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packetizer: hevc: don't double store poc prev msb/lsb
[vlc.git] / modules / stream_out / rtp.c
blob1cc89b8c9cf4c03ffc75e838fd1950491f9dcc18
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 stream", 0 )
190 add_shortcut( "rtp", "vod" )
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", DEFAULT_PTS_DELAY / 1000,
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
243 add_submodule ()
244 set_shortname( N_("RTSP VoD" ) )
245 set_description( N_("RTSP VoD server") )
246 set_category( CAT_SOUT )
247 set_subcategory( SUBCAT_SOUT_VOD )
248 set_capability( "vod server", 10 )
249 set_callbacks( OpenVoD, CloseVoD )
250 add_shortcut( "rtsp" )
251 add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
252 RTSP_TIMEOUT_LONGTEXT, true )
253 add_string( "sout-rtsp-user", "",
254 RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
255 add_password( "sout-rtsp-pwd", "",
256 RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
257
258 vlc_module_end ()
259
260 /*****************************************************************************
261 * Exported prototypes
262 *****************************************************************************/
263 static const char *const ppsz_sout_options[] = {
264 "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
265 "mux", "sap", "description", "url", "email",
266 "proto", "rtcp-mux", "caching",
267 #ifdef HAVE_SRTP
268 "key", "salt",
269 #endif
270 "mp4a-latm", NULL
271 };
272
273 static sout_stream_id_sys_t *Add( sout_stream_t *, const es_format_t * );
274 static void Del ( sout_stream_t *, sout_stream_id_sys_t * );
275 static int Send( sout_stream_t *, sout_stream_id_sys_t *,
276 block_t* );
277 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *, const es_format_t * );
278 static void MuxDel ( sout_stream_t *, sout_stream_id_sys_t * );
279 static int MuxSend( sout_stream_t *, sout_stream_id_sys_t *,
280 block_t* );
281
282 static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
283 static void* ThreadSend( void * );
284 static void *rtp_listen_thread( void * );
285
286 static void SDPHandleUrl( sout_stream_t *, const char * );
287
288 static int SapSetup( sout_stream_t *p_stream );
289 static int FileSetup( sout_stream_t *p_stream );
290 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
291
292 static int64_t rtp_init_ts( const vod_media_t *p_media,
293 const char *psz_vod_session );
294
295 struct sout_stream_sys_t
296 {
297 /* SDP */
298 char *psz_sdp;
299 vlc_mutex_t lock_sdp;
300
301 /* SDP to disk */
302 char *psz_sdp_file;
303
304 /* SDP via SAP */
305 bool b_export_sap;
306 session_descriptor_t *p_session;
307
308 /* SDP via HTTP */
309 httpd_host_t *p_httpd_host;
310 httpd_file_t *p_httpd_file;
311
312 /* RTSP */
313 rtsp_stream_t *rtsp;
314
315 /* RTSP NPT and timestamp computations */
316 mtime_t i_npt_zero; /* when NPT=0 packet is sent */
317 int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
318 int64_t i_pts_offset; /* matches actual PTS to prediction */
319 vlc_mutex_t lock_ts;
320
321 /* */
322 char *psz_destination;
323 uint16_t i_port;
324 uint16_t i_port_audio;
325 uint16_t i_port_video;
326 uint8_t proto;
327 bool rtcp_mux;
328 bool b_latm;
329
330 /* VoD */
331 vod_media_t *p_vod_media;
332 char *psz_vod_session;
333
334 /* in case we do TS/PS over rtp */
335 sout_mux_t *p_mux;
336 sout_access_out_t *p_grab;
337 block_t *packet;
338
339 /* */
340 vlc_mutex_t lock_es;
341 int i_es;
342 sout_stream_id_sys_t **es;
343 };
344
345 typedef struct rtp_sink_t
346 {
347 int rtp_fd;
348 rtcp_sender_t *rtcp;
349 } rtp_sink_t;
350
351 struct sout_stream_id_sys_t
352 {
353 sout_stream_t *p_stream;
354 /* rtp field */
355 /* For RFC 4175, seqnum is extended to 32-bits */
356 uint32_t i_sequence;
357 bool b_first_packet;
358 bool b_ts_init;
359 uint32_t i_ts_offset;
360 uint8_t ssrc[4];
361
362 /* for rtsp */
363 uint16_t i_seq_sent_next;
364
365 /* for sdp */
366 rtp_format_t rtp_fmt;
367 int i_port;
368
369 /* Packetizer specific fields */
370 int i_mtu;
371 #ifdef HAVE_SRTP
372 srtp_session_t *srtp;
373 #endif
374
375 /* Packets sinks */
376 vlc_thread_t thread;
377 vlc_mutex_t lock_sink;
378 int sinkc;
379 rtp_sink_t *sinkv;
380 rtsp_stream_id_t *rtsp_id;
381 struct {
382 int *fd;
383 vlc_thread_t thread;
384 } listen;
385
386 block_fifo_t *p_fifo;
387 int64_t i_caching;
388 };
389
390 /*****************************************************************************
391 * Open:
392 *****************************************************************************/
393 static int Open( vlc_object_t *p_this )
394 {
395 sout_stream_t *p_stream = (sout_stream_t*)p_this;
396 sout_stream_sys_t *p_sys = NULL;
397 char *psz;
398 bool b_rtsp = false;
399
400 config_ChainParse( p_stream, SOUT_CFG_PREFIX,
401 ppsz_sout_options, p_stream->p_cfg );
402
403 p_sys = malloc( sizeof( sout_stream_sys_t ) );
404 if( p_sys == NULL )
405 return VLC_ENOMEM;
406
407 p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
408
409 p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
410 p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
411 p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
412 p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
413
414 if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
415 {
416 msg_Err( p_stream, "audio and video RTP port must be distinct" );
