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rtsp.c
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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/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
32 #include "avformat.h"
33 #include "avio_internal.h"
34 
35 #if HAVE_POLL_H
36 #include <poll.h>
37 #endif
38 #include "internal.h"
39 #include "network.h"
40 #include "os_support.h"
41 #include "http.h"
42 #include "rtsp.h"
43 
44 #include "rtpdec.h"
45 #include "rdt.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
48 #include "url.h"
49 #include "rtpenc.h"
50 #include "mpegts.h"
51 
52 //#define DEBUG
53 
54 /* Timeout values for socket poll, in ms,
55  * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
62 
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 
67 #define RTSP_FLAG_OPTS(name, longname) \
68  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69  { "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" }, \
70  { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73  { 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" }, \
74  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 
78 #define RTSP_REORDERING_OPTS() \
79  { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
80 
82  { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84  { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89  RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91  { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92  { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93  { "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 },
95  { NULL },
96 };
97 
98 static const AVOption sdp_options[] = {
99  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100  { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
103  { NULL },
104 };
105 
106 static const AVOption rtp_options[] = {
107  RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
109  { NULL },
110 };
111 
112 static void get_word_until_chars(char *buf, int buf_size,
113  const char *sep, const char **pp)
114 {
115  const char *p;
116  char *q;
117 
118  p = *pp;
119  p += strspn(p, SPACE_CHARS);
120  q = buf;
121  while (!strchr(sep, *p) && *p != '\0') {
122  if ((q - buf) < buf_size - 1)
123  *q++ = *p;
124  p++;
125  }
126  if (buf_size > 0)
127  *q = '\0';
128  *pp = p;
129 }
130 
131 static void get_word_sep(char *buf, int buf_size, const char *sep,
132  const char **pp)
133 {
134  if (**pp == '/') (*pp)++;
135  get_word_until_chars(buf, buf_size, sep, pp);
136 }
137 
138 static void get_word(char *buf, int buf_size, const char **pp)
139 {
140  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
141 }
142 
143 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144  * and end time.
145  * Used for seeking in the rtp stream.
146  */
147 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
148 {
149  char buf[256];
150 
151  p += strspn(p, SPACE_CHARS);
152  if (!av_stristart(p, "npt=", &p))
153  return;
154 
155  *start = AV_NOPTS_VALUE;
156  *end = AV_NOPTS_VALUE;
157 
158  get_word_sep(buf, sizeof(buf), "-", &p);
159  av_parse_time(start, buf, 1);
160  if (*p == '-') {
161  p++;
162  get_word_sep(buf, sizeof(buf), "-", &p);
163  av_parse_time(end, buf, 1);
164  }
165 }
166 
167 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
168 {
169  struct addrinfo hints = { 0 }, *ai = NULL;
170  hints.ai_flags = AI_NUMERICHOST;
171  if (getaddrinfo(buf, NULL, &hints, &ai))
172  return -1;
173  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
174  freeaddrinfo(ai);
175  return 0;
176 }
177 
178 #if CONFIG_RTPDEC
179 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
180  RTSPStream *rtsp_st, AVCodecContext *codec)
181 {
182  if (!handler)
183  return;
184  if (codec)
185  codec->codec_id = handler->codec_id;
186  rtsp_st->dynamic_handler = handler;
187  if (handler->alloc) {
188  rtsp_st->dynamic_protocol_context = handler->alloc();
189  if (!rtsp_st->dynamic_protocol_context)
190  rtsp_st->dynamic_handler = NULL;
191  }
192 }
193 
194 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
195 static int sdp_parse_rtpmap(AVFormatContext *s,
196  AVStream *st, RTSPStream *rtsp_st,
197  int payload_type, const char *p)
198 {
199  AVCodecContext *codec = st->codec;
200  char buf[256];
201  int i;
202  AVCodec *c;
203  const char *c_name;
204 
205  /* See if we can handle this kind of payload.
206  * The space should normally not be there but some Real streams or
207  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
208  * have a trailing space. */
209  get_word_sep(buf, sizeof(buf), "/ ", &p);
210  if (payload_type < RTP_PT_PRIVATE) {
211  /* We are in a standard case
212  * (from http://www.iana.org/assignments/rtp-parameters). */
213  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
214  }
215 
216  if (codec->codec_id == AV_CODEC_ID_NONE) {
217  RTPDynamicProtocolHandler *handler =
219  init_rtp_handler(handler, rtsp_st, codec);
220  /* If no dynamic handler was found, check with the list of standard
221  * allocated types, if such a stream for some reason happens to
222  * use a private payload type. This isn't handled in rtpdec.c, since
223  * the format name from the rtpmap line never is passed into rtpdec. */
224  if (!rtsp_st->dynamic_handler)
225  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
226  }
227 
228  c = avcodec_find_decoder(codec->codec_id);
229  if (c && c->name)
230  c_name = c->name;
231  else
232  c_name = "(null)";
233 
234  get_word_sep(buf, sizeof(buf), "/", &p);
235  i = atoi(buf);
236  switch (codec->codec_type) {
237  case AVMEDIA_TYPE_AUDIO:
238  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
241  if (i > 0) {
242  codec->sample_rate = i;
243  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
244  get_word_sep(buf, sizeof(buf), "/", &p);
245  i = atoi(buf);
246  if (i > 0)
247  codec->channels = i;
248  }
249  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
250  codec->sample_rate);
251  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
252  codec->channels);
253  break;
254  case AVMEDIA_TYPE_VIDEO:
255  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
256  if (i > 0)
257  avpriv_set_pts_info(st, 32, 1, i);
258  break;
259  default:
260  break;
261  }
262  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
263  rtsp_st->dynamic_handler->init(s, st->index,
264  rtsp_st->dynamic_protocol_context);
265  return 0;
266 }
267 
268 /* parse the attribute line from the fmtp a line of an sdp response. This
269  * is broken out as a function because it is used in rtp_h264.c, which is
270  * forthcoming. */
271 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
272  char *value, int value_size)
273 {
274  *p += strspn(*p, SPACE_CHARS);
275  if (**p) {
276  get_word_sep(attr, attr_size, "=", p);
277  if (**p == '=')
278  (*p)++;
279  get_word_sep(value, value_size, ";", p);
280  if (**p == ';')
281  (*p)++;
282  return 1;
283  }
284  return 0;
285 }
286 
287 typedef struct SDPParseState {
288  /* SDP only */
289  struct sockaddr_storage default_ip;
290  int default_ttl;
291  int skip_media; ///< set if an unknown m= line occurs
292 } SDPParseState;
293 
294 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
295  int letter, const char *buf)
296 {
297  RTSPState *rt = s->priv_data;
298  char buf1[64], st_type[64];
299  const char *p;
300  enum AVMediaType codec_type;
301  int payload_type, i;
302  AVStream *st;
303  RTSPStream *rtsp_st;
304  struct sockaddr_storage sdp_ip;
305  int ttl;
306 
307  av_dlog(s, "sdp: %c='%s'\n", letter, buf);
