3ddf33ce644c84c7eb6b0efba3fca59bab8d01db
[libav.git] / libavformat / rtsp.c
1 /*
2 * RTSP/SDP client
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav 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 * Libav 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 Libav; 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/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
31 #include "avformat.h"
32 #include "avio_internal.h"
33
34 #if HAVE_POLL_H
35 #include <poll.h>
36 #endif
37 #include "internal.h"
38 #include "network.h"
39 #include "os_support.h"
40 #include "http.h"
41 #include "rtsp.h"
42
43 #include "rtpdec.h"
44 #include "rdt.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
47 #include "url.h"
48 #include "rtpenc.h"
49 #include "mpegts.h"
50
51 //#define DEBUG
52
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
61
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
70
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76
77 const AVOption ff_rtsp_options[] = {
78 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
79 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
80 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
81 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
82 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
84 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
85 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
86 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
88 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
89 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
90 { NULL },
91 };
92
93 static const AVOption sdp_options[] = {
94 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
95 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
96 { NULL },
97 };
98
99 static const AVOption rtp_options[] = {
100 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
101 { NULL },
102 };
103
104 static void get_word_until_chars(char *buf, int buf_size,
105 const char *sep, const char **pp)
106 {
107 const char *p;
108 char *q;
109
110 p = *pp;
111 p += strspn(p, SPACE_CHARS);
112 q = buf;
113 while (!strchr(sep, *p) && *p != '\0') {
114 if ((q - buf) < buf_size - 1)
115 *q++ = *p;
116 p++;
117 }
118 if (buf_size > 0)
119 *q = '\0';
120 *pp = p;
121 }
122
123 static void get_word_sep(char *buf, int buf_size, const char *sep,
124 const char **pp)
125 {
126 if (**pp == '/') (*pp)++;
127 get_word_until_chars(buf, buf_size, sep, pp);
128 }
129
130 static void get_word(char *buf, int buf_size, const char **pp)
131 {
132 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
133 }
134
135 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
136 * and end time.
137 * Used for seeking in the rtp stream.
138 */
139 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
140 {
141 char buf[256];
142
143 p += strspn(p, SPACE_CHARS);
144 if (!av_stristart(p, "npt=", &p))
145 return;
146
147 *start = AV_NOPTS_VALUE;
148 *end = AV_NOPTS_VALUE;
149
150 get_word_sep(buf, sizeof(buf), "-", &p);
151 av_parse_time(start, buf, 1);
152 if (*p == '-') {
153 p++;
154 get_word_sep(buf, sizeof(buf), "-", &p);
155 av_parse_time(end, buf, 1);
156 }
157 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
158 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
159 }
160
161 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
162 {
163 struct addrinfo hints = { 0 }, *ai = NULL;
164 hints.ai_flags = AI_NUMERICHOST;
165 if (getaddrinfo(buf, NULL, &hints, &ai))
166 return -1;
167 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
168 freeaddrinfo(ai);
169 return 0;
170 }
171
172 #if CONFIG_RTPDEC
173 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
174 RTSPStream *rtsp_st, AVCodecContext *codec)
175 {
176 if (!handler)
177 return;
178 codec->codec_id = handler->codec_id;
179 rtsp_st->dynamic_handler = handler;
180 if (handler->alloc) {
181 rtsp_st->dynamic_protocol_context = handler->alloc();
182 if (!rtsp_st->dynamic_protocol_context)
183 rtsp_st->dynamic_handler = NULL;
184 }
185 }
186
187 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
188 static int sdp_parse_rtpmap(AVFormatContext *s,
189 AVStream *st, RTSPStream *rtsp_st,
190 int payload_type, const char *p)
191 {
192 AVCodecContext *codec = st->codec;
193 char buf[256];
194 int i;
195 AVCodec *c;
196 const char *c_name;
197
198 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
199 * see if we can handle this kind of payload.
200 * The space should normally not be there but some Real streams or
201 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
202 * have a trailing space. */
203 get_word_sep(buf, sizeof(buf), "/ ", &p);
204 if (payload_type < RTP_PT_PRIVATE) {
205 /* We are in a standard case
206 * (from http://www.iana.org/assignments/rtp-parameters). */
207 /* search into AVRtpPayloadTypes[] */
208 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
209 }
210
211 if (codec->codec_id == AV_CODEC_ID_NONE) {
212 RTPDynamicProtocolHandler *handler =
213 ff_rtp_handler_find_by_name(buf, codec->codec_type);
214 init_rtp_handler(handler, rtsp_st, codec);
215 /* If no dynamic handler was found, check with the list of standard
216 * allocated types, if such a stream for some reason happens to
217 * use a private payload type. This isn't handled in rtpdec.c, since
218 * the format name from the rtpmap line never is passed into rtpdec. */
219 if (!rtsp_st->dynamic_handler)
220 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
221 }
222
223 c = avcodec_find_decoder(codec->codec_id);
224 if (c && c->name)
225 c_name = c->name;
226 else
227 c_name = "(null)";
228
229 get_word_sep(buf, sizeof(buf), "/", &p);
230 i = atoi(buf);
231 switch (codec->codec_type) {
232 case AVMEDIA_TYPE_AUDIO:
233 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
234 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
235 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
236 if (i > 0) {
237 codec->sample_rate = i;
238 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
239 get_word_sep(buf, sizeof(buf), "/", &p);
240 i = atoi(buf);
241 if (i > 0)
242 codec->channels = i;
243 // TODO: there is a bug here; if it is a mono stream, and
244 // less than 22000Hz, faad upconverts to stereo and twice
245 // the frequency. No problem, but the sample rate is being
246 // set here by the sdp line. Patch on its way. (rdm)
247 }
248 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
249 codec->sample_rate);
250 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251 codec->channels);
252 break;
253 case AVMEDIA_TYPE_VIDEO:
254 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
255 if (i > 0)
256 avpriv_set_pts_info(st, 32, 1, i);
257 break;
258 default:
259 break;
260 }
261 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262 rtsp_st->dynamic_handler->init(s, st->index,
263 rtsp_st->dynamic_protocol_context);
264 return 0;
265 }
266
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
269 * forthcoming. */
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271 char *value, int value_size)
272 {
273 *p += strspn(*p, SPACE_CHARS);
274 if (**p) {
275 get_word_sep(attr, attr_size, "=", p);
276 if (**p == '=')
277 (*p)++;
278 get_word_sep(value, value_size, ";", p);
279 if (**p == ';')
280 (*p)++;
281 return 1;
282 }
283 return 0;
284 }
285
286 typedef struct SDPParseState {
287 /* SDP only */
288 struct sockaddr_storage default_ip;