417 free( p_sys->psz_destination );
418 free( p_sys );
419 return VLC_EGENERIC;
420 }
421
422 for( config_chain_t *p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
423 {
424 if( !strcmp( p_cfg->psz_name, "sdp" )
425 && ( p_cfg->psz_value != NULL )
426 && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
427 {
428 b_rtsp = true;
429 break;
430 }
431 }
432 if( !b_rtsp )
433 {
434 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
435 if( psz != NULL )
436 {
437 if( !strncasecmp( psz, "rtsp:", 5 ) )
438 b_rtsp = true;
439 free( psz );
440 }
441 }
442
443 /* Transport protocol */
444 p_sys->proto = IPPROTO_UDP;
445 psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");
446
447 if ((psz == NULL) || !strcasecmp (psz, "udp"))
448 (void)0; /* default */
449 else
450 if (!strcasecmp (psz, "dccp"))
451 {
452 p_sys->proto = IPPROTO_DCCP;
453 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
454 }
455 #if 0
456 else
457 if (!strcasecmp (psz, "sctp"))
458 {
459 p_sys->proto = IPPROTO_TCP;
460 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
461 }
462 #endif
463 #if 0
464 else
465 if (!strcasecmp (psz, "tcp"))
466 {
467 p_sys->proto = IPPROTO_TCP;
468 p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
469 }
470 #endif
471 else
472 if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
473 p_sys->proto = IPPROTO_UDPLITE;
474 else
475 msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
476 psz);
477 free (psz);
478 var_Create (p_this, "dccp-service", VLC_VAR_STRING);
479
480 p_sys->p_vod_media = NULL;
481 p_sys->psz_vod_session = NULL;
482
483 if (! strcmp(p_stream->psz_name, "vod"))
484 {
485 /* The VLM stops all instances before deleting a media, so this
486 * reference will remain valid during the lifetime of the rtp
487 * stream output. */
488 p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
489
490 if (p_sys->p_vod_media != NULL)
491 {
492 p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
493 if (p_sys->psz_vod_session == NULL)
494 {
495 msg_Err(p_stream, "missing VoD session");
496 free(p_sys);
497 return VLC_EGENERIC;
498 }
499
500 const char *mux = vod_get_mux(p_sys->p_vod_media);
501 var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
502 }
503 }
504
505 if( p_sys->psz_destination == NULL && !b_rtsp
506 && p_sys->p_vod_media == NULL )
507 {
508 msg_Err( p_stream, "missing destination and not in RTSP mode" );
509 free( p_sys );
510 return VLC_EGENERIC;
511 }
512
513 int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
514 if( i_ttl != -1 )
515 {
516 var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
517 var_SetInteger( p_stream, "ttl", i_ttl );
518 }
519
520 p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
521
522 /* NPT=0 time will be determined when we packetize the first packet
523 * (of any ES). But we want to be able to report rtptime in RTSP
524 * without waiting (and already did in the VoD case). So until then,
525 * we use an arbitrary reference PTS for timestamp computations, and
526 * then actual PTS will catch up using offsets. */
527 p_sys->i_npt_zero = VLC_TS_INVALID;
528 p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
529 p_sys->psz_vod_session);
530 p_sys->i_es = 0;
531 p_sys->es = NULL;
532 p_sys->rtsp = NULL;
533 p_sys->psz_sdp = NULL;
534
535 p_sys->b_export_sap = false;
536 p_sys->p_session = NULL;
537 p_sys->psz_sdp_file = NULL;
538
539 p_sys->p_httpd_host = NULL;
540 p_sys->p_httpd_file = NULL;
541
542 p_stream->p_sys = p_sys;
543
544 vlc_mutex_init( &p_sys->lock_sdp );
545 vlc_mutex_init( &p_sys->lock_ts );
546 vlc_mutex_init( &p_sys->lock_es );
547
548 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
549 if( psz != NULL )
550 {
551 /* Check muxer type */
552 if( strncasecmp( psz, "ps", 2 )
553 && strncasecmp( psz, "mpeg1", 5 )
554 && strncasecmp( psz, "ts", 2 ) )
555 {
556 msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
557 free( psz );
558 vlc_mutex_destroy( &p_sys->lock_sdp );
559 vlc_mutex_destroy( &p_sys->lock_ts );
560 vlc_mutex_destroy( &p_sys->lock_es );
561 free( p_sys->psz_vod_session );
562 free( p_sys->psz_destination );
563 free( p_sys );
564 return VLC_EGENERIC;
565 }
566
567 p_sys->p_grab = GrabberCreate( p_stream );
568 p_sys->p_mux = sout_MuxNew( p_stream->p_sout, psz, p_sys->p_grab );
569 free( psz );
570
571 if( p_sys->p_mux == NULL )
572 {
573 msg_Err( p_stream, "cannot create muxer" );
574 sout_AccessOutDelete( p_sys->p_grab );
575 vlc_mutex_destroy( &p_sys->lock_sdp );
576 vlc_mutex_destroy( &p_sys->lock_ts );
577 vlc_mutex_destroy( &p_sys->lock_es );
578 free( p_sys->psz_vod_session );
579 free( p_sys->psz_destination );
580 free( p_sys );
581 return VLC_EGENERIC;
582 }
583
584 p_sys->packet = NULL;
585
586 p_stream->pf_add = MuxAdd;
587 p_stream->pf_del = MuxDel;
588 p_stream->pf_send = MuxSend;
589 }
590 else
591 {
592 p_sys->p_mux = NULL;
593 p_sys->p_grab = NULL;
594
595 p_stream->pf_add = Add;
596 p_stream->pf_del = Del;
597 p_stream->pf_send = Send;
598 }
599 p_stream->pace_nocontrol = true;
600
601 if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
602 SDPHandleUrl( p_stream, "sap" );
603
604 psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
605 if( psz != NULL )
606 {
607 config_chain_t *p_cfg;
608
609 SDPHandleUrl( p_stream, psz );
610
611 for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
612 {
613 if( !strcmp( p_cfg->psz_name, "sdp" ) )
