p_sys->es[0];
1610
1611 vlc_tick_t i_dts = p_buffer->i_dts;
1612
1613 uint8_t *p_data = p_buffer->p_buffer;
1614 size_t i_data = p_buffer->i_buffer;
1615 size_t i_max = id->i_mtu - 12;
1616 bool b_dis = (p_buffer->i_flags & BLOCK_FLAG_DISCONTINUITY);
1617
1618 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1619
1620 while( i_data > 0 )
1621 {
1622 size_t i_size;
1623
1624 /* output complete packet */
1625 if( p_sys->packet &&
1626 p_sys->packet->i_buffer + i_data > i_max )
1627 {
1628 rtp_packetize_send( id, p_sys->packet );
1629 p_sys->packet = NULL;
1630 }
1631
1632 if( p_sys->packet == NULL )
1633 {
1634 /* allocate a new packet */
1635 p_sys->packet = block_Alloc( id->i_mtu );
1636 /* m-bit is discontinuity for MPEG1/2 PS and TS, RFC2250 2.1 */
1637 rtp_packetize_common( id, p_sys->packet, b_dis, i_dts );
1638 p_sys->packet->i_buffer = 12;
1639 p_sys->packet->i_dts = i_dts;
1640 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1641 i_dts += p_sys->packet->i_length;
1642 b_dis = false;
1643 }
1644
1645 i_size = __MIN( i_data,
1646 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1647
1648 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1649 p_data, i_size );
1650
1651 p_sys->packet->i_buffer += i_size;
1652 p_data += i_size;
1653 i_data -= i_size;
1654 }
1655
1656 return VLC_SUCCESS;
1657 }
1658
1659
1660 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1661 block_t *p_buffer )
1662 {
1663 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1664
1665 while( p_buffer )
1666 {
1667 block_t *p_next;
1668
1669 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1670
1671 p_next = p_buffer->p_next;
1672 block_Release( p_buffer );
1673 p_buffer = p_next;
1674 }
1675
1676 return VLC_SUCCESS;
1677 }
1678
1679
1680 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1681 {
1682 sout_access_out_t *p_grab;
1683
1684 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1685 if( p_grab == NULL )
1686 return NULL;
1687
1688 p_grab->p_module = NULL;
1689 p_grab->psz_access = strdup( "grab" );
1690 p_grab->p_cfg = NULL;
1691 p_grab->psz_path = strdup( "" );
1692 p_grab->p_sys = p_stream;
1693 p_grab->pf_seek = NULL;
1694 p_grab->pf_write = AccessOutGrabberWrite;
1695 return p_grab;
1696 }
1697
1698 void rtp_get_video_geometry( sout_stream_id_sys_t *id, int *width, int *height )
1699 {
1700 int ret = sscanf( id->rtp_fmt.fmtp, "%*s width=%d; height=%d; ", width, height );
1701 assert( ret == 2 );
1702 }
VLC Development tree mirror