289 int default_ttl;
290 int skip_media; ///< set if an unknown m= line occurs
291 } SDPParseState;
292
293 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
294 int letter, const char *buf)
295 {
296 RTSPState *rt = s->priv_data;
297 char buf1[64], st_type[64];
298 const char *p;
299 enum AVMediaType codec_type;
300 int payload_type, i;
301 AVStream *st;
302 RTSPStream *rtsp_st;
303 struct sockaddr_storage sdp_ip;
304 int ttl;
305
306 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
307
308 p = buf;
309 if (s1->skip_media && letter != 'm')
310 return;
311 switch (letter) {
312 case 'c':
313 get_word(buf1, sizeof(buf1), &p);
314 if (strcmp(buf1, "IN") != 0)
315 return;
316 get_word(buf1, sizeof(buf1), &p);
317 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
318 return;
319 get_word_sep(buf1, sizeof(buf1), "/", &p);
320 if (get_sockaddr(buf1, &sdp_ip))
321 return;
322 ttl = 16;
323 if (*p == '/') {
324 p++;
325 get_word_sep(buf1, sizeof(buf1), "/", &p);
326 ttl = atoi(buf1);
327 }
328 if (s->nb_streams == 0) {
329 s1->default_ip = sdp_ip;
330 s1->default_ttl = ttl;
331 } else {
332 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
333 rtsp_st->sdp_ip = sdp_ip;
334 rtsp_st->sdp_ttl = ttl;
335 }
336 break;
337 case 's':
338 av_dict_set(&s->metadata, "title", p, 0);
339 break;
340 case 'i':
341 if (s->nb_streams == 0) {
342 av_dict_set(&s->metadata, "comment", p, 0);
343 break;
344 }
345 break;
346 case 'm':
347 /* new stream */
348 s1->skip_media = 0;
349 codec_type = AVMEDIA_TYPE_UNKNOWN;
350 get_word(st_type, sizeof(st_type), &p);
351 if (!strcmp(st_type, "audio")) {
352 codec_type = AVMEDIA_TYPE_AUDIO;
353 } else if (!strcmp(st_type, "video")) {
354 codec_type = AVMEDIA_TYPE_VIDEO;
355 } else if (!strcmp(st_type, "application")) {
356 codec_type = AVMEDIA_TYPE_DATA;
357 }
358 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
359 s1->skip_media = 1;
360 return;
361 }
362 rtsp_st = av_mallocz(sizeof(RTSPStream));
363 if (!rtsp_st)
364 return;
365 rtsp_st->stream_index = -1;
366 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
367
368 rtsp_st->sdp_ip = s1->default_ip;
369 rtsp_st->sdp_ttl = s1->default_ttl;
370
371 get_word(buf1, sizeof(buf1), &p); /* port */
372 rtsp_st->sdp_port = atoi(buf1);
373
374 get_word(buf1, sizeof(buf1), &p); /* protocol */
375 if (!strcmp(buf1, "udp"))
376 rt->transport = RTSP_TRANSPORT_RAW;
377
378 /* XXX: handle list of formats */
379 get_word(buf1, sizeof(buf1), &p); /* format list */
380 rtsp_st->sdp_payload_type = atoi(buf1);
381
382 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
383 /* no corresponding stream */
384 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
385 rt->ts = ff_mpegts_parse_open(s);
386 } else if (rt->server_type == RTSP_SERVER_WMS &&
387 codec_type == AVMEDIA_TYPE_DATA) {
388 /* RTX stream, a stream that carries all the other actual
389 * audio/video streams. Don't expose this to the callers. */
390 } else {
391 st = avformat_new_stream(s, NULL);
392 if (!st)
393 return;
394 st->id = rt->nb_rtsp_streams - 1;
395 rtsp_st->stream_index = st->index;
396 st->codec->codec_type = codec_type;
397 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
398 RTPDynamicProtocolHandler *handler;
399 /* if standard payload type, we can find the codec right now */
400 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
401 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
402 st->codec->sample_rate > 0)
403 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
404 /* Even static payload types may need a custom depacketizer */
405 handler = ff_rtp_handler_find_by_id(
406 rtsp_st->sdp_payload_type, st->codec->codec_type);
407 init_rtp_handler(handler, rtsp_st, st->codec);
408 if (handler && handler->init)
409 handler->init(s, st->index,
410 rtsp_st->dynamic_protocol_context);
411 }
412 }
413 /* put a default control url */
414 av_strlcpy(rtsp_st->control_url, rt->control_uri,
415 sizeof(rtsp_st->control_url));
416 break;
417 case 'a':
418 if (av_strstart(p, "control:", &p)) {
419 if (s->nb_streams == 0) {
420 if (!strncmp(p, "rtsp://", 7))
421 av_strlcpy(rt->control_uri, p,
422 sizeof(rt->control_uri));
423 } else {
424 char proto[32];
425 /* get the control url */
426 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
427
428 /* XXX: may need to add full url resolution */
429 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
430 NULL, NULL, 0, p);
431 if (proto[0] == '\0') {
432 /* relative control URL */
433 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
434 av_strlcat(rtsp_st->control_url, "/",
435 sizeof(rtsp_st->control_url));
436 av_strlcat(rtsp_st->control_url, p,
437 sizeof(rtsp_st->control_url));
438 } else
439 av_strlcpy(rtsp_st->control_url, p,
440 sizeof(rtsp_st->control_url));
441 }
442 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
443 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
444 get_word(buf1, sizeof(buf1), &p);
445 payload_type = atoi(buf1);
446 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
447 if (rtsp_st->stream_index >= 0) {
448 st = s->streams[rtsp_st->stream_index];
449 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
450 }
451 } else if (av_strstart(p, "fmtp:", &p) ||
452 av_strstart(p, "framesize:", &p)) {
453 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
454 // let dynamic protocol handlers have a stab at the line.
455 get_word(buf1, sizeof(buf1), &p);
456 payload_type = atoi(buf1);
457 for (i = 0; i < rt->nb_rtsp_streams; i++) {
458 rtsp_st = rt->rtsp_streams[i];
459 if (rtsp_st->sdp_payload_type == payload_type &&
460 rtsp_st->dynamic_handler &&
461 rtsp_st->dynamic_handler->parse_sdp_a_line)
462 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
463 rtsp_st->dynamic_protocol_context, buf);
464 }
465 } else if (av_strstart(p, "range:", &p)) {
466 int64_t start, end;
467
468 // this is so that seeking on a streamed file can work.
469 rtsp_parse_range_npt(p, &start, &end);
470 s->start_time = start;
471 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
472 s->duration = (end == AV_NOPTS_VALUE) ?
473 AV_NOPTS_VALUE : end - start;
474 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
475 if (atoi(p) == 1)
476 rt->transport = RTSP_TRANSPORT_RDT;
477 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
478 s->nb_streams > 0) {
479 st = s->streams[s->nb_streams - 1];
480 st->codec->sample_rate = atoi(p);
481 } else {
482 if (rt->server_type == RTSP_SERVER_WMS)
483 ff_wms_parse_sdp_a_line(s, p);
484 if (s->nb_streams > 0) {
485 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
486
487 if (rt->server_type == RTSP_SERVER_REAL)
488 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
489
490 if (rtsp_st->dynamic_handler &&
491 rtsp_st->dynamic_handler->parse_sdp_a_line)
492 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
493 rtsp_st->stream_index,
494 rtsp_st->dynamic_protocol_context, buf);
495 }
496 }
497 break;
498 }
499 }
500
501 int ff_sdp_parse(AVFormatContext *s, const char *content)
502 {
503 RTSPState *rt = s->priv_data;
504 const char *p;
505 int letter;
506 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
507 * contain long SDP lines containing complete ASF Headers (several
508 * kB) or arrays of MDPR (RM stream descriptor) headers plus
509 * "rulebooks" describing their properties. Therefore, the SDP line
510 * buffer is large.
