ARM: update struct offsets
[libav.git] / libavformat / rtsp.c
1 /*
2 * RTSP/SDP client
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/random_seed.h"
26 #include "avformat.h"
27
28 #include <sys/time.h>
29 #if HAVE_SYS_SELECT_H
30 #include <sys/select.h>
31 #endif
32 #include <strings.h>
33 #include "internal.h"
34 #include "network.h"
35 #include "os_support.h"
36 #include "http.h"
37 #include "rtsp.h"
38
39 #include "rtpdec.h"
40 #include "rdt.h"
41 #include "rtpdec_formats.h"
42
43 //#define DEBUG
44 //#define DEBUG_RTP_TCP
45
46 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
47 int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
48 #endif
49
50 /* Timeout values for socket select, in ms,
51 * and read_packet(), in seconds */
52 #define SELECT_TIMEOUT_MS 100
53 #define READ_PACKET_TIMEOUT_S 10
54 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
55 #define SDP_MAX_SIZE 16384
56
57 static void get_word_until_chars(char *buf, int buf_size,
58 const char *sep, const char **pp)
59 {
60 const char *p;
61 char *q;
62
63 p = *pp;
64 p += strspn(p, SPACE_CHARS);
65 q = buf;
66 while (!strchr(sep, *p) && *p != '\0') {
67 if ((q - buf) < buf_size - 1)
68 *q++ = *p;
69 p++;
70 }
71 if (buf_size > 0)
72 *q = '\0';
73 *pp = p;
74 }
75
76 static void get_word_sep(char *buf, int buf_size, const char *sep,
77 const char **pp)
78 {
79 if (**pp == '/') (*pp)++;
80 get_word_until_chars(buf, buf_size, sep, pp);
81 }
82
83 static void get_word(char *buf, int buf_size, const char **pp)
84 {
85 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
86 }
87
88 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
89 static int sdp_parse_rtpmap(AVFormatContext *s,
90 AVCodecContext *codec, RTSPStream *rtsp_st,
91 int payload_type, const char *p)
92 {
93 char buf[256];
94 int i;
95 AVCodec *c;
96 const char *c_name;
97
98 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
99 * see if we can handle this kind of payload.
100 * The space should normally not be there but some Real streams or
101 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
102 * have a trailing space. */
103 get_word_sep(buf, sizeof(buf), "/ ", &p);
104 if (payload_type >= RTP_PT_PRIVATE) {
105 RTPDynamicProtocolHandler *handler;
106 for (handler = RTPFirstDynamicPayloadHandler;
107 handler; handler = handler->next) {
108 if (!strcasecmp(buf, handler->enc_name) &&
109 codec->codec_type == handler->codec_type) {
110 codec->codec_id = handler->codec_id;
111 rtsp_st->dynamic_handler = handler;
112 if (handler->open)
113 rtsp_st->dynamic_protocol_context = handler->open();
114 break;
115 }
116 }
117 } else {
118 /* We are in a standard case
119 * (from http://www.iana.org/assignments/rtp-parameters). */
120 /* search into AVRtpPayloadTypes[] */
121 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
122 }
123
124 c = avcodec_find_decoder(codec->codec_id);
125 if (c && c->name)
126 c_name = c->name;
127 else
128 c_name = "(null)";
129
130 get_word_sep(buf, sizeof(buf), "/", &p);
131 i = atoi(buf);
132 switch (codec->codec_type) {
133 case AVMEDIA_TYPE_AUDIO:
134 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
135 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
136 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
137 if (i > 0) {
138 codec->sample_rate = i;
139 get_word_sep(buf, sizeof(buf), "/", &p);
140 i = atoi(buf);
141 if (i > 0)
142 codec->channels = i;
143 // TODO: there is a bug here; if it is a mono stream, and
144 // less than 22000Hz, faad upconverts to stereo and twice
145 // the frequency. No problem, but the sample rate is being
146 // set here by the sdp line. Patch on its way. (rdm)
147 }
148 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
149 codec->sample_rate);
150 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
151 codec->channels);
152 break;
153 case AVMEDIA_TYPE_VIDEO:
154 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
155 break;
156 default:
157 break;
158 }
159 return 0;
160 }
161
162 /* parse the attribute line from the fmtp a line of an sdp response. This
163 * is broken out as a function because it is used in rtp_h264.c, which is
164 * forthcoming. */
165 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
166 char *value, int value_size)
167 {
168 *p += strspn(*p, SPACE_CHARS);
169 if (**p) {
170 get_word_sep(attr, attr_size, "=", p);
171 if (**p == '=')
172 (*p)++;
173 get_word_sep(value, value_size, ";", p);
174 if (**p == ';')
175 (*p)++;
176 return 1;
177 }
178 return 0;
179 }
180
181 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
182 * and end time.
183 * Used for seeking in the rtp stream.
184 */
185 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
186 {
187 char buf[256];
188
189 p += strspn(p, SPACE_CHARS);
190 if (!av_stristart(p, "npt=", &p))
191 return;
192
193 *start = AV_NOPTS_VALUE;
194 *end = AV_NOPTS_VALUE;
195
196 get_word_sep(buf, sizeof(buf), "-", &p);
197 *start = parse_date(buf, 1);
198 if (*p == '-') {
199 p++;
200 get_word_sep(buf, sizeof(buf), "-", &p);
201 *end = parse_date(buf, 1);
202 }
203 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
204 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
205 }
206
207 typedef struct SDPParseState {
208 /* SDP only */
209 struct in_addr default_ip;
210 int default_ttl;
211 int skip_media; ///< set if an unknown m= line occurs
212 } SDPParseState;
213
214 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
215 int letter, const char *buf)
216 {
217 RTSPState *rt = s->priv_data;
218 char buf1[64], st_type[64];
219 const char *p;
220 enum AVMediaType codec_type;
221 int payload_type, i;
222 AVStream *st;
223 RTSPStream *rtsp_st;
224 struct in_addr sdp_ip;
225 int ttl;
226
227 dprintf(s, "sdp: %c='%s'\n", letter, buf);
228
229 p = buf;
230 if (s1->skip_media && letter != 'm')
231 return;
232 switch (letter) {
233 case 'c':
234 get_word(buf1, sizeof(buf1), &p);
235 if (strcmp(buf1, "IN") != 0)
236 return;
237 get_word(buf1, sizeof(buf1), &p);
238 if (strcmp(buf1, "IP4") != 0)
239 return;
240 get_word_sep(buf1, sizeof(buf1), "/", &p);
241 if (ff_inet_aton(buf1, &sdp_ip) == 0)
242 return;
243 ttl = 16;
244 if (*p == '/') {
245 p++;
246 get_word_sep(buf1, sizeof(buf1), "/", &p);
247 ttl = atoi(buf1);
248 }
249 if (s->nb_streams == 0) {
250 s1->default_ip = sdp_ip;
251 s1->default_ttl = ttl;
252 } else {
253 st = s->streams[s->nb_streams - 1];
254 rtsp_st = st->priv_data;
255 rtsp_st->sdp_ip = sdp_ip;
256 rtsp_st->sdp_ttl = ttl;
257 }
258 break;
259 case 's':
260 av_metadata_set2(&s->metadata, "title", p, 0);
261 break;
262 case 'i':
263 if (s->nb_streams == 0) {
264 av_metadata_set2(&s->metadata, "comment", p, 0);
265 break;
266 }
267 break;
268 case 'm':
269 /* new stream */
270 s1->skip_media = 0;
271 get_word(st_type, sizeof(st_type), &p);
272 if (!strcmp(st_type, "audio")) {
273 codec_type = AVMEDIA_TYPE_AUDIO;
274 } else if (!strcmp(st_type, "video")) {
275 codec_type = AVMEDIA_TYPE_VIDEO;
276 } else if (!strcmp(st_type, "application")) {
277 codec_type = AVMEDIA_TYPE_DATA;
278 } else {
279 s1->skip_media = 1;
280 return;
281 }
282 rtsp_st = av_mallocz(sizeof(RTSPStream));