308 
309  p = buf;
310  if (s1->skip_media && letter != 'm')
311  return;
312  switch (letter) {
313  case 'c':
314  get_word(buf1, sizeof(buf1), &p);
315  if (strcmp(buf1, "IN") != 0)
316  return;
317  get_word(buf1, sizeof(buf1), &p);
318  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
319  return;
320  get_word_sep(buf1, sizeof(buf1), "/", &p);
321  if (get_sockaddr(buf1, &sdp_ip))
322  return;
323  ttl = 16;
324  if (*p == '/') {
325  p++;
326  get_word_sep(buf1, sizeof(buf1), "/", &p);
327  ttl = atoi(buf1);
328  }
329  if (s->nb_streams == 0) {
330  s1->default_ip = sdp_ip;
331  s1->default_ttl = ttl;
332  } else {
333  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
334  rtsp_st->sdp_ip = sdp_ip;
335  rtsp_st->sdp_ttl = ttl;
336  }
337  break;
338  case 's':
339  av_dict_set(&s->metadata, "title", p, 0);
340  break;
341  case 'i':
342  if (s->nb_streams == 0) {
343  av_dict_set(&s->metadata, "comment", p, 0);
344  break;
345  }
346  break;
347  case 'm':
348  /* new stream */
349  s1->skip_media = 0;
350  codec_type = AVMEDIA_TYPE_UNKNOWN;
351  get_word(st_type, sizeof(st_type), &p);
352  if (!strcmp(st_type, "audio")) {
353  codec_type = AVMEDIA_TYPE_AUDIO;
354  } else if (!strcmp(st_type, "video")) {
355  codec_type = AVMEDIA_TYPE_VIDEO;
356  } else if (!strcmp(st_type, "application")) {
357  codec_type = AVMEDIA_TYPE_DATA;
358  }
359  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
360  s1->skip_media = 1;
361  return;
362  }
363  rtsp_st = av_mallocz(sizeof(RTSPStream));
364  if (!rtsp_st)
365  return;
366  rtsp_st->stream_index = -1;
367  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
368 
369  rtsp_st->sdp_ip = s1->default_ip;
370  rtsp_st->sdp_ttl = s1->default_ttl;
371 
372  get_word(buf1, sizeof(buf1), &p); /* port */
373  rtsp_st->sdp_port = atoi(buf1);
374 
375  get_word(buf1, sizeof(buf1), &p); /* protocol */
376  if (!strcmp(buf1, "udp"))
378  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
379  rtsp_st->feedback = 1;
380 
381  /* XXX: handle list of formats */
382  get_word(buf1, sizeof(buf1), &p); /* format list */
383  rtsp_st->sdp_payload_type = atoi(buf1);
384 
385  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
386  /* no corresponding stream */
387  if (rt->transport == RTSP_TRANSPORT_RAW) {
388  if (!rt->ts && CONFIG_RTPDEC)
389  rt->ts = ff_mpegts_parse_open(s);
390  } else {
391  RTPDynamicProtocolHandler *handler;
392  handler = ff_rtp_handler_find_by_id(
394  init_rtp_handler(handler, rtsp_st, NULL);
395  if (handler && handler->init)
396  handler->init(s, -1, rtsp_st->dynamic_protocol_context);
397  }
398  } else if (rt->server_type == RTSP_SERVER_WMS &&
399  codec_type == AVMEDIA_TYPE_DATA) {
400  /* RTX stream, a stream that carries all the other actual
401  * audio/video streams. Don't expose this to the callers. */
402  } else {
403  st = avformat_new_stream(s, NULL);
404  if (!st)
405  return;
406  st->id = rt->nb_rtsp_streams - 1;
407  rtsp_st->stream_index = st->index;
408  st->codec->codec_type = codec_type;
409  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
410  RTPDynamicProtocolHandler *handler;
411  /* if standard payload type, we can find the codec right now */
413  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
414  st->codec->sample_rate > 0)
415  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
416  /* Even static payload types may need a custom depacketizer */
417  handler = ff_rtp_handler_find_by_id(
418  rtsp_st->sdp_payload_type, st->codec->codec_type);
419  init_rtp_handler(handler, rtsp_st, st->codec);
420  if (handler && handler->init)
421  handler->init(s, st->index,
422  rtsp_st->dynamic_protocol_context);
423  }
424  }
425  /* put a default control url */
426  av_strlcpy(rtsp_st->control_url, rt->control_uri,
427  sizeof(rtsp_st->control_url));
428  break;
429  case 'a':
430  if (av_strstart(p, "control:", &p)) {
431  if (s->nb_streams == 0) {
432  if (!strncmp(p, "rtsp://", 7))
433  av_strlcpy(rt->control_uri, p,
434  sizeof(rt->control_uri));
435  } else {
436  char proto[32];
437  /* get the control url */
438  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
439 
440  /* XXX: may need to add full url resolution */
441  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
442  NULL, NULL, 0, p);
443  if (proto[0] == '\0') {
444  /* relative control URL */
445  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
446  av_strlcat(rtsp_st->control_url, "/",
447  sizeof(rtsp_st->control_url));
448  av_strlcat(rtsp_st->control_url, p,
449  sizeof(rtsp_st->control_url));
450  } else
451  av_strlcpy(rtsp_st->control_url, p,
452  sizeof(rtsp_st->control_url));
453  }
454  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
455  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
456  get_word(buf1, sizeof(buf1), &p);
457  payload_type = atoi(buf1);
458  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
459  if (rtsp_st->stream_index >= 0) {
460  st = s->streams[rtsp_st->stream_index];
461  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
462  }
463  } else if (av_strstart(p, "fmtp:", &p) ||
464  av_strstart(p, "framesize:", &p)) {
465  /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
466  // let dynamic protocol handlers have a stab at the line.
467  get_word(buf1, sizeof(buf1), &p);
468  payload_type = atoi(buf1);
469  for (i = 0; i < rt->nb_rtsp_streams; i++) {
470  rtsp_st = rt->rtsp_streams[i];
471  if (rtsp_st->sdp_payload_type == payload_type &&
472  rtsp_st->dynamic_handler &&
474  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
475  rtsp_st->dynamic_protocol_context, buf);
476  }
477  } else if (av_strstart(p, "range:", &p)) {
478  int64_t start, end;
479 
480  // this is so that seeking on a streamed file can work.
481  rtsp_parse_range_npt(p, &start, &end);
482  s->start_time = start;
483  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
484  s->duration = (end == AV_NOPTS_VALUE) ?
485  AV_NOPTS_VALUE : end - start;
486  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
487  if (atoi(p) == 1)
489  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
490  s->nb_streams > 0) {
491  st = s->streams[s->nb_streams - 1];
492  st->codec->sample_rate = atoi(p);
493  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
494  // RFC 4568
495  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
496  get_word(buf1, sizeof(buf1), &p); // ignore tag
497  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
498  p += strspn(p, SPACE_CHARS);
499  if (av_strstart(p, "inline:", &p))
500  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
501  } else {
502  if (rt->server_type == RTSP_SERVER_WMS)
504  if (s->nb_streams > 0) {
505  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
506 
507  if (rt->server_type == RTSP_SERVER_REAL)
508  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
509 
510  if (rtsp_st->dynamic_handler &&
512  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
513  rtsp_st->stream_index,
514  rtsp_st->dynamic_protocol_context, buf);
515  }
516  }
517  break;
518  }
519 }
520 
521 int ff_sdp_parse(AVFormatContext *s, const char *content)
522 {
523  RTSPState *rt = s->priv_data;
524  const char *p;
525  int letter;
526  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
527  * contain long SDP lines containing complete ASF Headers (several
528  * kB) or arrays of MDPR (RM stream descriptor) headers plus
529  * "rulebooks" describing their properties. Therefore, the SDP line
530  * buffer is large.