511 *
512 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
513 * in rtpdec_xiph.c. */
514 char buf[16384], *q;
515 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
516
517 p = content;
518 for (;;) {
519 p += strspn(p, SPACE_CHARS);
520 letter = *p;
521 if (letter == '\0')
522 break;
523 p++;
524 if (*p != '=')
525 goto next_line;
526 p++;
527 /* get the content */
528 q = buf;
529 while (*p != '\n' && *p != '\r' && *p != '\0') {
530 if ((q - buf) < sizeof(buf) - 1)
531 *q++ = *p;
532 p++;
533 }
534 *q = '\0';
535 sdp_parse_line(s, s1, letter, buf);
536 next_line:
537 while (*p != '\n' && *p != '\0')
538 p++;
539 if (*p == '\n')
540 p++;
541 }
542 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
543 if (!rt->p) return AVERROR(ENOMEM);
544 return 0;
545 }
546 #endif /* CONFIG_RTPDEC */
547
548 void ff_rtsp_undo_setup(AVFormatContext *s)
549 {
550 RTSPState *rt = s->priv_data;
551 int i;
552
553 for (i = 0; i < rt->nb_rtsp_streams; i++) {
554 RTSPStream *rtsp_st = rt->rtsp_streams[i];
555 if (!rtsp_st)
556 continue;
557 if (rtsp_st->transport_priv) {
558 if (s->oformat) {
559 AVFormatContext *rtpctx = rtsp_st->transport_priv;
560 av_write_trailer(rtpctx);
561 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
562 uint8_t *ptr;
563 avio_close_dyn_buf(rtpctx->pb, &ptr);
564 av_free(ptr);
565 } else {
566 avio_close(rtpctx->pb);
567 }
568 avformat_free_context(rtpctx);
569 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
570 ff_rdt_parse_close(rtsp_st->transport_priv);
571 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
572 ff_rtp_parse_close(rtsp_st->transport_priv);
573 }
574 rtsp_st->transport_priv = NULL;
575 if (rtsp_st->rtp_handle)
576 ffurl_close(rtsp_st->rtp_handle);
577 rtsp_st->rtp_handle = NULL;
578 }
579 }
580
581 /* close and free RTSP streams */
582 void ff_rtsp_close_streams(AVFormatContext *s)
583 {
584 RTSPState *rt = s->priv_data;
585 int i;
586 RTSPStream *rtsp_st;
587
588 ff_rtsp_undo_setup(s);
589 for (i = 0; i < rt->nb_rtsp_streams; i++) {
590 rtsp_st = rt->rtsp_streams[i];
591 if (rtsp_st) {
592 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
593 rtsp_st->dynamic_handler->free(
594 rtsp_st->dynamic_protocol_context);
595 av_free(rtsp_st);
596 }
597 }
598 av_free(rt->rtsp_streams);
599 if (rt->asf_ctx) {
600 avformat_close_input(&rt->asf_ctx);
601 }
602 if (rt->ts && CONFIG_RTPDEC)
603 ff_mpegts_parse_close(rt->ts);
604 av_free(rt->p);
605 av_free(rt->recvbuf);
606 }
607
608 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
609 {
610 RTSPState *rt = s->priv_data;
611 AVStream *st = NULL;
612
613 /* open the RTP context */
614 if (rtsp_st->stream_index >= 0)
615 st = s->streams[rtsp_st->stream_index];
616 if (!st)
617 s->ctx_flags |= AVFMTCTX_NOHEADER;
618
619 if (s->oformat && CONFIG_RTSP_MUXER) {
620 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
621 rtsp_st->rtp_handle,
622 RTSP_TCP_MAX_PACKET_SIZE);
623 /* Ownership of rtp_handle is passed to the rtp mux context */
624 rtsp_st->rtp_handle = NULL;
625 if (ret < 0)
626 return ret;
627 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
628 return 0; // Don't need to open any parser here
629 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
630 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
631 rtsp_st->dynamic_protocol_context,
632 rtsp_st->dynamic_handler);
633 else if (CONFIG_RTPDEC)
634 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
635 rtsp_st->sdp_payload_type,
636 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
637 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
638
639 if (!rtsp_st->transport_priv) {
640 return AVERROR(ENOMEM);
641 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
642 if (rtsp_st->dynamic_handler) {
643 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
644 rtsp_st->dynamic_protocol_context,
645 rtsp_st->dynamic_handler);
646 }
647 }
648
649 return 0;
650 }
651
652 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
653 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
654 {
655 const char *q;
656 char *p;
657 int v;
658
659 q = *pp;
660 q += strspn(q, SPACE_CHARS);
661 v = strtol(q, &p, 10);
662 if (*p == '-') {
663 p++;
664 *min_ptr = v;
665 v = strtol(p, &p, 10);
666 *max_ptr = v;
667 } else {
668 *min_ptr = v;
669 *max_ptr = v;
670 }
671 *pp = p;
672 }
673
674 /* XXX: only one transport specification is parsed */
675 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
676 {
677 char transport_protocol[16];
678 char profile[16];
679 char lower_transport[16];
680 char parameter[16];
681 RTSPTransportField *th;
682 char buf[256];
683
684 reply->nb_transports = 0;
685
686 for (;;) {
687 p += strspn(p, SPACE_CHARS);
688 if (*p == '\0')
689 break;
690
691 th = &reply->transports[reply->nb_transports];
692
693 get_word_sep(transport_protocol, sizeof(transport_protocol),
694 "/", &p);
695 if (!av_strcasecmp (transport_protocol, "rtp")) {
696 get_word_sep(profile, sizeof(profile), "/;,", &p);
697 lower_transport[0] = '\0';
698 /* rtp/avp/<protocol> */
699 if (*p == '/') {
700 get_word_sep(lower_transport, sizeof(lower_transport),
701 ";,", &p);
702 }
703 th->transport = RTSP_TRANSPORT_RTP;
704 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
705 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
706 /* x-pn-tng/<protocol> */
707 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
708 profile[0] = '\0';
709 th->transport = RTSP_TRANSPORT_RDT;
710 } else if (!av_strcasecmp(transport_protocol, "raw")) {
711 get_word_sep(profile, sizeof(profile), "/;,", &p);
712 lower_transport[0] = '\0';
713 /* raw/raw/<protocol> */
714 if (*p == '/') {
715 get_word_sep(lower_transport, sizeof(lower_transport),
716 ";,", &p);
717 }
718 th->transport = RTSP_TRANSPORT_RAW;
719 }
720 if (!av_strcasecmp(lower_transport, "TCP"))
721 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
722 else
723 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
724
725 if (*p == ';')
726 p++;
727 /* get each parameter */
728 while (*p != '\0' && *p != ',') {
729 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
730 if (!strcmp(parameter, "port")) {
731 if (*p == '=') {
732 p++;
733 rtsp_parse_range(&th->port_min, &th->port_max, &p);
734 }
735 } else if (!strcmp(parameter, "client_port")) {
736 if (*p == '=') {
737 p++;
738 rtsp_parse_range(&th->client_port_min,
739 &th->client_port_max, &p);
740 }
741 } else if (!strcmp(parameter, "server_port")) {
742 if (*p == '=') {
743 p++;
744 rtsp_parse_range(&th->server_port_min,
745 &th->server_port_max, &p);
746 }
747 } else if (!strcmp(parameter, "interleaved")) {
748 if (*p == '=') {
749 p++;
750 rtsp_parse_range(&th->interleaved_min,
751 &th->interleaved_max, &p);
752 }
753 } else if (!strcmp(parameter, "multicast")) {
754 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
755 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
756 } else if (!strcmp(parameter, "ttl")) {
757 if (*p == '=') {
758 p++;
759 th->ttl = strtol(p, (char **)&p, 10);
760 }
761 } else if (!strcmp(parameter, "destination")) {
762 if (*p == '=') {
763 p++;
764 get_word_sep(buf, sizeof(buf), ";,", &p);
765 get_sockaddr(buf, &th->destination);
766 }
767 } else if (!strcmp(parameter, "source")) {
768 if (*p == '=') {
769 p++;
770 get_word_sep(buf, sizeof(buf), ";,", &p);
771 av_strlcpy(th->source, buf, sizeof(th->source));
772 }
773 } else if (!strcmp(parameter, "mode")) {
774 if (*p == '=') {
775 p++;
776 get_word_sep(buf, sizeof(buf), ";, ", &p);
777 if (!strcmp(buf, "record") ||
778 !strcmp(buf, "receive"))
779 th->mode_record = 1;
780 }
781 }
782
783 while (*p != ';' && *p != '\0' && *p != ',')
784 p++;
785 if (*p == ';')
786 p++;
787 }
788 if (*p == ',')
789 p++;
790
791 reply->nb_transports++;
792 }
793 }
794
795 static void handle_rtp_info(RTSPState *rt, const char *url,
796 uint32_t seq, uint32_t rtptime)
797 {
798 int i;
799 if (!rtptime || !url[0])
800 return;
801 if (rt->transport != RTSP_TRANSPORT_RTP)
802 return;
803 for (i = 0; i < rt->nb_rtsp_streams; i++) {
804 RTSPStream *rtsp_st = rt->rtsp_streams[i];
805 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
806 if (!rtpctx)
807 continue;
808 if (!strcmp(rtsp_st->control_url, url)) {
809 rtpctx->base_timestamp = rtptime;
810 break;
811 }
812 }
813 }
814
815 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