614 {
615 if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
616 continue;
617
618 /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
619 if( !strcmp( p_cfg->psz_value, psz ) )
620 continue;
621
622 SDPHandleUrl( p_stream, p_cfg->psz_value );
623 }
624 }
625 free( psz );
626 }
627
628 if( p_sys->p_mux != NULL )
629 {
630 sout_stream_id_sys_t *id = Add( p_stream, NULL );
631 if( id == NULL )
632 {
633 Close( p_this );
634 return VLC_EGENERIC;
635 }
636 }
637
638 return VLC_SUCCESS;
639 }
640
641 /*****************************************************************************
642 * Close:
643 *****************************************************************************/
644 static void Close( vlc_object_t * p_this )
645 {
646 sout_stream_t *p_stream = (sout_stream_t*)p_this;
647 sout_stream_sys_t *p_sys = p_stream->p_sys;
648
649 if( p_sys->p_mux )
650 {
651 assert( p_sys->i_es <= 1 );
652
653 sout_MuxDelete( p_sys->p_mux );
654 if ( p_sys->i_es > 0 )
655 Del( p_stream, p_sys->es[0] );
656 sout_AccessOutDelete( p_sys->p_grab );
657
658 if( p_sys->packet )
659 {
660 block_Release( p_sys->packet );
661 }
662 }
663
664 if( p_sys->rtsp != NULL )
665 RtspUnsetup( p_sys->rtsp );
666
667 vlc_mutex_destroy( &p_sys->lock_sdp );
668 vlc_mutex_destroy( &p_sys->lock_ts );
669 vlc_mutex_destroy( &p_sys->lock_es );
670
671 if( p_sys->p_httpd_file )
672 httpd_FileDelete( p_sys->p_httpd_file );
673
674 if( p_sys->p_httpd_host )
675 httpd_HostDelete( p_sys->p_httpd_host );
676
677 free( p_sys->psz_sdp );
678
679 if( p_sys->psz_sdp_file != NULL )
680 {
681 unlink( p_sys->psz_sdp_file );
682 free( p_sys->psz_sdp_file );
683 }
684 free( p_sys->psz_vod_session );
685 free( p_sys->psz_destination );
686 free( p_sys );
687 }
688
689 /*****************************************************************************
690 * SDPHandleUrl:
691 *****************************************************************************/
692 static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
693 {
694 sout_stream_sys_t *p_sys = p_stream->p_sys;
695 vlc_url_t url;
696
697 vlc_UrlParse( &url, psz_url );
698 if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
699 {
700 if( p_sys->p_httpd_file )
701 {
702 msg_Err( p_stream, "you can use sdp=http:// only once" );
703 goto out;
704 }
705
706 if( HttpSetup( p_stream, &url ) )
707 {
708 msg_Err( p_stream, "cannot export SDP as HTTP" );
709 }
710 }
711 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
712 {
713 if( p_sys->rtsp != NULL )
714 {
715 msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
716 goto out;
717 }
718
719 if( url.psz_host != NULL && *url.psz_host )
720 {
721 msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
722 "multiple-host configurations, use at your own risks.",
723 url.psz_host );
724 msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
725 "command line instead." );
726
727 var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
728 var_SetString( p_stream, "rtsp-host", url.psz_host );
729 }
730 if( url.i_port != 0 )
731 {
732 /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
733 "the command line instead.", url.i_port ); */
734
735 var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
736 var_SetInteger( p_stream, "rtsp-port", url.i_port );
737 }
738
739 p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
740 if( p_sys->rtsp == NULL )
741 msg_Err( p_stream, "cannot export SDP as RTSP" );
742 }
743 else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
744 ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
745 {
746 p_sys->b_export_sap = true;
747 SapSetup( p_stream );
748 }
749 else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
750 {
751 if( p_sys->psz_sdp_file != NULL )
752 {
753 msg_Err( p_stream, "you can use sdp=file:// only once" );
754 goto out;
755 }
756 p_sys->psz_sdp_file = vlc_uri2path( psz_url );
757 if( p_sys->psz_sdp_file == NULL )
758 goto out;
759 FileSetup( p_stream );
760 }
761 else
762 {
763 msg_Warn( p_stream, "unknown protocol for SDP (%s)",
764 url.psz_protocol );
765 }
766
767 out:
768 vlc_UrlClean( &url );
769 }
770
771 /*****************************************************************************
772 * SDPGenerate
773 *****************************************************************************/
774 /*static*/
775 char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
776 {
777 sout_stream_sys_t *p_sys = p_stream->p_sys;
778 struct vlc_memstream sdp;
779 struct sockaddr_storage dst;
780 char *psz_sdp = NULL;
781 socklen_t dstlen;
782 int i;
783 /*
784 * When we have a fixed destination (typically when we do multicast),
785 * we need to put the actual port numbers in the SDP.
786 * When there is no fixed destination, we only support RTSP unicast
787 * on-demand setup, so we should rather let the clients decide which ports
788 * to use.
789 * When there is both a fixed destination and RTSP unicast, we need to
790 * put port numbers used by the fixed destination, otherwise the SDP would
791 * become totally incorrect for multicast use. It should be noted that
792 * port numbers from SDP with RTSP are only "recommendation" from the
793 * server to the clients (per RFC2326), so only broken clients will fail
794 * to handle this properly. There is no solution but to use two differents
795 * output chain with two different RTSP URLs if you need to handle this
796 * scenario.
