RTSP: Move more SDP/FMTP stuff from rtsp.c to rtpdec_mpeg4.c
[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_asf.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
56 static void get_word_until_chars(char *buf, int buf_size,
57 const char *sep, const char **pp)
58 {
59 const char *p;
60 char *q;
61
62 p = *pp;
63 p += strspn(p, SPACE_CHARS);
64 q = buf;
65 while (!strchr(sep, *p) && *p != '\0') {
66 if ((q - buf) < buf_size - 1)
67 *q++ = *p;
68 p++;
69 }
70 if (buf_size > 0)
71 *q = '\0';
72 *pp = p;
73 }
74
75 static void get_word_sep(char *buf, int buf_size, const char *sep,
76 const char **pp)
77 {
78 if (**pp == '/') (*pp)++;
79 get_word_until_chars(buf, buf_size, sep, pp);
80 }
81
82 static void get_word(char *buf, int buf_size, const char **pp)
83 {
84 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
85 }
86
87 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
88 static int sdp_parse_rtpmap(AVFormatContext *s,
89 AVCodecContext *codec, RTSPStream *rtsp_st,
90 int payload_type, const char *p)
91 {
92 char buf[256];
93 int i;
94 AVCodec *c;
95 const char *c_name;
96
97 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
98 * see if we can handle this kind of payload.
99 * The space should normally not be there but some Real streams or
100 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
101 * have a trailing space. */
102 get_word_sep(buf, sizeof(buf), "/ ", &p);
103 if (payload_type >= RTP_PT_PRIVATE) {
104 RTPDynamicProtocolHandler *handler;
105 for (handler = RTPFirstDynamicPayloadHandler;
106 handler; handler = handler->next) {
107 if (!strcasecmp(buf, handler->enc_name) &&
108 codec->codec_type == handler->codec_type) {
109 codec->codec_id = handler->codec_id;
110 rtsp_st->dynamic_handler = handler;
111 if (handler->open)
112 rtsp_st->dynamic_protocol_context = handler->open();
113 break;
114 }
115 }
116 } else {
117 /* We are in a standard case
118 * (from http://www.iana.org/assignments/rtp-parameters). */
119 /* search into AVRtpPayloadTypes[] */
120 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
121 }
122
123 c = avcodec_find_decoder(codec->codec_id);
124 if (c && c->name)
125 c_name = c->name;
126 else
127 c_name = "(null)";
128
129 get_word_sep(buf, sizeof(buf), "/", &p);
130 i = atoi(buf);
131 switch (codec->codec_type) {
132 case AVMEDIA_TYPE_AUDIO:
133 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
134 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
135 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
136 if (i > 0) {
137 codec->sample_rate = i;
138 get_word_sep(buf, sizeof(buf), "/", &p);
139 i = atoi(buf);
140 if (i > 0)
141 codec->channels = i;
142 // TODO: there is a bug here; if it is a mono stream, and
143 // less than 22000Hz, faad upconverts to stereo and twice
144 // the frequency. No problem, but the sample rate is being
145 // set here by the sdp line. Patch on its way. (rdm)
146 }
147 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
148 codec->sample_rate);
149 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
150 codec->channels);
151 break;
152 case AVMEDIA_TYPE_VIDEO:
153 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
154 break;
155 default:
156 break;
157 }
158 return 0;
159 }
160
161 /* parse the attribute line from the fmtp a line of an sdp response. This
162 * is broken out as a function because it is used in rtp_h264.c, which is
163 * forthcoming. */
164 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
165 char *value, int value_size)
166 {
167 *p += strspn(*p, SPACE_CHARS);
168 if (**p) {
169 get_word_sep(attr, attr_size, "=", p);
170 if (**p == '=')
171 (*p)++;
172 get_word_sep(value, value_size, ";", p);
173 if (**p == ';')
174 (*p)++;
175 return 1;
176 }
177 return 0;
178 }
179
180 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
181 * and end time.
182 * Used for seeking in the rtp stream.
183 */
184 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
185 {
186 char buf[256];
187
188 p += strspn(p, SPACE_CHARS);
189 if (!av_stristart(p, "npt=", &p))
190 return;
191
192 *start = AV_NOPTS_VALUE;
193 *end = AV_NOPTS_VALUE;
194
195 get_word_sep(buf, sizeof(buf), "-", &p);
196 *start = parse_date(buf, 1);
197 if (*p == '-') {
198 p++;
199 get_word_sep(buf, sizeof(buf), "-", &p);
200 *end = parse_date(buf, 1);
201 }
202 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
203 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
204 }
205
206 typedef struct SDPParseState {
207 /* SDP only */
208 struct in_addr default_ip;
209 int default_ttl;
210 int skip_media; ///< set if an unknown m= line occurs
211 } SDPParseState;
212
213 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
214 int letter, const char *buf)
215 {
216 RTSPState *rt = s->priv_data;
217 char buf1[64], st_type[64];
218 const char *p;
219 enum AVMediaType codec_type;
220 int payload_type, i;
221 AVStream *st;
222 RTSPStream *rtsp_st;
223 struct in_addr sdp_ip;
224 int ttl;
225
226 dprintf(s, "sdp: %c='%s'\n", letter, buf);
227
228 p = buf;
229 if (s1->skip_media && letter != 'm')
230 return;
231 switch (letter) {
232 case 'c':
233 get_word(buf1, sizeof(buf1), &p);
234 if (strcmp(buf1, "IN") != 0)
235 return;
236 get_word(buf1, sizeof(buf1), &p);
237 if (strcmp(buf1, "IP4") != 0)
238 return;
239 get_word_sep(buf1, sizeof(buf1), "/", &p);
240 if (ff_inet_aton(buf1, &sdp_ip) == 0)
241 return;
242 ttl = 16;
243 if (*p == '/') {
244 p++;
245 get_word_sep(buf1, sizeof(buf1), "/", &p);
246 ttl = atoi(buf1);
247 }
248 if (s->nb_streams == 0) {
249 s1->default_ip = sdp_ip;
250 s1->default_ttl = ttl;
251 } else {
252 st = s->streams[s->nb_streams - 1];
253 rtsp_st = st->priv_data;
254 rtsp_st->sdp_ip = sdp_ip;
255 rtsp_st->sdp_ttl = ttl;
256 }
257 break;
258 case 's':
259 av_metadata_set2(&s->metadata, "title", p, 0);
260 break;
261 case 'i':
262 if (s->nb_streams == 0) {
263 av_metadata_set2(&s->metadata, "comment", p, 0);
264 break;
265 }
266 break;
267 case 'm':
268 /* new stream */
269 s1->skip_media = 0;
270 get_word(st_type, sizeof(st_type), &p);
271 if (!strcmp(st_type, "audio")) {
272 codec_type = AVMEDIA_TYPE_AUDIO;
273 } else if (!strcmp(st_type, "video")) {
274 codec_type = AVMEDIA_TYPE_VIDEO;
275 } else if (!strcmp(st_type, "application")) {
276 codec_type = AVMEDIA_TYPE_DATA;
277 } else {
278 s1->skip_media = 1;
279 return;
280 }
281 rtsp_st = av_mallocz(sizeof(RTSPStream));
282 if (!rtsp_st)
283 return;
284 rtsp_st->stream_index = -1;
285 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
286
287 rtsp_st->sdp_ip = s1->default_ip;
288 rtsp_st->sdp_ttl = s1->default_ttl;
289
290 get_word(buf1, sizeof(buf1), &p); /* port */
291 rtsp_st->sdp_port = atoi(buf1);
292
293 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
294
295 /* XXX: handle list of formats */
