Libav
rtsp.c
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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 "rtpproto.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 /* Timeout values for socket poll, in ms,
53  * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
60 
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 
65 #define RTSP_FLAG_OPTS(name, longname) \
66  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67  { "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" }
68 
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70  { 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" }, \
71  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 
75 #define RTSP_REORDERING_OPTS() \
76  { "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 }
77 
79  { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81  { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86  RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87  { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
88  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89  { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90  { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91  { "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 },
93  { NULL },
94 };
95 
96 static const AVOption sdp_options[] = {
97  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
98  { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
99  { "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
100  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102  { NULL },
103 };
104 
105 static const AVOption rtp_options[] = {
106  RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108  { NULL },
109 };
110 
111 static void get_word_until_chars(char *buf, int buf_size,
112  const char *sep, const char **pp)
113 {
114  const char *p;
115  char *q;
116 
117  p = *pp;
118  p += strspn(p, SPACE_CHARS);
119  q = buf;
120  while (!strchr(sep, *p) && *p != '\0') {
121  if ((q - buf) < buf_size - 1)
122  *q++ = *p;
123  p++;
124  }
125  if (buf_size > 0)
126  *q = '\0';
127  *pp = p;
128 }
129 
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
131  const char **pp)
132 {
133  if (**pp == '/') (*pp)++;
134  get_word_until_chars(buf, buf_size, sep, pp);
135 }
136 
137 static void get_word(char *buf, int buf_size, const char **pp)
138 {
139  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
140 }
141 
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
147 {
148  char buf[256];
149 
150  p += strspn(p, SPACE_CHARS);
151  if (!av_stristart(p, "npt=", &p))
152  return;
153 
154  *start = AV_NOPTS_VALUE;
155  *end = AV_NOPTS_VALUE;
156 
157  get_word_sep(buf, sizeof(buf), "-", &p);
158  av_parse_time(start, buf, 1);
159  if (*p == '-') {
160  p++;
161  get_word_sep(buf, sizeof(buf), "-", &p);
162  av_parse_time(end, buf, 1);
163  }
164 }
165 
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
167 {
168  struct addrinfo hints = { 0 }, *ai = NULL;
169  hints.ai_flags = AI_NUMERICHOST;
170  if (getaddrinfo(buf, NULL, &hints, &ai))
171  return -1;
172  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
173  freeaddrinfo(ai);
174  return 0;
175 }
176 
177 #if CONFIG_RTPDEC
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179  RTSPStream *rtsp_st, AVCodecContext *codec)
180 {
181  if (!handler)
182  return;
183  if (codec)
184  codec->codec_id = handler->codec_id;
185  rtsp_st->dynamic_handler = handler;
186  if (handler->alloc) {
187  rtsp_st->dynamic_protocol_context = handler->alloc();
188  if (!rtsp_st->dynamic_protocol_context)
189  rtsp_st->dynamic_handler = NULL;
190  }
191 }
192 
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195  AVStream *st, RTSPStream *rtsp_st,
196  int payload_type, const char *p)
197 {
198  AVCodecContext *codec = st->codec;
199  char buf[256];
200  int i;
201  AVCodec *c;
202  const char *c_name;
203 
204  /* See if we can handle this kind of payload.
205  * The space should normally not be there but some Real streams or
206  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207  * have a trailing space. */
208  get_word_sep(buf, sizeof(buf), "/ ", &p);
209  if (payload_type < RTP_PT_PRIVATE) {
210  /* We are in a standard case
211  * (from http://www.iana.org/assignments/rtp-parameters). */
212  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
213  }
214 
215  if (codec->codec_id == AV_CODEC_ID_NONE) {
216  RTPDynamicProtocolHandler *handler =
218  init_rtp_handler(handler, rtsp_st, codec);
219  /* If no dynamic handler was found, check with the list of standard
220  * allocated types, if such a stream for some reason happens to
221  * use a private payload type. This isn't handled in rtpdec.c, since
222  * the format name from the rtpmap line never is passed into rtpdec. */
223  if (!rtsp_st->dynamic_handler)
224  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
225  }
226 
227  c = avcodec_find_decoder(codec->codec_id);
228  if (c && c->name)
229  c_name = c->name;
230  else
231  c_name = "(null)";
232 
233  get_word_sep(buf, sizeof(buf), "/", &p);
234  i = atoi(buf);
235  switch (codec->codec_type) {
236  case AVMEDIA_TYPE_AUDIO:
237  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240  if (i > 0) {
241  codec->sample_rate = i;
242  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243  get_word_sep(buf, sizeof(buf), "/", &p);
244  i = atoi(buf);
245  if (i > 0)
246  codec->channels = i;
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;
291  int nb_default_include_source_addrs;
292  struct RTSPSource **default_include_source_addrs;
293  int nb_default_exclude_source_addrs;
294  struct RTSPSource **default_exclude_source_addrs;
295 } SDPParseState;
296 
297 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
298  struct RTSPSource ***dest, int *dest_count)
299 {
300  RTSPSource *rtsp_src, *rtsp_src2;
301  int i;
302  for (i = 0; i < count; i++) {
303  rtsp_src = addrs[i];
304  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
305  if (!rtsp_src2)
306  continue;
307  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
308  dynarray_add(dest, dest_count, rtsp_src2);
309  }
310 }
311 
312 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
313  int letter, const char *buf)
314 {
315  RTSPState *rt = s->priv_data;
316  char buf1[64], st_type[64];
317  const char *p;
318  enum AVMediaType codec_type;
319  int payload_type, i;
320  AVStream *st;
321  RTSPStream *rtsp_st;
322  RTSPSource *rtsp_src;
323  struct sockaddr_storage sdp_ip;
324  int ttl;
325 
326  av_dlog(s, "sdp: %c='%s'\n", letter, buf);
327 
328  p = buf;
329  if (s1->skip_media && letter != 'm')
330  return;
331  switch (letter) {
332  case 'c':
333  get_word(buf1, sizeof(buf1), &p);
334  if (strcmp(buf1, "IN") != 0)
335  return;
336  get_word(buf1, sizeof(buf1), &p);
337  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
338  return;
339  get_word_sep(buf1, sizeof(buf1), "/", &p);
340  if (get_sockaddr(buf1, &sdp_ip))
341  return;
342  ttl = 16;
343  if (*p == '/') {
344  p++;
345  get_word_sep(buf1, sizeof(buf1), "/", &p);
346  ttl = atoi(buf1);
347  }
348  if (s->nb_streams == 0) {
349  s1->default_ip = sdp_ip;
350  s1->default_ttl = ttl;
351  } else {
352  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
353  rtsp_st->sdp_ip = sdp_ip;
354  rtsp_st->sdp_ttl = ttl;
355  }
356  break;
357  case 's':
358  av_dict_set(&s->metadata, "title", p, 0);
359  break;
360  case 'i':
361  if (s->nb_streams == 0) {
362  av_dict_set(&s->metadata, "comment", p, 0);
363  break;
364  }
365  break;
366  case 'm':
367  /* new stream */
368  s1->skip_media = 0;
369  codec_type = AVMEDIA_TYPE_UNKNOWN;
370  get_word(st_type, sizeof(st_type), &p);
371  if (!strcmp(st_type, "audio")) {
372  codec_type = AVMEDIA_TYPE_AUDIO;
373  } else if (!strcmp(st_type, "video")) {
374  codec_type = AVMEDIA_TYPE_VIDEO;
375  } else if (!strcmp(st_type, "application")) {
376  codec_type = AVMEDIA_TYPE_DATA;
377  }
378  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
379  s1->skip_media = 1;
380  return;
381  }
382  rtsp_st = av_mallocz(sizeof(RTSPStream));
383  if (!rtsp_st)
384  return;
385  rtsp_st->stream_index = -1;
386  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
387 
388  rtsp_st->sdp_ip = s1->default_ip;
389  rtsp_st->sdp_ttl = s1->default_ttl;
390 
391  copy_default_source_addrs(s1->default_include_source_addrs,
392  s1->nb_default_include_source_addrs,
393  &rtsp_st->include_source_addrs,
394  &rtsp_st->nb_include_source_addrs);
395  copy_default_source_addrs(s1->default_exclude_source_addrs,
396  s1->nb_default_exclude_source_addrs,
397  &rtsp_st->exclude_source_addrs,
398  &rtsp_st->nb_exclude_source_addrs);
399 
400  get_word(buf1, sizeof(buf1), &p); /* port */
401  rtsp_st->sdp_port = atoi(buf1);
402 
403  get_word(buf1, sizeof(buf1), &p); /* protocol */
404  if (!strcmp(buf1, "udp"))
406  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
407  rtsp_st->feedback = 1;
408 
409  /* XXX: handle list of formats */
410  get_word(buf1, sizeof(buf1), &p); /* format list */
411  rtsp_st->sdp_payload_type = atoi(buf1);
412 
413  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
414  /* no corresponding stream */
415  if (rt->transport == RTSP_TRANSPORT_RAW) {
416  if (!rt->ts && CONFIG_RTPDEC)
417  rt->ts = ff_mpegts_parse_open(s);
418  } else {
419  RTPDynamicProtocolHandler *handler;
420  handler = ff_rtp_handler_find_by_id(
422  init_rtp_handler(handler, rtsp_st, NULL);
423  if (handler && handler->init)
424  handler->init(s, -1, rtsp_st->dynamic_protocol_context);
425  }
426  } else if (rt->server_type == RTSP_SERVER_WMS &&
427  codec_type == AVMEDIA_TYPE_DATA) {
428  /* RTX stream, a stream that carries all the other actual
429  * audio/video streams. Don't expose this to the callers. */
430  } else {
431  st = avformat_new_stream(s, NULL);
432  if (!st)
433  return;
434  st->id = rt->nb_rtsp_streams - 1;
435  rtsp_st->stream_index = st->index;
436  st->codec->codec_type = codec_type;
437  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
438  RTPDynamicProtocolHandler *handler;
439  /* if standard payload type, we can find the codec right now */
441  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
442  st->codec->sample_rate > 0)
443  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
444  /* Even static payload types may need a custom depacketizer */
445  handler = ff_rtp_handler_find_by_id(
446  rtsp_st->sdp_payload_type, st->codec->codec_type);
447  init_rtp_handler(handler, rtsp_st, st->codec);
448  if (handler && handler->init)
449  handler->init(s, st->index,
450  rtsp_st->dynamic_protocol_context);
451  }
452  }
453  /* put a default control url */
454  av_strlcpy(rtsp_st->control_url, rt->control_uri,
455  sizeof(rtsp_st->control_url));
456  break;
457  case 'a':
458  if (av_strstart(p, "control:", &p)) {
459  if (s->nb_streams == 0) {
460  if (!strncmp(p, "rtsp://", 7))
461  av_strlcpy(rt->control_uri, p,
462  sizeof(rt->control_uri));
463  } else {
464  char proto[32];
465  /* get the control url */
466  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
467 
468  /* XXX: may need to add full url resolution */
469  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
470  NULL, NULL, 0, p);
471  if (proto[0] == '\0') {
472  /* relative control URL */
473  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
474  av_strlcat(rtsp_st->control_url, "/",
475  sizeof(rtsp_st->control_url));
476  av_strlcat(rtsp_st->control_url, p,
477  sizeof(rtsp_st->control_url));
478  } else
479  av_strlcpy(rtsp_st->control_url, p,
480  sizeof(rtsp_st->control_url));
481  }
482  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
483  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
484  get_word(buf1, sizeof(buf1), &p);
485  payload_type = atoi(buf1);
486  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
487  if (rtsp_st->stream_index >= 0) {
488  st = s->streams[rtsp_st->stream_index];
489  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
490  }
491  } else if (av_strstart(p, "fmtp:", &p) ||
492  av_strstart(p, "framesize:", &p)) {
493  /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
494  // let dynamic protocol handlers have a stab at the line.
