FFmpeg  4.3.6
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/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
33 #include "avformat.h"
34 #include "avio_internal.h"
35 
36 #if HAVE_POLL_H
37 #include <poll.h>
38 #endif
39 #include "internal.h"
40 #include "network.h"
41 #include "os_support.h"
42 #include "http.h"
43 #include "rtsp.h"
44 
45 #include "rtpdec.h"
46 #include "rtpproto.h"
47 #include "rdt.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
50 #include "url.h"
51 #include "rtpenc.h"
52 #include "mpegts.h"
53 
54 /* Timeout values for socket poll, in ms,
55  * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
62 
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 
67 #define RTSP_FLAG_OPTS(name, longname) \
68  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69  { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
70 
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72  { 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" }, \
73  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
76  { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
77 
78 #define COMMON_OPTS() \
79  { "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 }, \
80  { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
81  { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
82 
83 
85  { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
86  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
87  { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
88  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
89  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
90  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
91  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
92  { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
93  RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
94  { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
95  { "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" },
96  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
97  { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
98  { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
99  { "listen_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 },
100 #if FF_API_OLD_RTSP_OPTIONS
101  { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
102  { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
103 #else
104  { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
105 #endif
106  COMMON_OPTS(),
107  { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
108 #if FF_API_OLD_RTSP_OPTIONS
109  { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
110 #endif
111  { NULL },
112 };
113 
114 static const AVOption sdp_options[] = {
115  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
116  { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
117  { "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" },
118  RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
119  COMMON_OPTS(),
120  { NULL },
121 };
122 
123 static const AVOption rtp_options[] = {
124  RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
125  COMMON_OPTS(),
126  { NULL },
127 };
128 
129 
131 {
133  char buf[256];
134 
135  snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
136  av_dict_set(&opts, "buffer_size", buf, 0);
137  snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
138  av_dict_set(&opts, "pkt_size", buf, 0);
139 
140  return opts;
141 }
142 
143 static void get_word_until_chars(char *buf, int buf_size,
144  const char *sep, const char **pp)
145 {
146  const char *p;
147  char *q;
148 
149  p = *pp;
150  p += strspn(p, SPACE_CHARS);
151  q = buf;
152  while (!strchr(sep, *p) && *p != '\0') {
153  if ((q - buf) < buf_size - 1)
154  *q++ = *p;
155  p++;
156  }
157  if (buf_size > 0)
158  *q = '\0';
159  *pp = p;
160 }
161 
162 static void get_word_sep(char *buf, int buf_size, const char *sep,
163  const char **pp)
164 {
165  if (**pp == '/') (*pp)++;
166  get_word_until_chars(buf, buf_size, sep, pp);
167 }
168 
169 static void get_word(char *buf, int buf_size, const char **pp)
170 {
171  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
172 }
173 
174 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
175  * and end time.
176  * Used for seeking in the rtp stream.
177  */
178 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
179 {
180  char buf[256];
181 
182  p += strspn(p, SPACE_CHARS);
183  if (!av_stristart(p, "npt=", &p))
184  return;
185 
186  *start = AV_NOPTS_VALUE;
187  *end = AV_NOPTS_VALUE;
188 
189  get_word_sep(buf, sizeof(buf), "-", &p);
190  if (av_parse_time(start, buf, 1) < 0)
191  return;
192  if (*p == '-') {
193  p++;
194  get_word_sep(buf, sizeof(buf), "-", &p);
195  if (av_parse_time(end, buf, 1) < 0)
196  av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
197  }
198 }
199 
201  const char *buf, struct sockaddr_storage *sock)
202 {
203  struct addrinfo hints = { 0 }, *ai = NULL;
204  int ret;
205 
206  hints.ai_flags = AI_NUMERICHOST;
207  if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
208  av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
209  buf,
210  gai_strerror(ret));
211  return -1;
212  }
213  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
214  freeaddrinfo(ai);
215  return 0;
216 }
217 
218 #if CONFIG_RTPDEC
219 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
220  RTSPStream *rtsp_st, AVStream *st)
221 {
222  AVCodecParameters *par = st ? st->codecpar : NULL;
223  if (!handler)
224  return;
225  if (par)
226  par->codec_id = handler->codec_id;
227  rtsp_st->dynamic_handler = handler;
228  if (st)
229  st->need_parsing = handler->need_parsing;
230  if (handler->priv_data_size) {
232  if (!rtsp_st->dynamic_protocol_context)
233  rtsp_st->dynamic_handler = NULL;
234  }
235 }
236 
237 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
238  AVStream *st)
239 {
240  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
241  int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
242  rtsp_st->dynamic_protocol_context);
243  if (ret < 0) {
244  if (rtsp_st->dynamic_protocol_context) {
245  if (rtsp_st->dynamic_handler->close)
246  rtsp_st->dynamic_handler->close(
247  rtsp_st->dynamic_protocol_context);
249  }
250  rtsp_st->dynamic_protocol_context = NULL;
251  rtsp_st->dynamic_handler = NULL;
252  }
253  }
254 }
255 
256 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
257 static int sdp_parse_rtpmap(AVFormatContext *s,
258  AVStream *st, RTSPStream *rtsp_st,
259  int payload_type, const char *p)
260 {
261  AVCodecParameters *par = st->codecpar;
262  char buf[256];
263  int i;
264  const AVCodecDescriptor *desc;
265  const char *c_name;
266 
267  /* See if we can handle this kind of payload.
