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rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
32 #include "avformat.h"
33 #include "avio_internal.h"
34 
35 #if HAVE_POLL_H
36 #include <poll.h>
37 #endif
38 #include "internal.h"
39 #include "network.h"
40 #include "os_support.h"
41 #include "http.h"
42 #include "rtsp.h"
43 
44 #include "rtpdec.h"
45 #include "rtpproto.h"
46 #include "rdt.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
49 #include "url.h"
50 #include "rtpenc.h"
51 #include "mpegts.h"
52 
53 /* Timeout values for socket poll, in ms,
54  * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
61 
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 
66 #define RTSP_FLAG_OPTS(name, longname) \
67  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68  { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71  { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75  { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
76 
77 #define COMMON_OPTS() \
78  { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79  { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
80 
81 
83  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
84  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92  { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
98  COMMON_OPTS(),
99  { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
100  { NULL },
101 };
102 
103 static const AVOption sdp_options[] = {
104  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106  { "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" },
107  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
108  COMMON_OPTS(),
109  { NULL },
110 };
111 
112 static const AVOption rtp_options[] = {
113  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
114  COMMON_OPTS(),
115  { NULL },
116 };
117 
118 
120 {
122  char buf[256];
123 
124  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125  av_dict_set(&opts, "buffer_size", buf, 0);
126 
127  return opts;
128 }
129 
130 static void get_word_until_chars(char *buf, int buf_size,
131  const char *sep, const char **pp)
132 {
133  const char *p;
134  char *q;
135 
136  p = *pp;
137  p += strspn(p, SPACE_CHARS);
138  q = buf;
139  while (!strchr(sep, *p) && *p != '\0') {
140  if ((q - buf) < buf_size - 1)
141  *q++ = *p;
142  p++;
143  }
144  if (buf_size > 0)
145  *q = '\0';
146  *pp = p;
147 }
148 
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
150  const char **pp)
151 {
152  if (**pp == '/') (*pp)++;
153  get_word_until_chars(buf, buf_size, sep, pp);
154 }
155 
156 static void get_word(char *buf, int buf_size, const char **pp)
157 {
158  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
159 }
160 
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
162  * and end time.
163  * Used for seeking in the rtp stream.
164  */
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
166 {
167  char buf[256];
168 
169  p += strspn(p, SPACE_CHARS);
170  if (!av_stristart(p, "npt=", &p))
171  return;
172 
173  *start = AV_NOPTS_VALUE;
174  *end = AV_NOPTS_VALUE;
175 
176  get_word_sep(buf, sizeof(buf), "-", &p);
177  if (av_parse_time(start, buf, 1) < 0)
178  return;
179  if (*p == '-') {
180  p++;
181  get_word_sep(buf, sizeof(buf), "-", &p);
182  if (av_parse_time(end, buf, 1) < 0)
183  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
184  }
185 }
186 
188  const char *buf, struct sockaddr_storage *sock)
189 {
190  struct addrinfo hints = { 0 }, *ai = NULL;
191  int ret;
192 
193  hints.ai_flags = AI_NUMERICHOST;
194  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
195  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
196  buf,
197  gai_strerror(ret));
198  return -1;
199  }
200  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
201  freeaddrinfo(ai);
202  return 0;
203 }
204 
205 #if CONFIG_RTPDEC
206 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
207  RTSPStream *rtsp_st, AVStream *st)
208 {
209  AVCodecParameters *par = st ? st->codecpar : NULL;
210  if (!handler)
211  return;
212  if (par)
213  par->codec_id = handler->codec_id;
214  rtsp_st->dynamic_handler = handler;
215  if (st)
216  st->need_parsing = handler->need_parsing;
217  if (handler->priv_data_size) {
219  if (!rtsp_st->dynamic_protocol_context)
220  rtsp_st->dynamic_handler = NULL;
221  }
222 }
223 
224 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
225  AVStream *st)
226 {
227  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
228  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
229  rtsp_st->dynamic_protocol_context);
230  if (ret < 0) {
231  if (rtsp_st->dynamic_protocol_context) {
232  if (rtsp_st->dynamic_handler->close)
233  rtsp_st->dynamic_handler->close(
234  rtsp_st->dynamic_protocol_context);
236  }
237  rtsp_st->dynamic_protocol_context = NULL;
238  rtsp_st->dynamic_handler = NULL;
239  }
240  }
241 }
242 
243 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
244 static int sdp_parse_rtpmap(AVFormatContext *s,
245  AVStream *st, RTSPStream *rtsp_st,
246  int payload_type, const char *p)
247 {
248  AVCodecParameters *par = st->codecpar;
249  char buf[256];
250  int i;
251  const AVCodecDescriptor *desc;
252  const char *c_name;
253 
254  /* See if we can handle this kind of payload.
255  * The space should normally not be there but some Real streams or
256  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
257  * have a trailing space. */
258  get_word_sep(buf, sizeof(buf), "/ ", &p);
259  if (payload_type < RTP_PT_PRIVATE) {
260  /* We are in a standard case
261  * (from http://www.iana.org/assignments/rtp-parameters). */
262  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
263  }
264 
265  if (par->codec_id == AV_CODEC_ID_NONE) {
266  RTPDynamicProtocolHandler *handler =
268  init_rtp_handler(handler, rtsp_st, st);
269  /* If no dynamic handler was found, check with the list of standard
270  * allocated types, if such a stream for some reason happens to
271  * use a private payload type. This isn't handled in rtpdec.c, since
272  * the format name from the rtpmap line never is passed into rtpdec. */
273  if (!rtsp_st->dynamic_handler)
274  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
275  }
276 
277  desc = avcodec_descriptor_get(par->codec_id);
278  if (desc && desc->name)
279  c_name = desc->name;
280  else
281  c_name = "(null)";
282 
283  get_word_sep(buf, sizeof(buf), "/", &p);
284  i = atoi(buf);
285  switch (par->codec_type) {
286  case AVMEDIA_TYPE_AUDIO:
287  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
290  if (i > 0) {
291  par->sample_rate = i;
292  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
293  get_word_sep(buf, sizeof(buf), "/", &p);
294  i = atoi(buf);
295  if (i > 0)
296  par->channels = i;
297  }
298  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
299  par->sample_rate);
300  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
301  par->channels);
302  break;
303  case AVMEDIA_TYPE_VIDEO:
304  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
305  if (i > 0)
306  avpriv_set_pts_info(st, 32, 1, i);
307  break;
308  default:
309  break;
310  }
311  finalize_rtp_handler_init(s, rtsp_st, st);
312  return 0;
313 }
314 
315 /* parse the attribute line from the fmtp a line of an sdp response. This
316  * is broken out as a function because it is used in rtp_h264.c, which is
317  * forthcoming. */
318 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
319  char *value, int value_size)
320 {
321  *p += strspn(*p, SPACE_CHARS);
322  if (**p) {
323  get_word_sep(attr, attr_size, "=", p);
324  if (**p == '=')
325  (*p)++;
326  get_word_sep(value, value_size, ";", p);
327  if (**p == ';')
328  (*p)++;
329  return 1;
330  }
331  return 0;
332 }
333 
334 typedef struct SDPParseState {
335  /* SDP only */
336  struct sockaddr_storage default_ip;
337  int default_ttl;
338  int skip_media; ///< set if an unknown m= line occurs
339  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
340  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
341  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
342  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
343  int seen_rtpmap;
344  int seen_fmtp;
345  char delayed_fmtp[2048];
346 } SDPParseState;
347 
348 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
349  struct RTSPSource ***dest, int *dest_count)
350 {
351  RTSPSource *rtsp_src, *rtsp_src2;
352  int i;
353  for (i = 0; i < count; i++) {
354  rtsp_src = addrs[i];
355  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
356  if (!rtsp_src2)
357  continue;
358  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
359  dynarray_add(dest, dest_count, rtsp_src2);
360  }
361 }
362 
363 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
364  int payload_type, const char *line)
365 {
366  int i;
367 
368  for (i = 0; i < rt->nb_rtsp_streams; i++) {
369  RTSPStream *rtsp_st = rt->rtsp_streams[i];
370  if (rtsp_st->sdp_payload_type == payload_type &&
371  rtsp_st->dynamic_handler &&
372  rtsp_st->dynamic_handler->parse_sdp_a_line) {
373  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
374  rtsp_st->dynamic_protocol_context, line);
375  }
376  }
377 }
378 
379 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
380  int letter, const char *buf)
381 {
382  RTSPState *rt = s->priv_data;
383  char buf1[64], st_type[64];
384  const char *p;
385  enum AVMediaType codec_type;
386  int payload_type;
387  AVStream *st;
388  RTSPStream *rtsp_st;
389  RTSPSource *rtsp_src;
390  struct sockaddr_storage sdp_ip;
391  int ttl;
392 
393  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
394 
395  p = buf;