797 */
798 int inclport;
799
800 vlc_mutex_lock( &p_sys->lock_es );
801 if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
802 goto out; /* hmm... */
803
804 if( p_sys->psz_destination != NULL )
805 {
806 inclport = 1;
807
808 /* Oh boy, this is really ugly! */
809 dstlen = sizeof( dst );
810 if( p_sys->es[0]->listen.fd != NULL )
811 getsockname( p_sys->es[0]->listen.fd[0],
812 (struct sockaddr *)&dst, &dstlen );
813 else
814 getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
815 (struct sockaddr *)&dst, &dstlen );
816 }
817 else
818 {
819 inclport = 0;
820
821 /* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
822 bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
823 && rtsp_url[7] == '[';
824
825 /* Dummy destination address for RTSP */
826 dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
827 : sizeof( struct sockaddr_in );
828 memset (&dst, 0, dstlen);
829 dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
830 #ifdef HAVE_SA_LEN
831 dst.ss_len = dstlen;
832 #endif
833 }
834
835 if( vlc_sdp_Start( &sdp, VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
836 NULL, 0, (struct sockaddr *)&dst, dstlen ) )
837 goto out;
838
839 /* TODO: a=source-filter */
840 if( p_sys->rtcp_mux )
841 sdp_AddAttribute( &sdp, "rtcp-mux", NULL );
842
843 if( rtsp_url != NULL )
844 sdp_AddAttribute ( &sdp, "control", "%s", rtsp_url );
845
846 const char *proto = "RTP/AVP"; /* protocol */
847 if( rtsp_url == NULL )
848 {
849 switch( p_sys->proto )
850 {
851 case IPPROTO_UDP:
852 break;
853 case IPPROTO_TCP:
854 proto = "TCP/RTP/AVP";
855 break;
856 case IPPROTO_DCCP:
857 proto = "DCCP/RTP/AVP";
858 break;
859 case IPPROTO_UDPLITE:
860 return psz_sdp;
861 }
862 }
863
864 for( i = 0; i < p_sys->i_es; i++ )
865 {
866 sout_stream_id_sys_t *id = p_sys->es[i];
867 rtp_format_t *rtp_fmt = &id->rtp_fmt;
868 const char *mime_major; /* major MIME type */
869
870 switch( rtp_fmt->cat )
871 {
872 case VIDEO_ES:
873 mime_major = "video";
874 break;
875 case AUDIO_ES:
876 mime_major = "audio";
877 break;
878 case SPU_ES:
879 mime_major = "text";
880 break;
881 default:
882 continue;
883 }
884
885 sdp_AddMedia( &sdp, mime_major, proto, inclport * id->i_port,
886 rtp_fmt->payload_type, false, rtp_fmt->bitrate,
887 rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
888 rtp_fmt->fmtp);
889
890 /* cf RFC4566 §5.14 */
891 if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
892 sdp_AddAttribute( &sdp, "rtcp", "%u", id->i_port + 1 );
893
894 if( rtsp_url != NULL )
895 {
896 char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
897 if( track_url != NULL )
898 {
899 sdp_AddAttribute( &sdp, "control", "%s", track_url );
900 free( track_url );
901 }
902 }
903 else
904 {
905 if( id->listen.fd != NULL )
906 sdp_AddAttribute( &sdp, "setup", "passive" );
907 if( p_sys->proto == IPPROTO_DCCP )
908 sdp_AddAttribute( &sdp, "dccp-service-code", "SC:RTP%c",
909 toupper( (unsigned char)mime_major[0] ) );
910 }
911 }
912
913 if( vlc_memstream_close( &sdp ) == 0 )
914 psz_sdp = sdp.ptr;
915 out:
916 vlc_mutex_unlock( &p_sys->lock_es );
917 return psz_sdp;
918 }
919
920 /*****************************************************************************
921 * RTP mux
922 *****************************************************************************/
923
924 /**
925 * Shrink the MTU down to a fixed packetization time (for audio).
926 */
927 static void
928 rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes)
929 {
930 /* Samples per second */
931 size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
932 bytes *= id->rtp_fmt.channels;
933 spl *= bytes;
934
935 if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
936 id->i_mtu = 12 + spl;
937 else /* MTU is too small for ptime, align to a sample boundary */
938 id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
939 }
940
941 uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
942 {
943 /* This is an overflow-proof way of doing:
944 * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
945 *
946 * NOTE: this plays nice with offsets because the (equivalent)
947 * calculations are linear. */
948 lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
949 return q.quot * (int64_t)i_clock_rate
950 + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
951 }
952
953 /** Add an ES as a new RTP stream */
954 static sout_stream_id_sys_t *Add( sout_stream_t *p_stream,
955 const es_format_t *p_fmt )
956 {
957 /* NOTE: As a special case, if we use a non-RTP
958 * mux (TS/PS), then p_fmt is NULL. */
959 sout_stream_sys_t *p_sys = p_stream->p_sys;
960 char *psz_sdp;
961
962 sout_stream_id_sys_t *id = malloc( sizeof( *id ) );
963 if( unlikely(id == NULL) )
964 return NULL;
965 id->p_stream = p_stream;
966
967 id->i_mtu = var_InheritInteger( p_stream, "mtu" );
968 if( id->i_mtu <= 12 + 16 )
969 id->i_mtu = 576 - 20 - 8; /* pessimistic */
970 msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
971
972 #ifdef HAVE_SRTP
973 id->srtp = NULL;
974 #endif
975 vlc_mutex_init( &id->lock_sink );
976 id->sinkc = 0;
977 id->sinkv = NULL;
978 id->rtsp_id = NULL;
979 id->p_fifo = NULL;
980 id->listen.fd = NULL;
981
982 id->b_first_packet = true;
983 id->i_caching =
984 (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
985
986 vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
987 vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
988
989 bool format = false;
990
991 if (p_sys->p_vod_media != NULL)
992 {
993 id->rtp_fmt.ptname = NULL;
994 uint32_t ssrc;
995 int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
996 p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
997 &ssrc, &id->i_seq_sent_next);
998 if (val == VLC_SUCCESS)
999 {
1000 memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
1001 /* This is ugly, but id->i_seq_sent_next needs to be
1002 * initialized inside vod_init_id() to avoid race
1003 * conditions. */
1004 id->i_sequence = id->i_seq_sent_next;
1005 }
1006 /* vod_init_id() may fail either because the ES wasn't found in
1007 * the VoD media, or because the RTSP session is gone. In the
1008 * former case, id->rtp_fmt was left untouched. */
1009 format = (id->rtp_fmt.ptname != NULL);
1010 }
1011
1012 if (!format)
1013 {
1014 id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
1015 char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
1016 if (p_fmt == NULL && psz == NULL)
1017 goto error;
1018 int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
1019 free( psz );
1020 if (val != VLC_SUCCESS)
1021 goto error;
1022 }
1023
1024 #ifdef HAVE_SRTP
1025 char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
1026 if (key)
1027 {
1028 vlc_gcrypt_init ();
1029 id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
1030 SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
1031 if (id->srtp == NULL)
1032 {
1033 free (key);
1034 goto error;
1035 }
1036
1037 char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
1038 int val = srtp_setkeystring (id->srtp, key, salt ? salt : "");
1039 free (salt);
1040 free (key);
1041 if (val)
1042 {
1043 msg_Err (p_stream, "bad SRTP key/salt combination (%s)",
1044 vlc_strerror_c(val));
1045 goto error;
1046 }
1047 id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
1048 }
1049 #endif
1050
1051 id->i_seq_sent_next = id->i_sequence;
1052
1053 int mcast_fd = -1;
1054 if( p_sys->psz_destination != NULL )
1055 {
1056 /* Choose the port */
1057 uint16_t i_port = 0;
1058 if( p_fmt == NULL )
1059 ;
1060 else
1061 if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
1062 i_port = p_sys->i_port_audio;
1063 else
1064 if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
1065 i_port = p_sys->i_port_video;
1066
1067 /* We do not need the ES lock (p_sys->lock_es) here, because
1068 * this is the only one thread that can *modify* the ES table.