531  *
532  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
533  * in rtpdec_xiph.c. */
534  char buf[16384], *q;
535  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
536 
537  p = content;
538  for (;;) {
539  p += strspn(p, SPACE_CHARS);
540  letter = *p;
541  if (letter == '\0')
542  break;
543  p++;
544  if (*p != '=')
545  goto next_line;
546  p++;
547  /* get the content */
548  q = buf;
549  while (*p != '\n' && *p != '\r' && *p != '\0') {
550  if ((q - buf) < sizeof(buf) - 1)
551  *q++ = *p;
552  p++;
553  }
554  *q = '\0';
555  sdp_parse_line(s, s1, letter, buf);
556  next_line:
557  while (*p != '\n' && *p != '\0')
558  p++;
559  if (*p == '\n')
560  p++;
561  }
562  rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
563  if (!rt->p) return AVERROR(ENOMEM);
564  return 0;
565 }
566 #endif /* CONFIG_RTPDEC */
567 
569 {
570  RTSPState *rt = s->priv_data;
571  int i;
572 
573  for (i = 0; i < rt->nb_rtsp_streams; i++) {
574  RTSPStream *rtsp_st = rt->rtsp_streams[i];
575  if (!rtsp_st)
576  continue;
577  if (rtsp_st->transport_priv) {
578  if (s->oformat) {
579  AVFormatContext *rtpctx = rtsp_st->transport_priv;
580  av_write_trailer(rtpctx);
582  uint8_t *ptr;
583  avio_close_dyn_buf(rtpctx->pb, &ptr);
584  av_free(ptr);
585  } else {
586  avio_close(rtpctx->pb);
587  }
588  avformat_free_context(rtpctx);
589  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
591  else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
593  }
594  rtsp_st->transport_priv = NULL;
595  if (rtsp_st->rtp_handle)
596  ffurl_close(rtsp_st->rtp_handle);
597  rtsp_st->rtp_handle = NULL;
598  }
599 }
600 
601 /* close and free RTSP streams */
603 {
604  RTSPState *rt = s->priv_data;
605  int i;
606  RTSPStream *rtsp_st;
607 
609  for (i = 0; i < rt->nb_rtsp_streams; i++) {
610  rtsp_st = rt->rtsp_streams[i];
611  if (rtsp_st) {
612  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
613  rtsp_st->dynamic_handler->free(
614  rtsp_st->dynamic_protocol_context);
615  av_free(rtsp_st);
616  }
617  }
618  av_free(rt->rtsp_streams);
619  if (rt->asf_ctx) {
621  }
622  if (rt->ts && CONFIG_RTPDEC)
624  av_free(rt->p);
625  av_free(rt->recvbuf);
626 }
627 
629 {
630  RTSPState *rt = s->priv_data;
631  AVStream *st = NULL;
632  int reordering_queue_size = rt->reordering_queue_size;
633  if (reordering_queue_size < 0) {
635  reordering_queue_size = 0;
636  else
637  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
638  }
639 
640  /* open the RTP context */
641  if (rtsp_st->stream_index >= 0)
642  st = s->streams[rtsp_st->stream_index];
643  if (!st)
645 
646  if (s->oformat && CONFIG_RTSP_MUXER) {
647  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
648  rtsp_st->rtp_handle,
650  rtsp_st->stream_index);
651  /* Ownership of rtp_handle is passed to the rtp mux context */
652  rtsp_st->rtp_handle = NULL;
653  if (ret < 0)
654  return ret;
655  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
656  return 0; // Don't need to open any parser here
657  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
658  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
659  rtsp_st->dynamic_protocol_context,
660  rtsp_st->dynamic_handler);
661  else if (CONFIG_RTPDEC)
662  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
663  rtsp_st->sdp_payload_type,
664  reordering_queue_size);
665 
666  if (!rtsp_st->transport_priv) {
667  return AVERROR(ENOMEM);
668  } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
669  if (rtsp_st->dynamic_handler) {
671  rtsp_st->dynamic_protocol_context,
672  rtsp_st->dynamic_handler);
673  }
674  if (rtsp_st->crypto_suite[0])
676  rtsp_st->crypto_suite,
677  rtsp_st->crypto_params);
678  }
679 
680  return 0;
681 }
682 
683 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
684 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
685 {
686  const char *q;
687  char *p;
688  int v;
689 
690  q = *pp;
691  q += strspn(q, SPACE_CHARS);
692  v = strtol(q, &p, 10);
693  if (*p == '-') {
694  p++;
695  *min_ptr = v;
696  v = strtol(p, &p, 10);
697  *max_ptr = v;
698  } else {
699  *min_ptr = v;
700  *max_ptr = v;
701  }
702  *pp = p;
703 }
704 
705 /* XXX: only one transport specification is parsed */
706 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
707 {
708  char transport_protocol[16];
709  char profile[16];
710  char lower_transport[16];
711  char parameter[16];
712  RTSPTransportField *th;
713  char buf[256];
714 
715  reply->nb_transports = 0;
716 
717  for (;;) {
718  p += strspn(p, SPACE_CHARS);
719  if (*p == '\0')
720  break;
721 
722  th = &reply->transports[reply->nb_transports];
723 
724  get_word_sep(transport_protocol, sizeof(transport_protocol),
725  "/", &p);
726  if (!av_strcasecmp (transport_protocol, "rtp")) {
727  get_word_sep(profile, sizeof(profile), "/;,", &p);
728  lower_transport[0] = '\0';
729  /* rtp/avp/<protocol> */
730  if (*p == '/') {
731  get_word_sep(lower_transport, sizeof(lower_transport),
732  ";,", &p);
733  }
735  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
736  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
737  /* x-pn-tng/<protocol> */
738  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
739  profile[0] = '\0';
741  } else if (!av_strcasecmp(transport_protocol, "raw")) {
742  get_word_sep(profile, sizeof(profile), "/;,", &p);
743  lower_transport[0] = '\0';
744  /* raw/raw/<protocol> */
745  if (*p == '/') {
746  get_word_sep(lower_transport, sizeof(lower_transport),
747  ";,", &p);
748  }
750  }
751  if (!av_strcasecmp(lower_transport, "TCP"))
753  else
755 
756  if (*p == ';')
757  p++;
758  /* get each parameter */
759  while (*p != '\0' && *p != ',') {
760  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
761  if (!strcmp(parameter, "port")) {
762  if (*p == '=') {
763  p++;
764  rtsp_parse_range(&th->port_min, &th->port_max, &p);
765  }
766  } else if (!strcmp(parameter, "client_port")) {
767  if (*p == '=') {
768  p++;
769  rtsp_parse_range(&th->client_port_min,
770  &th->client_port_max, &p);
771  }
772  } else if (!strcmp(parameter, "server_port")) {
773  if (*p == '=') {
774  p++;
775  rtsp_parse_range(&th->server_port_min,
776  &th->server_port_max, &p);
777  }
778  } else if (!strcmp(parameter, "interleaved")) {
779  if (*p == '=') {
780  p++;
781  rtsp_parse_range(&th->interleaved_min,
782  &th->interleaved_max, &p);
783  }
784  } else if (!strcmp(parameter, "multicast")) {
787  } else if (!strcmp(parameter, "ttl")) {
788  if (*p == '=') {
789  char *end;
790  p++;
791  th->ttl = strtol(p, &end, 10);
792  p = end;
793  }
794  } else if (!strcmp(parameter, "destination")) {
795  if (*p == '=') {
796  p++;
797  get_word_sep(buf, sizeof(buf), ";,", &p);
798  get_sockaddr(buf, &th->destination);
799  }
800  } else if (!strcmp(parameter, "source")) {
801  if (*p == '=') {
802  p++;
803  get_word_sep(buf, sizeof(buf), ";,", &p);
804  av_strlcpy(th->source, buf, sizeof(th->source));
805  }
806  } else if (!strcmp(parameter, "mode")) {
807  if (*p == '=') {
808  p++;
809  get_word_sep(buf, sizeof(buf), ";, ", &p);
810  if (!strcmp(buf, "record") ||
811  !strcmp(buf, "receive"))
812  th->mode_record = 1;
813  }
814  }
815 
816  while (*p != ';' && *p != '\0' && *p != ',')
817  p++;
818  if (*p == ';')
819  p++;
820  }
821  if (*p == ',')
822  p++;
823 
824  reply->nb_transports++;
825  }
826 }
827 
828 static void handle_rtp_info(RTSPState *rt, const char *url,
829  uint32_t seq, uint32_t rtptime)
830 {
831  int i;
832  if (!rtptime || !url[0])
833  return;
834  if (rt->transport != RTSP_TRANSPORT_RTP)
835  return;
836  for (i = 0; i < rt->nb_rtsp_streams; i++) {
837  RTSPStream *rtsp_st = rt->rtsp_streams[i];
838  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
839  if (!rtpctx)
840  continue;
841  if (!strcmp(rtsp_st->control_url, url)) {
842  rtpctx->base_timestamp = rtptime;
843  break;
844  }
845  }
846 }
847 
848 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
849 {
850  int read = 0;
851  char key[20], value[1024], url[1024] = "";
852  uint32_t seq = 0, rtptime = 0;
853 
854  for (;;) {
855  p += strspn(p, SPACE_CHARS);
856  if (!*p)
857  break;
858  get_word_sep(key, sizeof(key), "=", &p);