495  get_word(buf1, sizeof(buf1), &p);
496  payload_type = atoi(buf1);
497  for (i = 0; i < rt->nb_rtsp_streams; i++) {
498  rtsp_st = rt->rtsp_streams[i];
499  if (rtsp_st->sdp_payload_type == payload_type &&
500  rtsp_st->dynamic_handler &&
502  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
503  rtsp_st->dynamic_protocol_context, buf);
504  }
505  } else if (av_strstart(p, "range:", &p)) {
506  int64_t start, end;
507 
508  // this is so that seeking on a streamed file can work.
509  rtsp_parse_range_npt(p, &start, &end);
510  s->start_time = start;
511  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
512  s->duration = (end == AV_NOPTS_VALUE) ?
513  AV_NOPTS_VALUE : end - start;
514  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
515  if (atoi(p) == 1)
517  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
518  s->nb_streams > 0) {
519  st = s->streams[s->nb_streams - 1];
520  st->codec->sample_rate = atoi(p);
521  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
522  // RFC 4568
523  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
524  get_word(buf1, sizeof(buf1), &p); // ignore tag
525  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
526  p += strspn(p, SPACE_CHARS);
527  if (av_strstart(p, "inline:", &p))
528  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
529  } else if (av_strstart(p, "source-filter:", &p)) {
530  int exclude = 0;
531  get_word(buf1, sizeof(buf1), &p);
532  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
533  return;
534  exclude = !strcmp(buf1, "excl");
535 
536  get_word(buf1, sizeof(buf1), &p);
537  if (strcmp(buf1, "IN") != 0)
538  return;
539  get_word(buf1, sizeof(buf1), &p);
540  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
541  return;
542  // not checking that the destination address actually matches or is wildcard
543  get_word(buf1, sizeof(buf1), &p);
544 
545  while (*p != '\0') {
546  rtsp_src = av_mallocz(sizeof(*rtsp_src));
547  if (!rtsp_src)
548  return;
549  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
550  if (exclude) {
551  if (s->nb_streams == 0) {
552  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
553  } else {
554  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
555  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
556  }
557  } else {
558  if (s->nb_streams == 0) {
559  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
560  } else {
561  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
562  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
563  }
564  }
565  }
566  } else {
567  if (rt->server_type == RTSP_SERVER_WMS)
569  if (s->nb_streams > 0) {
570  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
571 
572  if (rt->server_type == RTSP_SERVER_REAL)
573  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
574 
575  if (rtsp_st->dynamic_handler &&
577  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
578  rtsp_st->stream_index,
579  rtsp_st->dynamic_protocol_context, buf);
580  }
581  }
582  break;
583  }
584 }
585 
586 int ff_sdp_parse(AVFormatContext *s, const char *content)
587 {
588  RTSPState *rt = s->priv_data;
589  const char *p;
590  int letter, i;
591  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
592  * contain long SDP lines containing complete ASF Headers (several
593  * kB) or arrays of MDPR (RM stream descriptor) headers plus
594  * "rulebooks" describing their properties. Therefore, the SDP line
595  * buffer is large.
596  *
597  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
598  * in rtpdec_xiph.c. */
599  char buf[16384], *q;
600  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
601 
602  p = content;
603  for (;;) {
604  p += strspn(p, SPACE_CHARS);
605  letter = *p;
606  if (letter == '\0')
607  break;
608  p++;
609  if (*p != '=')
610  goto next_line;
611  p++;
612  /* get the content */
613  q = buf;
614  while (*p != '\n' && *p != '\r' && *p != '\0') {
615  if ((q - buf) < sizeof(buf) - 1)
616  *q++ = *p;
617  p++;
618  }
619  *q = '\0';
620  sdp_parse_line(s, s1, letter, buf);
621  next_line:
622  while (*p != '\n' && *p != '\0')
623  p++;
624  if (*p == '\n')
625  p++;
626  }
627 
628  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
629  av_free(s1->default_include_source_addrs[i]);
630  av_freep(&s1->default_include_source_addrs);
631  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
632  av_free(s1->default_exclude_source_addrs[i]);
633  av_freep(&s1->default_exclude_source_addrs);
634 
635  rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
636  if (!rt->p) return AVERROR(ENOMEM);
637  return 0;
638 }
639 #endif /* CONFIG_RTPDEC */
640 
641 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
642 {
643  RTSPState *rt = s->priv_data;
644  int i;
645 
646  for (i = 0; i < rt->nb_rtsp_streams; i++) {
647  RTSPStream *rtsp_st = rt->rtsp_streams[i];
648  if (!rtsp_st)
649  continue;
650  if (rtsp_st->transport_priv) {
651  if (s->oformat) {
652  AVFormatContext *rtpctx = rtsp_st->transport_priv;
653  av_write_trailer(rtpctx);
655  uint8_t *ptr;
656  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
657  ff_rtsp_tcp_write_packet(s, rtsp_st);
658  avio_close_dyn_buf(rtpctx->pb, &ptr);
659  av_free(ptr);
660  } else {
661  avio_close(rtpctx->pb);
662  }
663  avformat_free_context(rtpctx);
664  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
666  else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
668  }
669  rtsp_st->transport_priv = NULL;
670  if (rtsp_st->rtp_handle)
671  ffurl_close(rtsp_st->rtp_handle);
672  rtsp_st->rtp_handle = NULL;
673  }
674 }
675 
676 /* close and free RTSP streams */
678 {
679  RTSPState *rt = s->priv_data;
680  int i, j;
681  RTSPStream *rtsp_st;
682 
683  ff_rtsp_undo_setup(s, 0);
684  for (i = 0; i < rt->nb_rtsp_streams; i++) {
685  rtsp_st = rt->rtsp_streams[i];
686  if (rtsp_st) {
687  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
688  rtsp_st->dynamic_handler->free(
689  rtsp_st->dynamic_protocol_context);
690  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
691  av_free(rtsp_st->include_source_addrs[j]);
692  av_freep(&rtsp_st->include_source_addrs);
693  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
694  av_free(rtsp_st->exclude_source_addrs[j]);
695  av_freep(&rtsp_st->exclude_source_addrs);
696 
697  av_free(rtsp_st);
698  }
699  }
700  av_free(rt->rtsp_streams);
701  if (rt->asf_ctx) {
703  }
704  if (rt->ts && CONFIG_RTPDEC)
706  av_free(rt->p);
707  av_free(rt->recvbuf);
708 }
709 
711 {
712  RTSPState *rt = s->priv_data;
713  AVStream *st = NULL;
714  int reordering_queue_size = rt->reordering_queue_size;
715  if (reordering_queue_size < 0) {
717  reordering_queue_size = 0;
718  else
719  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
720  }
721 
722  /* open the RTP context */
723  if (rtsp_st->stream_index >= 0)
724  st = s->streams[rtsp_st->stream_index];
725  if (!st)
727 
728  if (s->oformat && CONFIG_RTSP_MUXER) {
729  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
730  s, st, rtsp_st->rtp_handle,
732  rtsp_st->stream_index);
733  /* Ownership of rtp_handle is passed to the rtp mux context */
734  rtsp_st->rtp_handle = NULL;
735  if (ret < 0)
736  return ret;
737  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
738  return 0; // Don't need to open any parser here
739  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
740  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
741  rtsp_st->dynamic_protocol_context,
742  rtsp_st->dynamic_handler);
743  else if (CONFIG_RTPDEC)
744  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
745  rtsp_st->sdp_payload_type,
746  reordering_queue_size);
747 
748  if (!rtsp_st->transport_priv) {
749  return AVERROR(ENOMEM);
750  } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
751  if (rtsp_st->dynamic_handler) {
753  rtsp_st->dynamic_protocol_context,
754  rtsp_st->dynamic_handler);
755  }
756  if (rtsp_st->crypto_suite[0])
758  rtsp_st->crypto_suite,
759  rtsp_st->crypto_params);
760  }
761 
762  return 0;
763 }
764 
765 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
766 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
767 {
768  const char *q;
769  char *p;
770  int v;
771 
772  q = *pp;
773  q += strspn(q, SPACE_CHARS);
774  v = strtol(q, &p, 10);
775  if (*p == '-') {
776  p++;
777  *min_ptr = v;
778  v = strtol(p, &p, 10);
779  *max_ptr = v;
780  } else {
781  *min_ptr = v;
782  *max_ptr = v;
783  }
784  *pp = p;
785 }
786 