283 if (!rtsp_st)
284 return;
285 rtsp_st->stream_index = -1;
286 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
287
288 rtsp_st->sdp_ip = s1->default_ip;
289 rtsp_st->sdp_ttl = s1->default_ttl;
290
291 get_word(buf1, sizeof(buf1), &p); /* port */
292 rtsp_st->sdp_port = atoi(buf1);
293
294 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
295
296 /* XXX: handle list of formats */
297 get_word(buf1, sizeof(buf1), &p); /* format list */
298 rtsp_st->sdp_payload_type = atoi(buf1);
299
300 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
301 /* no corresponding stream */
302 } else {
303 st = av_new_stream(s, 0);
304 if (!st)
305 return;
306 st->priv_data = rtsp_st;
307 rtsp_st->stream_index = st->index;
308 st->codec->codec_type = codec_type;
309 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
310 /* if standard payload type, we can find the codec right now */
311 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
312 }
313 }
314 /* put a default control url */
315 av_strlcpy(rtsp_st->control_url, rt->control_uri,
316 sizeof(rtsp_st->control_url));
317 break;
318 case 'a':
319 if (av_strstart(p, "control:", &p)) {
320 if (s->nb_streams == 0) {
321 if (!strncmp(p, "rtsp://", 7))
322 av_strlcpy(rt->control_uri, p,
323 sizeof(rt->control_uri));
324 } else {
325 char proto[32];
326 /* get the control url */
327 st = s->streams[s->nb_streams - 1];
328 rtsp_st = st->priv_data;
329
330 /* XXX: may need to add full url resolution */
331 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
332 NULL, NULL, 0, p);
333 if (proto[0] == '\0') {
334 /* relative control URL */
335 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
336 av_strlcat(rtsp_st->control_url, "/",
337 sizeof(rtsp_st->control_url));
338 av_strlcat(rtsp_st->control_url, p,
339 sizeof(rtsp_st->control_url));
340 } else
341 av_strlcpy(rtsp_st->control_url, p,
342 sizeof(rtsp_st->control_url));
343 }
344 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
345 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
346 get_word(buf1, sizeof(buf1), &p);
347 payload_type = atoi(buf1);
348 st = s->streams[s->nb_streams - 1];
349 rtsp_st = st->priv_data;
350 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
351 } else if (av_strstart(p, "fmtp:", &p) ||
352 av_strstart(p, "framesize:", &p)) {
353 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
354 // let dynamic protocol handlers have a stab at the line.
355 get_word(buf1, sizeof(buf1), &p);
356 payload_type = atoi(buf1);
357 for (i = 0; i < s->nb_streams; i++) {
358 st = s->streams[i];
359 rtsp_st = st->priv_data;
360 if (rtsp_st->sdp_payload_type == payload_type &&
361 rtsp_st->dynamic_handler &&
362 rtsp_st->dynamic_handler->parse_sdp_a_line)
363 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
364 rtsp_st->dynamic_protocol_context, buf);
365 }
366 } else if (av_strstart(p, "range:", &p)) {
367 int64_t start, end;
368
369 // this is so that seeking on a streamed file can work.
370 rtsp_parse_range_npt(p, &start, &end);
371 s->start_time = start;
372 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
373 s->duration = (end == AV_NOPTS_VALUE) ?
374 AV_NOPTS_VALUE : end - start;
375 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
376 if (atoi(p) == 1)
377 rt->transport = RTSP_TRANSPORT_RDT;
378 } else {
379 if (rt->server_type == RTSP_SERVER_WMS)
380 ff_wms_parse_sdp_a_line(s, p);
381 if (s->nb_streams > 0) {
382 if (rt->server_type == RTSP_SERVER_REAL)
383 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
384
385 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
386 if (rtsp_st->dynamic_handler &&
387 rtsp_st->dynamic_handler->parse_sdp_a_line)
388 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
389 s->nb_streams - 1,
390 rtsp_st->dynamic_protocol_context, buf);
391 }
392 }
393 break;
394 }
395 }
396
397 static int sdp_parse(AVFormatContext *s, const char *content)
398 {
399 const char *p;
400 int letter;
401 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
402 * contain long SDP lines containing complete ASF Headers (several
403 * kB) or arrays of MDPR (RM stream descriptor) headers plus
404 * "rulebooks" describing their properties. Therefore, the SDP line
405 * buffer is large.
406 *
407 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
408 * in rtpdec_xiph.c. */
409 char buf[16384], *q;
410 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
411
412 memset(s1, 0, sizeof(SDPParseState));
413 p = content;
414 for (;;) {
415 p += strspn(p, SPACE_CHARS);
416 letter = *p;
417 if (letter == '\0')
418 break;
419 p++;
420 if (*p != '=')
421 goto next_line;
422 p++;
423 /* get the content */
424 q = buf;
425 while (*p != '\n' && *p != '\r' && *p != '\0') {
426 if ((q - buf) < sizeof(buf) - 1)
427 *q++ = *p;
428 p++;
429 }
430 *q = '\0';
431 sdp_parse_line(s, s1, letter, buf);
432 next_line:
433 while (*p != '\n' && *p != '\0')
434 p++;
435 if (*p == '\n')
436 p++;
437 }
438 return 0;
439 }
440
441 /* close and free RTSP streams */
442 void ff_rtsp_close_streams(AVFormatContext *s)
443 {
444 RTSPState *rt = s->priv_data;
445 int i;
446 RTSPStream *rtsp_st;
447
448 for (i = 0; i < rt->nb_rtsp_streams; i++) {
449 rtsp_st = rt->rtsp_streams[i];
450 if (rtsp_st) {
451 if (rtsp_st->transport_priv) {
452 if (s->oformat) {
453 AVFormatContext *rtpctx = rtsp_st->transport_priv;
454 av_write_trailer(rtpctx);
455 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
456 uint8_t *ptr;
457 url_close_dyn_buf(rtpctx->pb, &ptr);
458 av_free(ptr);
459 } else {
460 url_fclose(rtpctx->pb);
461 }
462 av_metadata_free(&rtpctx->streams[0]->metadata);
463 av_metadata_free(&rtpctx->metadata);
464 av_free(rtpctx->streams[0]);
465 av_free(rtpctx);
466 } else if (rt->transport == RTSP_TRANSPORT_RDT)
467 ff_rdt_parse_close(rtsp_st->transport_priv);
468 else
469 rtp_parse_close(rtsp_st->transport_priv);
470 }
471 if (rtsp_st->rtp_handle)
472 url_close(rtsp_st->rtp_handle);
473 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
474 rtsp_st->dynamic_handler->close(
475 rtsp_st->dynamic_protocol_context);
476 }
477 }
478 av_free(rt->rtsp_streams);
479 if (rt->asf_ctx) {
480 av_close_input_stream (rt->asf_ctx);
481 rt->asf_ctx = NULL;
482 }
483 }
484
485 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
486 URLContext *handle)
487 {
488 RTSPState *rt = s->priv_data;
489 AVFormatContext *rtpctx;
490 int ret;
491 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
492
493 if (!rtp_format)
494 return NULL;
495
496 /* Allocate an AVFormatContext for each output stream */
497 rtpctx = avformat_alloc_context();
498 if (!rtpctx)
499 return NULL;
500
501 rtpctx->oformat = rtp_format;
502 if (!av_new_stream(rtpctx, 0)) {
503 av_free(rtpctx);
504 return NULL;
505 }
506 /* Copy the max delay setting; the rtp muxer reads this. */
507 rtpctx->max_delay = s->max_delay;
508 /* Copy other stream parameters. */
509 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
510
511 /* Set the synchronized start time. */
512 rtpctx->start_time_realtime = rt->start_time;
513
514 /* Remove the local codec, link to the original codec