859  if (*p != '=')
860  break;
861  p++;
862  get_word_sep(value, sizeof(value), ";, ", &p);
863  read++;
864  if (!strcmp(key, "url"))
865  av_strlcpy(url, value, sizeof(url));
866  else if (!strcmp(key, "seq"))
867  seq = strtoul(value, NULL, 10);
868  else if (!strcmp(key, "rtptime"))
869  rtptime = strtoul(value, NULL, 10);
870  if (*p == ',') {
871  handle_rtp_info(rt, url, seq, rtptime);
872  url[0] = '\0';
873  seq = rtptime = 0;
874  read = 0;
875  }
876  if (*p)
877  p++;
878  }
879  if (read > 0)
880  handle_rtp_info(rt, url, seq, rtptime);
881 }
882 
883 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
884  RTSPState *rt, const char *method)
885 {
886  const char *p;
887 
888  /* NOTE: we do case independent match for broken servers */
889  p = buf;
890  if (av_stristart(p, "Session:", &p)) {
891  int t;
892  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
893  if (av_stristart(p, ";timeout=", &p) &&
894  (t = strtol(p, NULL, 10)) > 0) {
895  reply->timeout = t;
896  }
897  } else if (av_stristart(p, "Content-Length:", &p)) {
898  reply->content_length = strtol(p, NULL, 10);
899  } else if (av_stristart(p, "Transport:", &p)) {
900  rtsp_parse_transport(reply, p);
901  } else if (av_stristart(p, "CSeq:", &p)) {
902  reply->seq = strtol(p, NULL, 10);
903  } else if (av_stristart(p, "Range:", &p)) {
904  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
905  } else if (av_stristart(p, "RealChallenge1:", &p)) {
906  p += strspn(p, SPACE_CHARS);
907  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
908  } else if (av_stristart(p, "Server:", &p)) {
909  p += strspn(p, SPACE_CHARS);
910  av_strlcpy(reply->server, p, sizeof(reply->server));
911  } else if (av_stristart(p, "Notice:", &p) ||
912  av_stristart(p, "X-Notice:", &p)) {
913  reply->notice = strtol(p, NULL, 10);
914  } else if (av_stristart(p, "Location:", &p)) {
915  p += strspn(p, SPACE_CHARS);
916  av_strlcpy(reply->location, p , sizeof(reply->location));
917  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
918  p += strspn(p, SPACE_CHARS);
919  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
920  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
921  p += strspn(p, SPACE_CHARS);
922  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
923  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
924  p += strspn(p, SPACE_CHARS);
925  if (method && !strcmp(method, "DESCRIBE"))
926  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
927  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
928  p += strspn(p, SPACE_CHARS);
929  if (method && !strcmp(method, "PLAY"))
930  rtsp_parse_rtp_info(rt, p);
931  } else if (av_stristart(p, "Public:", &p) && rt) {
932  if (strstr(p, "GET_PARAMETER") &&
933  method && !strcmp(method, "OPTIONS"))
934  rt->get_parameter_supported = 1;
935  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
936  p += strspn(p, SPACE_CHARS);
937  rt->accept_dynamic_rate = atoi(p);
938  } else if (av_stristart(p, "Content-Type:", &p)) {
939  p += strspn(p, SPACE_CHARS);
940  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
941  }
942 }
943 
944 /* skip a RTP/TCP interleaved packet */
946 {
947  RTSPState *rt = s->priv_data;
948  int ret, len, len1;
949  uint8_t buf[1024];
950 
951  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
952  if (ret != 3)
953  return;
954  len = AV_RB16(buf + 1);
955 
956  av_dlog(s, "skipping RTP packet len=%d\n", len);
957 
958  /* skip payload */
959  while (len > 0) {
960  len1 = len;
961  if (len1 > sizeof(buf))
962  len1 = sizeof(buf);
963  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
964  if (ret != len1)
965  return;
966  len -= len1;
967  }
968 }
969 
971  unsigned char **content_ptr,
972  int return_on_interleaved_data, const char *method)
973 {
974  RTSPState *rt = s->priv_data;
975  char buf[4096], buf1[1024], *q;
976  unsigned char ch;
977  const char *p;
978  int ret, content_length, line_count = 0, request = 0;
979  unsigned char *content = NULL;
980 
981 start:
982  line_count = 0;
983  request = 0;
984  content = NULL;
985  memset(reply, 0, sizeof(*reply));
986 
987  /* parse reply (XXX: use buffers) */
988  rt->last_reply[0] = '\0';
989  for (;;) {
990  q = buf;
991  for (;;) {
992  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
993  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
994  if (ret != 1)
995  return AVERROR_EOF;
996  if (ch == '\n')
997  break;
998  if (ch == '$') {
999  /* XXX: only parse it if first char on line ? */
1000  if (return_on_interleaved_data) {
1001  return 1;
1002  } else
1004  } else if (ch != '\r') {
1005  if ((q - buf) < sizeof(buf) - 1)
1006  *q++ = ch;
1007  }
1008  }
1009  *q = '\0';
1010 
1011  av_dlog(s, "line='%s'\n", buf);
1012 
1013  /* test if last line */
1014  if (buf[0] == '\0')
1015  break;
1016  p = buf;
1017  if (line_count == 0) {
1018  /* get reply code */
1019  get_word(buf1, sizeof(buf1), &p);
1020  if (!strncmp(buf1, "RTSP/", 5)) {
1021  get_word(buf1, sizeof(buf1), &p);
1022  reply->status_code = atoi(buf1);
1023  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1024  } else {
1025  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1026  get_word(buf1, sizeof(buf1), &p); // object
1027  request = 1;
1028  }
1029  } else {
1030  ff_rtsp_parse_line(reply, p, rt, method);
1031  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1032  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1033  }
1034  line_count++;
1035  }
1036 
1037  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1038  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1039 
1040  content_length = reply->content_length;
1041  if (content_length > 0) {
1042  /* leave some room for a trailing '\0' (useful for simple parsing) */
1043  content = av_malloc(content_length + 1);
1044  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1045  content[content_length] = '\0';
1046  }
1047  if (content_ptr)
1048  *content_ptr = content;
1049  else
1050  av_free(content);
1051 
1052  if (request) {
1053  char buf[1024];
1054  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1055  const char* ptr = buf;
1056 
1057  if (!strcmp(reply->reason, "OPTIONS")) {
1058  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1059  if (reply->seq)
1060  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1061  if (reply->session_id[0])
1062  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1063  reply->session_id);
1064  } else {
1065  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1066  }
1067  av_strlcat(buf, "\r\n", sizeof(buf));
1068 
1069  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1070  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1071  ptr = base64buf;
1072  }
1073  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1074 
1075  rt->last_cmd_time = av_gettime();
1076  /* Even if the request from the server had data, it is not the data
1077  * that the caller wants or expects. The memory could also be leaked
1078  * if the actual following reply has content data. */
1079  if (content_ptr)
1080  av_freep(content_ptr);
1081  /* If method is set, this is called from ff_rtsp_send_cmd,
1082  * where a reply to exactly this request is awaited. For
1083  * callers from within packet receiving, we just want to
1084  * return to the caller and go back to receiving packets. */
1085  if (method)
1086  goto start;
1087  return 0;
1088  }
1089 
1090  if (rt->seq != reply->seq) {
1091  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1092  rt->seq, reply->seq);
1093  }
1094 
1095  /* EOS */
1096  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1097  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1098  reply->notice == 2306 /* Continuous Feed Terminated */) {
1099  rt->state = RTSP_STATE_IDLE;
1100  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1101  return AVERROR(EIO); /* data or server error */
1102  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1103  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1104  return AVERROR(EPERM);
1105 
1106  return 0;
1107 }
1108 
1109 /**
1110  * Send a command to the RTSP server without waiting for the reply.