787 /* XXX: only one transport specification is parsed */
788 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
789 {
790  char transport_protocol[16];
791  char profile[16];
792  char lower_transport[16];
793  char parameter[16];
794  RTSPTransportField *th;
795  char buf[256];
796 
797  reply->nb_transports = 0;
798 
799  for (;;) {
800  p += strspn(p, SPACE_CHARS);
801  if (*p == '\0')
802  break;
803 
804  th = &reply->transports[reply->nb_transports];
805 
806  get_word_sep(transport_protocol, sizeof(transport_protocol),
807  "/", &p);
808  if (!av_strcasecmp (transport_protocol, "rtp")) {
809  get_word_sep(profile, sizeof(profile), "/;,", &p);
810  lower_transport[0] = '\0';
811  /* rtp/avp/<protocol> */
812  if (*p == '/') {
813  get_word_sep(lower_transport, sizeof(lower_transport),
814  ";,", &p);
815  }
817  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
818  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
819  /* x-pn-tng/<protocol> */
820  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
821  profile[0] = '\0';
823  } else if (!av_strcasecmp(transport_protocol, "raw")) {
824  get_word_sep(profile, sizeof(profile), "/;,", &p);
825  lower_transport[0] = '\0';
826  /* raw/raw/<protocol> */
827  if (*p == '/') {
828  get_word_sep(lower_transport, sizeof(lower_transport),
829  ";,", &p);
830  }
832  }
833  if (!av_strcasecmp(lower_transport, "TCP"))
835  else
837 
838  if (*p == ';')
839  p++;
840  /* get each parameter */
841  while (*p != '\0' && *p != ',') {
842  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
843  if (!strcmp(parameter, "port")) {
844  if (*p == '=') {
845  p++;
846  rtsp_parse_range(&th->port_min, &th->port_max, &p);
847  }
848  } else if (!strcmp(parameter, "client_port")) {
849  if (*p == '=') {
850  p++;
851  rtsp_parse_range(&th->client_port_min,
852  &th->client_port_max, &p);
853  }
854  } else if (!strcmp(parameter, "server_port")) {
855  if (*p == '=') {
856  p++;
857  rtsp_parse_range(&th->server_port_min,
858  &th->server_port_max, &p);
859  }
860  } else if (!strcmp(parameter, "interleaved")) {
861  if (*p == '=') {
862  p++;
863  rtsp_parse_range(&th->interleaved_min,
864  &th->interleaved_max, &p);
865  }
866  } else if (!strcmp(parameter, "multicast")) {
869  } else if (!strcmp(parameter, "ttl")) {
870  if (*p == '=') {
871  char *end;
872  p++;
873  th->ttl = strtol(p, &end, 10);
874  p = end;
875  }
876  } else if (!strcmp(parameter, "destination")) {
877  if (*p == '=') {
878  p++;
879  get_word_sep(buf, sizeof(buf), ";,", &p);
880  get_sockaddr(buf, &th->destination);
881  }
882  } else if (!strcmp(parameter, "source")) {
883  if (*p == '=') {
884  p++;
885  get_word_sep(buf, sizeof(buf), ";,", &p);
886  av_strlcpy(th->source, buf, sizeof(th->source));
887  }
888  } else if (!strcmp(parameter, "mode")) {
889  if (*p == '=') {
890  p++;
891  get_word_sep(buf, sizeof(buf), ";, ", &p);
892  if (!strcmp(buf, "record") ||
893  !strcmp(buf, "receive"))
894  th->mode_record = 1;
895  }
896  }
897 
898  while (*p != ';' && *p != '\0' && *p != ',')
899  p++;
900  if (*p == ';')
901  p++;
902  }
903  if (*p == ',')
904  p++;
905 
906  reply->nb_transports++;
907  }
908 }
909 
910 static void handle_rtp_info(RTSPState *rt, const char *url,
911  uint32_t seq, uint32_t rtptime)
912 {
913  int i;
914  if (!rtptime || !url[0])
915  return;
916  if (rt->transport != RTSP_TRANSPORT_RTP)
917  return;
918  for (i = 0; i < rt->nb_rtsp_streams; i++) {
919  RTSPStream *rtsp_st = rt->rtsp_streams[i];
920  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
921  if (!rtpctx)
922  continue;
923  if (!strcmp(rtsp_st->control_url, url)) {
924  rtpctx->base_timestamp = rtptime;
925  break;
926  }
927  }
928 }
929 
930 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
931 {
932  int read = 0;
933  char key[20], value[1024], url[1024] = "";
934  uint32_t seq = 0, rtptime = 0;
935 
936  for (;;) {
937  p += strspn(p, SPACE_CHARS);
938  if (!*p)
939  break;
940  get_word_sep(key, sizeof(key), "=", &p);
941  if (*p != '=')
942  break;
943  p++;
944  get_word_sep(value, sizeof(value), ";, ", &p);
945  read++;
946  if (!strcmp(key, "url"))
947  av_strlcpy(url, value, sizeof(url));
948  else if (!strcmp(key, "seq"))
949  seq = strtoul(value, NULL, 10);
950  else if (!strcmp(key, "rtptime"))
951  rtptime = strtoul(value, NULL, 10);
952  if (*p == ',') {
953  handle_rtp_info(rt, url, seq, rtptime);
954  url[0] = '\0';
955  seq = rtptime = 0;
956  read = 0;
957  }
958  if (*p)
959  p++;
960  }
961  if (read > 0)
962  handle_rtp_info(rt, url, seq, rtptime);
963 }
964 
965 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
966  RTSPState *rt, const char *method)
967 {
968  const char *p;
969 
970  /* NOTE: we do case independent match for broken servers */
971  p = buf;
972  if (av_stristart(p, "Session:", &p)) {
973  int t;
974  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
975  if (av_stristart(p, ";timeout=", &p) &&
976  (t = strtol(p, NULL, 10)) > 0) {
977  reply->timeout = t;
978  }
979  } else if (av_stristart(p, "Content-Length:", &p)) {
980  reply->content_length = strtol(p, NULL, 10);
981  } else if (av_stristart(p, "Transport:", &p)) {
982  rtsp_parse_transport(reply, p);
983  } else if (av_stristart(p, "CSeq:", &p)) {
984  reply->seq = strtol(p, NULL, 10);
985  } else if (av_stristart(p, "Range:", &p)) {
986  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
987  } else if (av_stristart(p, "RealChallenge1:", &p)) {
988  p += strspn(p, SPACE_CHARS);
989  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
990  } else if (av_stristart(p, "Server:", &p)) {
991  p += strspn(p, SPACE_CHARS);
992  av_strlcpy(reply->server, p, sizeof(reply->server));
993  } else if (av_stristart(p, "Notice:", &p) ||
994  av_stristart(p, "X-Notice:", &p)) {
995  reply->notice = strtol(p, NULL, 10);
996  } else if (av_stristart(p, "Location:", &p)) {
997  p += strspn(p, SPACE_CHARS);
998  av_strlcpy(reply->location, p , sizeof(reply->location));
999  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1000  p += strspn(p, SPACE_CHARS);
1001  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1002  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1003  p += strspn(p, SPACE_CHARS);
1004  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1005  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1006  p += strspn(p, SPACE_CHARS);
1007  if (method && !strcmp(method, "DESCRIBE"))
1008  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1009  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1010  p += strspn(p, SPACE_CHARS);
1011  if (method && !strcmp(method, "PLAY"))
1012  rtsp_parse_rtp_info(rt, p);
1013  } else if (av_stristart(p, "Public:", &p) && rt) {
1014  if (strstr(p, "GET_PARAMETER") &&
1015  method && !strcmp(method, "OPTIONS"))
1016  rt->get_parameter_supported = 1;
1017  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1018  p += strspn(p, SPACE_CHARS);
1019  rt->accept_dynamic_rate = atoi(p);
1020  } else if (av_stristart(p, "Content-Type:", &p)) {
1021  p += strspn(p, SPACE_CHARS);
1022  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1023  }
1024 }
1025 
1026 /* skip a RTP/TCP interleaved packet */
1028 {
1029  RTSPState *rt = s->priv_data;
1030  int ret, len, len1;
1031  uint8_t buf[1024];
1032 
1033  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1034  if (ret != 3)
1035  return;
1036  len = AV_RB16(buf + 1);
1037 
1038  av_dlog(s, "skipping RTP packet len=%d\n", len);
1039 
1040  /* skip payload */
1041  while (len > 0) {
1042  len1 = len;
1043  if (len1 > sizeof(buf))
1044  len1 = sizeof(buf);
1045  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1046  if (ret != len1)
1047  return;
1048  len -= len1;
1049  }
1050 }
1051 
1053  unsigned char **content_ptr,
1054  int return_on_interleaved_data, const char *method)
1055 {
1056  RTSPState *rt = s->priv_data;
1057  char buf[4096], buf1[1024], *q;
1058  unsigned char ch;
1059  const char *p;
1060  int ret, content_length, line_count = 0, request = 0;
1061  unsigned char *content = NULL;
1062 
1063 start:
1064  line_count = 0;
1065  request = 0;
1066  content = NULL;
1067  memset(reply, 0, sizeof(*reply));
1068 
1069  /* parse reply (XXX: use buffers) */
1070  rt->last_reply[0] = '\0';
1071  for (;;) {
1072  q = buf;
1073  for (;;) {
1074  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1075  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1076  if (ret != 1)
1077  return AVERROR_EOF;