1069 * The ES lock protects the other threads from our modifications
1070 * (TAB_APPEND, TAB_REMOVE). */
1071 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1072 if (i_port == p_sys->es[i]->i_port)
1073 i_port = 0; /* Port already in use! */
1074 for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
1075 {
1076 if (p == 0)
1077 {
1078 msg_Err (p_stream, "too many RTP elementary streams");
1079 goto error;
1080 }
1081 i_port = p;
1082 for (int i = 0; i_port && (i < p_sys->i_es); i++)
1083 if (p == p_sys->es[i]->i_port)
1084 i_port = 0;
1085 }
1086
1087 id->i_port = i_port;
1088
1089 int type = SOCK_STREAM;
1090
1091 switch( p_sys->proto )
1092 {
1093 #ifdef SOCK_DCCP
1094 case IPPROTO_DCCP:
1095 {
1096 const char *code;
1097 switch (id->rtp_fmt.cat)
1098 {
1099 case VIDEO_ES: code = "RTPV"; break;
1100 case AUDIO_ES: code = "RTPARTPV"; break;
1101 case SPU_ES: code = "RTPTRTPV"; break;
1102 default: code = "RTPORTPV"; break;
1103 }
1104 var_SetString (p_stream, "dccp-service", code);
1105 type = SOCK_DCCP;
1106 }
1107 #endif
1108 /* fall through */
1109 case IPPROTO_TCP:
1110 id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
1111 p_sys->psz_destination, i_port,
1112 type, p_sys->proto );
1113 if( id->listen.fd == NULL )
1114 {
1115 msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
1116 goto error;
1117 }
1118 if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
1119 VLC_THREAD_PRIORITY_LOW ) )
1120 {
1121 net_ListenClose( id->listen.fd );
1122 id->listen.fd = NULL;
1123 goto error;
1124 }
1125 break;
1126
1127 default:
1128 {
1129 int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
1130 i_port, -1, p_sys->proto );
1131 if( fd == -1 )
1132 {
1133 msg_Err( p_stream, "cannot create RTP socket" );
1134 goto error;
1135 }
1136 /* Ignore any unexpected incoming packet (including RTCP-RR
1137 * packets in case of rtcp-mux) */
1138 setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
1139 sizeof (int));
1140 rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
1141 /* FIXME: test if this is multicast */
1142 mcast_fd = fd;
1143 }
1144 }
1145 }
1146
1147 if( p_fmt != NULL )
1148 switch( p_fmt->i_codec )
1149 {
1150 case VLC_CODEC_MULAW:
1151 case VLC_CODEC_ALAW:
1152 case VLC_CODEC_U8:
1153 rtp_set_ptime (id, 20, 1);
1154 break;
1155 case VLC_CODEC_S16B:
1156 case VLC_CODEC_S16L:
1157 rtp_set_ptime (id, 20, 2);
1158 break;
1159 case VLC_CODEC_S24B:
1160 rtp_set_ptime (id, 20, 3);
1161 break;
1162 default:
1163 break;
1164 }
1165
1166 #if 0 /* No payload formats sets this at the moment */
1167 int cscov = -1;
1168 if( cscov != -1 )
1169 cscov += 8 /* UDP */ + 12 /* RTP */;
1170 if( id->sinkc > 0 )
1171 net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
1172 #endif
1173
1174 vlc_mutex_lock( &p_sys->lock_ts );
1175 id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
1176 vlc_mutex_unlock( &p_sys->lock_ts );
1177 if( id->b_ts_init )
1178 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1179 p_sys->i_pts_offset );
1180
1181 if( p_sys->rtsp != NULL )
1182 id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
1183 id->rtp_fmt.clock_rate, mcast_fd );
1184
1185 id->p_fifo = block_FifoNew();
1186 if( unlikely(id->p_fifo == NULL) )
1187 goto error;
1188 if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
1189 {
1190 block_FifoRelease( id->p_fifo );
1191 id->p_fifo = NULL;
1192 goto error;
1193 }
1194
1195 /* Update p_sys context */
1196 vlc_mutex_lock( &p_sys->lock_es );
1197 TAB_APPEND( p_sys->i_es, p_sys->es, id );
1198 vlc_mutex_unlock( &p_sys->lock_es );
1199
1200 psz_sdp = SDPGenerate( p_stream, NULL );
1201
1202 vlc_mutex_lock( &p_sys->lock_sdp );
1203 free( p_sys->psz_sdp );
1204 p_sys->psz_sdp = psz_sdp;
1205 vlc_mutex_unlock( &p_sys->lock_sdp );
1206
1207 msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
1208
1209 /* Update SDP (sap/file) */
1210 if( p_sys->b_export_sap ) SapSetup( p_stream );
1211 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1212
1213 return id;
1214
1215 error:
1216 Del( p_stream, id );
1217 return NULL;
1218 }
1219
1220 static void Del( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1221 {
1222 sout_stream_sys_t *p_sys = p_stream->p_sys;
1223
1224 vlc_mutex_lock( &p_sys->lock_es );
1225 TAB_REMOVE( p_sys->i_es, p_sys->es, id );
1226 vlc_mutex_unlock( &p_sys->lock_es );
1227
1228 if( likely(id->p_fifo != NULL) )
1229 {
1230 vlc_cancel( id->thread );
1231 vlc_join( id->thread, NULL );
1232 block_FifoRelease( id->p_fifo );
1233 }
1234
1235 free( id->rtp_fmt.fmtp );
1236
1237 if (p_sys->p_vod_media != NULL)
1238 vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
1239 if( id->rtsp_id )
1240 RtspDelId( p_sys->rtsp, id->rtsp_id );
1241 if( id->listen.fd != NULL )
1242 {
1243 vlc_cancel( id->listen.thread );
1244 vlc_join( id->listen.thread, NULL );
1245 net_ListenClose( id->listen.fd );