396  if (s1->skip_media && letter != 'm')
397  return;
398  switch (letter) {
399  case 'c':
400  get_word(buf1, sizeof(buf1), &p);
401  if (strcmp(buf1, "IN") != 0)
402  return;
403  get_word(buf1, sizeof(buf1), &p);
404  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
405  return;
406  get_word_sep(buf1, sizeof(buf1), "/", &p);
407  if (get_sockaddr(s, buf1, &sdp_ip))
408  return;
409  ttl = 16;
410  if (*p == '/') {
411  p++;
412  get_word_sep(buf1, sizeof(buf1), "/", &p);
413  ttl = atoi(buf1);
414  }
415  if (s->nb_streams == 0) {
416  s1->default_ip = sdp_ip;
417  s1->default_ttl = ttl;
418  } else {
419  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420  rtsp_st->sdp_ip = sdp_ip;
421  rtsp_st->sdp_ttl = ttl;
422  }
423  break;
424  case 's':
425  av_dict_set(&s->metadata, "title", p, 0);
426  break;
427  case 'i':
428  if (s->nb_streams == 0) {
429  av_dict_set(&s->metadata, "comment", p, 0);
430  break;
431  }
432  break;
433  case 'm':
434  /* new stream */
435  s1->skip_media = 0;
436  s1->seen_fmtp = 0;
437  s1->seen_rtpmap = 0;
438  codec_type = AVMEDIA_TYPE_UNKNOWN;
439  get_word(st_type, sizeof(st_type), &p);
440  if (!strcmp(st_type, "audio")) {
441  codec_type = AVMEDIA_TYPE_AUDIO;
442  } else if (!strcmp(st_type, "video")) {
443  codec_type = AVMEDIA_TYPE_VIDEO;
444  } else if (!strcmp(st_type, "application")) {
445  codec_type = AVMEDIA_TYPE_DATA;
446  } else if (!strcmp(st_type, "text")) {
447  codec_type = AVMEDIA_TYPE_SUBTITLE;
448  }
449  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
450  s1->skip_media = 1;
451  return;
452  }
453  rtsp_st = av_mallocz(sizeof(RTSPStream));
454  if (!rtsp_st)
455  return;
456  rtsp_st->stream_index = -1;
457  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
458 
459  rtsp_st->sdp_ip = s1->default_ip;
460  rtsp_st->sdp_ttl = s1->default_ttl;
461 
462  copy_default_source_addrs(s1->default_include_source_addrs,
463  s1->nb_default_include_source_addrs,
464  &rtsp_st->include_source_addrs,
465  &rtsp_st->nb_include_source_addrs);
466  copy_default_source_addrs(s1->default_exclude_source_addrs,
467  s1->nb_default_exclude_source_addrs,
468  &rtsp_st->exclude_source_addrs,
469  &rtsp_st->nb_exclude_source_addrs);
470 
471  get_word(buf1, sizeof(buf1), &p); /* port */
472  rtsp_st->sdp_port = atoi(buf1);
473 
474  get_word(buf1, sizeof(buf1), &p); /* protocol */
475  if (!strcmp(buf1, "udp"))
477  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
478  rtsp_st->feedback = 1;
479 
480  /* XXX: handle list of formats */
481  get_word(buf1, sizeof(buf1), &p); /* format list */
482  rtsp_st->sdp_payload_type = atoi(buf1);
483 
484  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
485  /* no corresponding stream */
486  if (rt->transport == RTSP_TRANSPORT_RAW) {
487  if (CONFIG_RTPDEC && !rt->ts)
488  rt->ts = avpriv_mpegts_parse_open(s);
489  } else {
491  handler = ff_rtp_handler_find_by_id(
493  init_rtp_handler(handler, rtsp_st, NULL);
494  finalize_rtp_handler_init(s, rtsp_st, NULL);
495  }
496  } else if (rt->server_type == RTSP_SERVER_WMS &&
497  codec_type == AVMEDIA_TYPE_DATA) {
498  /* RTX stream, a stream that carries all the other actual
499  * audio/video streams. Don't expose this to the callers. */
500  } else {
501  st = avformat_new_stream(s, NULL);
502  if (!st)
503  return;
504  st->id = rt->nb_rtsp_streams - 1;
505  rtsp_st->stream_index = st->index;
507  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
509  /* if standard payload type, we can find the codec right now */
511  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
512  st->codecpar->sample_rate > 0)
513  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
514  /* Even static payload types may need a custom depacketizer */
515  handler = ff_rtp_handler_find_by_id(
516  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
517  init_rtp_handler(handler, rtsp_st, st);
518  finalize_rtp_handler_init(s, rtsp_st, st);
519  }
520  if (rt->default_lang[0])
521  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
522  }
523  /* put a default control url */
524  av_strlcpy(rtsp_st->control_url, rt->control_uri,
525  sizeof(rtsp_st->control_url));
526  break;
527  case 'a':
528  if (av_strstart(p, "control:", &p)) {
529  if (s->nb_streams == 0) {
530  if (!strncmp(p, "rtsp://", 7))
531  av_strlcpy(rt->control_uri, p,
532  sizeof(rt->control_uri));
533  } else {
534  char proto[32];
535  /* get the control url */
536  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
537 
538  /* XXX: may need to add full url resolution */
539  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
540  NULL, NULL, 0, p);
541  if (proto[0] == '\0') {
542  /* relative control URL */
543  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
544  av_strlcat(rtsp_st->control_url, "/",
545  sizeof(rtsp_st->control_url));
546  av_strlcat(rtsp_st->control_url, p,
547  sizeof(rtsp_st->control_url));
548  } else
549  av_strlcpy(rtsp_st->control_url, p,
550  sizeof(rtsp_st->control_url));
551  }
552  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
553  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
554  get_word(buf1, sizeof(buf1), &p);
555  payload_type = atoi(buf1);
556  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
557  if (rtsp_st->stream_index >= 0) {
558  st = s->streams[rtsp_st->stream_index];
559  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
560  }
561  s1->seen_rtpmap = 1;
562  if (s1->seen_fmtp) {
563  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
564  }
565  } else if (av_strstart(p, "fmtp:", &p) ||
566  av_strstart(p, "framesize:", &p)) {
567  // let dynamic protocol handlers have a stab at the line.
568  get_word(buf1, sizeof(buf1), &p);
569  payload_type = atoi(buf1);
570  if (s1->seen_rtpmap) {
571  parse_fmtp(s, rt, payload_type, buf);
572  } else {
573  s1->seen_fmtp = 1;
574  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
575  }
576  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
577  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
578  get_word(buf1, sizeof(buf1), &p);
579  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
580  } else if (av_strstart(p, "range:", &p)) {
581  int64_t start, end;
582 
583  // this is so that seeking on a streamed file can work.
584  rtsp_parse_range_npt(p, &start, &end);
585  s->start_time = start;
586  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
587  s->duration = (end == AV_NOPTS_VALUE) ?
588  AV_NOPTS_VALUE : end - start;
589  } else if (av_strstart(p, "lang:", &p)) {
590  if (s->nb_streams > 0) {
591  get_word(buf1, sizeof(buf1), &p);
592  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593  if (rtsp_st->stream_index >= 0) {
594  st = s->streams[rtsp_st->stream_index];
595  av_dict_set(&st->metadata, "language", buf1, 0);
596  }
597  } else
598  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
599  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
600  if (atoi(p) == 1)
602  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
603  s->nb_streams > 0) {
604  st = s->streams[s->nb_streams - 1];
605  st->codecpar->sample_rate = atoi(p);
606  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
607  // RFC 4568
608  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609  get_word(buf1, sizeof(buf1), &p); // ignore tag
610  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
611  p += strspn(p, SPACE_CHARS);
612  if (av_strstart(p, "inline:", &p))
613  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
614  } else if (av_strstart(p, "source-filter:", &p)) {
615  int exclude = 0;
616  get_word(buf1, sizeof(buf1), &p);
617  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
618  return;
619  exclude = !strcmp(buf1, "excl");
620 
621  get_word(buf1, sizeof(buf1), &p);
622  if (strcmp(buf1, "IN") != 0)
623  return;
624  get_word(buf1, sizeof(buf1), &p);
625  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
626  return;
627  // not checking that the destination address actually matches or is wildcard
628  get_word(buf1, sizeof(buf1), &p);
629 
630  while (*p != '\0') {
631  rtsp_src = av_mallocz(sizeof(*rtsp_src));
632  if (!rtsp_src)
633  return;
634  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
635  if (exclude) {
636  if (s->nb_streams == 0) {
637  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
638  } else {
639  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
640  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
641  }
642  } else {
643  if (s->nb_streams == 0) {
644  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
645  } else {
646  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
647  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
648  }
649  }
650  }
651  } else {
652  if (rt->server_type == RTSP_SERVER_WMS)
654  if (s->nb_streams > 0) {
655  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
656 
657  if (rt->server_type == RTSP_SERVER_REAL)
658  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
659 
660  if (rtsp_st->dynamic_handler &&
662  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
663  rtsp_st->stream_index,
664  rtsp_st->dynamic_protocol_context, buf);
665  }
666  }
667  break;
668  }
669 }
670 
671 int ff_sdp_parse(AVFormatContext *s, const char *content)
672 {
673  RTSPState *rt = s->priv_data;
674  const char *p;
675  int letter, i;
676  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
677  * contain long SDP lines containing complete ASF Headers (several
678  * kB) or arrays of MDPR (RM stream descriptor) headers plus
679  * "rulebooks" describing their properties. Therefore, the SDP line
680  * buffer is large.