1078  if (ch == '\n')
1079  break;
1080  if (ch == '$') {
1081  /* XXX: only parse it if first char on line ? */
1082  if (return_on_interleaved_data) {
1083  return 1;
1084  } else
1086  } else if (ch != '\r') {
1087  if ((q - buf) < sizeof(buf) - 1)
1088  *q++ = ch;
1089  }
1090  }
1091  *q = '\0';
1092 
1093  av_dlog(s, "line='%s'\n", buf);
1094 
1095  /* test if last line */
1096  if (buf[0] == '\0')
1097  break;
1098  p = buf;
1099  if (line_count == 0) {
1100  /* get reply code */
1101  get_word(buf1, sizeof(buf1), &p);
1102  if (!strncmp(buf1, "RTSP/", 5)) {
1103  get_word(buf1, sizeof(buf1), &p);
1104  reply->status_code = atoi(buf1);
1105  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1106  } else {
1107  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1108  get_word(buf1, sizeof(buf1), &p); // object
1109  request = 1;
1110  }
1111  } else {
1112  ff_rtsp_parse_line(reply, p, rt, method);
1113  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1114  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1115  }
1116  line_count++;
1117  }
1118 
1119  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1120  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1121 
1122  content_length = reply->content_length;
1123  if (content_length > 0) {
1124  /* leave some room for a trailing '\0' (useful for simple parsing) */
1125  content = av_malloc(content_length + 1);
1126  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1127  content[content_length] = '\0';
1128  }
1129  if (content_ptr)
1130  *content_ptr = content;
1131  else
1132  av_free(content);
1133 
1134  if (request) {
1135  char buf[1024];
1136  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1137  const char* ptr = buf;
1138 
1139  if (!strcmp(reply->reason, "OPTIONS")) {
1140  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1141  if (reply->seq)
1142  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1143  if (reply->session_id[0])
1144  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1145  reply->session_id);
1146  } else {
1147  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1148  }
1149  av_strlcat(buf, "\r\n", sizeof(buf));
1150 
1151  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1152  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1153  ptr = base64buf;
1154  }
1155  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1156 
1157  rt->last_cmd_time = av_gettime();
1158  /* Even if the request from the server had data, it is not the data
1159  * that the caller wants or expects. The memory could also be leaked
1160  * if the actual following reply has content data. */
1161  if (content_ptr)
1162  av_freep(content_ptr);
1163  /* If method is set, this is called from ff_rtsp_send_cmd,
1164  * where a reply to exactly this request is awaited. For
1165  * callers from within packet receiving, we just want to
1166  * return to the caller and go back to receiving packets. */
1167  if (method)
1168  goto start;
1169  return 0;
1170  }
1171 
1172  if (rt->seq != reply->seq) {
1173  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1174  rt->seq, reply->seq);
1175  }
1176 
1177  /* EOS */
1178  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1179  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1180  reply->notice == 2306 /* Continuous Feed Terminated */) {
1181  rt->state = RTSP_STATE_IDLE;
1182  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1183  return AVERROR(EIO); /* data or server error */
1184  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1185  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1186  return AVERROR(EPERM);
1187 
1188  return 0;
1189 }
1190 
1204 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1205  const char *method, const char *url,
1206  const char *headers,
1207  const unsigned char *send_content,
1208  int send_content_length)
1209 {
1210  RTSPState *rt = s->priv_data;
1211  char buf[4096], *out_buf;
1212  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1213 
1214  /* Add in RTSP headers */
1215  out_buf = buf;
1216  rt->seq++;
1217  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1218  if (headers)
1219  av_strlcat(buf, headers, sizeof(buf));
1220  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1221  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1222  if (rt->session_id[0] != '\0' && (!headers ||
1223  !strstr(headers, "\nIf-Match:"))) {
1224  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1225  }
1226  if (rt->auth[0]) {
1227  char *str = ff_http_auth_create_response(&rt->auth_state,
1228  rt->auth, url, method);
1229  if (str)
1230  av_strlcat(buf, str, sizeof(buf));
1231  av_free(str);
1232  }
1233  if (send_content_length > 0 && send_content)
1234  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1235  av_strlcat(buf, "\r\n", sizeof(buf));
1236 
1237  /* base64 encode rtsp if tunneling */
1238  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1239  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1240  out_buf = base64buf;
1241  }
1242 
1243  av_dlog(s, "Sending:\n%s--\n", buf);
1244 
1245  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1246  if (send_content_length > 0 && send_content) {
1247  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1248  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1249  "with content data not supported\n");
1250  return AVERROR_PATCHWELCOME;
1251  }
1252  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1253  }
1254  rt->last_cmd_time = av_gettime();
1255 
1256  return 0;
1257 }
1258 
1259 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1260  const char *url, const char *headers)
1261 {
1262  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1263 }
1264 
1265 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1266  const char *headers, RTSPMessageHeader *reply,
1267  unsigned char **content_ptr)
1268 {
1269  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1270  content_ptr, NULL, 0);
1271 }
1272 
1274  const char *method, const char *url,
1275  const char *header,
1276  RTSPMessageHeader *reply,
1277  unsigned char **content_ptr,
1278  const unsigned char *send_content,
1279  int send_content_length)
1280 {
1281  RTSPState *rt = s->priv_data;
1282  HTTPAuthType cur_auth_type;
1283  int ret, attempts = 0;
1284 
1285 retry:
1286  cur_auth_type = rt->auth_state.auth_type;
1287  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1288  send_content,
1289  send_content_length)))
1290  return ret;
1291 
1292  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1293  return ret;
1294  attempts++;
1295 
1296  if (reply->status_code == 401 &&
1297  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1298  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1299  goto retry;
1300 
1301  if (reply->status_code > 400){
1302  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1303  method,
1304  reply->status_code,
1305  reply->reason);
1306  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1307  }
1308 
1309  return 0;
1310 }
1311 
1312 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1313  int lower_transport, const char *real_challenge)
1314 {
1315  RTSPState *rt = s->priv_data;
1316  int rtx = 0, j, i, err, interleave = 0, port_off;
1317  RTSPStream *rtsp_st;
1318  RTSPMessageHeader reply1, *reply = &reply1;
1319  char cmd[2048];
1320  const char *trans_pref;
1321 
1322  if (rt->transport == RTSP_TRANSPORT_RDT)
1323  trans_pref = "x-pn-tng";
1324  else if (rt->transport == RTSP_TRANSPORT_RAW)
1325  trans_pref = "RAW/RAW";
1326  else
1327  trans_pref = "RTP/AVP";
1328 
1329  /* default timeout: 1 minute */
1330  rt->timeout = 60;
1331 
1332  /* for each stream, make the setup request */
1333  /* XXX: we assume the same server is used for the control of each
1334  * RTSP stream */
1335 
1336  /* Choose a random starting offset within the first half of the
1337  * port range, to allow for a number of ports to try even if the offset
1338  * happens to be at the end of the random range. */
1339  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1340  /* even random offset */
1341  port_off -= port_off & 0x01;
1342 
1343  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1344  char transport[2048];
1345 
1346  /*
1347  * WMS serves all UDP data over a single connection, the RTX, which
1348  * isn't necessarily the first in the SDP but has to be the first
1349  * to be set up, else the second/third SETUP will fail with a 461.