1246 }
1247 /* Delete remaining sinks (incoming connections or explicit
1248 * outgoing dst=) */
1249 while( id->sinkc > 0 )
1250 rtp_del_sink( id, id->sinkv[0].rtp_fd );
1251 #ifdef HAVE_SRTP
1252 if( id->srtp != NULL )
1253 srtp_destroy( id->srtp );
1254 #endif
1255
1256 vlc_mutex_destroy( &id->lock_sink );
1257
1258 /* Update SDP (sap/file) */
1259 if( p_sys->b_export_sap ) SapSetup( p_stream );
1260 if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
1261
1262 free( id );
1263 }
1264
1265 static int Send( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1266 block_t *p_buffer )
1267 {
1268 assert( p_stream->p_sys->p_mux == NULL );
1269 (void)p_stream;
1270
1271 while( p_buffer != NULL )
1272 {
1273 block_t *p_next = p_buffer->p_next;
1274 p_buffer->p_next = NULL;
1275
1276 /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
1277 * as the first packet of the stream */
1278 if (id->b_first_packet)
1279 {
1280 id->b_first_packet = false;
1281 if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
1282 !strcmp(id->rtp_fmt.ptname, "theora"))
1283 rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
1284 p_buffer->i_pts);
1285 }
1286
1287 if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
1288 break;
1289
1290 p_buffer = p_next;
1291 }
1292 return VLC_SUCCESS;
1293 }
1294
1295 /****************************************************************************
1296 * SAP:
1297 ****************************************************************************/
1298 static int SapSetup( sout_stream_t *p_stream )
1299 {
1300 sout_stream_sys_t *p_sys = p_stream->p_sys;
1301
1302 /* Remove the previous session */
1303 if( p_sys->p_session != NULL)
1304 {
1305 sout_AnnounceUnRegister( p_stream, p_sys->p_session);
1306 p_sys->p_session = NULL;
1307 }
1308
1309 if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
1310 p_sys->p_session = sout_AnnounceRegisterSDP( p_stream,
1311 p_sys->psz_sdp,
1312 p_sys->psz_destination );
1313
1314 return VLC_SUCCESS;
1315 }
1316
1317 /****************************************************************************
1318 * File:
1319 ****************************************************************************/
1320 static int FileSetup( sout_stream_t *p_stream )
1321 {
1322 sout_stream_sys_t *p_sys = p_stream->p_sys;
1323 FILE *f;
1324
1325 if( p_sys->psz_sdp == NULL )
1326 return VLC_EGENERIC; /* too early */
1327
1328 if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
1329 {
1330 msg_Err( p_stream, "cannot open file '%s' (%s)",
1331 p_sys->psz_sdp_file, vlc_strerror_c(errno) );
1332 return VLC_EGENERIC;
1333 }
1334
1335 fputs( p_sys->psz_sdp, f );
1336 fclose( f );
1337
1338 return VLC_SUCCESS;
1339 }
1340
1341 /****************************************************************************
1342 * HTTP:
1343 ****************************************************************************/
1344 static int HttpCallback( httpd_file_sys_t *p_args,
1345 httpd_file_t *, uint8_t *p_request,
1346 uint8_t **pp_data, int *pi_data );
1347
1348 static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
1349 {
1350 sout_stream_sys_t *p_sys = p_stream->p_sys;
1351
1352 p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
1353 if( p_sys->p_httpd_host )
1354 {
1355 p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
1356 url->psz_path ? url->psz_path : "/",
1357 "application/sdp",
1358 NULL, NULL,
1359 HttpCallback, (void*)p_sys );
1360 }
1361 if( p_sys->p_httpd_file == NULL )
1362 {
1363 return VLC_EGENERIC;
1364 }
1365 return VLC_SUCCESS;
1366 }
1367
1368 static int HttpCallback( httpd_file_sys_t *p_args,
1369 httpd_file_t *f, uint8_t *p_request,
1370 uint8_t **pp_data, int *pi_data )
1371 {
1372 VLC_UNUSED(f); VLC_UNUSED(p_request);
1373 sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
1374
1375 vlc_mutex_lock( &p_sys->lock_sdp );
1376 if( p_sys->psz_sdp && *p_sys->psz_sdp )
1377 {
1378 *pi_data = strlen( p_sys->psz_sdp );
1379 *pp_data = malloc( *pi_data );
1380 memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
1381 }
1382 else
1383 {
1384 *pp_data = NULL;
1385 *pi_data = 0;
1386 }
1387 vlc_mutex_unlock( &p_sys->lock_sdp );
1388
1389 return VLC_SUCCESS;
1390 }
1391
1392 /****************************************************************************
1393 * RTP send
1394 ****************************************************************************/
1395 static void* ThreadSend( void *data )
1396 {
1397 #ifdef _WIN32
1398 # define ENOBUFS WSAENOBUFS
1399 # define EAGAIN WSAEWOULDBLOCK
1400 # define EWOULDBLOCK WSAEWOULDBLOCK
1401 #endif
1402 sout_stream_id_sys_t *id = data;
1403 unsigned i_caching = id->i_caching;
1404
1405 for (;;)
1406 {
1407 block_t *out = block_FifoGet( id->p_fifo );
1408 block_cleanup_push (out);
1409
1410 #ifdef HAVE_SRTP
1411 if( id->srtp )
1412 { /* FIXME: this is awfully inefficient */