681  *
682  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
683  * in rtpdec_xiph.c. */
684  char buf[16384], *q;
685  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
686 
687  p = content;
688  for (;;) {
689  p += strspn(p, SPACE_CHARS);
690  letter = *p;
691  if (letter == '\0')
692  break;
693  p++;
694  if (*p != '=')
695  goto next_line;
696  p++;
697  /* get the content */
698  q = buf;
699  while (*p != '\n' && *p != '\r' && *p != '\0') {
700  if ((q - buf) < sizeof(buf) - 1)
701  *q++ = *p;
702  p++;
703  }
704  *q = '\0';
705  sdp_parse_line(s, s1, letter, buf);
706  next_line:
707  while (*p != '\n' && *p != '\0')
708  p++;
709  if (*p == '\n')
710  p++;
711  }
712 
713  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
714  av_freep(&s1->default_include_source_addrs[i]);
715  av_freep(&s1->default_include_source_addrs);
716  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
717  av_freep(&s1->default_exclude_source_addrs[i]);
718  av_freep(&s1->default_exclude_source_addrs);
719 
720  rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
721  if (!rt->p) return AVERROR(ENOMEM);
722  return 0;
723 }
724 #endif /* CONFIG_RTPDEC */
725 
726 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
727 {
728  RTSPState *rt = s->priv_data;
729  int i;
730 
731  for (i = 0; i < rt->nb_rtsp_streams; i++) {
732  RTSPStream *rtsp_st = rt->rtsp_streams[i];
733  if (!rtsp_st)
734  continue;
735  if (rtsp_st->transport_priv) {
736  if (s->oformat) {
737  AVFormatContext *rtpctx = rtsp_st->transport_priv;
738  av_write_trailer(rtpctx);
740  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
741  ff_rtsp_tcp_write_packet(s, rtsp_st);
742  ffio_free_dyn_buf(&rtpctx->pb);
743  } else {
744  avio_closep(&rtpctx->pb);
745  }
746  avformat_free_context(rtpctx);
747  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
749  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
751  }
752  rtsp_st->transport_priv = NULL;
753  if (rtsp_st->rtp_handle)
754  ffurl_close(rtsp_st->rtp_handle);
755  rtsp_st->rtp_handle = NULL;
756  }
757 }
758 
759 /* close and free RTSP streams */
761 {
762  RTSPState *rt = s->priv_data;
763  int i, j;
764  RTSPStream *rtsp_st;
765 
766  ff_rtsp_undo_setup(s, 0);
767  for (i = 0; i < rt->nb_rtsp_streams; i++) {
768  rtsp_st = rt->rtsp_streams[i];
769  if (rtsp_st) {
770  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
771  if (rtsp_st->dynamic_handler->close)
772  rtsp_st->dynamic_handler->close(
773  rtsp_st->dynamic_protocol_context);
775  }
776  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
777  av_freep(&rtsp_st->include_source_addrs[j]);
778  av_freep(&rtsp_st->include_source_addrs);
779  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
780  av_freep(&rtsp_st->exclude_source_addrs[j]);
781  av_freep(&rtsp_st->exclude_source_addrs);
782 
783  av_freep(&rtsp_st);
784  }
785  }
786  av_freep(&rt->rtsp_streams);
787  if (rt->asf_ctx) {
789  }
790  if (CONFIG_RTPDEC && rt->ts)
792  av_freep(&rt->p);
793  av_freep(&rt->recvbuf);
794 }
795 
797 {
798  RTSPState *rt = s->priv_data;
799  AVStream *st = NULL;
800  int reordering_queue_size = rt->reordering_queue_size;
801  if (reordering_queue_size < 0) {
803  reordering_queue_size = 0;
804  else
805  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
806  }
807 
808  /* open the RTP context */
809  if (rtsp_st->stream_index >= 0)
810  st = s->streams[rtsp_st->stream_index];
811  if (!st)
813 
814  if (CONFIG_RTSP_MUXER && s->oformat && st) {
815  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
816  s, st, rtsp_st->rtp_handle,
818  rtsp_st->stream_index);
819  /* Ownership of rtp_handle is passed to the rtp mux context */
820  rtsp_st->rtp_handle = NULL;
821  if (ret < 0)
822  return ret;
823  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
824  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
825  return 0; // Don't need to open any parser here
826  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
827  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
828  rtsp_st->dynamic_protocol_context,
829  rtsp_st->dynamic_handler);
830  else if (CONFIG_RTPDEC)
831  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
832  rtsp_st->sdp_payload_type,
833  reordering_queue_size);
834 
835  if (!rtsp_st->transport_priv) {
836  return AVERROR(ENOMEM);
837  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
838  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
839  rtpctx->ssrc = rtsp_st->ssrc;
840  if (rtsp_st->dynamic_handler) {
842  rtsp_st->dynamic_protocol_context,
843  rtsp_st->dynamic_handler);
844  }
845  if (rtsp_st->crypto_suite[0])
847  rtsp_st->crypto_suite,
848  rtsp_st->crypto_params);
849  }
850 
851  return 0;
852 }
853 
854 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
855 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
856 {
857  const char *q;
858  char *p;
859  int v;
860 
861  q = *pp;
862  q += strspn(q, SPACE_CHARS);
863  v = strtol(q, &p, 10);
864  if (*p == '-') {
865  p++;
866  *min_ptr = v;
867  v = strtol(p, &p, 10);
868  *max_ptr = v;
869  } else {
870  *min_ptr = v;
871  *max_ptr = v;
872  }
873  *pp = p;
874 }
875 
876 /* XXX: only one transport specification is parsed */
877 static void rtsp_parse_transport(AVFormatContext *s,
878  RTSPMessageHeader *reply, const char *p)
879 {
880  char transport_protocol[16];
881  char profile[16];
882  char lower_transport[16];
883  char parameter[16];
885  char buf[256];
886 
887  reply->nb_transports = 0;
888 
889  for (;;) {
890  p += strspn(p, SPACE_CHARS);
891  if (*p == '\0')
892  break;
893 
894  th = &reply->transports[reply->nb_transports];
895 
896  get_word_sep(transport_protocol, sizeof(transport_protocol),
897  "/", &p);
898  if (!av_strcasecmp (transport_protocol, "rtp")) {
899  get_word_sep(profile, sizeof(profile), "/;,", &p);
900  lower_transport[0] = '\0';
901  /* rtp/avp/<protocol> */
902  if (*p == '/') {
903  get_word_sep(lower_transport, sizeof(lower_transport),
904  ";,", &p);
905  }
907  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
908  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
909  /* x-pn-tng/<protocol> */
910  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
911  profile[0] = '\0';
913  } else if (!av_strcasecmp(transport_protocol, "raw")) {
914  get_word_sep(profile, sizeof(profile), "/;,", &p);
915  lower_transport[0] = '\0';
916  /* raw/raw/<protocol> */
917  if (*p == '/') {
918  get_word_sep(lower_transport, sizeof(lower_transport),
919  ";,", &p);
920  }
922  }
923  if (!av_strcasecmp(lower_transport, "TCP"))
925  else
927 
928  if (*p == ';')
929  p++;
930  /* get each parameter */
931  while (*p != '\0' && *p != ',') {
932  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
933  if (!strcmp(parameter, "port")) {
934  if (*p == '=') {
935  p++;
936  rtsp_parse_range(&th->port_min, &th->port_max, &p);
937  }
938  } else if (!strcmp(parameter, "client_port")) {
939  if (*p == '=') {
940  p++;
941  rtsp_parse_range(&th->client_port_min,
942  &th->client_port_max, &p);
943  }
944  } else if (!strcmp(parameter, "server_port")) {
945  if (*p == '=') {
946  p++;
947  rtsp_parse_range(&th->server_port_min,
948  &th->server_port_max, &p);
949  }
950  } else if (!strcmp(parameter, "interleaved")) {
951  if (*p == '=') {
952  p++;
953  rtsp_parse_range(&th->interleaved_min,
954  &th->interleaved_max, &p);
955  }
956  } else if (!strcmp(parameter, "multicast")) {
959  } else if (!strcmp(parameter, "ttl")) {
960  if (*p == '=') {
961  char *end;
962  p++;
963  th->ttl = strtol(p, &end, 10);
964  p = end;
965  }
966  } else if (!strcmp(parameter, "destination")) {
967  if (*p == '=') {
968  p++;
969  get_word_sep(buf, sizeof(buf), ";,", &p);
970  get_sockaddr(s, buf, &th->destination);
971  }
972  } else if (!strcmp(parameter, "source")) {
973  if (*p == '=') {
974  p++;
975  get_word_sep(buf, sizeof(buf), ";,", &p);
976  av_strlcpy(th->source, buf, sizeof(th->source));
977  }
978  } else if (!strcmp(parameter, "mode")) {
979  if (*p == '=') {
980  p++;
981  get_word_sep(buf, sizeof(buf), ";, ", &p);
982  if (!strcmp(buf, "record") ||
983  !strcmp(buf, "receive"))
984  th->mode_record = 1;
985  }
986  }
987 
988  while (*p != ';' && *p != '\0' && *p != ',')
989  p++;
990  if (*p == ';')
991  p++;
992  }
993  if (*p == ',')
994  p++;
995 
996  reply->nb_transports++;
997  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
998  break;
999  }
1000 }
1001 
1002 static void handle_rtp_info(RTSPState *rt, const char *url,
1003  uint32_t seq, uint32_t rtptime)
1004 {
1005  int i;
1006  if (!rtptime || !url[0])
1007  return;
1008  if (rt->transport != RTSP_TRANSPORT_RTP)
1009  return;
1010  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1011  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1012  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1013  if (!rtpctx)
1014  continue;
1015  if (!strcmp(rtsp_st->control_url, url)) {
1016  rtpctx->base_timestamp = rtptime;
1017  break;
1018  }
1019  }
1020 }
1021 
1022 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1023 {
1024  int read = 0;
1025  char key[20], value[1024], url[1024] = "";
1026  uint32_t seq = 0, rtptime = 0;
1027 
1028  for (;;) {
1029  p += strspn(p, SPACE_CHARS);
1030  if (!*p)
1031  break;
1032  get_word_sep(key, sizeof(key), "=", &p);
1033  if (*p != '=')
1034  break;
1035  p++;
1036  get_word_sep(value, sizeof(value), ";, ", &p);
1037  read++;
1038  if (!strcmp(key, "url"))
1039  av_strlcpy(url, value, sizeof(url));
1040  else if (!strcmp(key, "seq"))
1041  seq = strtoul(value, NULL, 10);
1042  else if (!strcmp(key, "rtptime"))
1043  rtptime = strtoul(value, NULL, 10);
1044  if (*p == ',') {
1045  handle_rtp_info(rt, url, seq, rtptime);
1046  url[0] = '\0';
1047  seq = rtptime = 0;
1048  read = 0;
1049  }
1050  if (*p)
1051  p++;
1052  }
1053  if (read > 0)
1054  handle_rtp_info(rt, url, seq, rtptime);
1055 }
1056 
1058  RTSPMessageHeader *reply, const char *buf,
1059  RTSPState *rt, const char *method)
1060 {
1061  const char *p;
1062 
1063  /* NOTE: we do case independent match for broken servers */
1064  p = buf;
1065  if (av_stristart(p, "Session:", &p)) {
1066  int t;
1067  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1068  if (av_stristart(p, ";timeout=", &p) &&
1069  (t = strtol(p, NULL, 10)) > 0) {
1070  reply->timeout = t;
1071  }
1072  } else if (av_stristart(p, "Content-Length:", &p)) {
1073  reply->content_length = strtol(p, NULL, 10);
1074  } else if (av_stristart(p, "Transport:", &p)) {
1075  rtsp_parse_transport(s, reply, p);
1076  } else if (av_stristart(p, "CSeq:", &p)) {
1077  reply->seq = strtol(p, NULL, 10);
1078  } else if (av_stristart(p, "Range:", &p)) {
1079  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1080  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1081  p += strspn(p, SPACE_CHARS);
1082  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1083  } else if (av_stristart(p, "Server:", &p)) {
1084  p += strspn(p, SPACE_CHARS);
1085  av_strlcpy(reply->server, p, sizeof(reply->server));
1086  } else if (av_stristart(p, "Notice:", &p) ||
1087  av_stristart(p, "X-Notice:", &p)) {
1088  reply->notice = strtol(p, NULL, 10);
1089  } else if (av_stristart(p, "Location:", &p)) {
1090  p += strspn(p, SPACE_CHARS);
1091  av_strlcpy(reply->location, p , sizeof(reply->location));