296 get_word(buf1, sizeof(buf1), &p); /* format list */
297 rtsp_st->sdp_payload_type = atoi(buf1);
298
299 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
300 /* no corresponding stream */
301 } else {
302 st = av_new_stream(s, 0);
303 if (!st)
304 return;
305 st->priv_data = rtsp_st;
306 rtsp_st->stream_index = st->index;
307 st->codec->codec_type = codec_type;
308 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
309 /* if standard payload type, we can find the codec right now */
310 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
311 }
312 }
313 /* put a default control url */
314 av_strlcpy(rtsp_st->control_url, rt->control_uri,
315 sizeof(rtsp_st->control_url));
316 break;
317 case 'a':
318 if (av_strstart(p, "control:", &p)) {
319 if (s->nb_streams == 0) {
320 if (!strncmp(p, "rtsp://", 7))
321 av_strlcpy(rt->control_uri, p,
322 sizeof(rt->control_uri));
323 } else {
324 char proto[32];
325 /* get the control url */
326 st = s->streams[s->nb_streams - 1];
327 rtsp_st = st->priv_data;
328
329 /* XXX: may need to add full url resolution */
330 ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
331 NULL, NULL, 0, p);
332 if (proto[0] == '\0') {
333 /* relative control URL */
334 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
335 av_strlcat(rtsp_st->control_url, "/",
336 sizeof(rtsp_st->control_url));
337 av_strlcat(rtsp_st->control_url, p,
338 sizeof(rtsp_st->control_url));
339 } else
340 av_strlcpy(rtsp_st->control_url, p,
341 sizeof(rtsp_st->control_url));
342 }
343 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
344 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
345 get_word(buf1, sizeof(buf1), &p);
346 payload_type = atoi(buf1);
347 st = s->streams[s->nb_streams - 1];
348 rtsp_st = st->priv_data;
349 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
350 } else if (av_strstart(p, "fmtp:", &p) ||
351 av_strstart(p, "framesize:", &p)) {
352 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
353 // let dynamic protocol handlers have a stab at the line.
354 get_word(buf1, sizeof(buf1), &p);
355 payload_type = atoi(buf1);
356 for (i = 0; i < s->nb_streams; i++) {
357 st = s->streams[i];
358 rtsp_st = st->priv_data;
359 if (rtsp_st->sdp_payload_type == payload_type &&
360 rtsp_st->dynamic_handler &&
361 rtsp_st->dynamic_handler->parse_sdp_a_line)
362 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
363 rtsp_st->dynamic_protocol_context, buf);
364 }
365 } else if (av_strstart(p, "range:", &p)) {
366 int64_t start, end;
367
368 // this is so that seeking on a streamed file can work.
369 rtsp_parse_range_npt(p, &start, &end);
370 s->start_time = start;
371 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
372 s->duration = (end == AV_NOPTS_VALUE) ?
373 AV_NOPTS_VALUE : end - start;
374 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
375 if (atoi(p) == 1)
376 rt->transport = RTSP_TRANSPORT_RDT;
377 } else {
378 if (rt->server_type == RTSP_SERVER_WMS)
379 ff_wms_parse_sdp_a_line(s, p);
380 if (s->nb_streams > 0) {
381 if (rt->server_type == RTSP_SERVER_REAL)
382 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
383
384 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
385 if (rtsp_st->dynamic_handler &&
386 rtsp_st->dynamic_handler->parse_sdp_a_line)
387 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
388 s->nb_streams - 1,
389 rtsp_st->dynamic_protocol_context, buf);
390 }
391 }
392 break;
393 }
394 }
395
396 static int sdp_parse(AVFormatContext *s, const char *content)
397 {
398 const char *p;
399 int letter;
400 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
401 * contain long SDP lines containing complete ASF Headers (several
402 * kB) or arrays of MDPR (RM stream descriptor) headers plus
403 * "rulebooks" describing their properties. Therefore, the SDP line
404 * buffer is large.
405 *
406 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
407 * in rtpdec_xiph.c. */
408 char buf[16384], *q;
409 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
410
411 memset(s1, 0, sizeof(SDPParseState));
412 p = content;
413 for (;;) {
414 p += strspn(p, SPACE_CHARS);
415 letter = *p;
416 if (letter == '\0')
417 break;
418 p++;
419 if (*p != '=')
420 goto next_line;
421 p++;
422 /* get the content */
423 q = buf;
424 while (*p != '\n' && *p != '\r' && *p != '\0') {
425 if ((q - buf) < sizeof(buf) - 1)
426 *q++ = *p;
427 p++;
428 }
429 *q = '\0';
430 sdp_parse_line(s, s1, letter, buf);
431 next_line:
432 while (*p != '\n' && *p != '\0')
433 p++;
434 if (*p == '\n')
435 p++;
436 }
437 return 0;
438 }
439
440 /* close and free RTSP streams */
441 void ff_rtsp_close_streams(AVFormatContext *s)
442 {
443 RTSPState *rt = s->priv_data;
444 int i;
445 RTSPStream *rtsp_st;
446
447 for (i = 0; i < rt->nb_rtsp_streams; i++) {
448 rtsp_st = rt->rtsp_streams[i];
449 if (rtsp_st) {
450 if (rtsp_st->transport_priv) {
451 if (s->oformat) {
452 AVFormatContext *rtpctx = rtsp_st->transport_priv;
453 av_write_trailer(rtpctx);
454 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
455 uint8_t *ptr;
456 url_close_dyn_buf(rtpctx->pb, &ptr);
457 av_free(ptr);
458 } else {
459 url_fclose(rtpctx->pb);
460 }
461 av_metadata_free(&rtpctx->streams[0]->metadata);
462 av_metadata_free(&rtpctx->metadata);
463 av_free(rtpctx->streams[0]);
464 av_free(rtpctx);
465 } else if (rt->transport == RTSP_TRANSPORT_RDT)
466 ff_rdt_parse_close(rtsp_st->transport_priv);
467 else
468 rtp_parse_close(rtsp_st->transport_priv);
469 }
470 if (rtsp_st->rtp_handle)
471 url_close(rtsp_st->rtp_handle);
472 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
473 rtsp_st->dynamic_handler->close(
474 rtsp_st->dynamic_protocol_context);
475 }
476 }
477 av_free(rt->rtsp_streams);
478 if (rt->asf_ctx) {
479 av_close_input_stream (rt->asf_ctx);
480 rt->asf_ctx = NULL;
481 }
482 }
483
484 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
485 URLContext *handle)
486 {
487 RTSPState *rt = s->priv_data;
488 AVFormatContext *rtpctx;
489 int ret;
490 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
491
492 if (!rtp_format)
493 return NULL;
494
495 /* Allocate an AVFormatContext for each output stream */
496 rtpctx = avformat_alloc_context();
497 if (!rtpctx)
498 return NULL;
499
500 rtpctx->oformat = rtp_format;
501 if (!av_new_stream(rtpctx, 0)) {
502 av_free(rtpctx);
503 return NULL;
504 }
505 /* Copy the max delay setting; the rtp muxer reads this. */
506 rtpctx->max_delay = s->max_delay;
507 /* Copy other stream parameters. */
508 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
509
510 /* Set the synchronized start time. */
511 rtpctx->start_time_realtime = rt->start_time;
512
513 /* Remove the local codec, link to the original codec
514 * context instead, to give the rtp muxer access to
515 * codec parameters. */
516 av_free(rtpctx->streams[0]->codec);