515 * context instead, to give the rtp muxer access to
516 * codec parameters. */
517 av_free(rtpctx->streams[0]->codec);
518 rtpctx->streams[0]->codec = st->codec;
519
520 if (handle) {
521 url_fdopen(&rtpctx->pb, handle);
522 } else
523 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
524 ret = av_write_header(rtpctx);
525
526 if (ret) {
527 if (handle) {
528 url_fclose(rtpctx->pb);
529 } else {
530 uint8_t *ptr;
531 url_close_dyn_buf(rtpctx->pb, &ptr);
532 av_free(ptr);
533 }
534 av_free(rtpctx->streams[0]);
535 av_free(rtpctx);
536 return NULL;
537 }
538
539 /* Copy the RTP AVStream timebase back to the original AVStream */
540 st->time_base = rtpctx->streams[0]->time_base;
541 return rtpctx;
542 }
543
544 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
545 {
546 RTSPState *rt = s->priv_data;
547 AVStream *st = NULL;
548
549 /* open the RTP context */
550 if (rtsp_st->stream_index >= 0)
551 st = s->streams[rtsp_st->stream_index];
552 if (!st)
553 s->ctx_flags |= AVFMTCTX_NOHEADER;
554
555 if (s->oformat) {
556 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
557 /* Ownership of rtp_handle is passed to the rtp mux context */
558 rtsp_st->rtp_handle = NULL;
559 } else if (rt->transport == RTSP_TRANSPORT_RDT)
560 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
561 rtsp_st->dynamic_protocol_context,
562 rtsp_st->dynamic_handler);
563 else
564 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
565 rtsp_st->sdp_payload_type);
566
567 if (!rtsp_st->transport_priv) {
568 return AVERROR(ENOMEM);
569 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
570 if (rtsp_st->dynamic_handler) {
571 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
572 rtsp_st->dynamic_protocol_context,
573 rtsp_st->dynamic_handler);
574 }
575 }
576
577 return 0;
578 }
579
580 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
581 static int rtsp_probe(AVProbeData *p)
582 {
583 if (av_strstart(p->filename, "rtsp:", NULL))
584 return AVPROBE_SCORE_MAX;
585 return 0;
586 }
587
588 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
589 {
590 const char *p;
591 int v;
592
593 p = *pp;
594 p += strspn(p, SPACE_CHARS);
595 v = strtol(p, (char **)&p, 10);
596 if (*p == '-') {
597 p++;
598 *min_ptr = v;
599 v = strtol(p, (char **)&p, 10);
600 *max_ptr = v;
601 } else {
602 *min_ptr = v;
603 *max_ptr = v;
604 }
605 *pp = p;
606 }
607
608 /* XXX: only one transport specification is parsed */
609 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
610 {
611 char transport_protocol[16];
612 char profile[16];
613 char lower_transport[16];
614 char parameter[16];
615 RTSPTransportField *th;
616 char buf[256];
617
618 reply->nb_transports = 0;
619
620 for (;;) {
621 p += strspn(p, SPACE_CHARS);
622 if (*p == '\0')
623 break;
624
625 th = &reply->transports[reply->nb_transports];
626
627 get_word_sep(transport_protocol, sizeof(transport_protocol),
628 "/", &p);
629 if (!strcasecmp (transport_protocol, "rtp")) {
630 get_word_sep(profile, sizeof(profile), "/;,", &p);
631 lower_transport[0] = '\0';
632 /* rtp/avp/<protocol> */
633 if (*p == '/') {
634 get_word_sep(lower_transport, sizeof(lower_transport),
635 ";,", &p);
636 }
637 th->transport = RTSP_TRANSPORT_RTP;
638 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
639 !strcasecmp (transport_protocol, "x-real-rdt")) {
640 /* x-pn-tng/<protocol> */
641 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
642 profile[0] = '\0';
643 th->transport = RTSP_TRANSPORT_RDT;
644 }
645 if (!strcasecmp(lower_transport, "TCP"))
646 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
647 else
648 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
649
650 if (*p == ';')
651 p++;
652 /* get each parameter */
653 while (*p != '\0' && *p != ',') {
654 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
655 if (!strcmp(parameter, "port")) {
656 if (*p == '=') {
657 p++;
658 rtsp_parse_range(&th->port_min, &th->port_max, &p);
659 }
660 } else if (!strcmp(parameter, "client_port")) {
661 if (*p == '=') {
662 p++;
663 rtsp_parse_range(&th->client_port_min,
664 &th->client_port_max, &p);
665 }
666 } else if (!strcmp(parameter, "server_port")) {
667 if (*p == '=') {
668 p++;
669 rtsp_parse_range(&th->server_port_min,
670 &th->server_port_max, &p);
671 }
672 } else if (!strcmp(parameter, "interleaved")) {
673 if (*p == '=') {
674 p++;
675 rtsp_parse_range(&th->interleaved_min,
676 &th->interleaved_max, &p);
677 }
678 } else if (!strcmp(parameter, "multicast")) {
679 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
680 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
681 } else if (!strcmp(parameter, "ttl")) {
682 if (*p == '=') {
683 p++;
684 th->ttl = strtol(p, (char **)&p, 10);
685 }
686 } else if (!strcmp(parameter, "destination")) {
687 struct in_addr ipaddr;
688
689 if (*p == '=') {
690 p++;
691 get_word_sep(buf, sizeof(buf), ";,", &p);
692 if (ff_inet_aton(buf, &ipaddr))
693 th->destination = ntohl(ipaddr.s_addr);
694 }
695 }
696 while (*p != ';' && *p != '\0' && *p != ',')
697 p++;
698 if (*p == ';')
699 p++;
700 }
701 if (*p == ',')
702 p++;
703
704 reply->nb_transports++;
705 }
706 }
707
708 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
709 HTTPAuthState *auth_state)
710 {
711 const char *p;
712
713 /* NOTE: we do case independent match for broken servers */
714 p = buf;
715 if (av_stristart(p, "Session:", &p)) {
716 int t;
717 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
718 if (av_stristart(p, ";timeout=", &p) &&
719 (t = strtol(p, NULL, 10)) > 0) {
720 reply->timeout = t;
721 }
722 } else if (av_stristart(p, "Content-Length:", &p)) {
723 reply->content_length = strtol(p, NULL, 10);
724 } else if (av_stristart(p, "Transport:", &p)) {
725 rtsp_parse_transport(reply, p);
726 } else if (av_stristart(p, "CSeq:", &p)) {
727 reply->seq = strtol(p, NULL, 10);
728 } else if (av_stristart(p, "Range:", &p)) {
729 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
730 } else if (av_stristart(p, "RealChallenge1:", &p)) {
731 p += strspn(p, SPACE_CHARS);
732 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
733 } else if (av_stristart(p, "Server:", &p)) {
734 p += strspn(p, SPACE_CHARS);
735 av_strlcpy(reply->server, p, sizeof(reply->server));
736 } else if (av_stristart(p, "Notice:", &p) ||
737 av_stristart(p, "X-Notice:", &p)) {
738 reply->notice = strtol(p, NULL, 10);
739 } else if (av_stristart(p, "Location:", &p)) {
740 p += strspn(p, SPACE_CHARS);
741 av_strlcpy(reply->location, p , sizeof(reply->location));
742 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
743 p += strspn(p, SPACE_CHARS);
744 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
745 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
746 p += strspn(p, SPACE_CHARS);
747 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
748 }
749 }
750
751 /* skip a RTP/TCP interleaved packet */
752 void ff_rtsp_skip_packet(AVFormatContext *s)
753 {
754 RTSPState *rt = s->priv_data;
755 int ret, len, len1;
756 uint8_t buf[1024];
757
758 ret = url_read_complete(rt->rtsp_hd, buf, 3);
759 if (ret != 3)
760 return;
761 len = AV_RB16(buf + 1);
762