1092  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1093  p += strspn(p, SPACE_CHARS);
1094  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1095  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1096  p += strspn(p, SPACE_CHARS);
1097  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1098  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1099  p += strspn(p, SPACE_CHARS);
1100  if (method && !strcmp(method, "DESCRIBE"))
1101  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1102  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1103  p += strspn(p, SPACE_CHARS);
1104  if (method && !strcmp(method, "PLAY"))
1105  rtsp_parse_rtp_info(rt, p);
1106  } else if (av_stristart(p, "Public:", &p) && rt) {
1107  if (strstr(p, "GET_PARAMETER") &&
1108  method && !strcmp(method, "OPTIONS"))
1109  rt->get_parameter_supported = 1;
1110  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1111  p += strspn(p, SPACE_CHARS);
1112  rt->accept_dynamic_rate = atoi(p);
1113  } else if (av_stristart(p, "Content-Type:", &p)) {
1114  p += strspn(p, SPACE_CHARS);
1115  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1116  }
1117 }
1118 
1119 /* skip a RTP/TCP interleaved packet */
1121 {
1122  RTSPState *rt = s->priv_data;
1123  int ret, len, len1;
1124  uint8_t buf[1024];
1125 
1126  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1127  if (ret != 3)
1128  return;
1129  len = AV_RB16(buf + 1);
1130 
1131  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1132 
1133  /* skip payload */
1134  while (len > 0) {
1135  len1 = len;
1136  if (len1 > sizeof(buf))
1137  len1 = sizeof(buf);
1138  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1139  if (ret != len1)
1140  return;
1141  len -= len1;
1142  }
1143 }
1144 
1146  unsigned char **content_ptr,
1147  int return_on_interleaved_data, const char *method)
1148 {
1149  RTSPState *rt = s->priv_data;
1150  char buf[4096], buf1[1024], *q;
1151  unsigned char ch;
1152  const char *p;
1153  int ret, content_length, line_count = 0, request = 0;
1154  unsigned char *content = NULL;
1155 
1156 start:
1157  line_count = 0;
1158  request = 0;
1159  content = NULL;
1160  memset(reply, 0, sizeof(*reply));
1161 
1162  /* parse reply (XXX: use buffers) */
1163  rt->last_reply[0] = '\0';
1164  for (;;) {
1165  q = buf;
1166  for (;;) {
1167  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1168  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1169  if (ret != 1)
1170  return AVERROR_EOF;
1171  if (ch == '\n')
1172  break;
1173  if (ch == '$' && q == buf) {
1174  if (return_on_interleaved_data) {
1175  return 1;
1176  } else
1178  } else if (ch != '\r') {
1179  if ((q - buf) < sizeof(buf) - 1)
1180  *q++ = ch;
1181  }
1182  }
1183  *q = '\0';
1184 
1185  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1186 
1187  /* test if last line */
1188  if (buf[0] == '\0')
1189  break;
1190  p = buf;
1191  if (line_count == 0) {
1192  /* get reply code */
1193  get_word(buf1, sizeof(buf1), &p);
1194  if (!strncmp(buf1, "RTSP/", 5)) {
1195  get_word(buf1, sizeof(buf1), &p);
1196  reply->status_code = atoi(buf1);
1197  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1198  } else {
1199  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1200  get_word(buf1, sizeof(buf1), &p); // object
1201  request = 1;
1202  }
1203  } else {
1204  ff_rtsp_parse_line(s, reply, p, rt, method);
1205  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1206  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1207  }
1208  line_count++;
1209  }
1210 
1211  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1212  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1213 
1214  content_length = reply->content_length;
1215  if (content_length > 0) {
1216  /* leave some room for a trailing '\0' (useful for simple parsing) */
1217  content = av_malloc(content_length + 1);
1218  if (!content)
1219  return AVERROR(ENOMEM);
1220  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1221  content[content_length] = '\0';
1222  }
1223  if (content_ptr)
1224  *content_ptr = content;
1225  else
1226  av_freep(&content);
1227 
1228  if (request) {
1229  char buf[1024];
1230  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1231  const char* ptr = buf;
1232 
1233  if (!strcmp(reply->reason, "OPTIONS")) {
1234  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1235  if (reply->seq)
1236  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1237  if (reply->session_id[0])
1238  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1239  reply->session_id);
1240  } else {
1241  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1242  }
1243  av_strlcat(buf, "\r\n", sizeof(buf));
1244 
1245  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1246  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1247  ptr = base64buf;
1248  }
1249  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1250 
1252  /* Even if the request from the server had data, it is not the data
1253  * that the caller wants or expects. The memory could also be leaked
1254  * if the actual following reply has content data. */
1255  if (content_ptr)
1256  av_freep(content_ptr);
1257  /* If method is set, this is called from ff_rtsp_send_cmd,
1258  * where a reply to exactly this request is awaited. For
1259  * callers from within packet receiving, we just want to
1260  * return to the caller and go back to receiving packets. */
1261  if (method)
1262  goto start;
1263  return 0;
1264  }
1265 
1266  if (rt->seq != reply->seq) {
1267  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1268  rt->seq, reply->seq);
1269  }
1270 
1271  /* EOS */
1272  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1273  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1274  reply->notice == 2306 /* Continuous Feed Terminated */) {
1275  rt->state = RTSP_STATE_IDLE;
1276  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1277  return AVERROR(EIO); /* data or server error */
1278  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1279  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1280  return AVERROR(EPERM);
1281 
1282  return 0;
1283 }
1284 
1285 /**
1286  * Send a command to the RTSP server without waiting for the reply.
1287  *
1288  * @param s RTSP (de)muxer context
1289  * @param method the method for the request
1290  * @param url the target url for the request
1291  * @param headers extra header lines to include in the request
1292  * @param send_content if non-null, the data to send as request body content
1293  * @param send_content_length the length of the send_content data, or 0 if
1294  * send_content is null
1295  *
1296  * @return zero if success, nonzero otherwise
1297  */
1298 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1299  const char *method, const char *url,
1300  const char *headers,
1301  const unsigned char *send_content,
1302  int send_content_length)
1303 {
1304  RTSPState *rt = s->priv_data;
1305  char buf[4096], *out_buf;
1306  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1307 
1308  /* Add in RTSP headers */
1309  out_buf = buf;
1310  rt->seq++;
1311  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1312  if (headers)
1313  av_strlcat(buf, headers, sizeof(buf));
1314  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1315  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1316  if (rt->session_id[0] != '\0' && (!headers ||
1317  !strstr(headers, "\nIf-Match:"))) {
1318  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1319  }
1320  if (rt->auth[0]) {
1321  char *str = ff_http_auth_create_response(&rt->auth_state,
1322  rt->auth, url, method);
1323  if (str)
1324  av_strlcat(buf, str, sizeof(buf));
1325  av_free(str);
1326  }
1327  if (send_content_length > 0 && send_content)
1328  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1329  av_strlcat(buf, "\r\n", sizeof(buf));
1330 
1331  /* base64 encode rtsp if tunneling */
1332  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1333  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1334  out_buf = base64buf;
1335  }
1336 
1337  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1338 
1339  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1340  if (send_content_length > 0 && send_content) {
1341  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1342  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1343  "with content data not supported\n");
1344  return AVERROR_PATCHWELCOME;
1345  }
1346  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1347  }
1349 
1350  return 0;
1351 }
1352 
1353 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1354  const char *url, const char *headers)
1355 {
1356  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1357 }
1358 
1359 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1360  const char *headers, RTSPMessageHeader *reply,
1361  unsigned char **content_ptr)
1362 {
1363  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1364  content_ptr, NULL, 0);
1365 }
1366 
1368  const char *method, const char *url,
1369  const char *header,
1370  RTSPMessageHeader *reply,
1371  unsigned char **content_ptr,
1372  const unsigned char *send_content,
1373  int send_content_length)
1374 {
1375  RTSPState *rt = s->priv_data;
1376  HTTPAuthType cur_auth_type;
1377  int ret, attempts = 0;
1378 
1379 retry:
1380  cur_auth_type = rt->auth_state.auth_type;
1381  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1382  send_content,
1383  send_content_length)))
1384  return ret;
1385 
1386  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1387  return ret;
1388  attempts++;
1389 
1390  if (reply->status_code == 401 &&
1391  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1392  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1393  goto retry;
1394 
1395  if (reply->status_code > 400){
1396  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1397  method,
1398  reply->status_code,
1399  reply->reason);
1400  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1401  }
1402 
1403  return 0;
1404 }
1405 
1406 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1407  int lower_transport, const char *real_challenge)
1408 {
1409  RTSPState *rt = s->priv_data;
1410  int rtx = 0, j, i, err, interleave = 0, port_off;
1411  RTSPStream *rtsp_st;
1412  RTSPMessageHeader reply1, *reply = &reply1;
1413  char cmd[2048];
1414  const char *trans_pref;
1415 
1416  if (rt->transport == RTSP_TRANSPORT_RDT)
1417  trans_pref = "x-pn-tng";
1418  else if (rt->transport == RTSP_TRANSPORT_RAW)
1419  trans_pref = "RAW/RAW";
1420  else
1421  trans_pref = "RTP/AVP";
1422 
1423  /* default timeout: 1 minute */
1424  rt->timeout = 60;
1425 
1426  /* Choose a random starting offset within the first half of the
1427  * port range, to allow for a number of ports to try even if the offset
1428  * happens to be at the end of the random range. */
1429  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1430  /* even random offset */
1431  port_off -= port_off & 0x01;
1432 
1433  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1434  char transport[2048];
1435 
1436  /*
1437  * WMS serves all UDP data over a single connection, the RTX, which
1438  * isn't necessarily the first in the SDP but has to be the first
1439  * to be set up, else the second/third SETUP will fail with a 461.