816 {
817 int read = 0;
818 char key[20], value[1024], url[1024] = "";
819 uint32_t seq = 0, rtptime = 0;
820
821 for (;;) {
822 p += strspn(p, SPACE_CHARS);
823 if (!*p)
824 break;
825 get_word_sep(key, sizeof(key), "=", &p);
826 if (*p != '=')
827 break;
828 p++;
829 get_word_sep(value, sizeof(value), ";, ", &p);
830 read++;
831 if (!strcmp(key, "url"))
832 av_strlcpy(url, value, sizeof(url));
833 else if (!strcmp(key, "seq"))
834 seq = strtoul(value, NULL, 10);
835 else if (!strcmp(key, "rtptime"))
836 rtptime = strtoul(value, NULL, 10);
837 if (*p == ',') {
838 handle_rtp_info(rt, url, seq, rtptime);
839 url[0] = '\0';
840 seq = rtptime = 0;
841 read = 0;
842 }
843 if (*p)
844 p++;
845 }
846 if (read > 0)
847 handle_rtp_info(rt, url, seq, rtptime);
848 }
849
850 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
851 RTSPState *rt, const char *method)
852 {
853 const char *p;
854
855 /* NOTE: we do case independent match for broken servers */
856 p = buf;
857 if (av_stristart(p, "Session:", &p)) {
858 int t;
859 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
860 if (av_stristart(p, ";timeout=", &p) &&
861 (t = strtol(p, NULL, 10)) > 0) {
862 reply->timeout = t;
863 }
864 } else if (av_stristart(p, "Content-Length:", &p)) {
865 reply->content_length = strtol(p, NULL, 10);
866 } else if (av_stristart(p, "Transport:", &p)) {
867 rtsp_parse_transport(reply, p);
868 } else if (av_stristart(p, "CSeq:", &p)) {
869 reply->seq = strtol(p, NULL, 10);
870 } else if (av_stristart(p, "Range:", &p)) {
871 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
872 } else if (av_stristart(p, "RealChallenge1:", &p)) {
873 p += strspn(p, SPACE_CHARS);
874 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
875 } else if (av_stristart(p, "Server:", &p)) {
876 p += strspn(p, SPACE_CHARS);
877 av_strlcpy(reply->server, p, sizeof(reply->server));
878 } else if (av_stristart(p, "Notice:", &p) ||
879 av_stristart(p, "X-Notice:", &p)) {
880 reply->notice = strtol(p, NULL, 10);
881 } else if (av_stristart(p, "Location:", &p)) {
882 p += strspn(p, SPACE_CHARS);
883 av_strlcpy(reply->location, p , sizeof(reply->location));
884 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
885 p += strspn(p, SPACE_CHARS);
886 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
887 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
888 p += strspn(p, SPACE_CHARS);
889 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
890 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
891 p += strspn(p, SPACE_CHARS);
892 if (method && !strcmp(method, "DESCRIBE"))
893 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
894 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
895 p += strspn(p, SPACE_CHARS);
896 if (method && !strcmp(method, "PLAY"))
897 rtsp_parse_rtp_info(rt, p);
898 } else if (av_stristart(p, "Public:", &p) && rt) {
899 if (strstr(p, "GET_PARAMETER") &&
900 method && !strcmp(method, "OPTIONS"))
901 rt->get_parameter_supported = 1;
902 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
903 p += strspn(p, SPACE_CHARS);
904 rt->accept_dynamic_rate = atoi(p);
905 } else if (av_stristart(p, "Content-Type:", &p)) {
906 p += strspn(p, SPACE_CHARS);
907 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
908 }
909 }
910
911 /* skip a RTP/TCP interleaved packet */
912 void ff_rtsp_skip_packet(AVFormatContext *s)
913 {
914 RTSPState *rt = s->priv_data;
915 int ret, len, len1;
916 uint8_t buf[1024];
917
918 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
919 if (ret != 3)
920 return;
921 len = AV_RB16(buf + 1);
922
923 av_dlog(s, "skipping RTP packet len=%d\n", len);
924
925 /* skip payload */
926 while (len > 0) {
927 len1 = len;
928 if (len1 > sizeof(buf))
929 len1 = sizeof(buf);
930 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
931 if (ret != len1)
932 return;
933 len -= len1;
934 }
935 }
936
937 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
938 unsigned char **content_ptr,
939 int return_on_interleaved_data, const char *method)
940 {
941 RTSPState *rt = s->priv_data;
942 char buf[4096], buf1[1024], *q;
943 unsigned char ch;
944 const char *p;
945 int ret, content_length, line_count = 0, request = 0;
946 unsigned char *content = NULL;
947
948 start:
949 line_count = 0;
950 request = 0;
951 content = NULL;
952 memset(reply, 0, sizeof(*reply));
953
954 /* parse reply (XXX: use buffers) */
955 rt->last_reply[0] = '\0';
956 for (;;) {
957 q = buf;
958 for (;;) {
959 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
960 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
961 if (ret != 1)
962 return AVERROR_EOF;
963 if (ch == '\n')
964 break;
965 if (ch == '$') {
966 /* XXX: only parse it if first char on line ? */
967 if (return_on_interleaved_data) {
968 return 1;
969 } else
970 ff_rtsp_skip_packet(s);
971 } else if (ch != '\r') {
972 if ((q - buf) < sizeof(buf) - 1)
973 *q++ = ch;
974 }
975 }
976 *q = '\0';
977
978 av_dlog(s, "line='%s'\n", buf);
979
980 /* test if last line */
981 if (buf[0] == '\0')
982 break;
983 p = buf;
984 if (line_count == 0) {
985 /* get reply code */
986 get_word(buf1, sizeof(buf1), &p);
987 if (!strncmp(buf1, "RTSP/", 5)) {
988 get_word(buf1, sizeof(buf1), &p);
989 reply->status_code = atoi(buf1);
990 av_strlcpy(reply->reason, p, sizeof(reply->reason));
991 } else {
992 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
993 get_word(buf1, sizeof(buf1), &p); // object
994 request = 1;
995 }
996 } else {
997 ff_rtsp_parse_line(reply, p, rt, method);
998 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
999 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1000 }
1001 line_count++;
1002 }
1003
1004 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1005 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1006
1007 content_length = reply->content_length;
1008 if (content_length > 0) {
1009 /* leave some room for a trailing '\0' (useful for simple parsing) */
1010 content = av_malloc(content_length + 1);
1011 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1012 content[content_length] = '\0';
1013 }
1014 if (content_ptr)
1015 *content_ptr = content;
1016 else
1017 av_free(content);
1018
1019 if (request) {
1020 char buf[1024];
1021 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1022 const char* ptr = buf;
1023
1024 if (!strcmp(reply->reason, "OPTIONS")) {
1025 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1026 if (reply->seq)
1027 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1028 if (reply->session_id[0])
1029 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1030 reply->session_id);
1031 } else {
1032 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1033 }
1034 av_strlcat(buf, "\r\n", sizeof(buf));
1035
1036 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1037 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1038 ptr = base64buf;
1039 }
1040 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1041
1042 rt->last_cmd_time = av_gettime();
1043 /* Even if the request from the server had data, it is not the data
1044 * that the caller wants or expects. The memory could also be leaked
1045 * if the actual following reply has content data. */
1046 if (content_ptr)
1047 av_freep(content_ptr);
1048 /* If method is set, this is called from ff_rtsp_send_cmd,
1049 * where a reply to exactly this request is awaited. For
1050 * callers from within packet receiving, we just want to
1051 * return to the caller and go back to receiving packets. */
1052 if (method)
1053 goto start;
1054 return 0;
1055 }
1056
1057 if (rt->seq != reply->seq) {
1058 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1059 rt->seq, reply->seq);
1060 }
1061
1062 /* EOS */
1063 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1064 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1065 reply->notice == 2306 /* Continuous Feed Terminated */) {
1066 rt->state = RTSP_STATE_IDLE;
1067 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1068 return AVERROR(EIO); /* data or server error */
1069 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1070 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1071 return AVERROR(EPERM);
1072
1073 return 0;
1074 }
1075
1076 /**
1077 * Send a command to the RTSP server without waiting for the reply.