763 dprintf(s, "skipping RTP packet len=%d\n", len);
764
765 /* skip payload */
766 while (len > 0) {
767 len1 = len;
768 if (len1 > sizeof(buf))
769 len1 = sizeof(buf);
770 ret = url_read_complete(rt->rtsp_hd, buf, len1);
771 if (ret != len1)
772 return;
773 len -= len1;
774 }
775 }
776
777 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
778 unsigned char **content_ptr,
779 int return_on_interleaved_data)
780 {
781 RTSPState *rt = s->priv_data;
782 char buf[4096], buf1[1024], *q;
783 unsigned char ch;
784 const char *p;
785 int ret, content_length, line_count = 0;
786 unsigned char *content = NULL;
787
788 memset(reply, 0, sizeof(*reply));
789
790 /* parse reply (XXX: use buffers) */
791 rt->last_reply[0] = '\0';
792 for (;;) {
793 q = buf;
794 for (;;) {
795 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
796 #ifdef DEBUG_RTP_TCP
797 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
798 #endif
799 if (ret != 1)
800 return AVERROR_EOF;
801 if (ch == '\n')
802 break;
803 if (ch == '$') {
804 /* XXX: only parse it if first char on line ? */
805 if (return_on_interleaved_data) {
806 return 1;
807 } else
808 ff_rtsp_skip_packet(s);
809 } else if (ch != '\r') {
810 if ((q - buf) < sizeof(buf) - 1)
811 *q++ = ch;
812 }
813 }
814 *q = '\0';
815
816 dprintf(s, "line='%s'\n", buf);
817
818 /* test if last line */
819 if (buf[0] == '\0')
820 break;
821 p = buf;
822 if (line_count == 0) {
823 /* get reply code */
824 get_word(buf1, sizeof(buf1), &p);
825 get_word(buf1, sizeof(buf1), &p);
826 reply->status_code = atoi(buf1);
827 av_strlcpy(reply->reason, p, sizeof(reply->reason));
828 } else {
829 ff_rtsp_parse_line(reply, p, &rt->auth_state);
830 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
831 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
832 }
833 line_count++;
834 }
835
836 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
837 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
838
839 content_length = reply->content_length;
840 if (content_length > 0) {
841 /* leave some room for a trailing '\0' (useful for simple parsing) */
842 content = av_malloc(content_length + 1);
843 (void)url_read_complete(rt->rtsp_hd, content, content_length);
844 content[content_length] = '\0';
845 }
846 if (content_ptr)
847 *content_ptr = content;
848 else
849 av_free(content);
850
851 if (rt->seq != reply->seq) {
852 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
853 rt->seq, reply->seq);
854 }
855
856 /* EOS */
857 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
858 reply->notice == 2104 /* Start-of-Stream Reached */ ||
859 reply->notice == 2306 /* Continuous Feed Terminated */) {
860 rt->state = RTSP_STATE_IDLE;
861 } else if (reply->notice >= 4400 && reply->notice < 5500) {
862 return AVERROR(EIO); /* data or server error */
863 } else if (reply->notice == 2401 /* Ticket Expired */ ||
864 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
865 return AVERROR(EPERM);
866
867 return 0;
868 }
869
870 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
871 const char *method, const char *url,
872 const char *headers,
873 const unsigned char *send_content,
874 int send_content_length)
875 {
876 RTSPState *rt = s->priv_data;
877 char buf[4096], *out_buf;
878 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
879
880 /* Add in RTSP headers */
881 out_buf = buf;
882 rt->seq++;
883 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
884 if (headers)
885 av_strlcat(buf, headers, sizeof(buf));
886 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
887 if (rt->session_id[0] != '\0' && (!headers ||
888 !strstr(headers, "\nIf-Match:"))) {
889 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
890 }
891 if (rt->auth[0]) {
892 char *str = ff_http_auth_create_response(&rt->auth_state,
893 rt->auth, url, method);
894 if (str)
895 av_strlcat(buf, str, sizeof(buf));
896 av_free(str);
897 }
898 if (send_content_length > 0 && send_content)
899 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
900 av_strlcat(buf, "\r\n", sizeof(buf));
901
902 /* base64 encode rtsp if tunneling */
903 if (rt->control_transport == RTSP_MODE_TUNNEL) {
904 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
905 out_buf = base64buf;
906 }
907
908 dprintf(s, "Sending:\n%s--\n", buf);
909
910 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
911 if (send_content_length > 0 && send_content) {
912 if (rt->control_transport == RTSP_MODE_TUNNEL) {
913 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
914 "with content data not supported\n");
915 return AVERROR_PATCHWELCOME;
916 }
917 url_write(rt->rtsp_hd_out, send_content, send_content_length);
918 }
919 rt->last_cmd_time = av_gettime();
920
921 return 0;
922 }
923
924 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
925 const char *url, const char *headers)
926 {
927 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
928 }
929
930 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
931 const char *headers, RTSPMessageHeader *reply,
932 unsigned char **content_ptr)
933 {
934 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
935 content_ptr, NULL, 0);
936 }
937
938 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
939 const char *method, const char *url,
940 const char *header,
941 RTSPMessageHeader *reply,
942 unsigned char **content_ptr,
943 const unsigned char *send_content,
944 int send_content_length)
945 {
946 RTSPState *rt = s->priv_data;
947 HTTPAuthType cur_auth_type;
948 int ret;
949
950 retry:
951 cur_auth_type = rt->auth_state.auth_type;
952 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
953 send_content,
954 send_content_length)))
955 return ret;
956
957 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
958 return ret;
959
960 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
961 rt->auth_state.auth_type != HTTP_AUTH_NONE)
962 goto retry;
963
964 if (reply->status_code > 400){
965 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
966 method,
967 reply->status_code,
968 reply->reason);
969 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
970 }
971
972 return 0;
973 }
974
975 /**
976 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
977 */
978 static int make_setup_request(AVFormatContext *s, const char *host, int port,
979 int lower_transport, const char *real_challenge)
980 {
981 RTSPState *rt = s->priv_data;
982 int rtx, j, i, err, interleave = 0;
983 RTSPStream *rtsp_st;
984 RTSPMessageHeader reply1, *reply = &reply1;
985 char cmd[2048];
986 const char *trans_pref;
987
988 if (rt->transport == RTSP_TRANSPORT_RDT)
989 trans_pref = "x-pn-tng";
990 else
991 trans_pref = "RTP/AVP";
992
993 /* default timeout: 1 minute */
994 rt->timeout = 60;
995
996 /* for each stream, make the setup request */
997 /* XXX: we assume the same server is used for the control of each
998 * RTSP stream */
999
1000 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1001 char transport[2048];
1002
1003 /**
1004 * WMS serves all UDP data over a single connection, the RTX, which
1005 * isn't necessarily the first in the SDP but has to be the first
1006 * to be set up, else the second/third SETUP will fail with a 461.