1350  */
1351  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1352  rt->server_type == RTSP_SERVER_WMS) {
1353  if (i == 0) {
1354  /* rtx first */
1355  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1356  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1357  if (len >= 4 &&
1358  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1359  "/rtx"))
1360  break;
1361  }
1362  if (rtx == rt->nb_rtsp_streams)
1363  return -1; /* no RTX found */
1364  rtsp_st = rt->rtsp_streams[rtx];
1365  } else
1366  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1367  } else
1368  rtsp_st = rt->rtsp_streams[i];
1369 
1370  /* RTP/UDP */
1371  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1372  char buf[256];
1373 
1374  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1375  port = reply->transports[0].client_port_min;
1376  goto have_port;
1377  }
1378 
1379  /* first try in specified port range */
1380  while (j <= rt->rtp_port_max) {
1381  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1382  "?localport=%d", j);
1383  /* we will use two ports per rtp stream (rtp and rtcp) */
1384  j += 2;
1385  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1386  &s->interrupt_callback, NULL))
1387  goto rtp_opened;
1388  }
1389 
1390  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1391  err = AVERROR(EIO);
1392  goto fail;
1393 
1394  rtp_opened:
1395  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1396  have_port:
1397  snprintf(transport, sizeof(transport) - 1,
1398  "%s/UDP;", trans_pref);
1399  if (rt->server_type != RTSP_SERVER_REAL)
1400  av_strlcat(transport, "unicast;", sizeof(transport));
1401  av_strlcatf(transport, sizeof(transport),
1402  "client_port=%d", port);
1403  if (rt->transport == RTSP_TRANSPORT_RTP &&
1404  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1405  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1406  }
1407 
1408  /* RTP/TCP */
1409  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1410  /* For WMS streams, the application streams are only used for
1411  * UDP. When trying to set it up for TCP streams, the server
1412  * will return an error. Therefore, we skip those streams. */
1413  if (rt->server_type == RTSP_SERVER_WMS &&
1414  (rtsp_st->stream_index < 0 ||
1415  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1417  continue;
1418  snprintf(transport, sizeof(transport) - 1,
1419  "%s/TCP;", trans_pref);
1420  if (rt->transport != RTSP_TRANSPORT_RDT)
1421  av_strlcat(transport, "unicast;", sizeof(transport));
1422  av_strlcatf(transport, sizeof(transport),
1423  "interleaved=%d-%d",
1424  interleave, interleave + 1);
1425  interleave += 2;
1426  }
1427 
1428  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1429  snprintf(transport, sizeof(transport) - 1,
1430  "%s/UDP;multicast", trans_pref);
1431  }
1432  if (s->oformat) {
1433  av_strlcat(transport, ";mode=record", sizeof(transport));
1434  } else if (rt->server_type == RTSP_SERVER_REAL ||
1436  av_strlcat(transport, ";mode=play", sizeof(transport));
1437  snprintf(cmd, sizeof(cmd),
1438  "Transport: %s\r\n",
1439  transport);
1440  if (rt->accept_dynamic_rate)
1441  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1442  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1443  char real_res[41], real_csum[9];
1444  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1445  real_challenge);
1446  av_strlcatf(cmd, sizeof(cmd),
1447  "If-Match: %s\r\n"
1448  "RealChallenge2: %s, sd=%s\r\n",
1449  rt->session_id, real_res, real_csum);
1450  }
1451  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1452  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1453  err = 1;
1454  goto fail;
1455  } else if (reply->status_code != RTSP_STATUS_OK ||
1456  reply->nb_transports != 1) {
1457  err = AVERROR_INVALIDDATA;
1458  goto fail;
1459  }
1460 
1461  /* XXX: same protocol for all streams is required */
1462  if (i > 0) {
1463  if (reply->transports[0].lower_transport != rt->lower_transport ||
1464  reply->transports[0].transport != rt->transport) {
1465  err = AVERROR_INVALIDDATA;
1466  goto fail;
1467  }
1468  } else {
1469  rt->lower_transport = reply->transports[0].lower_transport;
1470  rt->transport = reply->transports[0].transport;
1471  }
1472 
1473  /* Fail if the server responded with another lower transport mode
1474  * than what we requested. */
1475  if (reply->transports[0].lower_transport != lower_transport) {
1476  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1477  err = AVERROR_INVALIDDATA;
1478  goto fail;
1479  }
1480 
1481  switch(reply->transports[0].lower_transport) {
1483  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1484  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1485  break;
1486 
1487  case RTSP_LOWER_TRANSPORT_UDP: {
1488  char url[1024], options[30] = "";
1489  const char *peer = host;
1490 
1491  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1492  av_strlcpy(options, "?connect=1", sizeof(options));
1493  /* Use source address if specified */
1494  if (reply->transports[0].source[0])
1495  peer = reply->transports[0].source;
1496  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1497  reply->transports[0].server_port_min, "%s", options);
1498  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1499  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1500  err = AVERROR_INVALIDDATA;
1501  goto fail;
1502  }
1503  /* Try to initialize the connection state in a
1504  * potential NAT router by sending dummy packets.
1505  * RTP/RTCP dummy packets are used for RDT, too.
1506  */
1507  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1508  CONFIG_RTPDEC)
1510  break;
1511  }
1513  char url[1024], namebuf[50], optbuf[20] = "";
1514  struct sockaddr_storage addr;
1515  int port, ttl;
1516 
1517  if (reply->transports[0].destination.ss_family) {
1518  addr = reply->transports[0].destination;
1519  port = reply->transports[0].port_min;
1520  ttl = reply->transports[0].ttl;
1521  } else {
1522  addr = rtsp_st->sdp_ip;
1523  port = rtsp_st->sdp_port;
1524  ttl = rtsp_st->sdp_ttl;
1525  }
1526  if (ttl > 0)
1527  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1528  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1529  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1530  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1531  port, "%s", optbuf);
1532  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1533  &s->interrupt_callback, NULL) < 0) {
1534  err = AVERROR_INVALIDDATA;
1535  goto fail;
1536  }
1537  break;
1538  }
1539  }
1540 
1541  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1542  goto fail;
1543  }
1544 
1545  if (rt->nb_rtsp_streams && reply->timeout > 0)
1546  rt->timeout = reply->timeout;
1547 
1548  if (rt->server_type == RTSP_SERVER_REAL)
1549  rt->need_subscription = 1;
1550 
1551  return 0;
1552 
1553 fail:
1554  ff_rtsp_undo_setup(s, 0);
1555  return err;
1556 }
1557 
1559 {
1560  RTSPState *rt = s->priv_data;
1561  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1562  ffurl_close(rt->rtsp_hd);
1563  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1564 }
1565 
1567 {
1568  RTSPState *rt = s->priv_data;
1569  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1570  int port, err, tcp_fd;
1571  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1572  int lower_transport_mask = 0;
1573  char real_challenge[64] = "";
1574  struct sockaddr_storage peer;
1575  socklen_t peer_len = sizeof(peer);
1576 
1577  if (rt->rtp_port_max < rt->rtp_port_min) {
1578  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1579  "than min port %d\n", rt->rtp_port_max,
1580  rt->rtp_port_min);
1581  return AVERROR(EINVAL);
1582  }
1583 
1584  if (!ff_network_init())
1585  return AVERROR(EIO);
1586 
1587  if (s->max_delay < 0) /* Not set by the caller */
1589 
1594  }
1595  /* Only pass through valid flags from here */
1597 
1598 redirect:
1599  lower_transport_mask = rt->lower_transport_mask;
1600  /* extract hostname and port */
1601  av_url_split(NULL, 0, auth, sizeof(auth),
1602  host, sizeof(host), &port, path, sizeof(path), s->filename);
1603  if (*auth) {
1604  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1605  }
1606  if (port < 0)
1607  port = RTSP_DEFAULT_PORT;
1608 
1609  if (!lower_transport_mask)
1610  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1611 
1612  if (s->oformat) {
1613  /* Only UDP or TCP - UDP multicast isn't supported. */
1614  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1615  (1 << RTSP_LOWER_TRANSPORT_TCP);
1616  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1617  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1618  "only UDP and TCP are supported for output.\n");
1619  err = AVERROR(EINVAL);
1620  goto fail;
1621  }
1622  }
1623 
1624  /* Construct the URI used in request; this is similar to s->filename,
1625  * but with authentication credentials removed and RTSP specific options
1626  * stripped out. */
1627  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1628  host, port, "%s", path);
1629 
1630  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1631  /* set up initial handshake for tunneling */
1632  char httpname[1024];
1633  char sessioncookie[17];
1634  char headers[1024];
1635 
1636  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1637  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1639 
1640  /* GET requests */
1641  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1642  &s->interrupt_callback) < 0) {
1643  err = AVERROR(EIO);
1644  goto fail;
1645  }
1646 
1647  /* generate GET headers */
1648  snprintf(headers, sizeof(headers),
1649  "x-sessioncookie: %s\r\n"
1650  "Accept: application/x-rtsp-tunnelled\r\n"
1651  "Pragma: no-cache\r\n"
1652  "Cache-Control: no-cache\r\n",
1653  sessioncookie);
1654  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1655 
1656  /* complete the connection */
1657  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1658  err = AVERROR(EIO);
1659  goto fail;
1660  }
1661 
1662  /* POST requests */
1663  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1664  &s->interrupt_callback) < 0 ) {
1665  err = AVERROR(EIO);
1666  goto fail;
1667  }
1668 
1669  /* generate POST headers */
1670  snprintf(headers, sizeof(headers),
1671  "x-sessioncookie: %s\r\n"
1672  "Content-Type: application/x-rtsp-tunnelled\r\n"
1673  "Pragma: no-cache\r\n"
1674  "Cache-Control: no-cache\r\n"
1675  "Content-Length: 32767\r\n"
1676  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1677  sessioncookie);
1678  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1679  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1680 
1681  /* Initialize the authentication state for the POST session. The HTTP
1682  * protocol implementation doesn't properly handle multi-pass
1683  * authentication for POST requests, since it would require one of
1684  * the following:
1685  * - implementing Expect: 100-continue, which many HTTP servers
1686  * don't support anyway, even less the RTSP servers that do HTTP
1687  * tunneling
1688  * - sending the whole POST data until getting a 401 reply specifying
1689  * what authentication method to use, then resending all that data
1690  * - waiting for potential 401 replies directly after sending the
1691  * POST header (waiting for some unspecified time)
1692  * Therefore, we copy the full auth state, which works for both basic
1693  * and digest. (For digest, we would have to synchronize the nonce
1694  * count variable between the two sessions, if we'd do more requests
1695  * with the original session, though.)