268  * The space should normally not be there but some Real streams or
269  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
270  * have a trailing space. */
271  get_word_sep(buf, sizeof(buf), "/ ", &p);
272  if (payload_type < RTP_PT_PRIVATE) {
273  /* We are in a standard case
274  * (from http://www.iana.org/assignments/rtp-parameters). */
275  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
276  }
277 
278  if (par->codec_id == AV_CODEC_ID_NONE) {
279  const RTPDynamicProtocolHandler *handler =
281  init_rtp_handler(handler, rtsp_st, st);
282  /* If no dynamic handler was found, check with the list of standard
283  * allocated types, if such a stream for some reason happens to
284  * use a private payload type. This isn't handled in rtpdec.c, since
285  * the format name from the rtpmap line never is passed into rtpdec. */
286  if (!rtsp_st->dynamic_handler)
287  par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
288  }
289 
290  desc = avcodec_descriptor_get(par->codec_id);
291  if (desc && desc->name)
292  c_name = desc->name;
293  else
294  c_name = "(null)";
295 
296  get_word_sep(buf, sizeof(buf), "/", &p);
297  i = atoi(buf);
298  switch (par->codec_type) {
299  case AVMEDIA_TYPE_AUDIO:
300  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
303  if (i > 0) {
304  par->sample_rate = i;
305  avpriv_set_pts_info(st, 32, 1, par->sample_rate);
306  get_word_sep(buf, sizeof(buf), "/", &p);
307  i = atoi(buf);
308  if (i > 0)
309  par->channels = i;
310  }
311  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
312  par->sample_rate);
313  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
314  par->channels);
315  break;
316  case AVMEDIA_TYPE_VIDEO:
317  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
318  if (i > 0)
319  avpriv_set_pts_info(st, 32, 1, i);
320  break;
321  default:
322  break;
323  }
324  finalize_rtp_handler_init(s, rtsp_st, st);
325  return 0;
326 }
327 
328 /* parse the attribute line from the fmtp a line of an sdp response. This
329  * is broken out as a function because it is used in rtp_h264.c, which is
330  * forthcoming. */
331 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
332  char *value, int value_size)
333 {
334  *p += strspn(*p, SPACE_CHARS);
335  if (**p) {
336  get_word_sep(attr, attr_size, "=", p);
337  if (**p == '=')
338  (*p)++;
339  get_word_sep(value, value_size, ";", p);
340  if (**p == ';')
341  (*p)++;
342  return 1;
343  }
344  return 0;
345 }
346 
347 typedef struct SDPParseState {
348  /* SDP only */
349  struct sockaddr_storage default_ip;
350  int default_ttl;
351  int skip_media; ///< set if an unknown m= line occurs
352  int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
353  struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
354  int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
355  struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
356  int seen_rtpmap;
357  int seen_fmtp;
358  char delayed_fmtp[2048];
359 } SDPParseState;
360 
361 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
362  struct RTSPSource ***dest, int *dest_count)
363 {
364  RTSPSource *rtsp_src, *rtsp_src2;
365  int i;
366  for (i = 0; i < count; i++) {
367  rtsp_src = addrs[i];
368  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
369  if (!rtsp_src2)
370  continue;
371  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
372  dynarray_add(dest, dest_count, rtsp_src2);
373  }
374 }
375 
376 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
377  int payload_type, const char *line)
378 {
379  int i;
380 
381  for (i = 0; i < rt->nb_rtsp_streams; i++) {
382  RTSPStream *rtsp_st = rt->rtsp_streams[i];
383  if (rtsp_st->sdp_payload_type == payload_type &&
384  rtsp_st->dynamic_handler &&
385  rtsp_st->dynamic_handler->parse_sdp_a_line) {
386  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
387  rtsp_st->dynamic_protocol_context, line);
388  }
389  }
390 }
391 
392 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
393  int letter, const char *buf)
394 {
395  RTSPState *rt = s->priv_data;
396  char buf1[64], st_type[64];
397  const char *p;
398  enum AVMediaType codec_type;
399  int payload_type;
400  AVStream *st;
401  RTSPStream *rtsp_st;
402  RTSPSource *rtsp_src;
403  struct sockaddr_storage sdp_ip;
404  int ttl;
405 
406  av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
407 
408  p = buf;
409  if (s1->skip_media && letter != 'm')
410  return;
411  switch (letter) {
412  case 'c':
413  get_word(buf1, sizeof(buf1), &p);
414  if (strcmp(buf1, "IN") != 0)
415  return;
416  get_word(buf1, sizeof(buf1), &p);
417  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
418  return;
419  get_word_sep(buf1, sizeof(buf1), "/", &p);
420  if (get_sockaddr(s, buf1, &sdp_ip))
421  return;
422  ttl = 16;
423  if (*p == '/') {
424  p++;
425  get_word_sep(buf1, sizeof(buf1), "/", &p);
426  ttl = atoi(buf1);
427  }
428  if (s->nb_streams == 0) {
429  s1->default_ip = sdp_ip;
430  s1->default_ttl = ttl;
431  } else {
432  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433  rtsp_st->sdp_ip = sdp_ip;