517 rtpctx->streams[0]->codec = st->codec;
518
519 if (handle) {
520 url_fdopen(&rtpctx->pb, handle);
521 } else
522 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
523 ret = av_write_header(rtpctx);
524
525 if (ret) {
526 if (handle) {
527 url_fclose(rtpctx->pb);
528 } else {
529 uint8_t *ptr;
530 url_close_dyn_buf(rtpctx->pb, &ptr);
531 av_free(ptr);
532 }
533 av_free(rtpctx->streams[0]);
534 av_free(rtpctx);
535 return NULL;
536 }
537
538 /* Copy the RTP AVStream timebase back to the original AVStream */
539 st->time_base = rtpctx->streams[0]->time_base;
540 return rtpctx;
541 }
542
543 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
544 {
545 RTSPState *rt = s->priv_data;
546 AVStream *st = NULL;
547
548 /* open the RTP context */
549 if (rtsp_st->stream_index >= 0)
550 st = s->streams[rtsp_st->stream_index];
551 if (!st)
552 s->ctx_flags |= AVFMTCTX_NOHEADER;
553
554 if (s->oformat) {
555 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
556 /* Ownership of rtp_handle is passed to the rtp mux context */
557 rtsp_st->rtp_handle = NULL;
558 } else if (rt->transport == RTSP_TRANSPORT_RDT)
559 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
560 rtsp_st->dynamic_protocol_context,
561 rtsp_st->dynamic_handler);
562 else
563 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
564 rtsp_st->sdp_payload_type,
565 &rtsp_st->rtp_payload_data);
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 -1;
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 } else {
828 ff_rtsp_parse_line(reply, p, &rt->auth_state);
829 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
830 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
831 }
832 line_count++;
833 }
834
835 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
836 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
837
838 content_length = reply->content_length;
839 if (content_length > 0) {
840 /* leave some room for a trailing '\0' (useful for simple parsing) */
841 content = av_malloc(content_length + 1);
842 (void)url_read_complete(rt->rtsp_hd, content, content_length);
843 content[content_length] = '\0';
844 }
845 if (content_ptr)
846 *content_ptr = content;
847 else
848 av_free(content);
849
850 if (rt->seq != reply->seq) {
851 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
852 rt->seq, reply->seq);
853 }
854
855 /* EOS */
856 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
857 reply->notice == 2104 /* Start-of-Stream Reached */ ||
858 reply->notice == 2306 /* Continuous Feed Terminated */) {
859 rt->state = RTSP_STATE_IDLE;
860 } else if (reply->notice >= 4400 && reply->notice < 5500) {
861 return AVERROR(EIO); /* data or server error */
862 } else if (reply->notice == 2401 /* Ticket Expired */ ||
863 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
864 return AVERROR(EPERM);
865
866 return 0;
867 }
868
869 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
870 const char *method, const char *url,
871 const char *headers,
872 const unsigned char *send_content,
873 int send_content_length)
874 {
875 RTSPState *rt = s->priv_data;
876 char buf[4096], *out_buf;
877 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
878
879 /* Add in RTSP headers */
880 out_buf = buf;
881 rt->seq++;
882 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
883 if (headers)
884 av_strlcat(buf, headers, sizeof(buf));
885 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
886 if (rt->session_id[0] != '\0' && (!headers ||
887 !strstr(headers, "\nIf-Match:"))) {
888 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
889 }
890 if (rt->auth[0]) {
891 char *str = ff_http_auth_create_response(&rt->auth_state,
892 rt->auth, url, method);
893 if (str)
894 av_strlcat(buf, str, sizeof(buf));
895 av_free(str);
896 }
897 if (send_content_length > 0 && send_content)
898 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
899 av_strlcat(buf, "\r\n", sizeof(buf));
900
901 /* base64 encode rtsp if tunneling */
902 if (rt->control_transport == RTSP_MODE_TUNNEL) {
903 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
904 out_buf = base64buf;
905 }
906
907 dprintf(s, "Sending:\n%s--\n", buf);
908
909 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
910 if (send_content_length > 0 && send_content) {
911 if (rt->control_transport == RTSP_MODE_TUNNEL) {
912 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
913 "with content data not supported\n");
914 return AVERROR_PATCHWELCOME;
915 }
916 url_write(rt->rtsp_hd_out, send_content, send_content_length);
917 }
918 rt->last_cmd_time = av_gettime();
919
920 return 0;
921 }
922
923 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
924 const char *url, const char *headers)
925 {
926 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
927 }
928
929 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
930 const char *headers, RTSPMessageHeader *reply,
931 unsigned char **content_ptr)
932 {
933 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
934 content_ptr, NULL, 0);
935 }
936
937 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
938 const char *method, const char *url,
939 const char *header,
940 RTSPMessageHeader *reply,
941 unsigned char **content_ptr,
942 const unsigned char *send_content,
943 int send_content_length)
944 {
945 RTSPState *rt = s->priv_data;
946 HTTPAuthType cur_auth_type;
947 int ret;
948
949 retry:
950 cur_auth_type = rt->auth_state.auth_type;
951 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
952 send_content,
953 send_content_length)))
954 return ret;
955
956 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
957 return ret;
958
959 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
960 rt->auth_state.auth_type != HTTP_AUTH_NONE)
961 goto retry;
962
963 return 0;
964 }
965
966 /**
967 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
968 */
969 static int make_setup_request(AVFormatContext *s, const char *host, int port,
970 int lower_transport, const char *real_challenge)
971 {
972 RTSPState *rt = s->priv_data;
973 int rtx, j, i, err, interleave = 0;
974 RTSPStream *rtsp_st;
975 RTSPMessageHeader reply1, *reply = &reply1;
976 char cmd[2048];
977 const char *trans_pref;
978
979 if (rt->transport == RTSP_TRANSPORT_RDT)
980 trans_pref = "x-pn-tng";
981 else
982 trans_pref = "RTP/AVP";
983
984 /* default timeout: 1 minute */
985 rt->timeout = 60;
986
987 /* for each stream, make the setup request */
988 /* XXX: we assume the same server is used for the control of each
989 * RTSP stream */
990
991 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
992 char transport[2048];
993
994 /**
995 * WMS serves all UDP data over a single connection, the RTX, which
996 * isn't necessarily the first in the SDP but has to be the first
997 * to be set up, else the second/third SETUP will fail with a 461.