1111  *
1112  * @param s RTSP (de)muxer context
1113  * @param method the method for the request
1114  * @param url the target url for the request
1115  * @param headers extra header lines to include in the request
1116  * @param send_content if non-null, the data to send as request body content
1117  * @param send_content_length the length of the send_content data, or 0 if
1118  * send_content is null
1119  *
1120  * @return zero if success, nonzero otherwise
1121  */
1122 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1123  const char *method, const char *url,
1124  const char *headers,
1125  const unsigned char *send_content,
1126  int send_content_length)
1127 {
1128  RTSPState *rt = s->priv_data;
1129  char buf[4096], *out_buf;
1130  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1131 
1132  /* Add in RTSP headers */
1133  out_buf = buf;
1134  rt->seq++;
1135  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1136  if (headers)
1137  av_strlcat(buf, headers, sizeof(buf));
1138  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1139  if (rt->session_id[0] != '\0' && (!headers ||
1140  !strstr(headers, "\nIf-Match:"))) {
1141  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1142  }
1143  if (rt->auth[0]) {
1144  char *str = ff_http_auth_create_response(&rt->auth_state,
1145  rt->auth, url, method);
1146  if (str)
1147  av_strlcat(buf, str, sizeof(buf));
1148  av_free(str);
1149  }
1150  if (send_content_length > 0 && send_content)
1151  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1152  av_strlcat(buf, "\r\n", sizeof(buf));
1153 
1154  /* base64 encode rtsp if tunneling */
1155  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1156  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1157  out_buf = base64buf;
1158  }
1159 
1160  av_dlog(s, "Sending:\n%s--\n", buf);
1161 
1162  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1163  if (send_content_length > 0 && send_content) {
1164  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1165  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1166  "with content data not supported\n");
1167  return AVERROR_PATCHWELCOME;
1168  }
1169  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1170  }
1171  rt->last_cmd_time = av_gettime();
1172 
1173  return 0;
1174 }
1175 
1176 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1177  const char *url, const char *headers)
1178 {
1179  return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1180 }
1181 
1182 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1183  const char *headers, RTSPMessageHeader *reply,
1184  unsigned char **content_ptr)
1185 {
1186  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1187  content_ptr, NULL, 0);
1188 }
1189 
1191  const char *method, const char *url,
1192  const char *header,
1193  RTSPMessageHeader *reply,
1194  unsigned char **content_ptr,
1195  const unsigned char *send_content,
1196  int send_content_length)
1197 {
1198  RTSPState *rt = s->priv_data;
1199  HTTPAuthType cur_auth_type;
1200  int ret, attempts = 0;
1201 
1202 retry:
1203  cur_auth_type = rt->auth_state.auth_type;
1204  if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1205  send_content,
1206  send_content_length)))
1207  return ret;
1208 
1209  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1210  return ret;
1211  attempts++;
1212 
1213  if (reply->status_code == 401 &&
1214  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1215  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1216  goto retry;
1217 
1218  if (reply->status_code > 400){
1219  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1220  method,
1221  reply->status_code,
1222  reply->reason);
1223  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1224  }
1225 
1226  return 0;
1227 }
1228 
1229 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1230  int lower_transport, const char *real_challenge)
1231 {
1232  RTSPState *rt = s->priv_data;
1233  int rtx = 0, j, i, err, interleave = 0, port_off;
1234  RTSPStream *rtsp_st;
1235  RTSPMessageHeader reply1, *reply = &reply1;
1236  char cmd[2048];
1237  const char *trans_pref;
1238 
1239  if (rt->transport == RTSP_TRANSPORT_RDT)
1240  trans_pref = "x-pn-tng";
1241  else if (rt->transport == RTSP_TRANSPORT_RAW)
1242  trans_pref = "RAW/RAW";
1243  else
1244  trans_pref = "RTP/AVP";
1245 
1246  /* default timeout: 1 minute */
1247  rt->timeout = 60;
1248 
1249  /* Choose a random starting offset within the first half of the
1250  * port range, to allow for a number of ports to try even if the offset
1251  * happens to be at the end of the random range. */
1252  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1253  /* even random offset */
1254  port_off -= port_off & 0x01;
1255 
1256  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1257  char transport[2048];
1258 
1259  /*
1260  * WMS serves all UDP data over a single connection, the RTX, which
1261  * isn't necessarily the first in the SDP but has to be the first
1262  * to be set up, else the second/third SETUP will fail with a 461.
1263  */
1264  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1265  rt->server_type == RTSP_SERVER_WMS) {
1266  if (i == 0) {
1267  /* rtx first */
1268  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1269  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1270  if (len >= 4 &&
1271  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1272  "/rtx"))
1273  break;
1274  }
1275  if (rtx == rt->nb_rtsp_streams)
1276  return -1; /* no RTX found */
1277  rtsp_st = rt->rtsp_streams[rtx];
1278  } else
1279  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1280  } else
1281  rtsp_st = rt->rtsp_streams[i];
1282 
1283  /* RTP/UDP */
1284  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1285  char buf[256];
1286 
1287  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1288  port = reply->transports[0].client_port_min;
1289  goto have_port;
1290  }
1291 
1292  /* first try in specified port range */
1293  while (j <= rt->rtp_port_max) {
1294  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1295  "?localport=%d", j);
1296  /* we will use two ports per rtp stream (rtp and rtcp) */
1297  j += 2;
1298  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1299  &s->interrupt_callback, NULL))
1300  goto rtp_opened;
1301  }
1302  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1303  err = AVERROR(EIO);
1304  goto fail;
1305 
1306  rtp_opened:
1307  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1308  have_port:
1309  snprintf(transport, sizeof(transport) - 1,
1310  "%s/UDP;", trans_pref);
1311  if (rt->server_type != RTSP_SERVER_REAL)
1312  av_strlcat(transport, "unicast;", sizeof(transport));
1313  av_strlcatf(transport, sizeof(transport),
1314  "client_port=%d", port);
1315  if (rt->transport == RTSP_TRANSPORT_RTP &&
1316  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1317  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1318  }
1319 
1320  /* RTP/TCP */
1321  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1322  /* For WMS streams, the application streams are only used for
1323  * UDP. When trying to set it up for TCP streams, the server
1324  * will return an error. Therefore, we skip those streams. */
1325  if (rt->server_type == RTSP_SERVER_WMS &&
1326  (rtsp_st->stream_index < 0 ||
1327  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1329  continue;
1330  snprintf(transport, sizeof(transport) - 1,
1331  "%s/TCP;", trans_pref);
1332  if (rt->transport != RTSP_TRANSPORT_RDT)
1333  av_strlcat(transport, "unicast;", sizeof(transport));
1334  av_strlcatf(transport, sizeof(transport),
1335  "interleaved=%d-%d",
1336  interleave, interleave + 1);
1337  interleave += 2;
1338  }
1339 
1340  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1341  snprintf(transport, sizeof(transport) - 1,
1342  "%s/UDP;multicast", trans_pref);
1343  }
1344  if (s->oformat) {
1345  av_strlcat(transport, ";mode=record", sizeof(transport));
1346  } else if (rt->server_type == RTSP_SERVER_REAL ||
1348  av_strlcat(transport, ";mode=play", sizeof(transport));
1349  snprintf(cmd, sizeof(cmd),
1350  "Transport: %s\r\n",
1351  transport);
1352  if (rt->accept_dynamic_rate)
1353  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1354  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1355  char real_res[41], real_csum[9];
1356  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1357  real_challenge);
1358  av_strlcatf(cmd, sizeof(cmd),
1359  "If-Match: %s\r\n"
1360  "RealChallenge2: %s, sd=%s\r\n",
1361  rt->session_id, real_res, real_csum);
1362  }
1363  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1364  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1365  err = 1;
1366  goto fail;
1367  } else if (reply->status_code != RTSP_STATUS_OK ||
1368  reply->nb_transports != 1) {
1369  err = AVERROR_INVALIDDATA;
1370  goto fail;
1371  }
1372 
1373  /* XXX: same protocol for all streams is required */
1374  if (i > 0) {
1375  if (reply->transports[0].lower_transport != rt->lower_transport ||
1376  reply->transports[0].transport != rt->transport) {
1377  err = AVERROR_INVALIDDATA;
1378  goto fail;
1379  }
1380  } else {
1381  rt->lower_transport = reply->transports[0].lower_transport;
1382  rt->transport = reply->transports[0].transport;
1383  }
1384 
1385  /* Fail if the server responded with another lower transport mode
1386  * than what we requested. */
1387  if (reply->transports[0].lower_transport != lower_transport) {
1388  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1389  err = AVERROR_INVALIDDATA;
1390  goto fail;
1391  }
1392 
1393  switch(reply->transports[0].lower_transport) {
1395  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1396  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1397  break;
1398 
1399  case RTSP_LOWER_TRANSPORT_UDP: {
1400  char url[1024], options[30] = "";
1401 
1402  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1403  av_strlcpy(options, "?connect=1", sizeof(options));
1404  /* Use source address if specified */
1405  if (reply->transports[0].source[0]) {
1406  ff_url_join(url, sizeof(url), "rtp", NULL,
1407  reply->transports[0].source,
1408  reply->transports[0].server_port_min, "%s", options);
1409  } else {
1410  ff_url_join(url, sizeof(url), "rtp", NULL, host,
1411  reply->transports[0].server_port_min, "%s", options);
1412  }
1413  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1414  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1415  err = AVERROR_INVALIDDATA;
1416  goto fail;
1417  }
1418  /* Try to initialize the connection state in a
1419  * potential NAT router by sending dummy packets.