434  rtsp_st->sdp_ttl = ttl;
435  }
436  break;
437  case 's':
438  av_dict_set(&s->metadata, "title", p, 0);
439  break;
440  case 'i':
441  if (s->nb_streams == 0) {
442  av_dict_set(&s->metadata, "comment", p, 0);
443  break;
444  }
445  break;
446  case 'm':
447  /* new stream */
448  s1->skip_media = 0;
449  s1->seen_fmtp = 0;
450  s1->seen_rtpmap = 0;
451  codec_type = AVMEDIA_TYPE_UNKNOWN;
452  get_word(st_type, sizeof(st_type), &p);
453  if (!strcmp(st_type, "audio")) {
454  codec_type = AVMEDIA_TYPE_AUDIO;
455  } else if (!strcmp(st_type, "video")) {
456  codec_type = AVMEDIA_TYPE_VIDEO;
457  } else if (!strcmp(st_type, "application")) {
458  codec_type = AVMEDIA_TYPE_DATA;
459  } else if (!strcmp(st_type, "text")) {
460  codec_type = AVMEDIA_TYPE_SUBTITLE;
461  }
462  if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
463  !(rt->media_type_mask & (1 << codec_type)) ||
464  rt->nb_rtsp_streams >= s->max_streams
465  ) {
466  s1->skip_media = 1;
467  return;
468  }
469  rtsp_st = av_mallocz(sizeof(RTSPStream));
470  if (!rtsp_st)
471  return;
472  rtsp_st->stream_index = -1;
473  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
474 
475  rtsp_st->sdp_ip = s1->default_ip;
476  rtsp_st->sdp_ttl = s1->default_ttl;
477 
478  copy_default_source_addrs(s1->default_include_source_addrs,
479  s1->nb_default_include_source_addrs,
480  &rtsp_st->include_source_addrs,
481  &rtsp_st->nb_include_source_addrs);
482  copy_default_source_addrs(s1->default_exclude_source_addrs,
483  s1->nb_default_exclude_source_addrs,
484  &rtsp_st->exclude_source_addrs,
485  &rtsp_st->nb_exclude_source_addrs);
486 
487  get_word(buf1, sizeof(buf1), &p); /* port */
488  rtsp_st->sdp_port = atoi(buf1);
489 
490  get_word(buf1, sizeof(buf1), &p); /* protocol */
491  if (!strcmp(buf1, "udp"))
493  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
494  rtsp_st->feedback = 1;
495 
496  /* XXX: handle list of formats */
497  get_word(buf1, sizeof(buf1), &p); /* format list */
498  rtsp_st->sdp_payload_type = atoi(buf1);
499 
500  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
501  /* no corresponding stream */
502  if (rt->transport == RTSP_TRANSPORT_RAW) {
503  if (CONFIG_RTPDEC && !rt->ts)
504  rt->ts = avpriv_mpegts_parse_open(s);
505  } else {
507  handler = ff_rtp_handler_find_by_id(
509  init_rtp_handler(handler, rtsp_st, NULL);
510  finalize_rtp_handler_init(s, rtsp_st, NULL);
511  }
512  } else if (rt->server_type == RTSP_SERVER_WMS &&
513  codec_type == AVMEDIA_TYPE_DATA) {
514  /* RTX stream, a stream that carries all the other actual
515  * audio/video streams. Don't expose this to the callers. */
516  } else {
517  st = avformat_new_stream(s, NULL);
518  if (!st)
519  return;
520  st->id = rt->nb_rtsp_streams - 1;
521  rtsp_st->stream_index = st->index;
523  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
525  /* if standard payload type, we can find the codec right now */
527  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
528  st->codecpar->sample_rate > 0)
529  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
530  /* Even static payload types may need a custom depacketizer */
531  handler = ff_rtp_handler_find_by_id(
532  rtsp_st->sdp_payload_type, st->codecpar->codec_type);
533  init_rtp_handler(handler, rtsp_st, st);
534  finalize_rtp_handler_init(s, rtsp_st, st);
535  }
536  if (rt->default_lang[0])
537  av_dict_set(&st->metadata, "language", rt->default_lang, 0);
538  }
539  /* put a default control url */
540  av_strlcpy(rtsp_st->control_url, rt->control_uri,
541  sizeof(rtsp_st->control_url));
542  break;
543  case 'a':
544  if (av_strstart(p, "control:", &p)) {
545  if (s->nb_streams == 0) {
546  if (!strncmp(p, "rtsp://", 7))
547  av_strlcpy(rt->control_uri, p,
548  sizeof(rt->control_uri));
549  } else {
550  char proto[32];
551  /* get the control url */
552  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
553 
554  /* XXX: may need to add full url resolution */
555  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
556  NULL, NULL, 0, p);
557  if (proto[0] == '\0') {
558  /* relative control URL */
559  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
560  av_strlcat(rtsp_st->control_url, "/",
561  sizeof(rtsp_st->control_url));
562  av_strlcat(rtsp_st->control_url, p,
563  sizeof(rtsp_st->control_url));
564  } else
565  av_strlcpy(rtsp_st->control_url, p,
566  sizeof(rtsp_st->control_url));
567  }
568  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
569  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
570  get_word(buf1, sizeof(buf1), &p);
571  payload_type = atoi(buf1);
572  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
573  if (rtsp_st->stream_index >= 0) {
574  st = s->streams[rtsp_st->stream_index];
575  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
576  }
577  s1->seen_rtpmap = 1;
578  if (s1->seen_fmtp) {
579  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
580  }
581  } else if (av_strstart(p, "fmtp:", &p) ||
582  av_strstart(p, "framesize:", &p)) {
583  // let dynamic protocol handlers have a stab at the line.