998 */
999 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1000 rt->server_type == RTSP_SERVER_WMS) {
1001 if (i == 0) {
1002 /* rtx first */
1003 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1004 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1005 if (len >= 4 &&
1006 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1007 "/rtx"))
1008 break;
1009 }
1010 if (rtx == rt->nb_rtsp_streams)
1011 return -1; /* no RTX found */
1012 rtsp_st = rt->rtsp_streams[rtx];
1013 } else
1014 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1015 } else
1016 rtsp_st = rt->rtsp_streams[i];
1017
1018 /* RTP/UDP */
1019 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1020 char buf[256];
1021
1022 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1023 port = reply->transports[0].client_port_min;
1024 goto have_port;
1025 }
1026
1027 /* first try in specified port range */
1028 if (RTSP_RTP_PORT_MIN != 0) {
1029 while (j <= RTSP_RTP_PORT_MAX) {
1030 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1031 "?localport=%d", j);
1032 /* we will use two ports per rtp stream (rtp and rtcp) */
1033 j += 2;
1034 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1035 goto rtp_opened;
1036 }
1037 }
1038
1039 #if 0
1040 /* then try on any port */
1041 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1042 err = AVERROR_INVALIDDATA;
1043 goto fail;
1044 }
1045 #endif
1046
1047 rtp_opened:
1048 port = rtp_get_local_port(rtsp_st->rtp_handle);
1049 have_port:
1050 snprintf(transport, sizeof(transport) - 1,
1051 "%s/UDP;", trans_pref);
1052 if (rt->server_type != RTSP_SERVER_REAL)
1053 av_strlcat(transport, "unicast;", sizeof(transport));
1054 av_strlcatf(transport, sizeof(transport),
1055 "client_port=%d", port);
1056 if (rt->transport == RTSP_TRANSPORT_RTP &&
1057 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1058 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1059 }
1060
1061 /* RTP/TCP */
1062 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1063 /** For WMS streams, the application streams are only used for
1064 * UDP. When trying to set it up for TCP streams, the server
1065 * will return an error. Therefore, we skip those streams. */
1066 if (rt->server_type == RTSP_SERVER_WMS &&
1067 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1068 AVMEDIA_TYPE_DATA)
1069 continue;
1070 snprintf(transport, sizeof(transport) - 1,
1071 "%s/TCP;", trans_pref);
1072 if (rt->server_type == RTSP_SERVER_WMS)
1073 av_strlcat(transport, "unicast;", sizeof(transport));
1074 av_strlcatf(transport, sizeof(transport),
1075 "interleaved=%d-%d",
1076 interleave, interleave + 1);
1077 interleave += 2;
1078 }
1079
1080 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1081 snprintf(transport, sizeof(transport) - 1,
1082 "%s/UDP;multicast", trans_pref);
1083 }
1084 if (s->oformat) {
1085 av_strlcat(transport, ";mode=receive", sizeof(transport));
1086 } else if (rt->server_type == RTSP_SERVER_REAL ||
1087 rt->server_type == RTSP_SERVER_WMS)
1088 av_strlcat(transport, ";mode=play", sizeof(transport));
1089 snprintf(cmd, sizeof(cmd),
1090 "Transport: %s\r\n",
1091 transport);
1092 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1093 char real_res[41], real_csum[9];
1094 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1095 real_challenge);
1096 av_strlcatf(cmd, sizeof(cmd),
1097 "If-Match: %s\r\n"
1098 "RealChallenge2: %s, sd=%s\r\n",
1099 rt->session_id, real_res, real_csum);
1100 }
1101 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1102 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1103 err = 1;
1104 goto fail;
1105 } else if (reply->status_code != RTSP_STATUS_OK ||
1106 reply->nb_transports != 1) {
1107 err = AVERROR_INVALIDDATA;
1108 goto fail;
1109 }
1110
1111 /* XXX: same protocol for all streams is required */
1112 if (i > 0) {
1113 if (reply->transports[0].lower_transport != rt->lower_transport ||
1114 reply->transports[0].transport != rt->transport) {
1115 err = AVERROR_INVALIDDATA;
1116 goto fail;
1117 }
1118 } else {
1119 rt->lower_transport = reply->transports[0].lower_transport;
1120 rt->transport = reply->transports[0].transport;
1121 }
1122
1123 /* close RTP connection if not choosen */
1124 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1125 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1126 url_close(rtsp_st->rtp_handle);
1127 rtsp_st->rtp_handle = NULL;
1128 }
1129
1130 switch(reply->transports[0].lower_transport) {
1131 case RTSP_LOWER_TRANSPORT_TCP:
1132 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1133 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1134 break;
1135
1136 case RTSP_LOWER_TRANSPORT_UDP: {
1137 char url[1024];
1138
1139 /* XXX: also use address if specified */
1140 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1141 reply->transports[0].server_port_min, NULL);
1142 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1143 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1144 err = AVERROR_INVALIDDATA;
1145 goto fail;
1146 }
1147 /* Try to initialize the connection state in a
1148 * potential NAT router by sending dummy packets.
1149 * RTP/RTCP dummy packets are used for RDT, too.