1078 *
1079 * @param s RTSP (de)muxer context
1080 * @param method the method for the request
1081 * @param url the target url for the request
1082 * @param headers extra header lines to include in the request
1083 * @param send_content if non-null, the data to send as request body content
1084 * @param send_content_length the length of the send_content data, or 0 if
1085 * send_content is null
1086 *
1087 * @return zero if success, nonzero otherwise
1088 */
1089 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1090 const char *method, const char *url,
1091 const char *headers,
1092 const unsigned char *send_content,
1093 int send_content_length)
1094 {
1095 RTSPState *rt = s->priv_data;
1096 char buf[4096], *out_buf;
1097 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1098
1099 /* Add in RTSP headers */
1100 out_buf = buf;
1101 rt->seq++;
1102 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1103 if (headers)
1104 av_strlcat(buf, headers, sizeof(buf));
1105 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1106 if (rt->session_id[0] != '\0' && (!headers ||
1107 !strstr(headers, "\nIf-Match:"))) {
1108 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1109 }
1110 if (rt->auth[0]) {
1111 char *str = ff_http_auth_create_response(&rt->auth_state,
1112 rt->auth, url, method);
1113 if (str)
1114 av_strlcat(buf, str, sizeof(buf));
1115 av_free(str);
1116 }
1117 if (send_content_length > 0 && send_content)
1118 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1119 av_strlcat(buf, "\r\n", sizeof(buf));
1120
1121 /* base64 encode rtsp if tunneling */
1122 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1123 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1124 out_buf = base64buf;
1125 }
1126
1127 av_dlog(s, "Sending:\n%s--\n", buf);
1128
1129 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1130 if (send_content_length > 0 && send_content) {
1131 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1132 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1133 "with content data not supported\n");
1134 return AVERROR_PATCHWELCOME;
1135 }
1136 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1137 }
1138 rt->last_cmd_time = av_gettime();
1139
1140 return 0;
1141 }
1142
1143 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1144 const char *url, const char *headers)
1145 {
1146 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1147 }
1148
1149 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1150 const char *headers, RTSPMessageHeader *reply,
1151 unsigned char **content_ptr)
1152 {
1153 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1154 content_ptr, NULL, 0);
1155 }
1156
1157 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1158 const char *method, const char *url,
1159 const char *header,
1160 RTSPMessageHeader *reply,
1161 unsigned char **content_ptr,
1162 const unsigned char *send_content,
1163 int send_content_length)
1164 {
1165 RTSPState *rt = s->priv_data;
1166 HTTPAuthType cur_auth_type;
1167 int ret, attempts = 0;
1168
1169 retry:
1170 cur_auth_type = rt->auth_state.auth_type;
1171 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1172 send_content,
1173 send_content_length)))
1174 return ret;
1175
1176 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1177 return ret;
1178 attempts++;
1179
1180 if (reply->status_code == 401 &&
1181 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1182 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1183 goto retry;
1184
1185 if (reply->status_code > 400){
1186 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1187 method,
1188 reply->status_code,
1189 reply->reason);
1190 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1191 }
1192
1193 return 0;
1194 }
1195
1196 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1197 int lower_transport, const char *real_challenge)
1198 {
1199 RTSPState *rt = s->priv_data;
1200 int rtx = 0, j, i, err, interleave = 0, port_off;
1201 RTSPStream *rtsp_st;
1202 RTSPMessageHeader reply1, *reply = &reply1;
1203 char cmd[2048];
1204 const char *trans_pref;
1205
1206 if (rt->transport == RTSP_TRANSPORT_RDT)
1207 trans_pref = "x-pn-tng";
1208 else if (rt->transport == RTSP_TRANSPORT_RAW)
1209 trans_pref = "RAW/RAW";
1210 else
1211 trans_pref = "RTP/AVP";
1212
1213 /* default timeout: 1 minute */
1214 rt->timeout = 60;
1215
1216 /* for each stream, make the setup request */
1217 /* XXX: we assume the same server is used for the control of each
1218 * RTSP stream */
1219
1220 /* Choose a random starting offset within the first half of the
1221 * port range, to allow for a number of ports to try even if the offset
1222 * happens to be at the end of the random range. */
1223 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1224 /* even random offset */
1225 port_off -= port_off & 0x01;
1226
1227 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1228 char transport[2048];
1229
1230 /*
1231 * WMS serves all UDP data over a single connection, the RTX, which
1232 * isn't necessarily the first in the SDP but has to be the first
1233 * to be set up, else the second/third SETUP will fail with a 461.
1234 */
1235 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1236 rt->server_type == RTSP_SERVER_WMS) {
1237 if (i == 0) {
1238 /* rtx first */
1239 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1240 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1241 if (len >= 4 &&
1242 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1243 "/rtx"))
1244 break;
1245 }
1246 if (rtx == rt->nb_rtsp_streams)
1247 return -1; /* no RTX found */
1248 rtsp_st = rt->rtsp_streams[rtx];
1249 } else
1250 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1251 } else
1252 rtsp_st = rt->rtsp_streams[i];
1253
1254 /* RTP/UDP */
1255 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1256 char buf[256];
1257
1258 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1259 port = reply->transports[0].client_port_min;
1260 goto have_port;
1261 }
1262
1263 /* first try in specified port range */
1264 while (j <= rt->rtp_port_max) {
1265 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1266 "?localport=%d", j);
1267 /* we will use two ports per rtp stream (rtp and rtcp) */
1268 j += 2;
1269 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1270 &s->interrupt_callback, NULL))
1271 goto rtp_opened;
1272 }
1273
1274 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1275 err = AVERROR(EIO);
1276 goto fail;
1277
1278 rtp_opened:
1279 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1280 have_port:
1281 snprintf(transport, sizeof(transport) - 1,
1282 "%s/UDP;", trans_pref);
1283 if (rt->server_type != RTSP_SERVER_REAL)
1284 av_strlcat(transport, "unicast;", sizeof(transport));
1285 av_strlcatf(transport, sizeof(transport),
1286 "client_port=%d", port);
1287 if (rt->transport == RTSP_TRANSPORT_RTP &&
1288 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1289 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1290 }
1291
1292 /* RTP/TCP */
1293 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1294 /* For WMS streams, the application streams are only used for
1295 * UDP. When trying to set it up for TCP streams, the server
1296 * will return an error. Therefore, we skip those streams. */
1297 if (rt->server_type == RTSP_SERVER_WMS &&
1298 (rtsp_st->stream_index < 0 ||
1299 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1300 AVMEDIA_TYPE_DATA))
1301 continue;
1302 snprintf(transport, sizeof(transport) - 1,
1303 "%s/TCP;", trans_pref);
1304 if (rt->transport != RTSP_TRANSPORT_RDT)
1305 av_strlcat(transport, "unicast;", sizeof(transport));
1306 av_strlcatf(transport, sizeof(transport),
1307 "interleaved=%d-%d",
1308 interleave, interleave + 1);
1309 interleave += 2;
1310 }
1311
1312 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1313 snprintf(transport, sizeof(transport) - 1,
1314 "%s/UDP;multicast", trans_pref);
1315 }
1316 if (s->oformat) {
1317 av_strlcat(transport, ";mode=record", sizeof(transport));
1318 } else if (rt->server_type == RTSP_SERVER_REAL ||
1319 rt->server_type == RTSP_SERVER_WMS)
1320 av_strlcat(transport, ";mode=play", sizeof(transport));
1321 snprintf(cmd, sizeof(cmd),
1322 "Transport: %s\r\n",
1323 transport);
1324 if (rt->accept_dynamic_rate)
1325 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1326 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1327 char real_res[41], real_csum[9];
1328 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1329 real_challenge);
1330 av_strlcatf(cmd, sizeof(cmd),
1331 "If-Match: %s\r\n"
1332 "RealChallenge2: %s, sd=%s\r\n",
1333 rt->session_id, real_res, real_csum);
1334 }
1335 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1336 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1337 err = 1;
1338 goto fail;
1339 } else if (reply->status_code != RTSP_STATUS_OK ||
1340 reply->nb_transports != 1) {
1341 err = AVERROR_INVALIDDATA;
1342 goto fail;
1343 }
1344
1345 /* XXX: same protocol for all streams is required */
1346 if (i > 0) {
1347 if (reply->transports[0].lower_transport != rt->lower_transport ||
1348 reply->transports[0].transport != rt->transport) {
1349 err = AVERROR_INVALIDDATA;
1350 goto fail;
1351 }
1352 } else {
1353 rt->lower_transport = reply->transports[0].lower_transport;
1354 rt->transport = reply->transports[0].transport;
1355 }
1356
1357 /* Fail if the server responded with another lower transport mode
1358 * than what we requested. */
1359 if (reply->transports[0].lower_transport != lower_transport) {
1360 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1361 err = AVERROR_INVALIDDATA;
1362 goto fail;
1363 }
1364
1365 switch(reply->transports[0].lower_transport) {
1366 case RTSP_LOWER_TRANSPORT_TCP:
1367 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1368 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1369 break;
1370
1371 case RTSP_LOWER_TRANSPORT_UDP: {
1372 char url[1024], options[30] = "";
1373
1374 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1375 av_strlcpy(options, "?connect=1", sizeof(options));
1376 /* Use source address if specified */
1377 if (reply->transports[0].source[0]) {
1378 ff_url_join(url, sizeof(url), "rtp", NULL,
1379 reply->transports[0].source,
1380 reply->transports[0].server_port_min, "%s", options);
1381 } else {
1382 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1383 reply->transports[0].server_port_min, "%s", options);
1384 }
1385 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1386 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1387 err = AVERROR_INVALIDDATA;
1388 goto fail;
1389 }
1390 /* Try to initialize the connection state in a
1391 * potential NAT router by sending dummy packets.