584  get_word(buf1, sizeof(buf1), &p);
585  payload_type = atoi(buf1);
586  if (s1->seen_rtpmap) {
587  parse_fmtp(s, rt, payload_type, buf);
588  } else {
589  s1->seen_fmtp = 1;
590  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
591  }
592  } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
593  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
594  get_word(buf1, sizeof(buf1), &p);
595  rtsp_st->ssrc = strtoll(buf1, NULL, 10);
596  } else if (av_strstart(p, "range:", &p)) {
597  int64_t start, end;
598 
599  // this is so that seeking on a streamed file can work.
600  rtsp_parse_range_npt(p, &start, &end);
601  s->start_time = start;
602  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
603  s->duration = (end == AV_NOPTS_VALUE) ?
604  AV_NOPTS_VALUE : end - start;
605  } else if (av_strstart(p, "lang:", &p)) {
606  if (s->nb_streams > 0) {
607  get_word(buf1, sizeof(buf1), &p);
608  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609  if (rtsp_st->stream_index >= 0) {
610  st = s->streams[rtsp_st->stream_index];
611  av_dict_set(&st->metadata, "language", buf1, 0);
612  }
613  } else
614  get_word(rt->default_lang, sizeof(rt->default_lang), &p);
615  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
616  if (atoi(p) == 1)
618  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
619  s->nb_streams > 0) {
620  st = s->streams[s->nb_streams - 1];
621  st->codecpar->sample_rate = atoi(p);
622  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
623  // RFC 4568
624  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
625  get_word(buf1, sizeof(buf1), &p); // ignore tag
626  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
627  p += strspn(p, SPACE_CHARS);
628  if (av_strstart(p, "inline:", &p))
629  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
630  } else if (av_strstart(p, "source-filter:", &p)) {
631  int exclude = 0;
632  get_word(buf1, sizeof(buf1), &p);
633  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
634  return;
635  exclude = !strcmp(buf1, "excl");
636 
637  get_word(buf1, sizeof(buf1), &p);
638  if (strcmp(buf1, "IN") != 0)
639  return;
640  get_word(buf1, sizeof(buf1), &p);
641  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
642  return;
643  // not checking that the destination address actually matches or is wildcard
644  get_word(buf1, sizeof(buf1), &p);
645 
646  while (*p != '\0') {
647  rtsp_src = av_mallocz(sizeof(*rtsp_src));
648  if (!rtsp_src)
649  return;
650  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
651  if (exclude) {
652  if (s->nb_streams == 0) {
653  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
654  } else {
655  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
656  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
657  }
658  } else {
659  if (s->nb_streams == 0) {
660  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
661  } else {
662  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
663  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
664  }
665  }
666  }
667  } else {
668  if (rt->server_type == RTSP_SERVER_WMS)
670  if (s->nb_streams > 0) {
671  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
672 
673  if (rt->server_type == RTSP_SERVER_REAL)
674  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
675 
676  if (rtsp_st->dynamic_handler &&
678  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
679  rtsp_st->stream_index,
680  rtsp_st->dynamic_protocol_context, buf);
681  }
682  }
683  break;
684  }
685 }
686 
687 int ff_sdp_parse(AVFormatContext *s, const char *content)
688 {
689  const char *p;
690  int letter, i;
691  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
692  * contain long SDP lines containing complete ASF Headers (several
693  * kB) or arrays of MDPR (RM stream descriptor) headers plus
694  * "rulebooks" describing their properties. Therefore, the SDP line
695  * buffer is large.