1696  */
1698 
1699  /* complete the connection */
1700  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1701  err = AVERROR(EIO);
1702  goto fail;
1703  }
1704  } else {
1705  /* open the tcp connection */
1706  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1707  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1708  &s->interrupt_callback, NULL) < 0) {
1709  err = AVERROR(EIO);
1710  goto fail;
1711  }
1712  rt->rtsp_hd_out = rt->rtsp_hd;
1713  }
1714  rt->seq = 0;
1715 
1716  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1717  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1718  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1719  NULL, 0, NI_NUMERICHOST);
1720  }
1721 
1722  /* request options supported by the server; this also detects server
1723  * type */
1724  for (rt->server_type = RTSP_SERVER_RTP;;) {
1725  cmd[0] = 0;
1726  if (rt->server_type == RTSP_SERVER_REAL)
1727  av_strlcat(cmd,
1728  /*
1729  * The following entries are required for proper
1730  * streaming from a Realmedia server. They are
1731  * interdependent in some way although we currently
1732  * don't quite understand how. Values were copied
1733  * from mplayer SVN r23589.
1734  * ClientChallenge is a 16-byte ID in hex
1735  * CompanyID is a 16-byte ID in base64
1736  */
1737  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1738  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1739  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1740  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1741  sizeof(cmd));
1742  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1743  if (reply->status_code != RTSP_STATUS_OK) {
1744  err = AVERROR_INVALIDDATA;
1745  goto fail;
1746  }
1747 
1748  /* detect server type if not standard-compliant RTP */
1749  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1751  continue;
1752  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1754  } else if (rt->server_type == RTSP_SERVER_REAL)
1755  strcpy(real_challenge, reply->real_challenge);
1756  break;
1757  }
1758 
1759  if (s->iformat && CONFIG_RTSP_DEMUXER)
1760  err = ff_rtsp_setup_input_streams(s, reply);
1761  else if (CONFIG_RTSP_MUXER)
1762  err = ff_rtsp_setup_output_streams(s, host);
1763  if (err)
1764  goto fail;
1765 
1766  do {
1767  int lower_transport = ff_log2_tab[lower_transport_mask &
1768  ~(lower_transport_mask - 1)];
1769 
1770  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1771  rt->server_type == RTSP_SERVER_REAL ?
1772  real_challenge : NULL);
1773  if (err < 0)
1774  goto fail;
1775  lower_transport_mask &= ~(1 << lower_transport);
1776  if (lower_transport_mask == 0 && err == 1) {
1777  err = AVERROR(EPROTONOSUPPORT);
1778  goto fail;
1779  }
1780  } while (err);
1781 
1782  rt->lower_transport_mask = lower_transport_mask;
1783  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1784  rt->state = RTSP_STATE_IDLE;
1785  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1786  return 0;
1787  fail:
1790  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1791  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1792  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1793  reply->status_code,
1794  s->filename);
1795  goto redirect;
1796  }
1797  ff_network_close();
1798  return err;
1799 }
1800 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1801 
1802 #if CONFIG_RTPDEC
1803 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1804  uint8_t *buf, int buf_size, int64_t wait_end)
1805 {
1806  RTSPState *rt = s->priv_data;
1807  RTSPStream *rtsp_st;
1808  int n, i, ret, tcp_fd, timeout_cnt = 0;
1809  int max_p = 0;
1810  struct pollfd *p = rt->p;
1811  int *fds = NULL, fdsnum, fdsidx;
1812 
1813  for (;;) {
1815  return AVERROR_EXIT;
1816  if (wait_end && wait_end - av_gettime() < 0)
1817  return AVERROR(EAGAIN);
1818  max_p = 0;
1819  if (rt->rtsp_hd) {
1820  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1821  p[max_p].fd = tcp_fd;
1822  p[max_p++].events = POLLIN;
1823  } else {
1824  tcp_fd = -1;
1825  }
1826  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1827  rtsp_st = rt->rtsp_streams[i];
1828  if (rtsp_st->rtp_handle) {
1829  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1830  &fds, &fdsnum)) {
1831  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1832  return ret;
1833  }
1834  if (fdsnum != 2) {
1835  av_log(s, AV_LOG_ERROR,
1836  "Number of fds %d not supported\n", fdsnum);
1837  return AVERROR_INVALIDDATA;
1838  }
1839  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1840  p[max_p].fd = fds[fdsidx];
1841  p[max_p++].events = POLLIN;
1842  }
1843  av_free(fds);
1844  }
1845  }
1846  n = poll(p, max_p, POLL_TIMEOUT_MS);
1847  if (n > 0) {
1848  int j = 1 - (tcp_fd == -1);
1849  timeout_cnt = 0;
1850  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1851  rtsp_st = rt->rtsp_streams[i];
1852  if (rtsp_st->rtp_handle) {
1853  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1854  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1855  if (ret > 0) {
1856  *prtsp_st = rtsp_st;
1857  return ret;
1858  }
1859  }
1860  j+=2;
1861  }
1862  }
1863 #if CONFIG_RTSP_DEMUXER
1864  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1865  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1866  if (rt->state == RTSP_STATE_STREAMING) {
1868  return AVERROR_EOF;
1869  else
1871  "Unable to answer to TEARDOWN\n");
1872  } else
1873  return 0;
1874  } else {
1875  RTSPMessageHeader reply;
1876  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1877  if (ret < 0)
1878  return ret;
1879  /* XXX: parse message */
1880  if (rt->state != RTSP_STATE_STREAMING)
1881  return 0;
1882  }
1883  }
1884 #endif
1885  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1886  return AVERROR(ETIMEDOUT);
1887  } else if (n < 0 && errno != EINTR)
1888  return AVERROR(errno);
1889  }
1890 }
1891 
1892 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1893  const uint8_t *buf, int len)
1894 {
1895  RTSPState *rt = s->priv_data;
1896  int i;
1897  if (len < 0)
1898  return len;
1899  if (rt->nb_rtsp_streams == 1) {
1900  *rtsp_st = rt->rtsp_streams[0];
1901  return len;
1902  }
1903  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1904  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1905  int no_ssrc = 0;
1906  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1907  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1908  if (!rtpctx)
1909  continue;
1910  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1911  *rtsp_st = rt->rtsp_streams[i];
1912  return len;
1913  }
1914  if (!rtpctx->ssrc)
1915  no_ssrc = 1;
1916  }
1917  if (no_ssrc) {
1919  "Unable to pick stream for packet - SSRC not known for "
1920  "all streams\n");
1921  return AVERROR(EAGAIN);
1922  }
1923  } else {
1924  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1925  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1926  *rtsp_st = rt->rtsp_streams[i];
1927  return len;
1928  }
1929  }
1930  }
1931  }
1932  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1933  return AVERROR(EAGAIN);
1934 }
1935 
1937 {
1938  RTSPState *rt = s->priv_data;
1939  int ret, len;
1940  RTSPStream *rtsp_st, *first_queue_st = NULL;
1941  int64_t wait_end = 0;
1942 
1943  if (rt->nb_byes == rt->nb_rtsp_streams)
1944  return AVERROR_EOF;
1945 
1946  /* get next frames from the same RTP packet */
1947  if (rt->cur_transport_priv) {
1948  if (rt->transport == RTSP_TRANSPORT_RDT) {
1949  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1950  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1951  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1952  } else if (rt->ts && CONFIG_RTPDEC) {
1953  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1954  if (ret >= 0) {
1955  rt->recvbuf_pos += ret;
1956  ret = rt->recvbuf_pos < rt->recvbuf_len;
1957  }
1958  } else
1959  ret = -1;
1960  if (ret == 0) {
1961  rt->cur_transport_priv = NULL;
1962  return 0;
1963  } else if (ret == 1) {
1964  return 0;
1965  } else
1966  rt->cur_transport_priv = NULL;
1967  }
1968 
1969 redo:
1970  if (rt->transport == RTSP_TRANSPORT_RTP) {
1971  int i;
1972  int64_t first_queue_time = 0;
1973  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1974  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1975  int64_t queue_time;
1976  if (!rtpctx)
1977  continue;
1978  queue_time = ff_rtp_queued_packet_time(rtpctx);
1979  if (queue_time && (queue_time - first_queue_time < 0 ||
1980  !first_queue_time)) {
1981  first_queue_time = queue_time;
1982  first_queue_st = rt->rtsp_streams[i];
1983  }
1984  }
1985  if (first_queue_time) {
1986  wait_end = first_queue_time + s->max_delay;
1987  } else {
1988  wait_end = 0;
1989  first_queue_st = NULL;
1990  }
1991  }
1992 
1993  /* read next RTP packet */
1994  if (!rt->recvbuf) {
1996  if (!rt->recvbuf)
1997  return AVERROR(ENOMEM);
1998  }
1999 
2000  switch(rt->lower_transport) {
2001  default:
2002 #if CONFIG_RTSP_DEMUXER
2004  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2005  break;
2006 #endif
2009  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2010  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2011  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2012  break;
2014  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2015  wait_end && wait_end < av_gettime())
2016  len = AVERROR(EAGAIN);
2017  else
2018  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2019  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2020  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2022  break;
2023  }
2024  if (len == AVERROR(EAGAIN) && first_queue_st &&
2025  rt->transport == RTSP_TRANSPORT_RTP) {
2026  rtsp_st = first_queue_st;