1150 */
1151 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1152 rtp_send_punch_packets(rtsp_st->rtp_handle);
1153 break;
1154 }
1155 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1156 char url[1024];
1157 struct in_addr in;
1158 int port, ttl;
1159
1160 if (reply->transports[0].destination) {
1161 in.s_addr = htonl(reply->transports[0].destination);
1162 port = reply->transports[0].port_min;
1163 ttl = reply->transports[0].ttl;
1164 } else {
1165 in = rtsp_st->sdp_ip;
1166 port = rtsp_st->sdp_port;
1167 ttl = rtsp_st->sdp_ttl;
1168 }
1169 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1170 port, "?ttl=%d", ttl);
1171 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1172 err = AVERROR_INVALIDDATA;
1173 goto fail;
1174 }
1175 break;
1176 }
1177 }
1178
1179 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1180 goto fail;
1181 }
1182
1183 if (reply->timeout > 0)
1184 rt->timeout = reply->timeout;
1185
1186 if (rt->server_type == RTSP_SERVER_REAL)
1187 rt->need_subscription = 1;
1188
1189 return 0;
1190
1191 fail:
1192 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1193 if (rt->rtsp_streams[i]->rtp_handle) {
1194 url_close(rt->rtsp_streams[i]->rtp_handle);
1195 rt->rtsp_streams[i]->rtp_handle = NULL;
1196 }
1197 }
1198 return err;
1199 }
1200
1201 static int rtsp_read_play(AVFormatContext *s)
1202 {
1203 RTSPState *rt = s->priv_data;
1204 RTSPMessageHeader reply1, *reply = &reply1;
1205 int i;
1206 char cmd[1024];
1207
1208 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1209
1210 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1211 if (rt->state == RTSP_STATE_PAUSED) {
1212 cmd[0] = 0;
1213 } else {
1214 snprintf(cmd, sizeof(cmd),
1215 "Range: npt=%0.3f-\r\n",
1216 (double)rt->seek_timestamp / AV_TIME_BASE);
1217 }
1218 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1219 if (reply->status_code != RTSP_STATUS_OK) {
1220 return -1;
1221 }
1222 if (reply->range_start != AV_NOPTS_VALUE &&
1223 rt->transport == RTSP_TRANSPORT_RTP) {
1224 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1225 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1226 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1227 AVStream *st = NULL;
1228 if (!rtpctx)
1229 continue;
1230 if (rtsp_st->stream_index >= 0)
1231 st = s->streams[rtsp_st->stream_index];
1232 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1233 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1234 if (st)
1235 rtpctx->range_start_offset = av_rescale_q(reply->range_start,
1236 AV_TIME_BASE_Q,
1237 st->time_base);
1238 }
1239 }
1240 }
1241 rt->state = RTSP_STATE_STREAMING;
1242 return 0;
1243 }
1244
1245 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1246 {
1247 RTSPState *rt = s->priv_data;
1248 char cmd[1024];
1249 unsigned char *content = NULL;
1250 int ret;
1251
1252 /* describe the stream */
1253 snprintf(cmd, sizeof(cmd),
1254 "Accept: application/sdp\r\n");
1255 if (rt->server_type == RTSP_SERVER_REAL) {
1256 /**
1257 * The Require: attribute is needed for proper streaming from
1258 * Realmedia servers.
1259 */
1260 av_strlcat(cmd,
1261 "Require: com.real.retain-entity-for-setup\r\n",
1262 sizeof(cmd));
1263 }
1264 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1265 if (!content)
1266 return AVERROR_INVALIDDATA;
1267 if (reply->status_code != RTSP_STATUS_OK) {
1268 av_freep(&content);
1269 return AVERROR_INVALIDDATA;
1270 }
1271
1272 /* now we got the SDP description, we parse it */
1273 ret = sdp_parse(s, (const char *)content);
1274 av_freep(&content);
1275 if (ret < 0)
1276 return AVERROR_INVALIDDATA;
1277
1278 return 0;
1279 }
1280
1281 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1282 {
1283 RTSPState *rt = s->priv_data;
1284 RTSPMessageHeader reply1, *reply = &reply1;
1285 int i;
1286 char *sdp;
1287 AVFormatContext sdp_ctx, *ctx_array[1];
1288
1289 rt->start_time = av_gettime();
1290
1291 /* Announce the stream */
1292 sdp = av_mallocz(8192);
1293 if (sdp == NULL)
1294 return AVERROR(ENOMEM);
1295 /* We create the SDP based on the RTSP AVFormatContext where we
1296 * aren't allowed to change the filename field. (We create the SDP
1297 * based on the RTSP context since the contexts for the RTP streams
1298 * don't exist yet.) In order to specify a custom URL with the actual
1299 * peer IP instead of the originally specified hostname, we create
1300 * a temporary copy of the AVFormatContext, where the custom URL is set.
1301 *
1302 * FIXME: Create the SDP without copying the AVFormatContext.
1303 * This either requires setting up the RTP stream AVFormatContexts
1304 * already here (complicating things immensely) or getting a more
1305 * flexible SDP creation interface.
1306 */
1307 sdp_ctx = *s;
1308 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1309 "rtsp", NULL, addr, -1, NULL);
1310 ctx_array[0] = &sdp_ctx;
1311 if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
1312 av_free(sdp);
1313 return AVERROR_INVALIDDATA;
1314 }
1315 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1316 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1317 "Content-Type: application/sdp\r\n",
1318 reply, NULL, sdp, strlen(sdp));
1319 av_free(sdp);
1320 if (reply->status_code != RTSP_STATUS_OK)
1321 return AVERROR_INVALIDDATA;
1322
1323 /* Set up the RTSPStreams for each AVStream */
1324 for (i = 0; i < s->nb_streams; i++) {
1325 RTSPStream *rtsp_st;
1326 AVStream *st = s->streams[i];
1327
1328 rtsp_st = av_mallocz(sizeof(RTSPStream));
1329 if (!rtsp_st)
1330 return AVERROR(ENOMEM);
1331 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1332
1333 st->priv_data = rtsp_st;
1334 rtsp_st->stream_index = i;
1335
1336 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1337 /* Note, this must match the relative uri set in the sdp content */
1338 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1339 "/streamid=%d", i);
1340 }
1341
1342 return 0;
1343 }
1344
1345 void ff_rtsp_close_connections(AVFormatContext *s)
1346 {
1347 RTSPState *rt = s->priv_data;
1348 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1349 url_close(rt->rtsp_hd);
1350 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1351 }
1352
1353 int ff_rtsp_connect(AVFormatContext *s)
1354 {
1355 RTSPState *rt = s->priv_data;
1356 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1357 char *option_list, *option, *filename;
1358 int port, err, tcp_fd;
1359 RTSPMessageHeader reply1 = {}, *reply = &reply1;
1360 int lower_transport_mask = 0;
1361 char real_challenge[64];
1362 struct sockaddr_storage peer;
1363 socklen_t peer_len = sizeof(peer);
1364
1365 if (!ff_network_init())
1366 return AVERROR(EIO);
1367 redirect:
1368 rt->control_transport = RTSP_MODE_PLAIN;
1369 /* extract hostname and port */
1370 ff_url_split(NULL, 0, auth, sizeof(auth),
1371 host, sizeof(host), &port, path, sizeof(path), s->filename);
1372 if (*auth) {
1373 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1374 }