696  *
697  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
698  * in rtpdec_xiph.c. */
699  char buf[16384], *q;
700  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
701 
702  p = content;
703  for (;;) {
704  p += strspn(p, SPACE_CHARS);
705  letter = *p;
706  if (letter == '\0')
707  break;
708  p++;
709  if (*p != '=')
710  goto next_line;
711  p++;
712  /* get the content */
713  q = buf;
714  while (*p != '\n' && *p != '\r' && *p != '\0') {
715  if ((q - buf) < sizeof(buf) - 1)
716  *q++ = *p;
717  p++;
718  }
719  *q = '\0';
720  sdp_parse_line(s, s1, letter, buf);
721  next_line:
722  while (*p != '\n' && *p != '\0')
723  p++;
724  if (*p == '\n')
725  p++;
726  }
727 
728  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
729  av_freep(&s1->default_include_source_addrs[i]);
730  av_freep(&s1->default_include_source_addrs);
731  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
732  av_freep(&s1->default_exclude_source_addrs[i]);
733  av_freep(&s1->default_exclude_source_addrs);
734 
735  return 0;
736 }
737 #endif /* CONFIG_RTPDEC */
738 
739 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
740 {
741  RTSPState *rt = s->priv_data;
742  int i;
743 
744  for (i = 0; i < rt->nb_rtsp_streams; i++) {
745  RTSPStream *rtsp_st = rt->rtsp_streams[i];
746  if (!rtsp_st)
747  continue;
748  if (rtsp_st->transport_priv) {
749  if (s->oformat) {
750  AVFormatContext *rtpctx = rtsp_st->transport_priv;
751  av_write_trailer(rtpctx);
753  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
754  ff_rtsp_tcp_write_packet(s, rtsp_st);
755  ffio_free_dyn_buf(&rtpctx->pb);
756  } else {
757  avio_closep(&rtpctx->pb);
758  }
759  avformat_free_context(rtpctx);
760  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
762  else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
764  }
765  rtsp_st->transport_priv = NULL;
766  ffurl_closep(&rtsp_st->rtp_handle);
767  }
768 }
769 
770 /* close and free RTSP streams */
772 {
773  RTSPState *rt = s->priv_data;
774  int i, j;
775  RTSPStream *rtsp_st;
776 
777  ff_rtsp_undo_setup(s, 0);
778  for (i = 0; i < rt->nb_rtsp_streams; i++) {
779  rtsp_st = rt->rtsp_streams[i];
780  if (rtsp_st) {
781  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
782  if (rtsp_st->dynamic_handler->close)
783  rtsp_st->dynamic_handler->close(
784  rtsp_st->dynamic_protocol_context);
786  }
787  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
788  av_freep(&rtsp_st->include_source_addrs[j]);
789  av_freep(&rtsp_st->include_source_addrs);
790  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
791  av_freep(&rtsp_st->exclude_source_addrs[j]);
792  av_freep(&rtsp_st->exclude_source_addrs);
793 
794  av_freep(&rtsp_st);
795  }
796  }
797  av_freep(&rt->rtsp_streams);
798  if (rt->asf_ctx) {
800  }
801  if (CONFIG_RTPDEC && rt->ts)
803  av_freep(&rt->p);
804  av_freep(&rt->recvbuf);
805 }
806 
808 {
809  RTSPState *rt = s->priv_data;
810  AVStream *st = NULL;
811  int reordering_queue_size = rt->reordering_queue_size;
812  if (reordering_queue_size < 0) {
814  reordering_queue_size = 0;
815  else
816  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
817  }
818 
819  /* open the RTP context */
820  if (rtsp_st->stream_index >= 0)
821  st = s->streams[rtsp_st->stream_index];
822  if (!st)
824 
825  if (CONFIG_RTSP_MUXER && s->oformat && st) {
826  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
827  s, st, rtsp_st->rtp_handle,
829  rtsp_st->stream_index);
830  /* Ownership of rtp_handle is passed to the rtp mux context */
831  rtsp_st->rtp_handle = NULL;
832  if (ret < 0)
833  return ret;
834  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
835  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
836  return 0; // Don't need to open any parser here
837  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
838  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
839  rtsp_st->dynamic_protocol_context,
840  rtsp_st->dynamic_handler);
841  else if (CONFIG_RTPDEC)
842  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
843  rtsp_st->sdp_payload_type,
844  reordering_queue_size);
845 
846  if (!rtsp_st->transport_priv) {
847  return AVERROR(ENOMEM);
848  } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
849  s->iformat) {
850  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
851  rtpctx->ssrc = rtsp_st->ssrc;
852  if (rtsp_st->dynamic_handler) {
854  rtsp_st->dynamic_protocol_context,
855  rtsp_st->dynamic_handler);
856  }
857  if (rtsp_st->crypto_suite[0])
859  rtsp_st->crypto_suite,
860  rtsp_st->crypto_params);
861  }
862 
863  return 0;
864 }
865 
866 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
867 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
868 {
869  const char *q;
870  char *p;
871  int v;
872 
873  q = *pp;
874  q += strspn(q, SPACE_CHARS);
875  v = strtol(q, &p, 10);
876  if (*p == '-') {
877  p++;
878  *min_ptr = v;
879  v = strtol(p, &p, 10);
880  *max_ptr = v;
881  } else {
882  *min_ptr = v;
883  *max_ptr = v;
884  }
885  *pp = p;
886 }
887 
888 /* XXX: only one transport specification is parsed */
889 static void rtsp_parse_transport(AVFormatContext *s,
890  RTSPMessageHeader *reply, const char *p)
891 {
892  char transport_protocol[16];
893  char profile[16];
894  char lower_transport[16];
895  char parameter[16];
897  char buf[256];
898 
899  reply->nb_transports = 0;
900 
901  for (;;) {
902  p += strspn(p, SPACE_CHARS);