1440  */
1441  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1442  rt->server_type == RTSP_SERVER_WMS) {
1443  if (i == 0) {
1444  /* rtx first */
1445  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1446  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1447  if (len >= 4 &&
1448  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1449  "/rtx"))
1450  break;
1451  }
1452  if (rtx == rt->nb_rtsp_streams)
1453  return -1; /* no RTX found */
1454  rtsp_st = rt->rtsp_streams[rtx];
1455  } else
1456  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1457  } else
1458  rtsp_st = rt->rtsp_streams[i];
1459 
1460  /* RTP/UDP */
1461  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1462  char buf[256];
1463 
1464  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1465  port = reply->transports[0].client_port_min;
1466  goto have_port;
1467  }
1468 
1469  /* first try in specified port range */
1470  while (j <= rt->rtp_port_max) {
1471  AVDictionary *opts = map_to_opts(rt);
1472 
1473  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1474  "?localport=%d", j);
1475  /* we will use two ports per rtp stream (rtp and rtcp) */
1476  j += 2;
1479 
1480  av_dict_free(&opts);
1481 
1482  if (!err)
1483  goto rtp_opened;
1484  }
1485  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1486  err = AVERROR(EIO);
1487  goto fail;
1488 
1489  rtp_opened:
1490  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1491  have_port:
1492  snprintf(transport, sizeof(transport) - 1,
1493  "%s/UDP;", trans_pref);
1494  if (rt->server_type != RTSP_SERVER_REAL)
1495  av_strlcat(transport, "unicast;", sizeof(transport));
1496  av_strlcatf(transport, sizeof(transport),
1497  "client_port=%d", port);
1498  if (rt->transport == RTSP_TRANSPORT_RTP &&
1499  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1500  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1501  }
1502 
1503  /* RTP/TCP */
1504  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1505  /* For WMS streams, the application streams are only used for
1506  * UDP. When trying to set it up for TCP streams, the server
1507  * will return an error. Therefore, we skip those streams. */
1508  if (rt->server_type == RTSP_SERVER_WMS &&
1509  (rtsp_st->stream_index < 0 ||
1510  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1512  continue;
1513  snprintf(transport, sizeof(transport) - 1,
1514  "%s/TCP;", trans_pref);
1515  if (rt->transport != RTSP_TRANSPORT_RDT)
1516  av_strlcat(transport, "unicast;", sizeof(transport));
1517  av_strlcatf(transport, sizeof(transport),
1518  "interleaved=%d-%d",
1519  interleave, interleave + 1);
1520  interleave += 2;
1521  }
1522 
1523  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1524  snprintf(transport, sizeof(transport) - 1,
1525  "%s/UDP;multicast", trans_pref);
1526  }
1527  if (s->oformat) {
1528  av_strlcat(transport, ";mode=record", sizeof(transport));
1529  } else if (rt->server_type == RTSP_SERVER_REAL ||
1531  av_strlcat(transport, ";mode=play", sizeof(transport));
1532  snprintf(cmd, sizeof(cmd),
1533  "Transport: %s\r\n",
1534  transport);
1535  if (rt->accept_dynamic_rate)
1536  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1537  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1538  char real_res[41], real_csum[9];
1539  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1540  real_challenge);
1541  av_strlcatf(cmd, sizeof(cmd),
1542  "If-Match: %s\r\n"
1543  "RealChallenge2: %s, sd=%s\r\n",
1544  rt->session_id, real_res, real_csum);
1545  }
1546  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1547  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1548  err = 1;
1549  goto fail;
1550  } else if (reply->status_code != RTSP_STATUS_OK ||
1551  reply->nb_transports != 1) {
1553  goto fail;
1554  }
1555 
1556  /* XXX: same protocol for all streams is required */
1557  if (i > 0) {
1558  if (reply->transports[0].lower_transport != rt->lower_transport ||
1559  reply->transports[0].transport != rt->transport) {
1560  err = AVERROR_INVALIDDATA;
1561  goto fail;
1562  }
1563  } else {
1564  rt->lower_transport = reply->transports[0].lower_transport;
1565  rt->transport = reply->transports[0].transport;
1566  }
1567 
1568  /* Fail if the server responded with another lower transport mode
1569  * than what we requested. */
1570  if (reply->transports[0].lower_transport != lower_transport) {
1571  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1572  err = AVERROR_INVALIDDATA;
1573  goto fail;
1574  }
1575 
1576  switch(reply->transports[0].lower_transport) {
1578  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1579  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1580  break;
1581 
1582  case RTSP_LOWER_TRANSPORT_UDP: {
1583  char url[1024], options[30] = "";
1584  const char *peer = host;
1585 
1586  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1587  av_strlcpy(options, "?connect=1", sizeof(options));
1588  /* Use source address if specified */
1589  if (reply->transports[0].source[0])
1590  peer = reply->transports[0].source;
1591  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1592  reply->transports[0].server_port_min, "%s", options);
1593  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1594  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1595  err = AVERROR_INVALIDDATA;
1596  goto fail;
1597  }
1598  break;
1599  }
1601  char url[1024], namebuf[50], optbuf[20] = "";
1602  struct sockaddr_storage addr;
1603  int port, ttl;
1604 
1605  if (reply->transports[0].destination.ss_family) {
1606  addr = reply->transports[0].destination;
1607  port = reply->transports[0].port_min;
1608  ttl = reply->transports[0].ttl;
1609  } else {
1610  addr = rtsp_st->sdp_ip;
1611  port = rtsp_st->sdp_port;
1612  ttl = rtsp_st->sdp_ttl;
1613  }
1614  if (ttl > 0)
1615  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1616  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1617  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1618  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1619  port, "%s", optbuf);
1622  err = AVERROR_INVALIDDATA;
1623  goto fail;
1624  }
1625  break;
1626  }
1627  }
1628 
1629  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1630  goto fail;
1631  }
1632 
1633  if (rt->nb_rtsp_streams && reply->timeout > 0)
1634  rt->timeout = reply->timeout;
1635 
1636  if (rt->server_type == RTSP_SERVER_REAL)
1637  rt->need_subscription = 1;
1638 
1639  return 0;
1640 
1641 fail:
1642  ff_rtsp_undo_setup(s, 0);
1643  return err;
1644 }
1645 
1647 {
1648  RTSPState *rt = s->priv_data;
1649  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1650  ffurl_close(rt->rtsp_hd);
1651  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1652 }
1653 
1655 {
1656  RTSPState *rt = s->priv_data;
1657  char proto[128], host[1024], path[1024];
1658  char tcpname[1024], cmd[2048], auth[128];
1659  const char *lower_rtsp_proto = "tcp";
1660  int port, err, tcp_fd;
1661  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1662  int lower_transport_mask = 0;
1663  int default_port = RTSP_DEFAULT_PORT;
1664  char real_challenge[64] = "";
1665  struct sockaddr_storage peer;
1666  socklen_t peer_len = sizeof(peer);
1667 
1668  if (rt->rtp_port_max < rt->rtp_port_min) {
1669  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1670  "than min port %d\n", rt->rtp_port_max,
1671  rt->rtp_port_min);
1672  return AVERROR(EINVAL);
1673  }
1674 
1675  if (!ff_network_init())
1676  return AVERROR(EIO);
1677 
1678  if (s->max_delay < 0) /* Not set by the caller */
1680 
1685  }
1686  /* Only pass through valid flags from here */
1688 
1689 redirect:
1690  /* extract hostname and port */
1691  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1692  host, sizeof(host), &port, path, sizeof(path), s->filename);
1693 
1694  if (!strcmp(proto, "rtsps")) {
1695  lower_rtsp_proto = "tls";
1696  default_port = RTSPS_DEFAULT_PORT;
1698  }
1699 
1700  if (*auth) {
1701  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1702  }
1703  if (port < 0)
1704  port = default_port;
1705 
1706  lower_transport_mask = rt->lower_transport_mask;
1707 
1708  if (!lower_transport_mask)
1709  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1710 
1711  if (s->oformat) {
1712  /* Only UDP or TCP - UDP multicast isn't supported. */
1713  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1714  (1 << RTSP_LOWER_TRANSPORT_TCP);
1715  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1716  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1717  "only UDP and TCP are supported for output.\n");
1718  err = AVERROR(EINVAL);
1719  goto fail;
1720  }
1721  }
1722 
1723  /* Construct the URI used in request; this is similar to s->filename,
1724  * but with authentication credentials removed and RTSP specific options
1725  * stripped out. */
1726  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1727  host, port, "%s", path);
1728 
1729  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1730  /* set up initial handshake for tunneling */
1731  char httpname[1024];
1732  char sessioncookie[17];
1733  char headers[1024];
1734 
1735  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1736  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1738 
1739  /* GET requests */
1740  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1741  &s->interrupt_callback) < 0) {
1742  err = AVERROR(EIO);
1743  goto fail;
1744  }
1745 
1746  /* generate GET headers */
1747  snprintf(headers, sizeof(headers),
1748  "x-sessioncookie: %s\r\n"
1749  "Accept: application/x-rtsp-tunnelled\r\n"
1750  "Pragma: no-cache\r\n"
1751  "Cache-Control: no-cache\r\n",
1752  sessioncookie);
1753  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1754 
1755  /* complete the connection */
1756  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1757  err = AVERROR(EIO);
1758  goto fail;
1759  }
1760 
1761  /* POST requests */
1762  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1763  &s->interrupt_callback) < 0 ) {
1764  err = AVERROR(EIO);
1765  goto fail;
1766  }
1767 
1768  /* generate POST headers */
1769  snprintf(headers, sizeof(headers),
1770  "x-sessioncookie: %s\r\n"
1771  "Content-Type: application/x-rtsp-tunnelled\r\n"
1772  "Pragma: no-cache\r\n"
1773  "Cache-Control: no-cache\r\n"
1774  "Content-Length: 32767\r\n"
1775  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1776  sessioncookie);
1777  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1778  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1779 
1780  /* Initialize the authentication state for the POST session. The HTTP
1781  * protocol implementation doesn't properly handle multi-pass
1782  * authentication for POST requests, since it would require one of
1783  * the following:
1784  * - implementing Expect: 100-continue, which many HTTP servers
1785  * don't support anyway, even less the RTSP servers that do HTTP
1786  * tunneling
1787  * - sending the whole POST data until getting a 401 reply specifying
1788  * what authentication method to use, then resending all that data
1789  * - waiting for potential 401 replies directly after sending the
1790  * POST header (waiting for some unspecified time)
1791  * Therefore, we copy the full auth state, which works for both basic
1792  * and digest. (For digest, we would have to synchronize the nonce
1793  * count variable between the two sessions, if we'd do more requests
1794  * with the original session, though.)