1413 size_t len = out->i_buffer;
1414 out = block_Realloc( out, 0, len + 10 );
1415 out->i_buffer = len;
1416
1417 int canc = vlc_savecancel ();
1418 int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
1419 vlc_restorecancel (canc);
1420 if( val )
1421 {
1422 msg_Dbg( id->p_stream, "SRTP sending error: %s",
1423 vlc_strerror_c(val) );
1424 block_Release( out );
1425 out = NULL;
1426 }
1427 else
1428 out->i_buffer = len;
1429 }
1430 if (out)
1431 mwait (out->i_dts + i_caching);
1432 vlc_cleanup_pop ();
1433 if (out == NULL)
1434 continue;
1435 #else
1436 mwait (out->i_dts + i_caching);
1437 vlc_cleanup_pop ();
1438 #endif
1439
1440 ssize_t len = out->i_buffer;
1441 int canc = vlc_savecancel ();
1442
1443 vlc_mutex_lock( &id->lock_sink );
1444 unsigned deadc = 0; /* How many dead sockets? */
1445 int deadv[id->sinkc ? id->sinkc : 1]; /* Dead sockets list */
1446
1447 for( int i = 0; i < id->sinkc; i++ )
1448 {
1449 #ifdef HAVE_SRTP
1450 if( !id->srtp ) /* FIXME: SRTCP support */
1451 #endif
1452 SendRTCP( id->sinkv[i].rtcp, out );
1453
1454 if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
1455 && net_errno != EAGAIN && net_errno != EWOULDBLOCK
1456 && net_errno != ENOBUFS && net_errno != ENOMEM )
1457 {
1458 int type;
1459 getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
1460 &type, &(socklen_t){ sizeof(type) });
1461 if( type == SOCK_DGRAM )
1462 /* ICMP soft error: ignore and retry */
1463 send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
1464 else
1465 /* Broken connection */
1466 deadv[deadc++] = id->sinkv[i].rtp_fd;
1467 }
1468 }
1469 id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
1470 vlc_mutex_unlock( &id->lock_sink );
1471 block_Release( out );
1472
1473 for( unsigned i = 0; i < deadc; i++ )
1474 {
1475 msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
1476 rtp_del_sink( id, deadv[i] );
1477 }
1478 vlc_restorecancel (canc);
1479 }
1480 return NULL;
1481 }
1482
1483
1484 /* This thread dequeues incoming connections (DCCP streaming) */
1485 static void *rtp_listen_thread( void *data )
1486 {
1487 sout_stream_id_sys_t *id = data;
1488
1489 assert( id->listen.fd != NULL );
1490
1491 for( ;; )
1492 {
1493 int fd = net_Accept( id->p_stream, id->listen.fd );
1494 if( fd == -1 )
1495 continue;
1496 int canc = vlc_savecancel( );
1497 rtp_add_sink( id, fd, true, NULL );
1498 vlc_restorecancel( canc );
1499 }
1500
1501 vlc_assert_unreachable();
1502 }
1503
1504
1505 int rtp_add_sink( sout_stream_id_sys_t *id, int fd, bool rtcp_mux, uint16_t *seq )
1506 {
1507 rtp_sink_t sink = { fd, NULL };
1508 sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
1509 rtcp_mux );
1510 if( sink.rtcp == NULL )
1511 msg_Err( id->p_stream, "RTCP failed!" );
1512
1513 vlc_mutex_lock( &id->lock_sink );
1514 TAB_APPEND(id->sinkc, id->sinkv, sink);
1515 if( seq != NULL )
1516 *seq = id->i_seq_sent_next;
1517 vlc_mutex_unlock( &id->lock_sink );
1518 return VLC_SUCCESS;
1519 }
1520
1521 void rtp_del_sink( sout_stream_id_sys_t *id, int fd )
1522 {
1523 rtp_sink_t sink = { fd, NULL };
1524
1525 /* NOTE: must be safe to use if fd is not included */
1526 vlc_mutex_lock( &id->lock_sink );
1527 for( int i = 0; i < id->sinkc; i++ )
1528 {
1529 if (id->sinkv[i].rtp_fd == fd)
1530 {
1531 sink = id->sinkv[i];
1532 TAB_ERASE(id->sinkc, id->sinkv, i);
1533 break;
1534 }
1535 }
1536 vlc_mutex_unlock( &id->lock_sink );
1537
1538 CloseRTCP( sink.rtcp );
1539 net_Close( sink.rtp_fd );
1540 }
1541
1542 uint16_t rtp_get_seq( sout_stream_id_sys_t *id )
1543 {
1544 /* This will return values for the next packet. */
1545 uint16_t seq;
1546
1547 vlc_mutex_lock( &id->lock_sink );
1548 seq = id->i_seq_sent_next;
1549 vlc_mutex_unlock( &id->lock_sink );
1550
1551 return seq;
1552 }
1553
1554 /* Return an arbitrary initial timestamp for RTP timestamp computations.
1555 * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
1556 * random (although we use the same reference for all the ES as a
1557 * feature). In the VoD case, this function is called independently
1558 * from several parts of the code, so we need to always return the same
1559 * value. */
1560 static int64_t rtp_init_ts( const vod_media_t *p_media,
1561 const char *psz_vod_session )
1562 {
1563 if (p_media == NULL || psz_vod_session == NULL)
1564 return mdate();
1565
1566 uint64_t i_ts_init;
1567 /* As per RFC 2326, session identifiers are at least 8 bytes long */
1568 strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
1569 i_ts_init ^= (uintptr_t)p_media;
1570 /* Limit the timestamp to 48 bits, this is enough and allows us
1571 * to stay away from overflows */
1572 i_ts_init &= 0xFFFFFFFFFFFF;
1573 return i_ts_init;
1574 }
1575
1576 /* Return a timestamp corresponding to packets being sent now, and that
1577 * can be passed to rtp_compute_ts() to get rtptime values for each ES.