903  if (*p == '\0')
904  break;
905 
906  th = &reply->transports[reply->nb_transports];
907 
908  get_word_sep(transport_protocol, sizeof(transport_protocol),
909  "/", &p);
910  if (!av_strcasecmp (transport_protocol, "rtp")) {
911  get_word_sep(profile, sizeof(profile), "/;,", &p);
912  lower_transport[0] = '\0';
913  /* rtp/avp/<protocol> */
914  if (*p == '/') {
915  get_word_sep(lower_transport, sizeof(lower_transport),
916  ";,", &p);
917  }
919  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
920  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
921  /* x-pn-tng/<protocol> */
922  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
923  profile[0] = '\0';
925  } else if (!av_strcasecmp(transport_protocol, "raw")) {
926  get_word_sep(profile, sizeof(profile), "/;,", &p);
927  lower_transport[0] = '\0';
928  /* raw/raw/<protocol> */
929  if (*p == '/') {
930  get_word_sep(lower_transport, sizeof(lower_transport),
931  ";,", &p);
932  }
934  } else {
935  break;
936  }
937  if (!av_strcasecmp(lower_transport, "TCP"))
939  else
941 
942  if (*p == ';')
943  p++;
944  /* get each parameter */
945  while (*p != '\0' && *p != ',') {
946  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
947  if (!strcmp(parameter, "port")) {
948  if (*p == '=') {
949  p++;
950  rtsp_parse_range(&th->port_min, &th->port_max, &p);
951  }
952  } else if (!strcmp(parameter, "client_port")) {
953  if (*p == '=') {
954  p++;
955  rtsp_parse_range(&th->client_port_min,
956  &th->client_port_max, &p);
957  }
958  } else if (!strcmp(parameter, "server_port")) {
959  if (*p == '=') {
960  p++;
961  rtsp_parse_range(&th->server_port_min,
962  &th->server_port_max, &p);
963  }
964  } else if (!strcmp(parameter, "interleaved")) {
965  if (*p == '=') {
966  p++;
967  rtsp_parse_range(&th->interleaved_min,
968  &th->interleaved_max, &p);
969  }
970  } else if (!strcmp(parameter, "multicast")) {
973  } else if (!strcmp(parameter, "ttl")) {
974  if (*p == '=') {
975  char *end;
976  p++;
977  th->ttl = strtol(p, &end, 10);
978  p = end;
979  }
980  } else if (!strcmp(parameter, "destination")) {
981  if (*p == '=') {
982  p++;
983  get_word_sep(buf, sizeof(buf), ";,", &p);
984  get_sockaddr(s, buf, &th->destination);
985  }
986  } else if (!strcmp(parameter, "source")) {
987  if (*p == '=') {
988  p++;
989  get_word_sep(buf, sizeof(buf), ";,", &p);
990  av_strlcpy(th->source, buf, sizeof(th->source));
991  }
992  } else if (!strcmp(parameter, "mode")) {
993  if (*p == '=') {
994  p++;
995  get_word_sep(buf, sizeof(buf), ";, ", &p);
996  if (!strcmp(buf, "record") ||
997  !strcmp(buf, "receive"))
998  th->mode_record = 1;
999  }
1000  }
1001 
1002  while (*p != ';' && *p != '\0' && *p != ',')
1003  p++;
1004  if (*p == ';')
1005  p++;
1006  }
1007  if (*p == ',')
1008  p++;
1009 
1010  reply->nb_transports++;
1011  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1012  break;
1013  }
1014 }
1015 
1016 static void handle_rtp_info(RTSPState *rt, const char *url,
1017  uint32_t seq, uint32_t rtptime)
1018 {
1019  int i;
1020  if (!rtptime || !url[0])
1021  return;
1022  if (rt->transport != RTSP_TRANSPORT_RTP)
1023  return;
1024  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1025  RTSPStream *rtsp_st = rt->rtsp_streams[i];
1026  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1027  if (!rtpctx)
1028  continue;
1029  if (!strcmp(rtsp_st->control_url, url)) {
1030  rtpctx->base_timestamp = rtptime;
1031  break;
1032  }
1033  }
1034 }
1035 
1036 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1037 {
1038  int read = 0;
1039  char key[20], value[1024], url[1024] = "";
1040  uint32_t seq = 0, rtptime = 0;
1041 
1042  for (;;) {
1043  p += strspn(p, SPACE_CHARS);
1044  if (!*p)
1045  break;
1046  get_word_sep(key, sizeof(key), "=", &p);
1047  if (*p != '=')
1048  break;
1049  p++;
1050  get_word_sep(value, sizeof(value), ";, ", &p);
1051  read++;
1052  if (!strcmp(key, "url"))
1053  av_strlcpy(url, value, sizeof(url));
1054  else if (!strcmp(key, "seq"))
1055  seq = strtoul(value, NULL, 10);
1056  else if (!strcmp(key, "rtptime"))
1057  rtptime = strtoul(value, NULL, 10);
1058  if (*p == ',') {
1059  handle_rtp_info(rt, url, seq, rtptime);
1060  url[0] = '\0';
1061  seq = rtptime = 0;
1062  read = 0;
1063  }
1064  if (*p)
1065  p++;
1066  }
1067  if (read > 0)
1068  handle_rtp_info(rt, url, seq, rtptime);
1069 }
1070 
1072  RTSPMessageHeader *reply, const char *buf,
1073  RTSPState *rt, const char *method)
1074 {
1075  const char *p;
1076 
1077  /* NOTE: we do case independent match for broken servers */
1078  p = buf;
1079  if (av_stristart(p, "Session:", &p)) {
1080  int t;
1081  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1082  if (av_stristart(p, ";timeout=", &p) &&
1083  (t = strtol(p, NULL, 10)) > 0) {
1084  reply->timeout = t;
1085  }
1086  } else if (av_stristart(p, "Content-Length:", &p)) {
1087  reply->content_length = strtol(p, NULL, 10);
1088  } else if (av_stristart(p, "Transport:", &p)) {
1089  rtsp_parse_transport(s, reply, p);
1090  } else if (av_stristart(p, "CSeq:", &p)) {
1091  reply->seq = strtol(p, NULL, 10);
1092  } else if (av_stristart(p, "Range:", &p)) {
1093  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1094  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1095  p += strspn(p, SPACE_CHARS);
1096  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1097  } else if (av_stristart(p, "Server:", &p)) {