1795  */
1797 
1798  /* complete the connection */
1799  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1800  err = AVERROR(EIO);
1801  goto fail;
1802  }
1803  } else {
1804  int ret;
1805  /* open the tcp connection */
1806  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1807  host, port,
1808  "?timeout=%d", rt->stimeout);
1809  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1811  err = ret;
1812  goto fail;
1813  }
1814  rt->rtsp_hd_out = rt->rtsp_hd;
1815  }
1816  rt->seq = 0;
1817 
1818  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1819  if (tcp_fd < 0) {
1820  err = tcp_fd;
1821  goto fail;
1822  }
1823  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1824  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1825  NULL, 0, NI_NUMERICHOST);
1826  }
1827 
1828  /* request options supported by the server; this also detects server
1829  * type */
1830  for (rt->server_type = RTSP_SERVER_RTP;;) {
1831  cmd[0] = 0;
1832  if (rt->server_type == RTSP_SERVER_REAL)
1833  av_strlcat(cmd,
1834  /*
1835  * The following entries are required for proper
1836  * streaming from a Realmedia server. They are
1837  * interdependent in some way although we currently
1838  * don't quite understand how. Values were copied
1839  * from mplayer SVN r23589.
1840  * ClientChallenge is a 16-byte ID in hex
1841  * CompanyID is a 16-byte ID in base64
1842  */
1843  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1844  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1845  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1846  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1847  sizeof(cmd));
1848  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1849  if (reply->status_code != RTSP_STATUS_OK) {
1851  goto fail;
1852  }
1853 
1854  /* detect server type if not standard-compliant RTP */
1855  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1857  continue;
1858  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1860  } else if (rt->server_type == RTSP_SERVER_REAL)
1861  strcpy(real_challenge, reply->real_challenge);
1862  break;
1863  }
1864 
1865  if (CONFIG_RTSP_DEMUXER && s->iformat)
1866  err = ff_rtsp_setup_input_streams(s, reply);
1867  else if (CONFIG_RTSP_MUXER)
1868  err = ff_rtsp_setup_output_streams(s, host);
1869  else
1870  av_assert0(0);
1871  if (err)
1872  goto fail;
1873 
1874  do {
1875  int lower_transport = ff_log2_tab[lower_transport_mask &
1876  ~(lower_transport_mask - 1)];
1877 
1878  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1879  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1880  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1881 
1882  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1883  rt->server_type == RTSP_SERVER_REAL ?
1884  real_challenge : NULL);
1885  if (err < 0)
1886  goto fail;
1887  lower_transport_mask &= ~(1 << lower_transport);
1888  if (lower_transport_mask == 0 && err == 1) {
1889  err = AVERROR(EPROTONOSUPPORT);
1890  goto fail;
1891  }
1892  } while (err);
1893 
1894  rt->lower_transport_mask = lower_transport_mask;
1895  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1896  rt->state = RTSP_STATE_IDLE;
1897  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1898  return 0;
1899  fail:
1902  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1903  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1904  rt->session_id[0] = '\0';
1905  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1906  reply->status_code,
1907  s->filename);
1908  goto redirect;
1909  }
1910  ff_network_close();
1911  return err;
1912 }
1913 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1914 
1915 #if CONFIG_RTPDEC
1916 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1917  uint8_t *buf, int buf_size, int64_t wait_end)
1918 {
1919  RTSPState *rt = s->priv_data;
1920  RTSPStream *rtsp_st;
1921  int n, i, ret, tcp_fd, timeout_cnt = 0;
1922  int max_p = 0;
1923  struct pollfd *p = rt->p;
1924  int *fds = NULL, fdsnum, fdsidx;
1925 
1926  for (;;) {
1928  return AVERROR_EXIT;
1929  if (wait_end && wait_end - av_gettime_relative() < 0)
1930  return AVERROR(EAGAIN);
1931  max_p = 0;
1932  if (rt->rtsp_hd) {
1933  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1934  p[max_p].fd = tcp_fd;
1935  p[max_p++].events = POLLIN;
1936  } else {
1937  tcp_fd = -1;
1938  }
1939  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1940  rtsp_st = rt->rtsp_streams[i];
1941  if (rtsp_st->rtp_handle) {
1942  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1943  &fds, &fdsnum)) {
1944  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1945  return ret;
1946  }
1947  if (fdsnum != 2) {
1948  av_log(s, AV_LOG_ERROR,
1949  "Number of fds %d not supported\n", fdsnum);
1950  return AVERROR_INVALIDDATA;
1951  }
1952  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1953  p[max_p].fd = fds[fdsidx];
1954  p[max_p++].events = POLLIN;
1955  }
1956  av_freep(&fds);
1957  }
1958  }
1959  n = poll(p, max_p, POLL_TIMEOUT_MS);
1960  if (n > 0) {
1961  int j = 1 - (tcp_fd == -1);
1962  timeout_cnt = 0;
1963  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1964  rtsp_st = rt->rtsp_streams[i];
1965  if (rtsp_st->rtp_handle) {
1966  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1967  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1968  if (ret > 0) {
1969  *prtsp_st = rtsp_st;
1970  return ret;
1971  }
1972  }
1973  j+=2;
1974  }
1975  }
1976 #if CONFIG_RTSP_DEMUXER
1977  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1978  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1979  if (rt->state == RTSP_STATE_STREAMING) {
1981  return AVERROR_EOF;
1982  else
1984  "Unable to answer to TEARDOWN\n");
1985  } else
1986  return 0;
1987  } else {
1988  RTSPMessageHeader reply;
1989  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1990  if (ret < 0)
1991  return ret;
1992  /* XXX: parse message */
1993  if (rt->state != RTSP_STATE_STREAMING)
1994  return 0;
1995  }
1996  }
1997 #endif
1998  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1999  return AVERROR(ETIMEDOUT);
2000  } else if (n < 0 && errno != EINTR)
2001  return AVERROR(errno);
2002  }
2003 }
2004 
2005 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2006  const uint8_t *buf, int len)
2007 {
2008  RTSPState *rt = s->priv_data;
2009  int i;
2010  if (len < 0)
2011  return len;
2012  if (rt->nb_rtsp_streams == 1) {
2013  *rtsp_st = rt->rtsp_streams[0];
2014  return len;
2015  }
2016  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2017  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2018  int no_ssrc = 0;
2019  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2020  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2021  if (!rtpctx)
2022  continue;
2023  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2024  *rtsp_st = rt->rtsp_streams[i];
2025  return len;
2026  }
2027  if (!rtpctx->ssrc)
2028  no_ssrc = 1;
2029  }
2030  if (no_ssrc) {
2032  "Unable to pick stream for packet - SSRC not known for "
2033  "all streams\n");
2034  return AVERROR(EAGAIN);
2035  }
2036  } else {
2037  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2038  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2039  *rtsp_st = rt->rtsp_streams[i];
2040  return len;
2041  }
2042  }
2043  }
2044  }
2045  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2046  return AVERROR(EAGAIN);
2047 }
2048 
2050 {
2051  RTSPState *rt = s->priv_data;
2052  int ret, len;
2053  RTSPStream *rtsp_st, *first_queue_st = NULL;
2054  int64_t wait_end = 0;
2055 
2056  if (rt->nb_byes == rt->nb_rtsp_streams)
2057  return AVERROR_EOF;
2058 
2059  /* get next frames from the same RTP packet */
2060  if (rt->cur_transport_priv) {
2061  if (rt->transport == RTSP_TRANSPORT_RDT) {
2062  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2063  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2064  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2065  } else if (CONFIG_RTPDEC && rt->ts) {
2066  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2067  if (ret >= 0) {
2068  rt->recvbuf_pos += ret;
2069  ret = rt->recvbuf_pos < rt->recvbuf_len;
2070  }
2071  } else
2072  ret = -1;
2073  if (ret == 0) {
2074  rt->cur_transport_priv = NULL;
2075  return 0;
2076  } else if (ret == 1) {
2077  return 0;
2078  } else
2079  rt->cur_transport_priv = NULL;
2080  }
2081 
2082 redo:
2083  if (rt->transport == RTSP_TRANSPORT_RTP) {
2084  int i;
2085  int64_t first_queue_time = 0;
2086  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2087  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2088  int64_t queue_time;
2089  if (!rtpctx)
2090  continue;
2091  queue_time = ff_rtp_queued_packet_time(rtpctx);
2092  if (queue_time && (queue_time - first_queue_time < 0 ||
2093  !first_queue_time)) {
2094  first_queue_time = queue_time;
2095  first_queue_st = rt->rtsp_streams[i];
2096  }
2097  }
2098  if (first_queue_time) {
2099  wait_end = first_queue_time + s->max_delay;
2100  } else {
2101  wait_end = 0;
2102  first_queue_st = NULL;
2103  }
2104  }
2105 
2106  /* read next RTP packet */
2107  if (!rt->recvbuf) {
2109  if (!rt->recvbuf)
2110  return AVERROR(ENOMEM);
2111  }
2112 
2113  switch(rt->lower_transport) {
2114  default:
2115 #if CONFIG_RTSP_DEMUXER
2117  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2118  break;
2119 #endif
2122  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2123  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2124  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2125  break;
2127  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2128  wait_end && wait_end < av_gettime_relative())
2129  len = AVERROR(EAGAIN);
2130  else
2131  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2132  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2133  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2135  break;
2136  }
2137  if (len == AVERROR(EAGAIN) && first_queue_st &&
2138  rt->transport == RTSP_TRANSPORT_RTP) {
2140  "max delay reached. need to consume packet\n");
2141  rtsp_st = first_queue_st;
2142  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2143  goto end;
2144  }
2145  if (len < 0)
2146  return len;
2147  if (len == 0)
2148  return AVERROR_EOF;
2149  if (rt->transport == RTSP_TRANSPORT_RDT) {