2027  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2028  goto end;
2029  }
2030  if (len < 0)
2031  return len;
2032  if (len == 0)
2033  return AVERROR_EOF;
2034  if (rt->transport == RTSP_TRANSPORT_RDT) {
2035  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2036  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2037  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2038  if (rtsp_st->feedback) {
2039  AVIOContext *pb = NULL;
2041  pb = s->pb;
2042  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2043  }
2044  if (ret < 0) {
2045  /* Either bad packet, or a RTCP packet. Check if the
2046  * first_rtcp_ntp_time field was initialized. */
2047  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2048  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2049  /* first_rtcp_ntp_time has been initialized for this stream,
2050  * copy the same value to all other uninitialized streams,
2051  * in order to map their timestamp origin to the same ntp time
2052  * as this one. */
2053  int i;
2054  AVStream *st = NULL;
2055  if (rtsp_st->stream_index >= 0)
2056  st = s->streams[rtsp_st->stream_index];
2057  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2058  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2059  AVStream *st2 = NULL;
2060  if (rt->rtsp_streams[i]->stream_index >= 0)
2061  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2062  if (rtpctx2 && st && st2 &&
2063  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2064  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2065  rtpctx2->rtcp_ts_offset = av_rescale_q(
2066  rtpctx->rtcp_ts_offset, st->time_base,
2067  st2->time_base);
2068  }
2069  }
2070  }
2071  if (ret == -RTCP_BYE) {
2072  rt->nb_byes++;
2073 
2074  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2075  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2076 
2077  if (rt->nb_byes == rt->nb_rtsp_streams)
2078  return AVERROR_EOF;
2079  }
2080  }
2081  } else if (rt->ts && CONFIG_RTPDEC) {
2082  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2083  if (ret >= 0) {
2084  if (ret < len) {
2085  rt->recvbuf_len = len;
2086  rt->recvbuf_pos = ret;
2087  rt->cur_transport_priv = rt->ts;
2088  return 1;
2089  } else {
2090  ret = 0;
2091  }
2092  }
2093  } else {
2094  return AVERROR_INVALIDDATA;
2095  }
2096 end:
2097  if (ret < 0)
2098  goto redo;
2099  if (ret == 1)
2100  /* more packets may follow, so we save the RTP context */
2101  rt->cur_transport_priv = rtsp_st->transport_priv;
2102 
2103  return ret;
2104 }
2105 #endif /* CONFIG_RTPDEC */
2106 
2107 #if CONFIG_SDP_DEMUXER
2108 static int sdp_probe(AVProbeData *p1)
2109 {
2110  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2111 
2112  /* we look for a line beginning "c=IN IP" */
2113  while (p < p_end && *p != '\0') {
2114  if (p + sizeof("c=IN IP") - 1 < p_end &&
2115  av_strstart(p, "c=IN IP", NULL))
2116  return AVPROBE_SCORE_EXTENSION;
2117 
2118  while (p < p_end - 1 && *p != '\n') p++;
2119  if (++p >= p_end)
2120  break;
2121  if (*p == '\r')
2122  p++;
2123  }
2124  return 0;
2125 }
2126 
2127 static void append_source_addrs(char *buf, int size, const char *name,
2128  int count, struct RTSPSource **addrs)
2129 {
2130  int i;
2131  if (!count)
2132  return;
2133  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2134  for (i = 1; i < count; i++)
2135  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2136 }
2137 
2138 static int sdp_read_header(AVFormatContext *s)
2139 {
2140  RTSPState *rt = s->priv_data;
2141  RTSPStream *rtsp_st;
2142  int size, i, err;
2143  char *content;
2144  char url[1024];
2145 
2146  if (!ff_network_init())
2147  return AVERROR(EIO);
2148 
2149  if (s->max_delay < 0) /* Not set by the caller */
2151  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2153 
2154  /* read the whole sdp file */
2155  /* XXX: better loading */
2156  content = av_malloc(SDP_MAX_SIZE);
2157  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2158  if (size <= 0) {
2159  av_free(content);
2160  return AVERROR_INVALIDDATA;
2161  }
2162  content[size] ='\0';
2163 
2164  err = ff_sdp_parse(s, content);
2165  av_free(content);
2166  if (err) goto fail;
2167 
2168  /* open each RTP stream */
2169  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2170  char namebuf[50];
2171  rtsp_st = rt->rtsp_streams[i];
2172 
2173  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2174  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2175  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2176  ff_url_join(url, sizeof(url), "rtp", NULL,
2177  namebuf, rtsp_st->sdp_port,
2178  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2179  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2180  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2181  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2182 
2183  append_source_addrs(url, sizeof(url), "sources",
2184  rtsp_st->nb_include_source_addrs,
2185  rtsp_st->include_source_addrs);
2186  append_source_addrs(url, sizeof(url), "block",
2187  rtsp_st->nb_exclude_source_addrs,
2188  rtsp_st->exclude_source_addrs);
2189  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2190  &s->interrupt_callback, NULL) < 0) {
2191  err = AVERROR_INVALIDDATA;
2192  goto fail;
2193  }
2194  }
2195  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2196  goto fail;
2197  }
2198  return 0;
2199 fail:
2201  ff_network_close();
2202  return err;
2203 }
2204 
2205 static int sdp_read_close(AVFormatContext *s)
2206 {
2208  ff_network_close();
2209  return 0;
2210 }
2211 
2212 static const AVClass sdp_demuxer_class = {
2213  .class_name = "SDP demuxer",
2214  .item_name = av_default_item_name,
2215  .option = sdp_options,
2216  .version = LIBAVUTIL_VERSION_INT,
2217 };
2218 
2219 AVInputFormat ff_sdp_demuxer = {
2220  .name = "sdp",
2221  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2222  .priv_data_size = sizeof(RTSPState),
2223  .read_probe = sdp_probe,
2224  .read_header = sdp_read_header,
2226  .read_close = sdp_read_close,
2227  .priv_class = &sdp_demuxer_class,
2228 };
2229 #endif /* CONFIG_SDP_DEMUXER */
2230 
2231 #if CONFIG_RTP_DEMUXER
2232 static int rtp_probe(AVProbeData *p)
2233 {
2234  if (av_strstart(p->filename, "rtp:", NULL))
2235  return AVPROBE_SCORE_MAX;
2236  return 0;
2237 }
2238 
2239 static int rtp_read_header(AVFormatContext *s)
2240 {
2241  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2242  char host[500], sdp[500];
2243  int ret, port;
2244  URLContext* in = NULL;
2245  int payload_type;
2246  AVCodecContext codec = { 0 };
2247  struct sockaddr_storage addr;
2248  AVIOContext pb;
2249  socklen_t addrlen = sizeof(addr);
2250  RTSPState *rt = s->priv_data;
2251 
2252  if (!ff_network_init())
2253  return AVERROR(EIO);
2254 
2255  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2256  &s->interrupt_callback, NULL);
2257  if (ret)
2258  goto fail;
2259 
2260  while (1) {
2261  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2262  if (ret == AVERROR(EAGAIN))
2263  continue;
2264  if (ret < 0)
2265  goto fail;
2266  if (ret < 12) {
2267  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2268  continue;
2269  }
2270 
2271  if ((recvbuf[0] & 0xc0) != 0x80) {
2272  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2273  "received\n");
2274  continue;
2275  }
2276 
2277  if (RTP_PT_IS_RTCP(recvbuf[1]))
2278  continue;
2279 
2280  payload_type = recvbuf[1] & 0x7f;
2281  break;
2282  }
2283  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2284  ffurl_close(in);
2285  in = NULL;
2286 
2287  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2288  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2289  "without an SDP file describing it\n",
2290  payload_type);
2291  goto fail;
2292  }
2293  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2294  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2295  "properly you need an SDP file "
2296  "describing it\n");
2297  }
2298 
2299  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2300  NULL, 0, s->filename);
2301 
2302  snprintf(sdp, sizeof(sdp),
2303  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2304  addr.ss_family == AF_INET ? 4 : 6, host,
2305  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2306  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2307  port, payload_type);
2308  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2309 
2310  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2311  s->pb = &pb;
2312 
2313  /* sdp_read_header initializes this again */
2314  ff_network_close();
2315 
2316  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2317 
2318  ret = sdp_read_header(s);
2319  s->pb = NULL;
2320  return ret;
2321 
2322 fail:
2323  if (in)
2324  ffurl_close(in);
2325  ff_network_close();
2326  return ret;
2327 }
2328 
2329 static const AVClass rtp_demuxer_class = {
2330  .class_name = "RTP demuxer",
2331  .item_name = av_default_item_name,
2332  .option = rtp_options,
2333  .version = LIBAVUTIL_VERSION_INT,
2334 };
2335 
2336 AVInputFormat ff_rtp_demuxer = {
2337  .name = "rtp",
2338  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2339  .priv_data_size = sizeof(RTSPState),
2340  .read_probe = rtp_probe,
2341  .read_header = rtp_read_header,
2343  .read_close = sdp_read_close,
2344  .flags = AVFMT_NOFILE,
2345  .priv_class = &rtp_demuxer_class,
2346 };
2347 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:272
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:92
void av_url_split(char *proto, int proto_size, char *authorization, int authorization_size, char *hostname, int hostname_size, int *port_ptr, char *path, int path_size, const char *url)
Split a URL string into components.