1098  p += strspn(p, SPACE_CHARS);
1099  av_strlcpy(reply->server, p, sizeof(reply->server));
1100  } else if (av_stristart(p, "Notice:", &p) ||
1101  av_stristart(p, "X-Notice:", &p)) {
1102  reply->notice = strtol(p, NULL, 10);
1103  } else if (av_stristart(p, "Location:", &p)) {
1104  p += strspn(p, SPACE_CHARS);
1105  av_strlcpy(reply->location, p , sizeof(reply->location));
1106  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1107  p += strspn(p, SPACE_CHARS);
1108  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1109  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1110  p += strspn(p, SPACE_CHARS);
1111  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1112  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1113  p += strspn(p, SPACE_CHARS);
1114  if (method && !strcmp(method, "DESCRIBE"))
1115  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1116  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1117  p += strspn(p, SPACE_CHARS);
1118  if (method && !strcmp(method, "PLAY"))
1119  rtsp_parse_rtp_info(rt, p);
1120  } else if (av_stristart(p, "Public:", &p) && rt) {
1121  if (strstr(p, "GET_PARAMETER") &&
1122  method && !strcmp(method, "OPTIONS"))
1123  rt->get_parameter_supported = 1;
1124  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1125  p += strspn(p, SPACE_CHARS);
1126  rt->accept_dynamic_rate = atoi(p);
1127  } else if (av_stristart(p, "Content-Type:", &p)) {
1128  p += strspn(p, SPACE_CHARS);
1129  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1130  }
1131 }
1132 
1133 /* skip a RTP/TCP interleaved packet */
1135 {
1136  RTSPState *rt = s->priv_data;
1137  int ret, len, len1;
1138  uint8_t buf[1024];
1139 
1140  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1141  if (ret != 3)
1142  return;
1143  len = AV_RB16(buf + 1);
1144 
1145  av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1146 
1147  /* skip payload */
1148  while (len > 0) {
1149  len1 = len;
1150  if (len1 > sizeof(buf))
1151  len1 = sizeof(buf);
1152  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1153  if (ret != len1)
1154  return;
1155  len -= len1;
1156  }
1157 }
1158 
1160  unsigned char **content_ptr,
1161  int return_on_interleaved_data, const char *method)
1162 {
1163  RTSPState *rt = s->priv_data;
1164  char buf[4096], buf1[1024], *q;
1165  unsigned char ch;
1166  const char *p;
1167  int ret, content_length, line_count = 0, request = 0;
1168  unsigned char *content = NULL;
1169 
1170 start:
1171  line_count = 0;
1172  request = 0;
1173  content = NULL;
1174  memset(reply, 0, sizeof(*reply));
1175 
1176  /* parse reply (XXX: use buffers) */
1177  rt->last_reply[0] = '\0';
1178  for (;;) {
1179  q = buf;
1180  for (;;) {
1181  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1182  av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1183  if (ret != 1)
1184  return AVERROR_EOF;
1185  if (ch == '\n')
1186  break;
1187  if (ch == '$' && q == buf) {
1188  if (return_on_interleaved_data) {
1189  return 1;
1190  } else
1192  } else if (ch != '\r') {
1193  if ((q - buf) < sizeof(buf) - 1)
1194  *q++ = ch;
1195  }
1196  }
1197  *q = '\0';
1198 
1199  av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1200 
1201  /* test if last line */
1202  if (buf[0] == '\0')
1203  break;
1204  p = buf;
1205  if (line_count == 0) {
1206  /* get reply code */
1207  get_word(buf1, sizeof(buf1), &p);
1208  if (!strncmp(buf1, "RTSP/", 5)) {
1209  get_word(buf1, sizeof(buf1), &p);
1210  reply->status_code = atoi(buf1);
1211  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1212  } else {
1213  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1214  get_word(buf1, sizeof(buf1), &p); // object
1215  request = 1;
1216  }
1217  } else {
1218  ff_rtsp_parse_line(s, reply, p, rt, method);
1219  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1220  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1221  }
1222  line_count++;
1223  }
1224 
1225  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1226  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1227 
1228  content_length = reply->content_length;
1229  if (content_length > 0) {
1230  /* leave some room for a trailing '\0' (useful for simple parsing) */
1231  content = av_malloc(content_length + 1);
1232  if (!content)
1233  return AVERROR(ENOMEM);
1234  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1235  content[content_length] = '\0';
1236  }
1237  if (content_ptr)
1238  *content_ptr = content;
1239  else
1240  av_freep(&content);
1241 
1242  if (request) {
1243  char buf[1024];
1244  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1245  const char* ptr = buf;
1246 
1247  if (!strcmp(reply->reason, "OPTIONS")) {
1248  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1249  if (reply->seq)
1250  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1251  if (reply->session_id[0])
1252  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1253  reply->session_id);
1254  } else {
1255  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1256  }
1257  av_strlcat(buf, "\r\n", sizeof(buf));
1258 
1259  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1260  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1261  ptr = base64buf;
1262  }
1263  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1264 
1266  /* Even if the request from the server had data, it is not the data