1007 */
1008 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1009 rt->server_type == RTSP_SERVER_WMS) {
1010 if (i == 0) {
1011 /* rtx first */
1012 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1013 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1014 if (len >= 4 &&
1015 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1016 "/rtx"))
1017 break;
1018 }
1019 if (rtx == rt->nb_rtsp_streams)
1020 return -1; /* no RTX found */
1021 rtsp_st = rt->rtsp_streams[rtx];
1022 } else
1023 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1024 } else
1025 rtsp_st = rt->rtsp_streams[i];
1026
1027 /* RTP/UDP */
1028 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1029 char buf[256];
1030
1031 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1032 port = reply->transports[0].client_port_min;
1033 goto have_port;
1034 }
1035
1036 /* first try in specified port range */
1037 if (RTSP_RTP_PORT_MIN != 0) {
1038 while (j <= RTSP_RTP_PORT_MAX) {
1039 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1040 "?localport=%d", j);
1041 /* we will use two ports per rtp stream (rtp and rtcp) */
1042 j += 2;
1043 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1044 goto rtp_opened;
1045 }
1046 }
1047
1048 #if 0
1049 /* then try on any port */
1050 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1051 err = AVERROR_INVALIDDATA;
1052 goto fail;
1053 }
1054 #endif
1055
1056 rtp_opened:
1057 port = rtp_get_local_port(rtsp_st->rtp_handle);
1058 have_port:
1059 snprintf(transport, sizeof(transport) - 1,
1060 "%s/UDP;", trans_pref);
1061 if (rt->server_type != RTSP_SERVER_REAL)
1062 av_strlcat(transport, "unicast;", sizeof(transport));
1063 av_strlcatf(transport, sizeof(transport),
1064 "client_port=%d", port);
1065 if (rt->transport == RTSP_TRANSPORT_RTP &&
1066 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1067 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1068 }
1069
1070 /* RTP/TCP */
1071 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1072 /** For WMS streams, the application streams are only used for
1073 * UDP. When trying to set it up for TCP streams, the server
1074 * will return an error. Therefore, we skip those streams. */
1075 if (rt->server_type == RTSP_SERVER_WMS &&
1076 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1077 AVMEDIA_TYPE_DATA)
1078 continue;
1079 snprintf(transport, sizeof(transport) - 1,
1080 "%s/TCP;", trans_pref);
1081 if (rt->server_type == RTSP_SERVER_WMS)
1082 av_strlcat(transport, "unicast;", sizeof(transport));
1083 av_strlcatf(transport, sizeof(transport),
1084 "interleaved=%d-%d",
1085 interleave, interleave + 1);
1086 interleave += 2;
1087 }
1088
1089 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1090 snprintf(transport, sizeof(transport) - 1,
1091 "%s/UDP;multicast", trans_pref);
1092 }
1093 if (s->oformat) {
1094 av_strlcat(transport, ";mode=receive", sizeof(transport));
1095 } else if (rt->server_type == RTSP_SERVER_REAL ||
1096 rt->server_type == RTSP_SERVER_WMS)
1097 av_strlcat(transport, ";mode=play", sizeof(transport));
1098 snprintf(cmd, sizeof(cmd),
1099 "Transport: %s\r\n",
1100 transport);
1101 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1102 char real_res[41], real_csum[9];
1103 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1104 real_challenge);
1105 av_strlcatf(cmd, sizeof(cmd),
1106 "If-Match: %s\r\n"
1107 "RealChallenge2: %s, sd=%s\r\n",
1108 rt->session_id, real_res, real_csum);
1109 }
1110 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1111 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1112 err = 1;
1113 goto fail;
1114 } else if (reply->status_code != RTSP_STATUS_OK ||
1115 reply->nb_transports != 1) {
1116 err = AVERROR_INVALIDDATA;
1117 goto fail;
1118 }
1119
1120 /* XXX: same protocol for all streams is required */
1121 if (i > 0) {
1122 if (reply->transports[0].lower_transport != rt->lower_transport ||
1123 reply->transports[0].transport != rt->transport) {
1124 err = AVERROR_INVALIDDATA;
1125 goto fail;
1126 }
1127 } else {
1128 rt->lower_transport = reply->transports[0].lower_transport;
1129 rt->transport = reply->transports[0].transport;
1130 }
1131
1132 /* close RTP connection if not chosen */
1133 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1134 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1135 url_close(rtsp_st->rtp_handle);
1136 rtsp_st->rtp_handle = NULL;
1137 }
1138
1139 switch(reply->transports[0].lower_transport) {
1140 case RTSP_LOWER_TRANSPORT_TCP:
1141 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1142 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1143 break;
1144
1145 case RTSP_LOWER_TRANSPORT_UDP: {
1146 char url[1024];
1147
1148 /* XXX: also use address if specified */
1149 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1150 reply->transports[0].server_port_min, NULL);
1151 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1152 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1153 err = AVERROR_INVALIDDATA;
1154 goto fail;
1155 }
1156 /* Try to initialize the connection state in a
1157 * potential NAT router by sending dummy packets.
1158 * RTP/RTCP dummy packets are used for RDT, too.
1159 */
1160 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1161 rtp_send_punch_packets(rtsp_st->rtp_handle);
1162 break;
1163 }
1164 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1165 char url[1024];
1166 struct in_addr in;
1167 int port, ttl;
1168
1169 if (reply->transports[0].destination) {
1170 in.s_addr = htonl(reply->transports[0].destination);
1171 port = reply->transports[0].port_min;
1172 ttl = reply->transports[0].ttl;
1173 } else {
1174 in = rtsp_st->sdp_ip;
1175 port = rtsp_st->sdp_port;
1176 ttl = rtsp_st->sdp_ttl;
1177 }
1178 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1179 port, "?ttl=%d", ttl);
1180 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1181 err = AVERROR_INVALIDDATA;
1182 goto fail;
1183 }
1184 break;
1185 }
1186 }
1187
1188 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1189 goto fail;
1190 }
1191
1192 if (reply->timeout > 0)
1193 rt->timeout = reply->timeout;
1194
1195 if (rt->server_type == RTSP_SERVER_REAL)
1196 rt->need_subscription = 1;
1197
1198 return 0;
1199
1200 fail:
1201 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1202 if (rt->rtsp_streams[i]->rtp_handle) {
1203 url_close(rt->rtsp_streams[i]->rtp_handle);
1204 rt->rtsp_streams[i]->rtp_handle = NULL;
1205 }
1206 }
1207 return err;
1208 }
1209
1210 static int rtsp_read_play(AVFormatContext *s)
1211 {
1212 RTSPState *rt = s->priv_data;
1213 RTSPMessageHeader reply1, *reply = &reply1;
1214 int i;
1215 char cmd[1024];
1216
1217 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1218
1219 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1220 if (rt->state == RTSP_STATE_PAUSED) {
1221 cmd[0] = 0;
1222 } else {
1223 snprintf(cmd, sizeof(cmd),
1224 "Range: npt=%0.3f-\r\n",
1225 (double)rt->seek_timestamp / AV_TIME_BASE);
1226 }
1227 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1228 if (reply->status_code != RTSP_STATUS_OK) {
1229 return -1;
1230 }
1231 if (reply->range_start != AV_NOPTS_VALUE &&
1232 rt->transport == RTSP_TRANSPORT_RTP) {
1233 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1234 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1235 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1236 AVStream *st = NULL;
1237 if (!rtpctx)
1238 continue;
1239 if (rtsp_st->stream_index >= 0)
1240 st = s->streams[rtsp_st->stream_index];
1241 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1242 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1243 if (st)
1244 rtpctx->range_start_offset = av_rescale_q(reply->range_start,
1245 AV_TIME_BASE_Q,
1246 st->time_base);
1247 }
1248 }
1249 }
1250 rt->state = RTSP_STATE_STREAMING;
1251 return 0;
1252 }
1253
1254 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1255 {
1256 RTSPState *rt = s->priv_data;
1257 char cmd[1024];
1258 unsigned char *content = NULL;
1259 int ret;
1260
1261 /* describe the stream */
1262 snprintf(cmd, sizeof(cmd),
1263 "Accept: application/sdp\r\n");
1264 if (rt->server_type == RTSP_SERVER_REAL) {
1265 /**
1266 * The Require: attribute is needed for proper streaming from
1267 * Realmedia servers.
1268 */
1269 av_strlcat(cmd,
1270 "Require: com.real.retain-entity-for-setup\r\n",
1271 sizeof(cmd));
1272 }
1273 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1274 if (!content)
1275 return AVERROR_INVALIDDATA;
1276 if (reply->status_code != RTSP_STATUS_OK) {
1277 av_freep(&content);
1278 return AVERROR_INVALIDDATA;
1279 }
1280
1281 /* now we got the SDP description, we parse it */
1282 ret = sdp_parse(s, (const char *)content);
1283 av_freep(&content);
1284 if (ret < 0)
1285 return AVERROR_INVALIDDATA;
1286
1287 return 0;
1288 }
1289
1290 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1291 {
1292 RTSPState *rt = s->priv_data;
1293 RTSPMessageHeader reply1, *reply = &reply1;
1294 int i;
1295 char *sdp;
1296 AVFormatContext sdp_ctx, *ctx_array[1];
1297
1298 rt->start_time = av_gettime();
1299
1300 /* Announce the stream */
1301 sdp = av_mallocz(SDP_MAX_SIZE);
1302 if (sdp == NULL)
1303 return AVERROR(ENOMEM);
1304 /* We create the SDP based on the RTSP AVFormatContext where we
1305 * aren't allowed to change the filename field. (We create the SDP
1306 * based on the RTSP context since the contexts for the RTP streams
1307 * don't exist yet.) In order to specify a custom URL with the actual
1308 * peer IP instead of the originally specified hostname, we create
1309 * a temporary copy of the AVFormatContext, where the custom URL is set.