1420  * RTP/RTCP dummy packets are used for RDT, too.
1421  */
1422  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1423  CONFIG_RTPDEC)
1425  break;
1426  }
1428  char url[1024], namebuf[50], optbuf[20] = "";
1429  struct sockaddr_storage addr;
1430  int port, ttl;
1431 
1432  if (reply->transports[0].destination.ss_family) {
1433  addr = reply->transports[0].destination;
1434  port = reply->transports[0].port_min;
1435  ttl = reply->transports[0].ttl;
1436  } else {
1437  addr = rtsp_st->sdp_ip;
1438  port = rtsp_st->sdp_port;
1439  ttl = rtsp_st->sdp_ttl;
1440  }
1441  if (ttl > 0)
1442  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1443  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1444  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1445  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1446  port, "%s", optbuf);
1447  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1448  &s->interrupt_callback, NULL) < 0) {
1449  err = AVERROR_INVALIDDATA;
1450  goto fail;
1451  }
1452  break;
1453  }
1454  }
1455 
1456  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1457  goto fail;
1458  }
1459 
1460  if (rt->nb_rtsp_streams && reply->timeout > 0)
1461  rt->timeout = reply->timeout;
1462 
1463  if (rt->server_type == RTSP_SERVER_REAL)
1464  rt->need_subscription = 1;
1465 
1466  return 0;
1467 
1468 fail:
1469  ff_rtsp_undo_setup(s);
1470  return err;
1471 }
1472 
1474 {
1475  RTSPState *rt = s->priv_data;
1476  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1477  ffurl_close(rt->rtsp_hd);
1478  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1479 }
1480 
1482 {
1483  RTSPState *rt = s->priv_data;
1484  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1485  int port, err, tcp_fd;
1486  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1487  int lower_transport_mask = 0;
1488  char real_challenge[64] = "";
1489  struct sockaddr_storage peer;
1490  socklen_t peer_len = sizeof(peer);
1491 
1492  if (rt->rtp_port_max < rt->rtp_port_min) {
1493  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1494  "than min port %d\n", rt->rtp_port_max,
1495  rt->rtp_port_min);
1496  return AVERROR(EINVAL);
1497  }
1498 
1499  if (!ff_network_init())
1500  return AVERROR(EIO);
1501 
1502  if (s->max_delay < 0) /* Not set by the caller */
1504 
1509  }
1510  /* Only pass through valid flags from here */
1512 
1513 redirect:
1514  lower_transport_mask = rt->lower_transport_mask;
1515  /* extract hostname and port */
1516  av_url_split(NULL, 0, auth, sizeof(auth),
1517  host, sizeof(host), &port, path, sizeof(path), s->filename);
1518  if (*auth) {
1519  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1520  }
1521  if (port < 0)
1522  port = RTSP_DEFAULT_PORT;
1523 
1524  if (!lower_transport_mask)
1525  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1526 
1527  if (s->oformat) {
1528  /* Only UDP or TCP - UDP multicast isn't supported. */
1529  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1530  (1 << RTSP_LOWER_TRANSPORT_TCP);
1531  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1532  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1533  "only UDP and TCP are supported for output.\n");
1534  err = AVERROR(EINVAL);
1535  goto fail;
1536  }
1537  }
1538 
1539  /* Construct the URI used in request; this is similar to s->filename,
1540  * but with authentication credentials removed and RTSP specific options
1541  * stripped out. */
1542  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1543  host, port, "%s", path);
1544 
1545  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1546  /* set up initial handshake for tunneling */
1547  char httpname[1024];
1548  char sessioncookie[17];
1549  char headers[1024];
1550 
1551  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1552  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1554 
1555  /* GET requests */
1556  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1557  &s->interrupt_callback) < 0) {
1558  err = AVERROR(EIO);
1559  goto fail;
1560  }
1561 
1562  /* generate GET headers */
1563  snprintf(headers, sizeof(headers),
1564  "x-sessioncookie: %s\r\n"
1565  "Accept: application/x-rtsp-tunnelled\r\n"
1566  "Pragma: no-cache\r\n"
1567  "Cache-Control: no-cache\r\n",
1568  sessioncookie);
1569  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1570 
1571  /* complete the connection */
1572  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1573  err = AVERROR(EIO);
1574  goto fail;
1575  }
1576 
1577  /* POST requests */
1578  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1579  &s->interrupt_callback) < 0 ) {
1580  err = AVERROR(EIO);
1581  goto fail;
1582  }
1583 
1584  /* generate POST headers */
1585  snprintf(headers, sizeof(headers),
1586  "x-sessioncookie: %s\r\n"
1587  "Content-Type: application/x-rtsp-tunnelled\r\n"
1588  "Pragma: no-cache\r\n"
1589  "Cache-Control: no-cache\r\n"
1590  "Content-Length: 32767\r\n"
1591  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1592  sessioncookie);
1593  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1594  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1595 
1596  /* Initialize the authentication state for the POST session. The HTTP
1597  * protocol implementation doesn't properly handle multi-pass
1598  * authentication for POST requests, since it would require one of
1599  * the following:
1600  * - implementing Expect: 100-continue, which many HTTP servers
1601  * don't support anyway, even less the RTSP servers that do HTTP
1602  * tunneling
1603  * - sending the whole POST data until getting a 401 reply specifying
1604  * what authentication method to use, then resending all that data
1605  * - waiting for potential 401 replies directly after sending the
1606  * POST header (waiting for some unspecified time)
1607  * Therefore, we copy the full auth state, which works for both basic
1608  * and digest. (For digest, we would have to synchronize the nonce
1609  * count variable between the two sessions, if we'd do more requests
1610  * with the original session, though.)
1611  */
1613 
1614  /* complete the connection */
1615  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1616  err = AVERROR(EIO);
1617  goto fail;
1618  }
1619  } else {
1620  /* open the tcp connection */
1621  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1622  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1623  &s->interrupt_callback, NULL) < 0) {
1624  err = AVERROR(EIO);
1625  goto fail;
1626  }
1627  rt->rtsp_hd_out = rt->rtsp_hd;
1628  }
1629  rt->seq = 0;
1630 
1631  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1632  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1633  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1634  NULL, 0, NI_NUMERICHOST);
1635  }
1636 
1637  /* request options supported by the server; this also detects server
1638  * type */
1639  for (rt->server_type = RTSP_SERVER_RTP;;) {
1640  cmd[0] = 0;
1641  if (rt->server_type == RTSP_SERVER_REAL)
1642  av_strlcat(cmd,
1643  /*
1644  * The following entries are required for proper
1645  * streaming from a Realmedia server. They are
1646  * interdependent in some way although we currently
1647  * don't quite understand how. Values were copied
1648  * from mplayer SVN r23589.
1649  * ClientChallenge is a 16-byte ID in hex
1650  * CompanyID is a 16-byte ID in base64
1651  */
1652  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1653  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1654  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1655  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1656  sizeof(cmd));
1657  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1658  if (reply->status_code != RTSP_STATUS_OK) {
1659  err = AVERROR_INVALIDDATA;
1660  goto fail;
1661  }
1662 
1663  /* detect server type if not standard-compliant RTP */
1664  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1666  continue;
1667  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1669  } else if (rt->server_type == RTSP_SERVER_REAL)
1670  strcpy(real_challenge, reply->real_challenge);
1671  break;
1672  }
1673 
1674  if (s->iformat && CONFIG_RTSP_DEMUXER)
1675  err = ff_rtsp_setup_input_streams(s, reply);
1676  else if (CONFIG_RTSP_MUXER)
1677  err = ff_rtsp_setup_output_streams(s, host);
1678  if (err)
1679  goto fail;
1680 
1681  do {
1682  int lower_transport = ff_log2_tab[lower_transport_mask &
1683  ~(lower_transport_mask - 1)];
1684 
1685  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1686  rt->server_type == RTSP_SERVER_REAL ?