1392 * RTP/RTCP dummy packets are used for RDT, too.
1393 */
1394 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1395 CONFIG_RTPDEC)
1396 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1397 break;
1398 }
1399 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1400 char url[1024], namebuf[50], optbuf[20] = "";
1401 struct sockaddr_storage addr;
1402 int port, ttl;
1403
1404 if (reply->transports[0].destination.ss_family) {
1405 addr = reply->transports[0].destination;
1406 port = reply->transports[0].port_min;
1407 ttl = reply->transports[0].ttl;
1408 } else {
1409 addr = rtsp_st->sdp_ip;
1410 port = rtsp_st->sdp_port;
1411 ttl = rtsp_st->sdp_ttl;
1412 }
1413 if (ttl > 0)
1414 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1415 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1416 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1417 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1418 port, "%s", optbuf);
1419 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1420 &s->interrupt_callback, NULL) < 0) {
1421 err = AVERROR_INVALIDDATA;
1422 goto fail;
1423 }
1424 break;
1425 }
1426 }
1427
1428 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1429 goto fail;
1430 }
1431
1432 if (rt->nb_rtsp_streams && reply->timeout > 0)
1433 rt->timeout = reply->timeout;
1434
1435 if (rt->server_type == RTSP_SERVER_REAL)
1436 rt->need_subscription = 1;
1437
1438 return 0;
1439
1440 fail:
1441 ff_rtsp_undo_setup(s);
1442 return err;
1443 }
1444
1445 void ff_rtsp_close_connections(AVFormatContext *s)
1446 {
1447 RTSPState *rt = s->priv_data;
1448 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1449 ffurl_close(rt->rtsp_hd);
1450 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1451 }
1452
1453 int ff_rtsp_connect(AVFormatContext *s)
1454 {
1455 RTSPState *rt = s->priv_data;
1456 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1457 int port, err, tcp_fd;
1458 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1459 int lower_transport_mask = 0;
1460 char real_challenge[64] = "";
1461 struct sockaddr_storage peer;
1462 socklen_t peer_len = sizeof(peer);
1463
1464 if (rt->rtp_port_max < rt->rtp_port_min) {
1465 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1466 "than min port %d\n", rt->rtp_port_max,
1467 rt->rtp_port_min);
1468 return AVERROR(EINVAL);
1469 }
1470
1471 if (!ff_network_init())
1472 return AVERROR(EIO);
1473
1474 if (s->max_delay < 0) /* Not set by the caller */
1475 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1476
1477 rt->control_transport = RTSP_MODE_PLAIN;
1478 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1479 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1480 rt->control_transport = RTSP_MODE_TUNNEL;
1481 }
1482 /* Only pass through valid flags from here */
1483 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1484
1485 redirect:
1486 lower_transport_mask = rt->lower_transport_mask;
1487 /* extract hostname and port */
1488 av_url_split(NULL, 0, auth, sizeof(auth),
1489 host, sizeof(host), &port, path, sizeof(path), s->filename);
1490 if (*auth) {
1491 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1492 }
1493 if (port < 0)
1494 port = RTSP_DEFAULT_PORT;
1495
1496 if (!lower_transport_mask)
1497 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1498
1499 if (s->oformat) {
1500 /* Only UDP or TCP - UDP multicast isn't supported. */
1501 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1502 (1 << RTSP_LOWER_TRANSPORT_TCP);
1503 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1504 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1505 "only UDP and TCP are supported for output.\n");
1506 err = AVERROR(EINVAL);
1507 goto fail;
1508 }
1509 }
1510
1511 /* Construct the URI used in request; this is similar to s->filename,
1512 * but with authentication credentials removed and RTSP specific options
1513 * stripped out. */
1514 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1515 host, port, "%s", path);
1516
1517 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1518 /* set up initial handshake for tunneling */
1519 char httpname[1024];
1520 char sessioncookie[17];
1521 char headers[1024];
1522
1523 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1524 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1525 av_get_random_seed(), av_get_random_seed());
1526
1527 /* GET requests */
1528 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1529 &s->interrupt_callback) < 0) {
1530 err = AVERROR(EIO);
1531 goto fail;
1532 }
1533
1534 /* generate GET headers */
1535 snprintf(headers, sizeof(headers),
1536 "x-sessioncookie: %s\r\n"
1537 "Accept: application/x-rtsp-tunnelled\r\n"
1538 "Pragma: no-cache\r\n"
1539 "Cache-Control: no-cache\r\n",
1540 sessioncookie);
1541 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1542
1543 /* complete the connection */
1544 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1545 err = AVERROR(EIO);
1546 goto fail;
1547 }
1548
1549 /* POST requests */
1550 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1551 &s->interrupt_callback) < 0 ) {
1552 err = AVERROR(EIO);
1553 goto fail;
1554 }
1555
1556 /* generate POST headers */
1557 snprintf(headers, sizeof(headers),
1558 "x-sessioncookie: %s\r\n"
1559 "Content-Type: application/x-rtsp-tunnelled\r\n"
1560 "Pragma: no-cache\r\n"
1561 "Cache-Control: no-cache\r\n"
1562 "Content-Length: 32767\r\n"
1563 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1564 sessioncookie);
1565 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1566 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1567
1568 /* Initialize the authentication state for the POST session. The HTTP
1569 * protocol implementation doesn't properly handle multi-pass
1570 * authentication for POST requests, since it would require one of
1571 * the following:
1572 * - implementing Expect: 100-continue, which many HTTP servers
1573 * don't support anyway, even less the RTSP servers that do HTTP
1574 * tunneling
1575 * - sending the whole POST data until getting a 401 reply specifying
1576 * what authentication method to use, then resending all that data
1577 * - waiting for potential 401 replies directly after sending the
1578 * POST header (waiting for some unspecified time)
1579 * Therefore, we copy the full auth state, which works for both basic
1580 * and digest. (For digest, we would have to synchronize the nonce
1581 * count variable between the two sessions, if we'd do more requests
1582 * with the original session, though.)
1583 */
1584 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1585
1586 /* complete the connection */
1587 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1588 err = AVERROR(EIO);
1589 goto fail;
1590 }
1591 } else {
1592 /* open the tcp connection */
1593 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1594 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1595 &s->interrupt_callback, NULL) < 0) {
1596 err = AVERROR(EIO);
1597 goto fail;
1598 }
1599 rt->rtsp_hd_out = rt->rtsp_hd;
1600 }
1601 rt->seq = 0;
1602
1603 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1604 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1605 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1606 NULL, 0, NI_NUMERICHOST);
1607 }
1608
1609 /* request options supported by the server; this also detects server
1610 * type */
1611 for (rt->server_type = RTSP_SERVER_RTP;;) {
1612 cmd[0] = 0;
1613 if (rt->server_type == RTSP_SERVER_REAL)
1614 av_strlcat(cmd,
1615 /*
1616 * The following entries are required for proper
1617 * streaming from a Realmedia server. They are
1618 * interdependent in some way although we currently
1619 * don't quite understand how. Values were copied
1620 * from mplayer SVN r23589.