2150  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2151  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2152  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2153  if (rtsp_st->feedback) {
2154  AVIOContext *pb = NULL;
2156  pb = s->pb;
2157  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2158  }
2159  if (ret < 0) {
2160  /* Either bad packet, or a RTCP packet. Check if the
2161  * first_rtcp_ntp_time field was initialized. */
2162  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2163  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2164  /* first_rtcp_ntp_time has been initialized for this stream,
2165  * copy the same value to all other uninitialized streams,
2166  * in order to map their timestamp origin to the same ntp time
2167  * as this one. */
2168  int i;
2169  AVStream *st = NULL;
2170  if (rtsp_st->stream_index >= 0)
2171  st = s->streams[rtsp_st->stream_index];
2172  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2173  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2174  AVStream *st2 = NULL;
2175  if (rt->rtsp_streams[i]->stream_index >= 0)
2176  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2177  if (rtpctx2 && st && st2 &&
2178  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2179  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2180  rtpctx2->rtcp_ts_offset = av_rescale_q(
2181  rtpctx->rtcp_ts_offset, st->time_base,
2182  st2->time_base);
2183  }
2184  }
2185  // Make real NTP start time available in AVFormatContext
2186  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2187  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2188  if (rtpctx->st) {
2189  s->start_time_realtime -=
2190  av_rescale (rtpctx->rtcp_ts_offset,
2191  (uint64_t) rtpctx->st->time_base.num * 1000000,
2192  rtpctx->st->time_base.den);
2193  }
2194  }
2195  }
2196  if (ret == -RTCP_BYE) {
2197  rt->nb_byes++;
2198 
2199  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2200  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2201 
2202  if (rt->nb_byes == rt->nb_rtsp_streams)
2203  return AVERROR_EOF;
2204  }
2205  }
2206  } else if (CONFIG_RTPDEC && rt->ts) {
2207  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2208  if (ret >= 0) {
2209  if (ret < len) {
2210  rt->recvbuf_len = len;
2211  rt->recvbuf_pos = ret;
2212  rt->cur_transport_priv = rt->ts;
2213  return 1;
2214  } else {
2215  ret = 0;
2216  }
2217  }
2218  } else {
2219  return AVERROR_INVALIDDATA;
2220  }
2221 end:
2222  if (ret < 0)
2223  goto redo;
2224  if (ret == 1)
2225  /* more packets may follow, so we save the RTP context */
2226  rt->cur_transport_priv = rtsp_st->transport_priv;
2227 
2228  return ret;
2229 }
2230 #endif /* CONFIG_RTPDEC */
2231 
2232 #if CONFIG_SDP_DEMUXER
2233 static int sdp_probe(AVProbeData *p1)
2234 {
2235  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2236 
2237  /* we look for a line beginning "c=IN IP" */
2238  while (p < p_end && *p != '\0') {
2239  if (sizeof("c=IN IP") - 1 < p_end - p &&
2240  av_strstart(p, "c=IN IP", NULL))
2241  return AVPROBE_SCORE_EXTENSION;
2242 
2243  while (p < p_end - 1 && *p != '\n') p++;
2244  if (++p >= p_end)
2245  break;
2246  if (*p == '\r')
2247  p++;
2248  }
2249  return 0;
2250 }
2251 
2252 static void append_source_addrs(char *buf, int size, const char *name,
2253  int count, struct RTSPSource **addrs)
2254 {
2255  int i;
2256  if (!count)
2257  return;
2258  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2259  for (i = 1; i < count; i++)
2260  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2261 }
2262 
2263 static int sdp_read_header(AVFormatContext *s)
2264 {
2265  RTSPState *rt = s->priv_data;
2266  RTSPStream *rtsp_st;
2267  int size, i, err;
2268  char *content;
2269  char url[1024];
2270 
2271  if (!ff_network_init())
2272  return AVERROR(EIO);
2273 
2274  if (s->max_delay < 0) /* Not set by the caller */
2276  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2278 
2279  /* read the whole sdp file */
2280  /* XXX: better loading */
2281  content = av_malloc(SDP_MAX_SIZE);
2282  if (!content)
2283  return AVERROR(ENOMEM);
2284  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2285  if (size <= 0) {
2286  av_free(content);
2287  return AVERROR_INVALIDDATA;
2288  }
2289  content[size] ='\0';
2290 
2291  err = ff_sdp_parse(s, content);
2292  av_freep(&content);
2293  if (err) goto fail;
2294 
2295  /* open each RTP stream */
2296  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2297  char namebuf[50];
2298  rtsp_st = rt->rtsp_streams[i];
2299 
2300  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2301  AVDictionary *opts = map_to_opts(rt);
2302 
2303  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2304  sizeof(rtsp_st->sdp_ip),
2305  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2306  if (err) {
2307  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2308  err = AVERROR(EIO);
2309  av_dict_free(&opts);
2310  goto fail;
2311  }
2312  ff_url_join(url, sizeof(url), "rtp", NULL,
2313  namebuf, rtsp_st->sdp_port,
2314  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2315  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2316  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2317  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2318 
2319  append_source_addrs(url, sizeof(url), "sources",
2320  rtsp_st->nb_include_source_addrs,
2321  rtsp_st->include_source_addrs);
2322  append_source_addrs(url, sizeof(url), "block",
2323  rtsp_st->nb_exclude_source_addrs,
2324  rtsp_st->exclude_source_addrs);
2325  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2327 
2328  av_dict_free(&opts);
2329 
2330  if (err < 0) {
2331  err = AVERROR_INVALIDDATA;
2332  goto fail;
2333  }
2334  }
2335  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2336  goto fail;
2337  }
2338  return 0;
2339 fail:
2341  ff_network_close();
2342  return err;
2343 }
2344 
2345 static int sdp_read_close(AVFormatContext *s)
2346 {
2348  ff_network_close();
2349  return 0;
2350 }
2351 
2352 static const AVClass sdp_demuxer_class = {
2353  .class_name = "SDP demuxer",
2354  .item_name = av_default_item_name,
2355  .option = sdp_options,
2356  .version = LIBAVUTIL_VERSION_INT,
2357 };
2358 
2359 AVInputFormat ff_sdp_demuxer = {
2360  .name = "sdp",
2361  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2362  .priv_data_size = sizeof(RTSPState),
2363  .read_probe = sdp_probe,
2364  .read_header = sdp_read_header,
2366  .read_close = sdp_read_close,
2367  .priv_class = &sdp_demuxer_class,
2368 };
2369 #endif /* CONFIG_SDP_DEMUXER */
2370 
2371 #if CONFIG_RTP_DEMUXER
2372 static int rtp_probe(AVProbeData *p)
2373 {
2374  if (av_strstart(p->filename, "rtp:", NULL))
2375  return AVPROBE_SCORE_MAX;
2376  return 0;
2377 }
2378 
2379 static int rtp_read_header(AVFormatContext *s)
2380 {
2381  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2382  char host[500], sdp[500];
2383  int ret, port;
2384  URLContext* in = NULL;
2385  int payload_type;
2386  AVCodecParameters *par = NULL;
2387  struct sockaddr_storage addr;
2388  AVIOContext pb;
2389  socklen_t addrlen = sizeof(addr);
2390  RTSPState *rt = s->priv_data;
2391 
2392  if (!ff_network_init())
2393  return AVERROR(EIO);
2394 
2397  if (ret)
2398  goto fail;
2399 
2400  while (1) {
2401  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2402  if (ret == AVERROR(EAGAIN))
2403  continue;
2404  if (ret < 0)
2405  goto fail;
2406  if (ret < 12) {
2407  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2408  continue;
2409  }
2410 
2411  if ((recvbuf[0] & 0xc0) != 0x80) {
2412  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2413  "received\n");
2414  continue;
2415  }
2416 
2417  if (RTP_PT_IS_RTCP(recvbuf[1]))
2418  continue;
2419 
2420  payload_type = recvbuf[1] & 0x7f;
2421  break;
2422  }
2423  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2424  ffurl_close(in);
2425  in = NULL;
2426 
2427  par = avcodec_parameters_alloc();
2428  if (!par) {
2429  ret = AVERROR(ENOMEM);
2430  goto fail;
2431  }
2432 
2433  if (ff_rtp_get_codec_info(par, payload_type)) {
2434  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2435  "without an SDP file describing it\n",
2436  payload_type);
2437  goto fail;
2438  }
2439  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2440  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2441  "properly you need an SDP file "
2442  "describing it\n");
2443  }
2444 
2445  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2446  NULL, 0, s->filename);
2447 
2448  snprintf(sdp, sizeof(sdp),
2449  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2450  addr.ss_family == AF_INET ? 4 : 6, host,
2451  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2452  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2453  port, payload_type);
2454  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2456 
2457  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2458  s->pb = &pb;
2459 
2460  /* sdp_read_header initializes this again */
2461  ff_network_close();
2462 
2463  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2464 
2465  ret = sdp_read_header(s);
2466  s->pb = NULL;
2467  return ret;
2468 
2469 fail:
2471  if (in)
2472  ffurl_close(in);
2473  ff_network_close();
2474  return ret;
2475 }
2476 
2477 static const AVClass rtp_demuxer_class = {
2478  .class_name = "RTP demuxer",
2479  .item_name = av_default_item_name,
2480  .option = rtp_options,
2481  .version = LIBAVUTIL_VERSION_INT,
2482 };
2483 
2484 AVInputFormat ff_rtp_demuxer = {
2485  .name = "rtp",
2486  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2487  .priv_data_size = sizeof(RTSPState),
2488  .read_probe = rtp_probe,
2489  .read_header = rtp_read_header,
2491  .read_close = sdp_read_close,
2492  .flags = AVFMT_NOFILE,
2493  .priv_class = &rtp_demuxer_class,
2494 };
2495 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:273
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:93
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:4307
char crypto_suite[40]
Definition: rtsp.h:475
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:387
#define NULL
Definition: coverity.c:32
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:100
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:552
const char * s
Definition: avisynth_c.h:631
Bytestream IO Context.