1687  real_challenge : NULL);
1688  if (err < 0)
1689  goto fail;
1690  lower_transport_mask &= ~(1 << lower_transport);
1691  if (lower_transport_mask == 0 && err == 1) {
1692  err = AVERROR(EPROTONOSUPPORT);
1693  goto fail;
1694  }
1695  } while (err);
1696 
1697  rt->lower_transport_mask = lower_transport_mask;
1698  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1699  rt->state = RTSP_STATE_IDLE;
1700  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1701  return 0;
1702  fail:
1705  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1706  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1707  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1708  reply->status_code,
1709  s->filename);
1710  goto redirect;
1711  }
1712  ff_network_close();
1713  return err;
1714 }
1715 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1716 
1717 #if CONFIG_RTPDEC
1718 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1719  uint8_t *buf, int buf_size, int64_t wait_end)
1720 {
1721  RTSPState *rt = s->priv_data;
1722  RTSPStream *rtsp_st;
1723  int n, i, ret, tcp_fd, timeout_cnt = 0;
1724  int max_p = 0;
1725  struct pollfd *p = rt->p;
1726  int *fds = NULL, fdsnum, fdsidx;
1727 
1728  for (;;) {
1730  return AVERROR_EXIT;
1731  if (wait_end && wait_end - av_gettime() < 0)
1732  return AVERROR(EAGAIN);
1733  max_p = 0;
1734  if (rt->rtsp_hd) {
1735  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1736  p[max_p].fd = tcp_fd;
1737  p[max_p++].events = POLLIN;
1738  } else {
1739  tcp_fd = -1;
1740  }
1741  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1742  rtsp_st = rt->rtsp_streams[i];
1743  if (rtsp_st->rtp_handle) {
1744  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1745  &fds, &fdsnum)) {
1746  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1747  return ret;
1748  }
1749  if (fdsnum != 2) {
1750  av_log(s, AV_LOG_ERROR,
1751  "Number of fds %d not supported\n", fdsnum);
1752  return AVERROR_INVALIDDATA;
1753  }
1754  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1755  p[max_p].fd = fds[fdsidx];
1756  p[max_p++].events = POLLIN;
1757  }
1758  av_free(fds);
1759  }
1760  }
1761  n = poll(p, max_p, POLL_TIMEOUT_MS);
1762  if (n > 0) {
1763  int j = 1 - (tcp_fd == -1);
1764  timeout_cnt = 0;
1765  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1766  rtsp_st = rt->rtsp_streams[i];
1767  if (rtsp_st->rtp_handle) {
1768  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1769  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1770  if (ret > 0) {
1771  *prtsp_st = rtsp_st;
1772  return ret;
1773  }
1774  }
1775  j+=2;
1776  }
1777  }
1778 #if CONFIG_RTSP_DEMUXER
1779  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1780  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1781  if (rt->state == RTSP_STATE_STREAMING) {
1783  return AVERROR_EOF;
1784  else
1786  "Unable to answer to TEARDOWN\n");
1787  } else
1788  return 0;
1789  } else {
1790  RTSPMessageHeader reply;
1791  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1792  if (ret < 0)
1793  return ret;
1794  /* XXX: parse message */
1795  if (rt->state != RTSP_STATE_STREAMING)
1796  return 0;
1797  }
1798  }
1799 #endif
1800  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1801  return AVERROR(ETIMEDOUT);
1802  } else if (n < 0 && errno != EINTR)
1803  return AVERROR(errno);
1804  }
1805 }
1806 
1807 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1808  const uint8_t *buf, int len)
1809 {
1810  RTSPState *rt = s->priv_data;
1811  int i;
1812  if (len < 0)
1813  return len;
1814  if (rt->nb_rtsp_streams == 1) {
1815  *rtsp_st = rt->rtsp_streams[0];
1816  return len;
1817  }
1818  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1819  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1820  int no_ssrc = 0;
1821  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1822  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1823  if (!rtpctx)
1824  continue;
1825  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1826  *rtsp_st = rt->rtsp_streams[i];
1827  return len;
1828  }
1829  if (!rtpctx->ssrc)
1830  no_ssrc = 1;
1831  }
1832  if (no_ssrc) {
1834  "Unable to pick stream for packet - SSRC not known for "
1835  "all streams\n");
1836  return AVERROR(EAGAIN);
1837  }
1838  } else {
1839  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1840  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1841  *rtsp_st = rt->rtsp_streams[i];
1842  return len;
1843  }
1844  }
1845  }
1846  }
1847  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1848  return AVERROR(EAGAIN);
1849 }
1850 
1852 {
1853  RTSPState *rt = s->priv_data;
1854  int ret, len;
1855  RTSPStream *rtsp_st, *first_queue_st = NULL;
1856  int64_t wait_end = 0;
1857 
1858  if (rt->nb_byes == rt->nb_rtsp_streams)
1859  return AVERROR_EOF;
1860 
1861  /* get next frames from the same RTP packet */
1862  if (rt->cur_transport_priv) {
1863  if (rt->transport == RTSP_TRANSPORT_RDT) {
1864  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1865  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1866  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1867  } else if (rt->ts && CONFIG_RTPDEC) {
1868  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1869  if (ret >= 0) {
1870  rt->recvbuf_pos += ret;
1871  ret = rt->recvbuf_pos < rt->recvbuf_len;
1872  }
1873  } else
1874  ret = -1;
1875  if (ret == 0) {
1876  rt->cur_transport_priv = NULL;
1877  return 0;
1878  } else if (ret == 1) {
1879  return 0;
1880  } else
1881  rt->cur_transport_priv = NULL;
1882  }
1883 
1884 redo:
1885  if (rt->transport == RTSP_TRANSPORT_RTP) {
1886  int i;
1887  int64_t first_queue_time = 0;
1888  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1889  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1890  int64_t queue_time;
1891  if (!rtpctx)
1892  continue;
1893  queue_time = ff_rtp_queued_packet_time(rtpctx);
1894  if (queue_time && (queue_time - first_queue_time < 0 ||
1895  !first_queue_time)) {
1896  first_queue_time = queue_time;
1897  first_queue_st = rt->rtsp_streams[i];
1898  }
1899  }
1900  if (first_queue_time) {
1901  wait_end = first_queue_time + s->max_delay;
1902  } else {
1903  wait_end = 0;
1904  first_queue_st = NULL;
1905  }
1906  }
1907 
1908  /* read next RTP packet */
1909  if (!rt->recvbuf) {
1911  if (!rt->recvbuf)
1912  return AVERROR(ENOMEM);
1913  }
1914 
1915  switch(rt->lower_transport) {
1916  default:
1917 #if CONFIG_RTSP_DEMUXER
1919  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1920  break;
1921 #endif
1924  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1925  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1926  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1927  break;
1929  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1930  wait_end && wait_end < av_gettime())
1931  len = AVERROR(EAGAIN);
1932  else
1933  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1934  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1935  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1937  break;
1938  }
1939  if (len == AVERROR(EAGAIN) && first_queue_st &&
1940  rt->transport == RTSP_TRANSPORT_RTP) {
1941  rtsp_st = first_queue_st;
1942  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1943  goto end;
1944  }
1945  if (len < 0)
1946  return len;
1947  if (len == 0)
1948  return AVERROR_EOF;
1949  if (rt->transport == RTSP_TRANSPORT_RDT) {
1950  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1951  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1952  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1953  if (rtsp_st->feedback) {
1954  AVIOContext *pb = NULL;
1956  pb = s->pb;
1957  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1958  }
1959  if (ret < 0) {
1960  /* Either bad packet, or a RTCP packet. Check if the
1961  * first_rtcp_ntp_time field was initialized. */