1310 *
1311 * FIXME: Create the SDP without copying the AVFormatContext.
1312 * This either requires setting up the RTP stream AVFormatContexts
1313 * already here (complicating things immensely) or getting a more
1314 * flexible SDP creation interface.
1315 */
1316 sdp_ctx = *s;
1317 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1318 "rtsp", NULL, addr, -1, NULL);
1319 ctx_array[0] = &sdp_ctx;
1320 if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
1321 av_free(sdp);
1322 return AVERROR_INVALIDDATA;
1323 }
1324 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1325 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1326 "Content-Type: application/sdp\r\n",
1327 reply, NULL, sdp, strlen(sdp));
1328 av_free(sdp);
1329 if (reply->status_code != RTSP_STATUS_OK)
1330 return AVERROR_INVALIDDATA;
1331
1332 /* Set up the RTSPStreams for each AVStream */
1333 for (i = 0; i < s->nb_streams; i++) {
1334 RTSPStream *rtsp_st;
1335 AVStream *st = s->streams[i];
1336
1337 rtsp_st = av_mallocz(sizeof(RTSPStream));
1338 if (!rtsp_st)
1339 return AVERROR(ENOMEM);
1340 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1341
1342 st->priv_data = rtsp_st;
1343 rtsp_st->stream_index = i;
1344
1345 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1346 /* Note, this must match the relative uri set in the sdp content */
1347 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1348 "/streamid=%d", i);
1349 }
1350
1351 return 0;
1352 }
1353
1354 void ff_rtsp_close_connections(AVFormatContext *s)
1355 {
1356 RTSPState *rt = s->priv_data;
1357 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1358 url_close(rt->rtsp_hd);
1359 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1360 }
1361
1362 int ff_rtsp_connect(AVFormatContext *s)
1363 {
1364 RTSPState *rt = s->priv_data;
1365 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1366 char *option_list, *option, *filename;
1367 int port, err, tcp_fd;
1368 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1369 int lower_transport_mask = 0;
1370 char real_challenge[64];
1371 struct sockaddr_storage peer;
1372 socklen_t peer_len = sizeof(peer);
1373
1374 if (!ff_network_init())
1375 return AVERROR(EIO);
1376 redirect:
1377 rt->control_transport = RTSP_MODE_PLAIN;
1378 /* extract hostname and port */
1379 av_url_split(NULL, 0, auth, sizeof(auth),
1380 host, sizeof(host), &port, path, sizeof(path), s->filename);
1381 if (*auth) {
1382 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1383 }
1384 if (port < 0)
1385 port = RTSP_DEFAULT_PORT;
1386
1387 /* search for options */
1388 option_list = strrchr(path, '?');
1389 if (option_list) {
1390 /* Strip out the RTSP specific options, write out the rest of
1391 * the options back into the same string. */
1392 filename = option_list;
1393 while (option_list) {
1394 /* move the option pointer */
1395 option = ++option_list;
1396 option_list = strchr(option_list, '&');
1397 if (option_list)
1398 *option_list = 0;
1399
1400 /* handle the options */
1401 if (!strcmp(option, "udp")) {
1402 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1403 } else if (!strcmp(option, "multicast")) {
1404 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1405 } else if (!strcmp(option, "tcp")) {
1406 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1407 } else if(!strcmp(option, "http")) {
1408 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1409 rt->control_transport = RTSP_MODE_TUNNEL;
1410 } else {
1411 /* Write options back into the buffer, using memmove instead
1412 * of strcpy since the strings may overlap. */
1413 int len = strlen(option);
1414 memmove(++filename, option, len);
1415 filename += len;
1416 if (option_list) *filename = '&';
1417 }
1418 }
1419 *filename = 0;
1420 }
1421
1422 if (!lower_transport_mask)
1423 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1424
1425 if (s->oformat) {
1426 /* Only UDP or TCP - UDP multicast isn't supported. */
1427 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1428 (1 << RTSP_LOWER_TRANSPORT_TCP);
1429 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1430 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1431 "only UDP and TCP are supported for output.\n");
1432 err = AVERROR(EINVAL);
1433 goto fail;
1434 }
1435 }
1436
1437 /* Construct the URI used in request; this is similar to s->filename,
1438 * but with authentication credentials removed and RTSP specific options
1439 * stripped out. */
1440 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1441 host, port, "%s", path);
1442
1443 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1444 /* set up initial handshake for tunneling */
1445 char httpname[1024];
1446 char sessioncookie[17];
1447 char headers[1024];
1448
1449 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1450 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1451 av_get_random_seed(), av_get_random_seed());
1452
1453 /* GET requests */
1454 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1455 err = AVERROR(EIO);
1456 goto fail;
1457 }
1458
1459 /* generate GET headers */
1460 snprintf(headers, sizeof(headers),
1461 "x-sessioncookie: %s\r\n"
1462 "Accept: application/x-rtsp-tunnelled\r\n"
1463 "Pragma: no-cache\r\n"
1464 "Cache-Control: no-cache\r\n",
1465 sessioncookie);
1466 ff_http_set_headers(rt->rtsp_hd, headers);
1467
1468 /* complete the connection */
1469 if (url_connect(rt->rtsp_hd)) {
1470 err = AVERROR(EIO);
1471 goto fail;
1472 }
1473
1474 /* POST requests */
1475 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1476 err = AVERROR(EIO);
1477 goto fail;
1478 }
1479
1480 /* generate POST headers */
1481 snprintf(headers, sizeof(headers),
1482 "x-sessioncookie: %s\r\n"
1483 "Content-Type: application/x-rtsp-tunnelled\r\n"
1484 "Pragma: no-cache\r\n"
1485 "Cache-Control: no-cache\r\n"
1486 "Content-Length: 32767\r\n"
1487 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1488 sessioncookie);
1489 ff_http_set_headers(rt->rtsp_hd_out, headers);
1490 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1491
1492 /* Initialize the authentication state for the POST session. The HTTP
1493 * protocol implementation doesn't properly handle multi-pass
1494 * authentication for POST requests, since it would require one of
1495 * the following:
1496 * - implementing Expect: 100-continue, which many HTTP servers
1497 * don't support anyway, even less the RTSP servers that do HTTP
1498 * tunneling
1499 * - sending the whole POST data until getting a 401 reply specifying
1500 * what authentication method to use, then resending all that data
1501 * - waiting for potential 401 replies directly after sending the
1502 * POST header (waiting for some unspecified time)
1503 * Therefore, we copy the full auth state, which works for both basic
1504 * and digest. (For digest, we would have to synchronize the nonce
1505 * count variable between the two sessions, if we'd do more requests
1506 * with the original session, though.)
1507 */
1508 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1509
1510 /* complete the connection */
1511 if (url_connect(rt->rtsp_hd_out)) {
1512 err = AVERROR(EIO);
1513 goto fail;
1514 }
1515 } else {
1516 /* open the tcp connection */
1517 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1518 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1519 err = AVERROR(EIO);
1520 goto fail;
1521 }
1522 rt->rtsp_hd_out = rt->rtsp_hd;
1523 }
1524 rt->seq = 0;
1525
1526 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1527 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1528 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1529 NULL, 0, NI_NUMERICHOST);
1530 }
1531
1532 /* request options supported by the server; this also detects server
1533 * type */
1534 for (rt->server_type = RTSP_SERVER_RTP;;) {
1535 cmd[0] = 0;
1536 if (rt->server_type == RTSP_SERVER_REAL)
1537 av_strlcat(cmd,
1538 /**
1539 * The following entries are required for proper
1540 * streaming from a Realmedia server. They are
1541 * interdependent in some way although we currently
1542 * don't quite understand how. Values were copied
1543 * from mplayer SVN r23589.