1375 if (port < 0)
1376 port = RTSP_DEFAULT_PORT;
1377
1378 /* search for options */
1379 option_list = strrchr(path, '?');
1380 if (option_list) {
1381 /* Strip out the RTSP specific options, write out the rest of
1382 * the options back into the same string. */
1383 filename = option_list;
1384 while (option_list) {
1385 /* move the option pointer */
1386 option = ++option_list;
1387 option_list = strchr(option_list, '&');
1388 if (option_list)
1389 *option_list = 0;
1390
1391 /* handle the options */
1392 if (!strcmp(option, "udp")) {
1393 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1394 } else if (!strcmp(option, "multicast")) {
1395 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1396 } else if (!strcmp(option, "tcp")) {
1397 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1398 } else if(!strcmp(option, "http")) {
1399 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1400 rt->control_transport = RTSP_MODE_TUNNEL;
1401 } else {
1402 /* Write options back into the buffer, using memmove instead
1403 * of strcpy since the strings may overlap. */
1404 int len = strlen(option);
1405 memmove(++filename, option, len);
1406 filename += len;
1407 if (option_list) *filename = '&';
1408 }
1409 }
1410 *filename = 0;
1411 }
1412
1413 if (!lower_transport_mask)
1414 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1415
1416 if (s->oformat) {
1417 /* Only UDP or TCP - UDP multicast isn't supported. */
1418 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1419 (1 << RTSP_LOWER_TRANSPORT_TCP);
1420 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1421 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1422 "only UDP and TCP are supported for output.\n");
1423 err = AVERROR(EINVAL);
1424 goto fail;
1425 }
1426 }
1427
1428 /* Construct the URI used in request; this is similar to s->filename,
1429 * but with authentication credentials removed and RTSP specific options
1430 * stripped out. */
1431 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1432 host, port, "%s", path);
1433
1434 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1435 /* set up initial handshake for tunneling */
1436 char httpname[1024];
1437 char sessioncookie[17];
1438 char headers[1024];
1439
1440 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1441 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1442 av_get_random_seed(), av_get_random_seed());
1443
1444 /* GET requests */
1445 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1446 err = AVERROR(EIO);
1447 goto fail;
1448 }
1449
1450 /* generate GET headers */
1451 snprintf(headers, sizeof(headers),
1452 "x-sessioncookie: %s\r\n"
1453 "Accept: application/x-rtsp-tunnelled\r\n"
1454 "Pragma: no-cache\r\n"
1455 "Cache-Control: no-cache\r\n",
1456 sessioncookie);
1457 ff_http_set_headers(rt->rtsp_hd, headers);
1458
1459 /* complete the connection */
1460 if (url_connect(rt->rtsp_hd)) {
1461 err = AVERROR(EIO);
1462 goto fail;
1463 }
1464
1465 /* POST requests */
1466 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1467 err = AVERROR(EIO);
1468 goto fail;
1469 }
1470
1471 /* generate POST headers */
1472 snprintf(headers, sizeof(headers),
1473 "x-sessioncookie: %s\r\n"
1474 "Content-Type: application/x-rtsp-tunnelled\r\n"
1475 "Pragma: no-cache\r\n"
1476 "Cache-Control: no-cache\r\n"
1477 "Content-Length: 32767\r\n"
1478 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1479 sessioncookie);
1480 ff_http_set_headers(rt->rtsp_hd_out, headers);
1481 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1482
1483 /* Initialize the authentication state for the POST session. The HTTP
1484 * protocol implementation doesn't properly handle multi-pass
1485 * authentication for POST requests, since it would require one of
1486 * the following:
1487 * - implementing Expect: 100-continue, which many HTTP servers
1488 * don't support anyway, even less the RTSP servers that do HTTP
1489 * tunneling
1490 * - sending the whole POST data until getting a 401 reply specifying
1491 * what authentication method to use, then resending all that data
1492 * - waiting for potential 401 replies directly after sending the
1493 * POST header (waiting for some unspecified time)
1494 * Therefore, we copy the full auth state, which works for both basic
1495 * and digest. (For digest, we would have to synchronize the nonce
1496 * count variable between the two sessions, if we'd do more requests
1497 * with the original session, though.)
1498 */
1499 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1500
1501 /* complete the connection */
1502 if (url_connect(rt->rtsp_hd_out)) {
1503 err = AVERROR(EIO);
1504 goto fail;
1505 }
1506 } else {
1507 /* open the tcp connection */
1508 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1509 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1510 err = AVERROR(EIO);
1511 goto fail;
1512 }
1513 rt->rtsp_hd_out = rt->rtsp_hd;
1514 }
1515 rt->seq = 0;
1516
1517 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1518 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1519 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1520 NULL, 0, NI_NUMERICHOST);
1521 }
1522
1523 /* request options supported by the server; this also detects server
1524 * type */
1525 for (rt->server_type = RTSP_SERVER_RTP;;) {
1526 cmd[0] = 0;
1527 if (rt->server_type == RTSP_SERVER_REAL)
1528 av_strlcat(cmd,
1529 /**
1530 * The following entries are required for proper
1531 * streaming from a Realmedia server. They are
1532 * interdependent in some way although we currently
1533 * don't quite understand how. Values were copied
1534 * from mplayer SVN r23589.
1535 * @param CompanyID is a 16-byte ID in base64
1536 * @param ClientChallenge is a 16-byte ID in hex
1537 */
1538 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1539 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1540 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1541 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1542 sizeof(cmd));
1543 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1544 if (reply->status_code != RTSP_STATUS_OK) {
1545 err = AVERROR_INVALIDDATA;
1546 goto fail;
1547 }
1548
1549 /* detect server type if not standard-compliant RTP */
1550 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1551 rt->server_type = RTSP_SERVER_REAL;
1552 continue;
1553 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1554 rt->server_type = RTSP_SERVER_WMS;
1555 } else if (rt->server_type == RTSP_SERVER_REAL)
1556 strcpy(real_challenge, reply->real_challenge);
1557 break;
1558 }
1559
1560 if (s->iformat)
1561 err = rtsp_setup_input_streams(s, reply);
1562 else
1563 err = rtsp_setup_output_streams(s, host);
1564 if (err)
1565 goto fail;
1566
1567 do {
1568 int lower_transport = ff_log2_tab[lower_transport_mask &
1569 ~(lower_transport_mask - 1)];
1570
1571 err = make_setup_request(s, host, port, lower_transport,
1572 rt->server_type == RTSP_SERVER_REAL ?