1578 * Also return the NPT corresponding to this timestamp. If the stream
1579 * output is not started, the initial timestamp that will be used with
1580 * the first packets for NPT=0 is returned instead. */
1581 int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_sys_t *id,
1582 const vod_media_t *p_media, const char *psz_vod_session,
1583 int64_t *p_npt )
1584 {
1585 if (p_npt != NULL)
1586 *p_npt = 0;
1587
1588 if (id != NULL)
1589 p_stream = id->p_stream;
1590
1591 if (p_stream == NULL)
1592 return rtp_init_ts(p_media, psz_vod_session);
1593
1594 sout_stream_sys_t *p_sys = p_stream->p_sys;
1595 mtime_t i_npt_zero;
1596 vlc_mutex_lock( &p_sys->lock_ts );
1597 i_npt_zero = p_sys->i_npt_zero;
1598 vlc_mutex_unlock( &p_sys->lock_ts );
1599
1600 if( i_npt_zero == VLC_TS_INVALID )
1601 return p_sys->i_pts_zero;
1602
1603 mtime_t now = mdate();
1604 if( now < i_npt_zero )
1605 return p_sys->i_pts_zero;
1606
1607 int64_t npt = now - i_npt_zero;
1608 if (p_npt != NULL)
1609 *p_npt = npt;
1610
1611 return p_sys->i_pts_zero + npt;
1612 }
1613
1614 void rtp_packetize_common( sout_stream_id_sys_t *id, block_t *out,
1615 bool b_m_bit, int64_t i_pts )
1616 {
1617 if( !id->b_ts_init )
1618 {
1619 sout_stream_sys_t *p_sys = id->p_stream->p_sys;
1620 vlc_mutex_lock( &p_sys->lock_ts );
1621 if( p_sys->i_npt_zero == VLC_TS_INVALID )
1622 {
1623 /* This is the first packet of any ES. We initialize the
1624 * NPT=0 time reference, and the offset to match the
1625 * arbitrary PTS reference. */
1626 p_sys->i_npt_zero = i_pts + id->i_caching;
1627 p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
1628 }
1629 vlc_mutex_unlock( &p_sys->lock_ts );
1630
1631 /* And in any case this is the first packet of this ES, so we
1632 * initialize the offset for this ES. */
1633 id->i_ts_offset = rtp_compute_ts( id->rtp_fmt.clock_rate,
1634 p_sys->i_pts_offset );
1635 id->b_ts_init = true;
1636 }
1637
1638 uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, i_pts )
1639 + id->i_ts_offset;
1640
1641 out->p_buffer[0] = 0x80;
1642 out->p_buffer[1] = (b_m_bit?0x80:0x00)|id->rtp_fmt.payload_type;
1643 out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
1644 out->p_buffer[3] = ( id->i_sequence )&0xff;
1645 out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
1646 out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
1647 out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
1648 out->p_buffer[7] = ( i_timestamp )&0xff;
1649
1650 memcpy( out->p_buffer + 8, id->ssrc, 4 );
1651
1652 id->i_sequence++;
1653 }
1654
1655 uint16_t rtp_get_extended_sequence( sout_stream_id_sys_t *id )
1656 {
1657 return id->i_sequence >> 16;
1658 }
1659
1660 void rtp_packetize_send( sout_stream_id_sys_t *id, block_t *out )
1661 {
1662 block_FifoPut( id->p_fifo, out );
1663 }
1664
1665 /**
1666 * @return configured max RTP payload size (including payload type-specific
1667 * headers, excluding RTP and transport headers)
1668 */
1669 size_t rtp_mtu (const sout_stream_id_sys_t *id)
1670 {
1671 return id->i_mtu - 12;
1672 }
1673
1674 /*****************************************************************************
1675 * Non-RTP mux
1676 *****************************************************************************/
1677
1678 /** Add an ES to a non-RTP muxed stream */
1679 static sout_stream_id_sys_t *MuxAdd( sout_stream_t *p_stream,
1680 const es_format_t *p_fmt )
1681 {
1682 sout_input_t *p_input;
1683 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1684 assert( p_mux != NULL );
1685
1686 p_input = sout_MuxAddStream( p_mux, p_fmt );
1687 if( p_input == NULL )
1688 {
1689 msg_Err( p_stream, "cannot add this stream to the muxer" );
1690 return NULL;
1691 }
1692
1693 return (sout_stream_id_sys_t *)p_input;
1694 }
1695
1696
1697 static int MuxSend( sout_stream_t *p_stream, sout_stream_id_sys_t *id,
1698 block_t *p_buffer )
1699 {
1700 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1701 assert( p_mux != NULL );
1702
1703 return sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
1704 }
1705
1706
1707 /** Remove an ES from a non-RTP muxed stream */
1708 static void MuxDel( sout_stream_t *p_stream, sout_stream_id_sys_t *id )
1709 {
1710 sout_mux_t *p_mux = p_stream->p_sys->p_mux;
1711 assert( p_mux != NULL );
1712
1713 sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
1714 }
1715
1716
1717 static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
1718 const block_t *p_buffer )
1719 {
1720 sout_stream_sys_t *p_sys = p_stream->p_sys;
1721 sout_stream_id_sys_t *id = p_sys->es[0];
1722
1723 int64_t i_dts = p_buffer->i_dts;
1724
1725 uint8_t *p_data = p_buffer->p_buffer;
1726 size_t i_data = p_buffer->i_buffer;
1727 size_t i_max = id->i_mtu - 12;
1728 bool b_dis = (p_buffer->i_flags & BLOCK_FLAG_DISCONTINUITY);
1729
1730 size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
1731
1732 while( i_data > 0 )
1733 {
1734 size_t i_size;
1735
1736 /* output complete packet */
1737 if( p_sys->packet &&
1738 p_sys->packet->i_buffer + i_data > i_max )
1739 {
1740 rtp_packetize_send( id, p_sys->packet );
1741 p_sys->packet = NULL;
1742 }
1743
1744 if( p_sys->packet == NULL )
1745 {
1746 /* allocate a new packet */
1747 p_sys->packet = block_Alloc( id->i_mtu );
1748 /* m-bit is discontinuity for MPEG1/2 PS and TS, RFC2250 2.1 */
1749 rtp_packetize_common( id, p_sys->packet, b_dis, i_dts );
1750 p_sys->packet->i_buffer = 12;
1751 p_sys->packet->i_dts = i_dts;
1752 p_sys->packet->i_length = p_buffer->i_length / i_packet;
1753 i_dts += p_sys->packet->i_length;
1754 b_dis = false;
1755 }
1756
1757 i_size = __MIN( i_data,
1758 (unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
1759
1760 memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
1761 p_data, i_size );
1762
1763 p_sys->packet->i_buffer += i_size;
1764 p_data += i_size;
1765 i_data -= i_size;
1766 }
1767
1768 return VLC_SUCCESS;
1769 }
1770
1771
1772 static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
1773 block_t *p_buffer )
1774 {
1775 sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
1776
1777 while( p_buffer )
1778 {
1779 block_t *p_next;
1780
1781 AccessOutGrabberWriteBuffer( p_stream, p_buffer );
1782
1783 p_next = p_buffer->p_next;
1784 block_Release( p_buffer );
1785 p_buffer = p_next;
1786 }
1787
1788 return VLC_SUCCESS;
1789 }
1790
1791
1792 static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
1793 {
1794 sout_access_out_t *p_grab;
1795
1796 p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
1797 if( p_grab == NULL )
1798 return NULL;
1799
1800 p_grab->p_module = NULL;
1801 p_grab->psz_access = strdup( "grab" );
1802 p_grab->p_cfg = NULL;
1803 p_grab->psz_path = strdup( "" );
1804 p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
1805 p_grab->pf_seek = NULL;
1806 p_grab->pf_write = AccessOutGrabberWrite;
1807 return p_grab;
1808 }
1809
1810 void rtp_get_video_geometry( sout_stream_id_sys_t *id, int *width, int *height )
1811 {
1812 int ret = sscanf( id->rtp_fmt.fmtp, "%*s width=%d; height=%d; ", width, height );
1813 assert( ret == 2 );
1814 }
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