1544 * @param CompanyID is a 16-byte ID in base64
1545 * @param ClientChallenge is a 16-byte ID in hex
1546 */
1547 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1548 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1549 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1550 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1551 sizeof(cmd));
1552 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1553 if (reply->status_code != RTSP_STATUS_OK) {
1554 err = AVERROR_INVALIDDATA;
1555 goto fail;
1556 }
1557
1558 /* detect server type if not standard-compliant RTP */
1559 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1560 rt->server_type = RTSP_SERVER_REAL;
1561 continue;
1562 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1563 rt->server_type = RTSP_SERVER_WMS;
1564 } else if (rt->server_type == RTSP_SERVER_REAL)
1565 strcpy(real_challenge, reply->real_challenge);
1566 break;
1567 }
1568
1569 if (s->iformat)
1570 err = rtsp_setup_input_streams(s, reply);
1571 else
1572 err = rtsp_setup_output_streams(s, host);
1573 if (err)
1574 goto fail;
1575
1576 do {
1577 int lower_transport = ff_log2_tab[lower_transport_mask &
1578 ~(lower_transport_mask - 1)];
1579
1580 err = make_setup_request(s, host, port, lower_transport,
1581 rt->server_type == RTSP_SERVER_REAL ?
1582 real_challenge : NULL);
1583 if (err < 0)
1584 goto fail;
1585 lower_transport_mask &= ~(1 << lower_transport);
1586 if (lower_transport_mask == 0 && err == 1) {
1587 err = FF_NETERROR(EPROTONOSUPPORT);
1588 goto fail;
1589 }
1590 } while (err);
1591
1592 rt->state = RTSP_STATE_IDLE;
1593 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1594 return 0;
1595 fail:
1596 ff_rtsp_close_streams(s);
1597 ff_rtsp_close_connections(s);
1598 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1599 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1600 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1601 reply->status_code,
1602 s->filename);
1603 goto redirect;
1604 }
1605 ff_network_close();
1606 return err;
1607 }
1608 #endif
1609
1610 #if CONFIG_RTSP_DEMUXER
1611 static int rtsp_read_header(AVFormatContext *s,
1612 AVFormatParameters *ap)
1613 {
1614 RTSPState *rt = s->priv_data;
1615 int ret;
1616
1617 ret = ff_rtsp_connect(s);
1618 if (ret)
1619 return ret;
1620
1621 rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
1622 if (!rt->real_setup_cache)
1623 return AVERROR(ENOMEM);
1624 rt->real_setup = rt->real_setup_cache + s->nb_streams * sizeof(*rt->real_setup);
1625
1626 if (ap->initial_pause) {
1627 /* do not start immediately */
1628 } else {
1629 if (rtsp_read_play(s) < 0) {
1630 ff_rtsp_close_streams(s);
1631 ff_rtsp_close_connections(s);
1632 return AVERROR_INVALIDDATA;
1633 }
1634 }
1635
1636 return 0;
1637 }
1638
1639 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1640 uint8_t *buf, int buf_size)
1641 {
1642 RTSPState *rt = s->priv_data;
1643 RTSPStream *rtsp_st;
1644 fd_set rfds;
1645 int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1646 struct timeval tv;
1647
1648 for (;;) {
1649 if (url_interrupt_cb())
1650 return AVERROR(EINTR);
1651 FD_ZERO(&rfds);
1652 if (rt->rtsp_hd) {
1653 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1654 FD_SET(tcp_fd, &rfds);
1655 } else {
1656 fd_max = 0;
1657 tcp_fd = -1;
1658 }
1659 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1660 rtsp_st = rt->rtsp_streams[i];
1661 if (rtsp_st->rtp_handle) {
1662 /* currently, we cannot probe RTCP handle because of
1663 * blocking restrictions */
1664 fd = url_get_file_handle(rtsp_st->rtp_handle);
1665 if (fd > fd_max)
1666 fd_max = fd;
1667 FD_SET(fd, &rfds);
1668 }
1669 }
1670 tv.tv_sec = 0;
1671 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1672 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1673 if (n > 0) {
1674 timeout_cnt = 0;
1675 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1676 rtsp_st = rt->rtsp_streams[i];
1677 if (rtsp_st->rtp_handle) {
1678 fd = url_get_file_handle(rtsp_st->rtp_handle);
1679 if (FD_ISSET(fd, &rfds)) {
1680 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1681 if (ret > 0) {
1682 *prtsp_st = rtsp_st;
1683 return ret;
1684 }
1685 }
1686 }
1687 }
1688 #if CONFIG_RTSP_DEMUXER
1689 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1690 RTSPMessageHeader reply;
1691
1692 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1693 if (ret < 0)
1694 return ret;
1695 /* XXX: parse message */
1696 if (rt->state != RTSP_STATE_STREAMING)
1697 return 0;
1698 }
1699 #endif
1700 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1701 return FF_NETERROR(ETIMEDOUT);
1702 } else if (n < 0 && errno != EINTR)
1703 return AVERROR(errno);
1704 }
1705 }
1706
1707 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1708 uint8_t *buf, int buf_size)
1709 {
1710 RTSPState *rt = s->priv_data;
1711 int id, len, i, ret;
1712 RTSPStream *rtsp_st;
1713
1714 #ifdef DEBUG_RTP_TCP
1715 dprintf(s, "tcp_read_packet:\n");
1716 #endif
1717 redo:
1718 for (;;) {
1719 RTSPMessageHeader reply;
1720
1721 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1722 if (ret < 0)
1723 return ret;
1724 if (ret == 1) /* received '$' */
1725 break;
1726 /* XXX: parse message */
1727 if (rt->state != RTSP_STATE_STREAMING)
1728 return 0;
1729 }
1730 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1731 if (ret != 3)
1732 return -1;
1733 id = buf[0];
1734 len = AV_RB16(buf + 1);
1735 #ifdef DEBUG_RTP_TCP
1736 dprintf(s, "id=%d len=%d\n", id, len);
1737 #endif
1738 if (len > buf_size || len < 12)
1739 goto redo;
1740 /* get the data */
1741 ret = url_read_complete(rt->rtsp_hd, buf, len);
1742 if (ret != len)
1743 return -1;
1744 if (rt->transport == RTSP_TRANSPORT_RDT &&
1745 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1746 return -1;
1747
1748 /* find the matching stream */
1749 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1750 rtsp_st = rt->rtsp_streams[i];
1751 if (id >= rtsp_st->interleaved_min &&
1752 id <= rtsp_st->interleaved_max)
1753 goto found;
1754 }
1755 goto redo;
1756 found:
1757 *prtsp_st = rtsp_st;
1758 return len;
1759 }
1760
1761 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1762 {
1763 RTSPState *rt = s->priv_data;
1764 int ret, len;
1765 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1766 RTSPStream *rtsp_st;
1767
1768 /* get next frames from the same RTP packet */
1769 if (rt->cur_transport_priv) {
1770 if (rt->transport == RTSP_TRANSPORT_RDT) {
1771 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1772 } else
1773 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1774 if (ret == 0) {
1775 rt->cur_transport_priv = NULL;
1776 return 0;
1777 } else if (ret == 1) {
1778 return 0;
1779 } else
1780 rt->cur_transport_priv = NULL;
1781 }
1782
1783 /* read next RTP packet */
1784 redo:
1785 switch(rt->lower_transport) {
1786 default:
1787 #if CONFIG_RTSP_DEMUXER
1788 case RTSP_LOWER_TRANSPORT_TCP:
1789 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1790 break;
1791 #endif
1792 case RTSP_LOWER_TRANSPORT_UDP:
1793 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1794 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1795 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1796 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1797 break;
1798 }
1799 if (len < 0)
1800 return len;
1801 if (len == 0)
1802 return AVERROR_EOF;
1803 if (rt->transport == RTSP_TRANSPORT_RDT) {