1573 real_challenge : NULL);
1574 if (err < 0)
1575 goto fail;
1576 lower_transport_mask &= ~(1 << lower_transport);
1577 if (lower_transport_mask == 0 && err == 1) {
1578 err = FF_NETERROR(EPROTONOSUPPORT);
1579 goto fail;
1580 }
1581 } while (err);
1582
1583 rt->state = RTSP_STATE_IDLE;
1584 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1585 return 0;
1586 fail:
1587 ff_rtsp_close_streams(s);
1588 ff_rtsp_close_connections(s);
1589 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1590 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1591 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1592 reply->status_code,
1593 s->filename);
1594 goto redirect;
1595 }
1596 ff_network_close();
1597 return err;
1598 }
1599 #endif
1600
1601 #if CONFIG_RTSP_DEMUXER
1602 static int rtsp_read_header(AVFormatContext *s,
1603 AVFormatParameters *ap)
1604 {
1605 int ret;
1606
1607 ret = ff_rtsp_connect(s);
1608 if (ret)
1609 return ret;
1610
1611 if (ap->initial_pause) {
1612 /* do not start immediately */
1613 } else {
1614 if (rtsp_read_play(s) < 0) {
1615 ff_rtsp_close_streams(s);
1616 ff_rtsp_close_connections(s);
1617 return AVERROR_INVALIDDATA;
1618 }
1619 }
1620
1621 return 0;
1622 }
1623
1624 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1625 uint8_t *buf, int buf_size)
1626 {
1627 RTSPState *rt = s->priv_data;
1628 RTSPStream *rtsp_st;
1629 fd_set rfds;
1630 int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1631 struct timeval tv;
1632
1633 for (;;) {
1634 if (url_interrupt_cb())
1635 return AVERROR(EINTR);
1636 FD_ZERO(&rfds);
1637 if (rt->rtsp_hd) {
1638 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1639 FD_SET(tcp_fd, &rfds);
1640 } else {
1641 fd_max = 0;
1642 tcp_fd = -1;
1643 }
1644 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1645 rtsp_st = rt->rtsp_streams[i];
1646 if (rtsp_st->rtp_handle) {
1647 /* currently, we cannot probe RTCP handle because of
1648 * blocking restrictions */
1649 fd = url_get_file_handle(rtsp_st->rtp_handle);
1650 if (fd > fd_max)
1651 fd_max = fd;
1652 FD_SET(fd, &rfds);
1653 }
1654 }
1655 tv.tv_sec = 0;
1656 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1657 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1658 if (n > 0) {
1659 timeout_cnt = 0;
1660 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1661 rtsp_st = rt->rtsp_streams[i];
1662 if (rtsp_st->rtp_handle) {
1663 fd = url_get_file_handle(rtsp_st->rtp_handle);
1664 if (FD_ISSET(fd, &rfds)) {
1665 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1666 if (ret > 0) {
1667 *prtsp_st = rtsp_st;
1668 return ret;
1669 }
1670 }
1671 }
1672 }
1673 #if CONFIG_RTSP_DEMUXER
1674 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1675 RTSPMessageHeader reply;
1676
1677 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1678 if (ret < 0)
1679 return ret;
1680 /* XXX: parse message */
1681 if (rt->state != RTSP_STATE_STREAMING)
1682 return 0;
1683 }
1684 #endif
1685 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1686 return FF_NETERROR(ETIMEDOUT);
1687 } else if (n < 0 && errno != EINTR)
1688 return AVERROR(errno);
1689 }
1690 }
1691
1692 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1693 uint8_t *buf, int buf_size)
1694 {
1695 RTSPState *rt = s->priv_data;
1696 int id, len, i, ret;
1697 RTSPStream *rtsp_st;
1698
1699 #ifdef DEBUG_RTP_TCP
1700 dprintf(s, "tcp_read_packet:\n");
1701 #endif
1702 redo:
1703 for (;;) {
1704 RTSPMessageHeader reply;
1705
1706 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1707 if (ret == -1)
1708 return -1;
1709 if (ret == 1) /* received '$' */
1710 break;
1711 /* XXX: parse message */
1712 if (rt->state != RTSP_STATE_STREAMING)
1713 return 0;
1714 }
1715 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1716 if (ret != 3)
1717 return -1;
1718 id = buf[0];
1719 len = AV_RB16(buf + 1);
1720 #ifdef DEBUG_RTP_TCP
1721 dprintf(s, "id=%d len=%d\n", id, len);
1722 #endif
1723 if (len > buf_size || len < 12)
1724 goto redo;
1725 /* get the data */
1726 ret = url_read_complete(rt->rtsp_hd, buf, len);
1727 if (ret != len)
1728 return -1;
1729 if (rt->transport == RTSP_TRANSPORT_RDT &&
1730 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1731 return -1;
1732
1733 /* find the matching stream */
1734 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1735 rtsp_st = rt->rtsp_streams[i];
1736 if (id >= rtsp_st->interleaved_min &&
1737 id <= rtsp_st->interleaved_max)
1738 goto found;
1739 }
1740 goto redo;
1741 found:
1742 *prtsp_st = rtsp_st;
1743 return len;
1744 }
1745
1746 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1747 {
1748 RTSPState *rt = s->priv_data;
1749 int ret, len;
1750 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1751 RTSPStream *rtsp_st;
1752
1753 /* get next frames from the same RTP packet */
1754 if (rt->cur_transport_priv) {
1755 if (rt->transport == RTSP_TRANSPORT_RDT) {
1756 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1757 } else
1758 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1759 if (ret == 0) {
1760 rt->cur_transport_priv = NULL;
1761 return 0;
1762 } else if (ret == 1) {
1763 return 0;
1764 } else
1765 rt->cur_transport_priv = NULL;
1766 }
1767
1768 /* read next RTP packet */
1769 redo:
1770 switch(rt->lower_transport) {
1771 default:
1772 #if CONFIG_RTSP_DEMUXER
1773 case RTSP_LOWER_TRANSPORT_TCP:
1774 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1775 break;
1776 #endif
1777 case RTSP_LOWER_TRANSPORT_UDP:
1778 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1779 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1780 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1781 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1782 break;
1783 }
1784 if (len < 0)
1785 return len;
1786 if (len == 0)
1787 return AVERROR_EOF;
1788 if (rt->transport == RTSP_TRANSPORT_RDT) {
1789 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1790 } else {
1791 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1792 if (ret < 0) {
1793 /* Either bad packet, or a RTCP packet. Check if the
1794 * first_rtcp_ntp_time field was initialized. */
1795 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1796 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1797 /* first_rtcp_ntp_time has been initialized for this stream,
1798 * copy the same value to all other uninitialized streams,
1799 * in order to map their timestamp origin to the same ntp time
1800 * as this one. */
1801 int i;
1802 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1803 RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
1804 if (rtpctx2 &&
1805 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1806 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1807 }