1621 * ClientChallenge is a 16-byte ID in hex
1622 * CompanyID is a 16-byte ID in base64
1623 */
1624 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1625 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1626 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1627 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1628 sizeof(cmd));
1629 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1630 if (reply->status_code != RTSP_STATUS_OK) {
1631 err = AVERROR_INVALIDDATA;
1632 goto fail;
1633 }
1634
1635 /* detect server type if not standard-compliant RTP */
1636 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1637 rt->server_type = RTSP_SERVER_REAL;
1638 continue;
1639 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1640 rt->server_type = RTSP_SERVER_WMS;
1641 } else if (rt->server_type == RTSP_SERVER_REAL)
1642 strcpy(real_challenge, reply->real_challenge);
1643 break;
1644 }
1645
1646 if (s->iformat && CONFIG_RTSP_DEMUXER)
1647 err = ff_rtsp_setup_input_streams(s, reply);
1648 else if (CONFIG_RTSP_MUXER)
1649 err = ff_rtsp_setup_output_streams(s, host);
1650 if (err)
1651 goto fail;
1652
1653 do {
1654 int lower_transport = ff_log2_tab[lower_transport_mask &
1655 ~(lower_transport_mask - 1)];
1656
1657 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1658 rt->server_type == RTSP_SERVER_REAL ?
1659 real_challenge : NULL);
1660 if (err < 0)
1661 goto fail;
1662 lower_transport_mask &= ~(1 << lower_transport);
1663 if (lower_transport_mask == 0 && err == 1) {
1664 err = AVERROR(EPROTONOSUPPORT);
1665 goto fail;
1666 }
1667 } while (err);
1668
1669 rt->lower_transport_mask = lower_transport_mask;
1670 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1671 rt->state = RTSP_STATE_IDLE;
1672 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1673 return 0;
1674 fail:
1675 ff_rtsp_close_streams(s);
1676 ff_rtsp_close_connections(s);
1677 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1678 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1679 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1680 reply->status_code,
1681 s->filename);
1682 goto redirect;
1683 }
1684 ff_network_close();
1685 return err;
1686 }
1687 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1688
1689 #if CONFIG_RTPDEC
1690 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1691 uint8_t *buf, int buf_size, int64_t wait_end)
1692 {
1693 RTSPState *rt = s->priv_data;
1694 RTSPStream *rtsp_st;
1695 int n, i, ret, tcp_fd, timeout_cnt = 0;
1696 int max_p = 0;
1697 struct pollfd *p = rt->p;
1698 int *fds = NULL, fdsnum, fdsidx;
1699
1700 for (;;) {
1701 if (ff_check_interrupt(&s->interrupt_callback))
1702 return AVERROR_EXIT;
1703 if (wait_end && wait_end - av_gettime() < 0)
1704 return AVERROR(EAGAIN);
1705 max_p = 0;
1706 if (rt->rtsp_hd) {
1707 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1708 p[max_p].fd = tcp_fd;
1709 p[max_p++].events = POLLIN;
1710 } else {
1711 tcp_fd = -1;
1712 }
1713 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1714 rtsp_st = rt->rtsp_streams[i];
1715 if (rtsp_st->rtp_handle) {
1716 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1717 &fds, &fdsnum)) {
1718 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1719 return ret;
1720 }
1721 if (fdsnum != 2) {
1722 av_log(s, AV_LOG_ERROR,
1723 "Number of fds %d not supported\n", fdsnum);
1724 return AVERROR_INVALIDDATA;
1725 }
1726 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1727 p[max_p].fd = fds[fdsidx];
1728 p[max_p++].events = POLLIN;
1729 }
1730 av_free(fds);
1731 }
1732 }
1733 n = poll(p, max_p, POLL_TIMEOUT_MS);
1734 if (n > 0) {
1735 int j = 1 - (tcp_fd == -1);
1736 timeout_cnt = 0;
1737 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1738 rtsp_st = rt->rtsp_streams[i];
1739 if (rtsp_st->rtp_handle) {
1740 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1741 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1742 if (ret > 0) {
1743 *prtsp_st = rtsp_st;
1744 return ret;
1745 }
1746 }
1747 j+=2;
1748 }
1749 }
1750 #if CONFIG_RTSP_DEMUXER
1751 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1752 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1753 if (rt->state == RTSP_STATE_STREAMING) {
1754 if (!ff_rtsp_parse_streaming_commands(s))
1755 return AVERROR_EOF;
1756 else
1757 av_log(s, AV_LOG_WARNING,
1758 "Unable to answer to TEARDOWN\n");
1759 } else
1760 return 0;
1761 } else {
1762 RTSPMessageHeader reply;
1763 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1764 if (ret < 0)
1765 return ret;
1766 /* XXX: parse message */
1767 if (rt->state != RTSP_STATE_STREAMING)
1768 return 0;
1769 }
1770 }
1771 #endif
1772 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1773 return AVERROR(ETIMEDOUT);
1774 } else if (n < 0 && errno != EINTR)
1775 return AVERROR(errno);
1776 }
1777 }
1778
1779 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1780 {
1781 RTSPState *rt = s->priv_data;
1782 int ret, len;
1783 RTSPStream *rtsp_st, *first_queue_st = NULL;
1784 int64_t wait_end = 0;
1785
1786 if (rt->nb_byes == rt->nb_rtsp_streams)
1787 return AVERROR_EOF;
1788
1789 /* get next frames from the same RTP packet */
1790 if (rt->cur_transport_priv) {
1791 if (rt->transport == RTSP_TRANSPORT_RDT) {
1792 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1793 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1794 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1795 } else if (rt->ts && CONFIG_RTPDEC) {
1796 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1797 if (ret >= 0) {
1798 rt->recvbuf_pos += ret;
1799 ret = rt->recvbuf_pos < rt->recvbuf_len;
1800 }
1801 }
1802 if (ret == 0) {
1803 rt->cur_transport_priv = NULL;
1804 return 0;
1805 } else if (ret == 1) {
1806 return 0;
1807 } else
1808 rt->cur_transport_priv = NULL;
1809 }
1810
1811 if (rt->transport == RTSP_TRANSPORT_RTP) {
1812 int i;
1813 int64_t first_queue_time = 0;
1814 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1815 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1816 int64_t queue_time;
1817 if (!rtpctx)
1818 continue;
1819 queue_time = ff_rtp_queued_packet_time(rtpctx);
1820 if (queue_time && (queue_time - first_queue_time < 0 ||
1821 !first_queue_time)) {
1822 first_queue_time = queue_time;
1823 first_queue_st = rt->rtsp_streams[i];
1824 }
1825 }
1826 if (first_queue_time)
1827 wait_end = first_queue_time + s->max_delay;
1828 }
1829
1830 /* read next RTP packet */
1831 redo:
1832 if (!rt->recvbuf) {
1833 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1834 if (!rt->recvbuf)
1835 return AVERROR(ENOMEM);
1836 }
1837
1838 switch(rt->lower_transport) {
1839 default:
1840 #if CONFIG_RTSP_DEMUXER
1841 case RTSP_LOWER_TRANSPORT_TCP:
1842 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1843 break;
1844 #endif
1845 case RTSP_LOWER_TRANSPORT_UDP:
1846 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1847 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1848 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1849 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1850 break;
1851 }
1852 if (len == AVERROR(EAGAIN) && first_queue_st &&
1853 rt->transport == RTSP_TRANSPORT_RTP) {
1854 rtsp_st = first_queue_st;
1855 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1856 goto end;
1857 }
1858 if (len < 0)
1859 return len;
1860 if (len == 0)
1861 return AVERROR_EOF;
1862 if (rt->transport == RTSP_TRANSPORT_RDT) {
1863 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1864 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1865 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1866 if (ret < 0) {
1867 /* Either bad packet, or a RTCP packet. Check if the
1868 * first_rtcp_ntp_time field was initialized. */