1804 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1805 } else {
1806 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1807 if (ret < 0) {
1808 /* Either bad packet, or a RTCP packet. Check if the
1809 * first_rtcp_ntp_time field was initialized. */
1810 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1811 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1812 /* first_rtcp_ntp_time has been initialized for this stream,
1813 * copy the same value to all other uninitialized streams,
1814 * in order to map their timestamp origin to the same ntp time
1815 * as this one. */
1816 int i;
1817 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1818 RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
1819 if (rtpctx2 &&
1820 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1821 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1822 }
1823 }
1824 }
1825 }
1826 if (ret < 0)
1827 goto redo;
1828 if (ret == 1)
1829 /* more packets may follow, so we save the RTP context */
1830 rt->cur_transport_priv = rtsp_st->transport_priv;
1831
1832 return ret;
1833 }
1834
1835 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1836 {
1837 RTSPState *rt = s->priv_data;
1838 int ret;
1839 RTSPMessageHeader reply1, *reply = &reply1;
1840 char cmd[1024];
1841
1842 if (rt->server_type == RTSP_SERVER_REAL) {
1843 int i;
1844
1845 for (i = 0; i < s->nb_streams; i++)
1846 rt->real_setup[i] = s->streams[i]->discard;
1847
1848 if (!rt->need_subscription) {
1849 if (memcmp (rt->real_setup, rt->real_setup_cache,
1850 sizeof(enum AVDiscard) * s->nb_streams)) {
1851 snprintf(cmd, sizeof(cmd),
1852 "Unsubscribe: %s\r\n",
1853 rt->last_subscription);
1854 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1855 cmd, reply, NULL);
1856 if (reply->status_code != RTSP_STATUS_OK)
1857 return AVERROR_INVALIDDATA;
1858 rt->need_subscription = 1;
1859 }
1860 }
1861
1862 if (rt->need_subscription) {
1863 int r, rule_nr, first = 1;
1864
1865 memcpy(rt->real_setup_cache, rt->real_setup,
1866 sizeof(enum AVDiscard) * s->nb_streams);
1867 rt->last_subscription[0] = 0;
1868
1869 snprintf(cmd, sizeof(cmd),
1870 "Subscribe: ");
1871 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1872 rule_nr = 0;
1873 for (r = 0; r < s->nb_streams; r++) {
1874 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1875 if (s->streams[r]->discard != AVDISCARD_ALL) {
1876 if (!first)
1877 av_strlcat(rt->last_subscription, ",",
1878 sizeof(rt->last_subscription));
1879 ff_rdt_subscribe_rule(
1880 rt->last_subscription,
1881 sizeof(rt->last_subscription), i, rule_nr);
1882 first = 0;
1883 }
1884 rule_nr++;
1885 }
1886 }
1887 }
1888 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1889 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1890 cmd, reply, NULL);
1891 if (reply->status_code != RTSP_STATUS_OK)
1892 return AVERROR_INVALIDDATA;
1893 rt->need_subscription = 0;
1894
1895 if (rt->state == RTSP_STATE_STREAMING)
1896 rtsp_read_play (s);
1897 }
1898 }
1899
1900 ret = rtsp_fetch_packet(s, pkt);
1901 if (ret < 0)
1902 return ret;
1903
1904 /* send dummy request to keep TCP connection alive */
1905 if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1906 if (rt->server_type == RTSP_SERVER_WMS) {
1907 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1908 } else {
1909 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1910 }
1911 }
1912
1913 return 0;
1914 }
1915
1916 /* pause the stream */
1917 static int rtsp_read_pause(AVFormatContext *s)
1918 {
1919 RTSPState *rt = s->priv_data;
1920 RTSPMessageHeader reply1, *reply = &reply1;
1921
1922 if (rt->state != RTSP_STATE_STREAMING)
1923 return 0;
1924 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1925 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1926 if (reply->status_code != RTSP_STATUS_OK) {
1927 return -1;
1928 }
1929 }
1930 rt->state = RTSP_STATE_PAUSED;
1931 return 0;
1932 }
1933
1934 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1935 int64_t timestamp, int flags)
1936 {
1937 RTSPState *rt = s->priv_data;
1938
1939 rt->seek_timestamp = av_rescale_q(timestamp,
1940 s->streams[stream_index]->time_base,
1941 AV_TIME_BASE_Q);
1942 switch(rt->state) {
1943 default:
1944 case RTSP_STATE_IDLE:
1945 break;
1946 case RTSP_STATE_STREAMING:
1947 if (rtsp_read_pause(s) != 0)
1948 return -1;
1949 rt->state = RTSP_STATE_SEEKING;
1950 if (rtsp_read_play(s) != 0)
1951 return -1;
1952 break;
1953 case RTSP_STATE_PAUSED:
1954 rt->state = RTSP_STATE_IDLE;
1955 break;
1956 }
1957 return 0;
1958 }
1959
1960 static int rtsp_read_close(AVFormatContext *s)
1961 {
1962 RTSPState *rt = s->priv_data;
1963
1964 #if 0
1965 /* NOTE: it is valid to flush the buffer here */
1966 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1967 url_fclose(&rt->rtsp_gb);
1968 }
1969 #endif
1970 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1971
1972 ff_rtsp_close_streams(s);
1973 ff_rtsp_close_connections(s);
1974 ff_network_close();
1975 rt->real_setup = NULL;
1976 av_freep(&rt->real_setup_cache);
1977 return 0;
1978 }
1979
1980 AVInputFormat rtsp_demuxer = {
1981 "rtsp",
1982 NULL_IF_CONFIG_SMALL("RTSP input format"),
1983 sizeof(RTSPState),
1984 rtsp_probe,
1985 rtsp_read_header,
1986 rtsp_read_packet,
1987 rtsp_read_close,
1988 rtsp_read_seek,
1989 .flags = AVFMT_NOFILE,
1990 .read_play = rtsp_read_play,
1991 .read_pause = rtsp_read_pause,
1992 };
1993 #endif
1994
1995 static int sdp_probe(AVProbeData *p1)
1996 {
1997 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1998
1999 /* we look for a line beginning "c=IN IP4" */
2000 while (p < p_end && *p != '\0') {
2001 if (p + sizeof("c=IN IP4") - 1 < p_end &&
2002 av_strstart(p, "c=IN IP4", NULL))
2003 return AVPROBE_SCORE_MAX / 2;
2004
2005 while (p < p_end - 1 && *p != '\n') p++;
2006 if (++p >= p_end)
2007 break;
2008 if (*p == '\r')
2009 p++;
2010 }
2011 return 0;
2012 }
2013
2014 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2015 {
2016 RTSPState *rt = s->priv_data;
2017 RTSPStream *rtsp_st;
2018 int size, i, err;
2019 char *content;
2020 char url[1024];
2021
2022 if (!ff_network_init())
2023 return AVERROR(EIO);
2024
2025 /* read the whole sdp file */
2026 /* XXX: better loading */
2027 content = av_malloc(SDP_MAX_SIZE);
2028 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2029 if (size <= 0) {
2030 av_free(content);
2031 return AVERROR_INVALIDDATA;
2032 }
2033 content[size] ='\0';
2034
2035 sdp_parse(s, content);
2036 av_free(content);
2037
2038 /* open each RTP stream */
2039 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2040 rtsp_st = rt->rtsp_streams[i];
2041
2042 ff_url_join(url, sizeof(url), "rtp", NULL,
2043 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2044 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2045 rtsp_st->sdp_ttl);
2046 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2047 err = AVERROR_INVALIDDATA;
2048 goto fail;
2049 }
2050 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2051 goto fail;
2052 }
2053 return 0;
2054 fail:
2055 ff_rtsp_close_streams(s);
2056 ff_network_close();
2057 return err;
2058 }
2059
2060 static int sdp_read_close(AVFormatContext *s)
2061 {
2062 ff_rtsp_close_streams(s);
2063 ff_network_close();
2064 return 0;
2065 }
2066
2067 AVInputFormat sdp_demuxer = {
2068 "sdp",
2069 NULL_IF_CONFIG_SMALL("SDP"),
2070 sizeof(RTSPState),
2071 sdp_probe,
2072 sdp_read_header,
2073 rtsp_fetch_packet,
2074 sdp_read_close,
2075 };