Definition: avio.h:147
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:573
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1552
int ffurl_open_whitelist(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options, const char *whitelist, const char *blacklist, URLContext *parent)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
Definition: avio.c:309
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1577
AVOption.
Definition: opt.h:245
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:115
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:187
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:70
const char * filename
Definition: avformat.h:460
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:165
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:317
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:4427
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:559
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:421
const char * desc
Definition: nvenc.c:89
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
Windows Media server.
Definition: rtsp.h:209
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:354
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:796
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2830
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:166
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:131
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3922
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 num
numerator
Definition: rational.h:44
int index
stream index in AVFormatContext
Definition: avformat.h:877
#define AVIO_FLAG_READ
read-only
Definition: avio.h:606
char * user_agent
User-Agent string.
Definition: rtsp.h:407
char location[4096]
the "Location:" field.
Definition: rtsp.h:152
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:607
int mode_record
transport set to record data
Definition: rtsp.h:112
enum AVMediaType codec_type
Definition: rtp.c:37
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:223
void ff_network_close(void)
Definition: network.c:102
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:420
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:79
#define freeaddrinfo
Definition: network.h:208
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:452
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1374
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:418
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:245
int auth_type
The currently chosen auth type.
Definition: httpauth.h:59
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:239
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:132
#define AI_NUMERICHOST
Definition: network.h:177
This struct describes the properties of an encoded stream.
Definition: avcodec.h:3914
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:121
This describes the server response to each RTSP command.
Definition: rtsp.h:127
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:509
#define RECVBUF_SIZE
Definition: rtsp.c:59
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:142
Format I/O context.
Definition: avformat.h:1325
#define RTP_PT_PRIVATE
Definition: rtp.h:77
#define COMMON_OPTS()
Definition: rtsp.c: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:143
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:207
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:402
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:72
#define RTSP_FLAG_PREFER_TCP
Try RTP via TCP first if possible.
Definition: rtsp.h:423
int recvbuf_len
Definition: rtsp.h:323
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:180
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:45
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:359
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:148
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:74
#define av_malloc(s)
Opaque data information usually continuous.
Definition: avutil.h:195
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:109
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:126
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:187
int ff_network_init(void)
Definition: network.c:55
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1278
AVOptions.
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: utils.c:4038
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:470
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
uint16_t ss_family
Definition: network.h:106
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int id
Format-specific stream ID.
Definition: avformat.h:883
enum AVStreamParseType need_parsing
Definition: avformat.h:1074
#define POLL_TIMEOUT_MS
Definition: rtsp.c:55
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:60
static void handler(vbi_event *ev, void *user_data)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4065
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1393
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:372
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:71
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:435
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
char * protocol_whitelist
',' separated list of allowed protocols.
Definition: avformat.h:1861
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
#define AVERROR_EOF
End of file.
Definition: error.h:55
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:166
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
ptrdiff_t size
Definition: opengl_enc.c:101
static const uint8_t header[24]
Definition: sdr2.c:67
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:465
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.
Normal RTSP.
Definition: rtsp.h:68
const OptionDef options[]
Definition: ffserver.c:3969
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
#define av_log(a,...)
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:134
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:598
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1344
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:177
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:77
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:142
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:455
Private data for the RTSP demuxer.
Definition: rtsp.h:218
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:255
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:292
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1539
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:633
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
av_default_item_name
#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:176
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:202
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3918
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:182
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:220
simple assert() macros that are a bit more flexible than ISO C assert().
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:331
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:453
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:667
GLsizei count
Definition: opengl_enc.c:109
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:138
void avcodec_parameters_free(AVCodecParameters **par)
Free an AVCodecParameters instance and everything associated with it and write NULL to the supplied p...
Definition: utils.c:4048
int64_t rtcp_ts_offset
Definition: rtpdec.h:182
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:83
#define fail()
Definition: checkasm.h:81
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:132
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:164
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:2956
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:440
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:462
int seq
RTSP command sequence number.
Definition: rtsp.h:241
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:461
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:339
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1381
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:419
AVDictionary * opts
Definition: movenc.c:50
#define NI_NUMERICHOST
Definition: network.h:185
#define th
Definition: regdef.h:75
#define LIBAVFORMAT_IDENT
Definition: version.h:46
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:307
int recvbuf_pos
Definition: rtsp.h:322
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:168
char filename[1024]
input or output filename
Definition: avformat.h:1401
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes to a null-terminated string.
Definition: base64.h:66
#define FFMIN(a, b)
Definition: common.h:96
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:283
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:129
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:172
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:75
enum AVStreamParseType need_parsing
Definition: rtpdec.h:119
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:88
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:130
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:140
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:156
int n
Definition: avisynth_c.h:547
AVDictionary * metadata
Definition: avformat.h:945
char crypto_params[100]
Definition: rtsp.h:476
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:192
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:128
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:626
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:64
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:450
Raw data (over UDP)
Definition: rtsp.h:59
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:321
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:23
int sdp_payload_type
payload type
Definition: rtsp.h:457
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:545
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:454
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1300
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:638
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:441
Stream structure.
Definition: avformat.h:876
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
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:336
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:426
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:451
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:726
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:655
int rtp_port_max
Definition: rtsp.h:387
#define NTP_OFFSET
Definition: internal.h:206
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1367
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:382
int server_port_max
Definition: rtsp.h:105
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:66
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
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:473
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=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);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_AARCH64) ff_audio_convert_init_aarch64(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_log(ac->avr, AV_LOG_TRACE,"%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
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:413
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:258
void * buf
Definition: avisynth_c.h:553
Definition: url.h:38
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:73
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
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:608
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:69
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:377
int client_port_max
Definition: rtsp.h:101
Describe the class of an AVClass context structure.
Definition: log.h:67
#define SDP_MAX_SIZE
Definition: rtsp.c:58
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:515
#define SPACE_CHARS
Definition: internal.h:272
void * priv_data
Definition: url.h:41
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:466
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:279
#define gai_strerror
Definition: network.h:215
not initialized
Definition: rtsp.h:196
int64_t range_end
Definition: rtsp.h:138
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:118
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:2849
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:155
AVMediaType
Definition: avutil.h:191
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:70
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:726
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:752
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:760
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:270
#define s1
Definition: regdef.h:38
const char * name
Name of the codec described by this descriptor.
Definition: avcodec.h:665
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:469
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4001
int buffer_size
Definition: rtsp.h:410
This structure contains the data a format has to probe a file.
Definition: avformat.h:459
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:76
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
static void interleave(short *output, short **input, int channels, int samples)
Definition: resample.c:161
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:93
int interleaved_max
Definition: rtsp.h:93
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
mfxU16 profile
Definition: qsvenc.c:42
This struct describes the properties of a single codec described by an AVCodecID. ...
Definition: avcodec.h:657
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:267
static int flags
Definition: cpu.c:47
int ffurl_close(URLContext *h)
Definition: avio.c:467
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:138
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1410
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
int sample_rate
Audio only.
Definition: avcodec.h:4032
#define DEC
Definition: rtsp.c:63
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:471
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:34
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:207
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:103
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:28
uint32_t ssrc
Definition: rtpdec.h:153
static AVDictionary * map_to_opts(RTSPState *rt)
Definition: rtsp.c:119
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:476
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:119
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:81
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:288
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:463
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:414
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
int den
denominator
Definition: rational.h:45
char default_lang[4]
Definition: rtsp.h:409
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1337
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4037
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:156
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...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:397
#define getnameinfo
Definition: network.h:209
#define av_free(p)
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:149
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:78
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:446
void * priv_data
Format private data.
Definition: avformat.h:1353
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:594
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:853
int channels
Audio only.
Definition: avcodec.h:4028
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:456
#define MAX_TIMEOUTS
Definition: rtsp.c:57
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1184
char * protocol_blacklist
',' separated list of disallowed protocols.
Definition: avformat.h:1896
int ai_flags
Definition: network.h:128
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1420
Realmedia-style server.
Definition: rtsp.h:208
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:344
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:553
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:660
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:114
AVCodecParameters * codecpar
Definition: avformat.h:1006
#define av_malloc_array(a, b)
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:913
int interleaved_max
Definition: rtsp.h:444
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:840
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:114
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:98
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2874
AVStream * st
Definition: rtpdec.h:151
#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:444
This structure stores compressed data.
Definition: avcodec.h:1557
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
Definition: aviobuf.c:1086
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:105
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:431
static const AVOption rtp_options[]
Definition: rtsp.c:112
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:407
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:252
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:436
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240
#define OFFSET(x)
Definition: rtsp.c:62
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:97
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:437
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:101
const char * name
Definition: opengl_enc.c:103