1869 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1870 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1871 /* first_rtcp_ntp_time has been initialized for this stream,
1872 * copy the same value to all other uninitialized streams,
1873 * in order to map their timestamp origin to the same ntp time
1874 * as this one. */
1875 int i;
1876 AVStream *st = NULL;
1877 if (rtsp_st->stream_index >= 0)
1878 st = s->streams[rtsp_st->stream_index];
1879 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1880 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1881 AVStream *st2 = NULL;
1882 if (rt->rtsp_streams[i]->stream_index >= 0)
1883 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1884 if (rtpctx2 && st && st2 &&
1885 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1886 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1887 rtpctx2->rtcp_ts_offset = av_rescale_q(
1888 rtpctx->rtcp_ts_offset, st->time_base,
1889 st2->time_base);
1890 }
1891 }
1892 }
1893 if (ret == -RTCP_BYE) {
1894 rt->nb_byes++;
1895
1896 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1897 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1898
1899 if (rt->nb_byes == rt->nb_rtsp_streams)
1900 return AVERROR_EOF;
1901 }
1902 }
1903 } else if (rt->ts && CONFIG_RTPDEC) {
1904 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1905 if (ret >= 0) {
1906 if (ret < len) {
1907 rt->recvbuf_len = len;
1908 rt->recvbuf_pos = ret;
1909 rt->cur_transport_priv = rt->ts;
1910 return 1;
1911 } else {
1912 ret = 0;
1913 }
1914 }
1915 } else {
1916 return AVERROR_INVALIDDATA;
1917 }
1918 end:
1919 if (ret < 0)
1920 goto redo;
1921 if (ret == 1)
1922 /* more packets may follow, so we save the RTP context */
1923 rt->cur_transport_priv = rtsp_st->transport_priv;
1924
1925 return ret;
1926 }
1927 #endif /* CONFIG_RTPDEC */
1928
1929 #if CONFIG_SDP_DEMUXER
1930 static int sdp_probe(AVProbeData *p1)
1931 {
1932 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1933
1934 /* we look for a line beginning "c=IN IP" */
1935 while (p < p_end && *p != '\0') {
1936 if (p + sizeof("c=IN IP") - 1 < p_end &&
1937 av_strstart(p, "c=IN IP", NULL))
1938 return AVPROBE_SCORE_MAX / 2;
1939
1940 while (p < p_end - 1 && *p != '\n') p++;
1941 if (++p >= p_end)
1942 break;
1943 if (*p == '\r')
1944 p++;
1945 }
1946 return 0;
1947 }
1948
1949 static int sdp_read_header(AVFormatContext *s)
1950 {
1951 RTSPState *rt = s->priv_data;
1952 RTSPStream *rtsp_st;
1953 int size, i, err;
1954 char *content;
1955 char url[1024];
1956
1957 if (!ff_network_init())
1958 return AVERROR(EIO);
1959
1960 if (s->max_delay < 0) /* Not set by the caller */
1961 s->max_delay = DEFAULT_REORDERING_DELAY;
1962
1963 /* read the whole sdp file */
1964 /* XXX: better loading */
1965 content = av_malloc(SDP_MAX_SIZE);
1966 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1967 if (size <= 0) {
1968 av_free(content);
1969 return AVERROR_INVALIDDATA;
1970 }
1971 content[size] ='\0';
1972
1973 err = ff_sdp_parse(s, content);
1974 av_free(content);
1975 if (err) goto fail;
1976
1977 /* open each RTP stream */
1978 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1979 char namebuf[50];
1980 rtsp_st = rt->rtsp_streams[i];
1981
1982 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1983 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1984 ff_url_join(url, sizeof(url), "rtp", NULL,
1985 namebuf, rtsp_st->sdp_port,
1986 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1987 rtsp_st->sdp_ttl,
1988 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1989 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1990 &s->interrupt_callback, NULL) < 0) {
1991 err = AVERROR_INVALIDDATA;
1992 goto fail;
1993 }
1994 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1995 goto fail;
1996 }
1997 return 0;
1998 fail:
1999 ff_rtsp_close_streams(s);
2000 ff_network_close();
2001 return err;
2002 }
2003
2004 static int sdp_read_close(AVFormatContext *s)
2005 {
2006 ff_rtsp_close_streams(s);
2007 ff_network_close();
2008 return 0;
2009 }
2010
2011 static const AVClass sdp_demuxer_class = {
2012 .class_name = "SDP demuxer",
2013 .item_name = av_default_item_name,
2014 .option = sdp_options,
2015 .version = LIBAVUTIL_VERSION_INT,
2016 };
2017
2018 AVInputFormat ff_sdp_demuxer = {
2019 .name = "sdp",
2020 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2021 .priv_data_size = sizeof(RTSPState),
2022 .read_probe = sdp_probe,
2023 .read_header = sdp_read_header,
2024 .read_packet = ff_rtsp_fetch_packet,
2025 .read_close = sdp_read_close,
2026 .priv_class = &sdp_demuxer_class,
2027 };
2028 #endif /* CONFIG_SDP_DEMUXER */
2029
2030 #if CONFIG_RTP_DEMUXER
2031 static int rtp_probe(AVProbeData *p)
2032 {
2033 if (av_strstart(p->filename, "rtp:", NULL))
2034 return AVPROBE_SCORE_MAX;
2035 return 0;
2036 }
2037
2038 static int rtp_read_header(AVFormatContext *s)
2039 {
2040 uint8_t recvbuf[1500];
2041 char host[500], sdp[500];
2042 int ret, port;
2043 URLContext* in = NULL;
2044 int payload_type;
2045 AVCodecContext codec = { 0 };
2046 struct sockaddr_storage addr;
2047 AVIOContext pb;
2048 socklen_t addrlen = sizeof(addr);
2049 RTSPState *rt = s->priv_data;
2050
2051 if (!ff_network_init())
2052 return AVERROR(EIO);
2053
2054 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2055 &s->interrupt_callback, NULL);
2056 if (ret)
2057 goto fail;
2058
2059 while (1) {
2060 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2061 if (ret == AVERROR(EAGAIN))
2062 continue;
2063 if (ret < 0)
2064 goto fail;
2065 if (ret < 12) {
2066 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2067 continue;
2068 }
2069
2070 if ((recvbuf[0] & 0xc0) != 0x80) {
2071 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2072 "received\n");
2073 continue;
2074 }
2075
2076 if (RTP_PT_IS_RTCP(recvbuf[1]))
2077 continue;
2078
2079 payload_type = recvbuf[1] & 0x7f;
2080 break;
2081 }
2082 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2083 ffurl_close(in);
2084 in = NULL;
2085
2086 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2087 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2088 "without an SDP file describing it\n",
2089 payload_type);
2090 goto fail;
2091 }
2092 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2093 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2094 "properly you need an SDP file "
2095 "describing it\n");
2096 }
2097
2098 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2099 NULL, 0, s->filename);
2100
2101 snprintf(sdp, sizeof(sdp),
2102 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2103 addr.ss_family == AF_INET ? 4 : 6, host,
2104 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2105 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2106 port, payload_type);
2107 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2108
2109 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2110 s->pb = &pb;
2111
2112 /* sdp_read_header initializes this again */
2113 ff_network_close();
2114
2115 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2116
2117 ret = sdp_read_header(s);
2118 s->pb = NULL;
2119 return ret;
2120
2121 fail:
2122 if (in)
2123 ffurl_close(in);
2124 ff_network_close();
2125 return ret;
2126 }
2127
2128 static const AVClass rtp_demuxer_class = {
2129 .class_name = "RTP demuxer",
2130 .item_name = av_default_item_name,
2131 .option = rtp_options,
2132 .version = LIBAVUTIL_VERSION_INT,
2133 };
2134
2135 AVInputFormat ff_rtp_demuxer = {
2136 .name = "rtp",
2137 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2138 .priv_data_size = sizeof(RTSPState),
2139 .read_probe = rtp_probe,
2140 .read_header = rtp_read_header,
2141 .read_packet = ff_rtsp_fetch_packet,
2142 .read_close = sdp_read_close,
2143 .flags = AVFMT_NOFILE,
2144 .priv_class = &rtp_demuxer_class,
2145 };
2146 #endif /* CONFIG_RTP_DEMUXER */