1962  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1963  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1964  /* first_rtcp_ntp_time has been initialized for this stream,
1965  * copy the same value to all other uninitialized streams,
1966  * in order to map their timestamp origin to the same ntp time
1967  * as this one. */
1968  int i;
1969  AVStream *st = NULL;
1970  if (rtsp_st->stream_index >= 0)
1971  st = s->streams[rtsp_st->stream_index];
1972  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1974  AVStream *st2 = NULL;
1975  if (rt->rtsp_streams[i]->stream_index >= 0)
1976  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1977  if (rtpctx2 && st && st2 &&
1978  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1979  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1980  rtpctx2->rtcp_ts_offset = av_rescale_q(
1981  rtpctx->rtcp_ts_offset, st->time_base,
1982  st2->time_base);
1983  }
1984  }
1985  }
1986  if (ret == -RTCP_BYE) {
1987  rt->nb_byes++;
1988 
1989  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1990  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1991 
1992  if (rt->nb_byes == rt->nb_rtsp_streams)
1993  return AVERROR_EOF;
1994  }
1995  }
1996  } else if (rt->ts && CONFIG_RTPDEC) {
1997  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1998  if (ret >= 0) {
1999  if (ret < len) {
2000  rt->recvbuf_len = len;
2001  rt->recvbuf_pos = ret;
2002  rt->cur_transport_priv = rt->ts;
2003  return 1;
2004  } else {
2005  ret = 0;
2006  }
2007  }
2008  } else {
2009  return AVERROR_INVALIDDATA;
2010  }
2011 end:
2012  if (ret < 0)
2013  goto redo;
2014  if (ret == 1)
2015  /* more packets may follow, so we save the RTP context */
2016  rt->cur_transport_priv = rtsp_st->transport_priv;
2017 
2018  return ret;
2019 }
2020 #endif /* CONFIG_RTPDEC */
2021 
2022 #if CONFIG_SDP_DEMUXER
2023 static int sdp_probe(AVProbeData *p1)
2024 {
2025  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2026 
2027  /* we look for a line beginning "c=IN IP" */
2028  while (p < p_end && *p != '\0') {
2029  if (p + sizeof("c=IN IP") - 1 < p_end &&
2030  av_strstart(p, "c=IN IP", NULL))
2031  return AVPROBE_SCORE_MAX / 2;
2032 
2033  while (p < p_end - 1 && *p != '\n') p++;
2034  if (++p >= p_end)
2035  break;
2036  if (*p == '\r')
2037  p++;
2038  }
2039  return 0;
2040 }
2041 
2042 static int sdp_read_header(AVFormatContext *s)
2043 {
2044  RTSPState *rt = s->priv_data;
2045  RTSPStream *rtsp_st;
2046  int size, i, err;
2047  char *content;
2048  char url[1024];
2049 
2050  if (!ff_network_init())
2051  return AVERROR(EIO);
2052 
2053  if (s->max_delay < 0) /* Not set by the caller */
2055  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2057 
2058  /* read the whole sdp file */
2059  /* XXX: better loading */
2060  content = av_malloc(SDP_MAX_SIZE);
2061  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2062  if (size <= 0) {
2063  av_free(content);
2064  return AVERROR_INVALIDDATA;
2065  }
2066  content[size] ='\0';
2067 
2068  err = ff_sdp_parse(s, content);
2069  av_free(content);
2070  if (err) goto fail;
2071 
2072  /* open each RTP stream */
2073  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074  char namebuf[50];
2075  rtsp_st = rt->rtsp_streams[i];
2076 
2077  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2078  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2079  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2080  ff_url_join(url, sizeof(url), "rtp", NULL,
2081  namebuf, rtsp_st->sdp_port,
2082  "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2083  rtsp_st->sdp_ttl,
2084  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2085  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2086  &s->interrupt_callback, NULL) < 0) {
2087  err = AVERROR_INVALIDDATA;
2088  goto fail;
2089  }
2090  }
2091  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2092  goto fail;
2093  }
2094  return 0;
2095 fail:
2097  ff_network_close();
2098  return err;
2099 }
2100 
2101 static int sdp_read_close(AVFormatContext *s)
2102 {
2104  ff_network_close();
2105  return 0;
2106 }
2107 
2108 static const AVClass sdp_demuxer_class = {
2109  .class_name = "SDP demuxer",
2110  .item_name = av_default_item_name,
2111  .option = sdp_options,
2112  .version = LIBAVUTIL_VERSION_INT,
2113 };
2114 
2115 AVInputFormat ff_sdp_demuxer = {
2116  .name = "sdp",
2117  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2118  .priv_data_size = sizeof(RTSPState),
2119  .read_probe = sdp_probe,
2120  .read_header = sdp_read_header,
2122  .read_close = sdp_read_close,
2123  .priv_class = &sdp_demuxer_class,
2124 };
2125 #endif /* CONFIG_SDP_DEMUXER */
2126 
2127 #if CONFIG_RTP_DEMUXER
2128 static int rtp_probe(AVProbeData *p)
2129 {
2130  if (av_strstart(p->filename, "rtp:", NULL))
2131  return AVPROBE_SCORE_MAX;
2132  return 0;
2133 }
2134 
2135 static int rtp_read_header(AVFormatContext *s)
2136 {
2137  uint8_t recvbuf[1500];
2138  char host[500], sdp[500];
2139  int ret, port;
2140  URLContext* in = NULL;
2141  int payload_type;
2142  AVCodecContext codec = { 0 };
2143  struct sockaddr_storage addr;
2144  AVIOContext pb;
2145  socklen_t addrlen = sizeof(addr);
2146  RTSPState *rt = s->priv_data;
2147 
2148  if (!ff_network_init())
2149  return AVERROR(EIO);
2150 
2151  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2152  &s->interrupt_callback, NULL);
2153  if (ret)
2154  goto fail;
2155 
2156  while (1) {
2157  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2158  if (ret == AVERROR(EAGAIN))
2159  continue;
2160  if (ret < 0)
2161  goto fail;
2162  if (ret < 12) {
2163  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2164  continue;
2165  }
2166 
2167  if ((recvbuf[0] & 0xc0) != 0x80) {
2168  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2169  "received\n");
2170  continue;
2171  }
2172 
2173  if (RTP_PT_IS_RTCP(recvbuf[1]))
2174  continue;
2175 
2176  payload_type = recvbuf[1] & 0x7f;
2177  break;
2178  }
2179  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2180  ffurl_close(in);
2181  in = NULL;
2182 
2183  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2184  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2185  "without an SDP file describing it\n",
2186  payload_type);
2187  goto fail;
2188  }
2189  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2190  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2191  "properly you need an SDP file "
2192  "describing it\n");
2193  }
2194 
2195  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2196  NULL, 0, s->filename);
2197 
2198  snprintf(sdp, sizeof(sdp),
2199  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2200  addr.ss_family == AF_INET ? 4 : 6, host,
2201  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2202  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2203  port, payload_type);
2204  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2205 
2206  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2207  s->pb = &pb;
2208 
2209  /* sdp_read_header initializes this again */
2210  ff_network_close();
2211 
2212  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2213 
2214  ret = sdp_read_header(s);
2215  s->pb = NULL;
2216  return ret;
2217 
2218 fail:
2219  if (in)
2220  ffurl_close(in);
2221  ff_network_close();
2222  return ret;
2223 }
2224 
2225 static const AVClass rtp_demuxer_class = {
2226  .class_name = "RTP demuxer",
2227  .item_name = av_default_item_name,
2228  .option = rtp_options,
2229  .version = LIBAVUTIL_VERSION_INT,
2230 };
2231 
2232 AVInputFormat ff_rtp_demuxer = {
2233  .name = "rtp",
2234  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2235  .priv_data_size = sizeof(RTSPState),
2236  .read_probe = rtp_probe,
2237  .read_header = rtp_read_header,
2239  .read_close = sdp_read_close,
2240  .flags = AVFMT_NOFILE,
2241  .priv_class = &rtp_demuxer_class,
2242 };
2243 #endif /* CONFIG_RTP_DEMUXER */