1808 }
1809 }
1810 }
1811 if (ret < 0)
1812 goto redo;
1813 if (ret == 1)
1814 /* more packets may follow, so we save the RTP context */
1815 rt->cur_transport_priv = rtsp_st->transport_priv;
1816
1817 return ret;
1818 }
1819
1820 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1821 {
1822 RTSPState *rt = s->priv_data;
1823 int ret;
1824 RTSPMessageHeader reply1, *reply = &reply1;
1825 char cmd[1024];
1826
1827 if (rt->server_type == RTSP_SERVER_REAL) {
1828 int i;
1829 enum AVDiscard cache[MAX_STREAMS];
1830
1831 for (i = 0; i < s->nb_streams; i++)
1832 cache[i] = s->streams[i]->discard;
1833
1834 if (!rt->need_subscription) {
1835 if (memcmp (cache, rt->real_setup_cache,
1836 sizeof(enum AVDiscard) * s->nb_streams)) {
1837 snprintf(cmd, sizeof(cmd),
1838 "Unsubscribe: %s\r\n",
1839 rt->last_subscription);
1840 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1841 cmd, reply, NULL);
1842 if (reply->status_code != RTSP_STATUS_OK)
1843 return AVERROR_INVALIDDATA;
1844 rt->need_subscription = 1;
1845 }
1846 }
1847
1848 if (rt->need_subscription) {
1849 int r, rule_nr, first = 1;
1850
1851 memcpy(rt->real_setup_cache, cache,
1852 sizeof(enum AVDiscard) * s->nb_streams);
1853 rt->last_subscription[0] = 0;
1854
1855 snprintf(cmd, sizeof(cmd),
1856 "Subscribe: ");
1857 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1858 rule_nr = 0;
1859 for (r = 0; r < s->nb_streams; r++) {
1860 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1861 if (s->streams[r]->discard != AVDISCARD_ALL) {
1862 if (!first)
1863 av_strlcat(rt->last_subscription, ",",
1864 sizeof(rt->last_subscription));
1865 ff_rdt_subscribe_rule(
1866 rt->last_subscription,
1867 sizeof(rt->last_subscription), i, rule_nr);
1868 first = 0;
1869 }
1870 rule_nr++;
1871 }
1872 }
1873 }
1874 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1875 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1876 cmd, reply, NULL);
1877 if (reply->status_code != RTSP_STATUS_OK)
1878 return AVERROR_INVALIDDATA;
1879 rt->need_subscription = 0;
1880
1881 if (rt->state == RTSP_STATE_STREAMING)
1882 rtsp_read_play (s);
1883 }
1884 }
1885
1886 ret = rtsp_fetch_packet(s, pkt);
1887 if (ret < 0)
1888 return ret;
1889
1890 /* send dummy request to keep TCP connection alive */
1891 if ((rt->server_type == RTSP_SERVER_WMS ||
1892 rt->server_type == RTSP_SERVER_REAL) &&
1893 (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1894 if (rt->server_type == RTSP_SERVER_WMS) {
1895 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1896 } else {
1897 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1898 }
1899 }
1900
1901 return 0;
1902 }
1903
1904 /* pause the stream */
1905 static int rtsp_read_pause(AVFormatContext *s)
1906 {
1907 RTSPState *rt = s->priv_data;
1908 RTSPMessageHeader reply1, *reply = &reply1;
1909
1910 if (rt->state != RTSP_STATE_STREAMING)
1911 return 0;
1912 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1913 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1914 if (reply->status_code != RTSP_STATUS_OK) {
1915 return -1;
1916 }
1917 }
1918 rt->state = RTSP_STATE_PAUSED;
1919 return 0;
1920 }
1921
1922 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1923 int64_t timestamp, int flags)
1924 {
1925 RTSPState *rt = s->priv_data;
1926
1927 rt->seek_timestamp = av_rescale_q(timestamp,
1928 s->streams[stream_index]->time_base,
1929 AV_TIME_BASE_Q);
1930 switch(rt->state) {
1931 default:
1932 case RTSP_STATE_IDLE:
1933 break;
1934 case RTSP_STATE_STREAMING:
1935 if (rtsp_read_pause(s) != 0)
1936 return -1;
1937 rt->state = RTSP_STATE_SEEKING;
1938 if (rtsp_read_play(s) != 0)
1939 return -1;
1940 break;
1941 case RTSP_STATE_PAUSED:
1942 rt->state = RTSP_STATE_IDLE;
1943 break;
1944 }
1945 return 0;
1946 }
1947
1948 static int rtsp_read_close(AVFormatContext *s)
1949 {
1950 RTSPState *rt = s->priv_data;
1951
1952 #if 0
1953 /* NOTE: it is valid to flush the buffer here */
1954 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1955 url_fclose(&rt->rtsp_gb);
1956 }
1957 #endif
1958 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1959
1960 ff_rtsp_close_streams(s);
1961 ff_rtsp_close_connections(s);
1962 ff_network_close();
1963 return 0;
1964 }
1965
1966 AVInputFormat rtsp_demuxer = {
1967 "rtsp",
1968 NULL_IF_CONFIG_SMALL("RTSP input format"),
1969 sizeof(RTSPState),
1970 rtsp_probe,
1971 rtsp_read_header,
1972 rtsp_read_packet,
1973 rtsp_read_close,
1974 rtsp_read_seek,
1975 .flags = AVFMT_NOFILE,
1976 .read_play = rtsp_read_play,
1977 .read_pause = rtsp_read_pause,
1978 };
1979 #endif
1980
1981 static int sdp_probe(AVProbeData *p1)
1982 {
1983 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1984
1985 /* we look for a line beginning "c=IN IP4" */
1986 while (p < p_end && *p != '\0') {
1987 if (p + sizeof("c=IN IP4") - 1 < p_end &&
1988 av_strstart(p, "c=IN IP4", NULL))
1989 return AVPROBE_SCORE_MAX / 2;
1990
1991 while (p < p_end - 1 && *p != '\n') p++;
1992 if (++p >= p_end)
1993 break;
1994 if (*p == '\r')
1995 p++;
1996 }
1997 return 0;
1998 }
1999
2000 #define SDP_MAX_SIZE 8192
2001
2002 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2003 {
2004 RTSPState *rt = s->priv_data;
2005 RTSPStream *rtsp_st;
2006 int size, i, err;
2007 char *content;
2008 char url[1024];
2009
2010 if (!ff_network_init())
2011 return AVERROR(EIO);
2012
2013 /* read the whole sdp file */
2014 /* XXX: better loading */
2015 content = av_malloc(SDP_MAX_SIZE);
2016 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2017 if (size <= 0) {
2018 av_free(content);
2019 return AVERROR_INVALIDDATA;
2020 }
2021 content[size] ='\0';
2022
2023 sdp_parse(s, content);
2024 av_free(content);
2025
2026 /* open each RTP stream */
2027 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2028 rtsp_st = rt->rtsp_streams[i];
2029
2030 ff_url_join(url, sizeof(url), "rtp", NULL,
2031 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2032 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2033 rtsp_st->sdp_ttl);
2034 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2035 err = AVERROR_INVALIDDATA;
2036 goto fail;
2037 }
2038 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2039 goto fail;
2040 }
2041 return 0;
2042 fail:
2043 ff_rtsp_close_streams(s);
2044 ff_network_close();
2045 return err;
2046 }
2047
2048 static int sdp_read_close(AVFormatContext *s)
2049 {
2050 ff_rtsp_close_streams(s);
2051 ff_network_close();
2052 return 0;
2053 }
2054
2055 AVInputFormat sdp_demuxer = {
2056 "sdp",
2057 NULL_IF_CONFIG_SMALL("SDP"),
2058 sizeof(RTSPState),
2059 sdp_probe,
2060 sdp_read_header,
2061 rtsp_fetch_packet,
2062 sdp_read_close,
2063 };