Definition: utils.c:3092
char crypto_suite[40]
Definition: rtsp.h:457
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:386
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:96
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:527
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
Bytestream IO Context.
Definition: avio.h:68
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:54
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:498
int size
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
Send a dummy packet on both port pairs to set up the connection state in potential NAT routers...
Definition: rtpdec.c:352
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1097
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
Definition: aviobuf.c:960
AVOption.
Definition: opt.h:233
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:114
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:186
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:129
const char * filename
Definition: avformat.h:389
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:146
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:316
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:3208
int av_parse_time(int64_t *timeval, const char *timestr, int duration)
Parse timestr and return in *time a corresponding number of microseconds.
Definition: parseutils.c:482
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:276
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
Windows Media server.
Definition: rtsp.h:208
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:353
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:710
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:159
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
int index
stream index in AVFormatContext
Definition: avformat.h:684
#define AVIO_FLAG_READ
read-only
Definition: avio.h:292
char location[4096]
the "Location:" field.
Definition: rtsp.h:151
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:293
int mode_record
transport set to record data
Definition: rtsp.h:111
enum AVMediaType codec_type
Definition: rtp.c:36
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:166
void ff_network_close(void)
Definition: network.c:150
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:143
initialized and sending/receiving data
Definition: rtsp.h:196
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:269
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:406
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:78
#define freeaddrinfo
Definition: network.h:183
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:437
int ctx_flags
Format-specific flags, see AVFMTCTX_xx.
Definition: avformat.h:916
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:404
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:244
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:238
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:131
AVCodec.
Definition: avcodec.h:2755
#define AI_NUMERICHOST
Definition: network.h:152
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:120
This describes the server response to each RTSP command.
Definition: rtsp.h:126
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:488
#define RECVBUF_SIZE
Definition: rtsp.c:58
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:141
Format I/O context.
Definition: avformat.h:871
#define RTP_PT_PRIVATE
Definition: rtp.h:77
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:142
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:206
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:396
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:38
int recvbuf_len
Definition: rtsp.h:322
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:43
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:358
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:147
Opaque data information usually continuous.
Definition: avutil.h:189
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:108
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:124
int ff_network_init(void)
Definition: network.c:123
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:850
AVOptions.
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:455
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:93
PayloadContext *(* alloc)(void)
Allocate any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:129
int id
Format-specific stream ID.
Definition: avformat.h:690
#define POLL_TIMEOUT_MS
Definition: rtsp.c:54
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:59
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
void(* free)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:131
const char * name
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:935
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:371
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:327
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:420
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
static int flags
Definition: log.c:44
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:130
#define AVERROR_EOF
End of file.
Definition: error.h:51
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:130
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:139
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:21
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:452
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
static float t
Definition: output.c:52
Normal RTSP.
Definition: rtsp.h:68
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:133
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:446
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:890
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:176
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:76
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:122
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
#define CONFIG_RTPDEC
Definition: config.h:378
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:440
Private data for the RTSP demuxer.
Definition: rtsp.h:217
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:254
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
Definition: avio.c:183
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1064
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:186
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:359
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:249
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:142
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int avio_close(AVIOContext *s)
Close the resource accessed by the AVIOContext s and free it.
Definition: aviobuf.c:794
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:144
const AVOption ff_rtsp_options[]
Definition: rtsp.c:78
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:181
URLContext * rtsp_hd
Definition: rtsp.h:219
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:148
AVStream * avformat_new_stream(AVFormatContext *s, AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:2662
const char * name
Name of the codec implementation.
Definition: avcodec.h:2762
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:330
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:438
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:505
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
Definition: base64.c:72
int64_t rtcp_ts_offset
Definition: rtpdec.h:180
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:81
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:110
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:224
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:163
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:70
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:425
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:702
MpegTSContext * ff_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2148
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:391
int seq
RTSP command sequence number.
Definition: rtsp.h:240
#define CONFIG_RTSP_DEMUXER
Definition: config.h:834
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:390
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:338
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:923
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:405
#define NI_NUMERICHOST
Definition: network.h:160
#define LIBAVFORMAT_IDENT
Definition: version.h:44
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:306
int recvbuf_pos
Definition: rtsp.h:321
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:64
char filename[1024]
input or output filename
Definition: avformat.h:943
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:222
int64_t first_rtcp_ntp_time
Definition: rtpdec.h:178
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes.
Definition: base64.h:59
#define FFMIN(a, b)
Definition: common.h:57
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:282
int av_strcasecmp(const char *a, const char *b)
Definition: avstring.c:156
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:128
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: [;ti...
Definition: rtsp.h:171
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:74
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:87
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:111
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:139
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:137
char crypto_params[100]
Definition: rtsp.h:458
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:186
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
Definition: rtpdec.h:126
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:352
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:52
#define ENC
Definition: rtsp.c:63
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:435
int ff_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:2164
Raw data (over UDP)
Definition: rtsp.h:59
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:320
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
int sdp_payload_type
payload type
Definition: rtsp.h:442
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:520
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:439
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:166
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:536
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:37
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:420
Stream structure.
Definition: avformat.h:683
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
Definition: error.h:57
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
int nb_byes
Definition: rtsp.h:335
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:261
NULL
Definition: eval.c:55
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:411
#define AV_LOG_INFO
Standard information.
Definition: log.h:134
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:436
enum AVMediaType codec_type
Definition: avcodec.h:1062
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:641
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:381
enum AVCodecID codec_id
Definition: avcodec.h:1065
int rtp_port_max
Definition: rtsp.h:386
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:1779
AVIOContext * pb
I/O context.
Definition: avformat.h:913
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:381
av_default_item_name
Definition: dnxhdenc.c:45
int server_port_max
Definition: rtsp.h:104
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:73
main external API structure.
Definition: avcodec.h:1054
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:1673
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:65
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:399
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:257
Definition: url.h:41
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
static int read_packet(AVFormatContext *ctx, AVPacket *pkt)
Definition: libcdio.c:114
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:294
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:62
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:376
int client_port_max
Definition: rtsp.h:100
Describe the class of an AVClass context structure.
Definition: log.h:33
#define SDP_MAX_SIZE
Definition: rtsp.c:57
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:511
#define SPACE_CHARS
Definition: internal.h:160
void * priv_data
Definition: url.h:44
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:451
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:278
not initialized
Definition: rtsp.h:195
int64_t range_end
Definition: rtsp.h:137
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:117
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:154
AVMediaType
Definition: avutil.h:185
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:99
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:69
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:699
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:714
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:677
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
Definition: rtpdec.c:249
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:394
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:2595
This structure contains the data a format has to probe a file.
Definition: avformat.h:388
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:75
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:242
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:91
int interleaved_max
Definition: rtsp.h:92
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:266
int ffurl_close(URLContext *h)
Close the resource accessed by the URLContext h, and free the memory used by it.
Definition: avio.c:297
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:137
#define CONFIG_RTSP_MUXER
Definition: config.h:1186
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:952
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:230
#define DEC
Definition: rtsp.c:62
const OptionDef options[]
Definition: avserver.c:4624
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:395
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:32
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:182
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:96
void ff_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2189
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:29
uint32_t ssrc
Definition: rtpdec.h:151
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:400
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:98
int ffio_init_context(AVIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
Definition: aviobuf.c:70
int ffurl_open(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
Definition: avio.c:211
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:287
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:448
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
Definition: avio.c:269
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
#define RTSP_REORDERING_OPTS()
Definition: rtsp.c:75
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:883
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:2640
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
uint32_t base_timestamp
Definition: rtpdec.h:154
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
#define getnameinfo
Definition: network.h:184
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:130
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:275
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:77
int channels
number of audio channels
Definition: avcodec.h:1780
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:431
void * priv_data
Format private data.
Definition: avformat.h:899
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:568
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:822
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:441
#define MAX_TIMEOUTS
Definition: rtsp.c:56
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:672
int ai_flags
Definition: network.h:103
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:962
HTTPAuthType auth_type
The currently chosen auth type.
Definition: httpauth.h:59
Realmedia-style server.
Definition: rtsp.h:207
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:343
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:516
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get random data.
Definition: random_seed.c:95
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:719
int interleaved_max
Definition: rtsp.h:429
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:809
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:113
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
Definition: rtpproto.c:63
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:429
This structure stores compressed data.
Definition: avcodec.h:950
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:104
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:210
static const AVOption rtp_options[]
Definition: rtsp.c:105
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:262
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:205
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:421
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:228
#define OFFSET(x)
Definition: rtsp.c:61
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:96
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:422
No authentication specified.
Definition: httpauth.h:29
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:100