1267  * that the caller wants or expects. The memory could also be leaked
1268  * if the actual following reply has content data. */
1269  if (content_ptr)
1270  av_freep(content_ptr);
1271  /* If method is set, this is called from ff_rtsp_send_cmd,
1272  * where a reply to exactly this request is awaited. For
1273  * callers from within packet receiving, we just want to
1274  * return to the caller and go back to receiving packets. */
1275  if (method)
1276  goto start;
1277  return 0;
1278  }
1279 
1280  if (rt->seq != reply->seq) {
1281  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1282  rt->seq, reply->seq);
1283  }
1284 
1285  /* EOS */
1286  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1287  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1288  reply->notice == 2306 /* Continuous Feed Terminated */) {
1289  rt->state = RTSP_STATE_IDLE;
1290  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1291  return AVERROR(EIO); /* data or server error */
1292  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1293  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1294  return AVERROR(EPERM);
1295 
1296  return 0;
1297 }
1298 
1299 /**
1300  * Send a command to the RTSP server without waiting for the reply.
1301  *
1302  * @param s RTSP (de)muxer context
1303  * @param method the method for the request
1304  * @param url the target url for the request
1305  * @param headers extra header lines to include in the request
1306  * @param send_content if non-null, the data to send as request body content
1307  * @param send_content_length the length of the send_content data, or 0 if
1308  * send_content is null
1309  *
1310  * @return zero if success, nonzero otherwise
1311  */
1312 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1313  const char *method, const char *url,
1314  const char *headers,
1315  const unsigned char *send_content,
1316  int send_content_length)
1317 {
1318  RTSPState *rt = s->priv_data;
1319  char buf[4096], *out_buf;
1320  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1321 
1322  if (!rt->rtsp_hd_out)
1323  return AVERROR(ENOTCONN);
1324 
1325  /* Add in RTSP headers */
1326  out_buf = buf;
1327  rt->seq++;
1328  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1329  if (headers)
1330  av_strlcat(buf, headers, sizeof(buf));
1331  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1332  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1333  if (rt->session_id[0] != '\0' && (!headers ||
1334  !strstr(headers, "\nIf-Match:"))) {
1335  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1336  }
1337  if (rt->auth[0]) {
1339  rt->auth, url, method);
1340  if (str)
1341  av_strlcat(buf, str, sizeof(buf));
1342  av_free(str);
1343  }
1344  if (send_content_length > 0 && send_content)
1345  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1346  av_strlcat(buf, "\r\n", sizeof(buf));
1347 
1348  /* base64 encode rtsp if tunneling */
1349  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1350  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1351  out_buf = base64buf;
1352  }
1353 
1354  av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1355 
1356  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1357  if (send_content_length > 0 && send_content) {
1358  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1359  avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1360  return AVERROR_PATCHWELCOME;
1361  }
1362  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1363  }
1365 
1366  return 0;
1367 }
1368 
1369 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1370  const char *url, const char *headers)
1371 {
1372  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1373 }
1374 
1375 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1376  const char *headers, RTSPMessageHeader *reply,
1377  unsigned char **content_ptr)
1378 {
1379  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1380  content_ptr, NULL, 0);
1381 }
1382 
1384  const char *method, const char *url,
1385  const char *header,
1386  RTSPMessageHeader *reply,
1387  unsigned char **content_ptr,
1388  const unsigned char *send_content,
1389  int send_content_length)
1390 {
1391  RTSPState *rt = s->priv_data;
1392  HTTPAuthType cur_auth_type;
1393  int ret, attempts = 0;
1394 
1395 retry:
1396  cur_auth_type = rt->auth_state.auth_type;
1397  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1398  send_content,
1399  send_content_length)))
1400  return ret;
1401 
1402  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1403  return ret;
1404  attempts++;
1405 
1406  if (reply->status_code == 401 &&
1407  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1408  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1409  goto retry;
1410 
1411  if (reply->status_code > 400){
1412  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1413  method,
1414  reply->status_code,
1415  reply->reason);
1416  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1417  }
1418 
1419  return 0;
1420 }
1421 
1422 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1423  int lower_transport, const char *real_challenge)
1424 {
1425  RTSPState *rt = s->priv_data;
1426  int rtx = 0, j, i, err, interleave = 0, port_off;
1427  RTSPStream *rtsp_st;
1428  RTSPMessageHeader reply1, *reply = &reply1;
1429  char cmd[2048];
1430  const char *trans_pref;
1431 
1432  if (rt->transport == RTSP_TRANSPORT_RDT)
1433  trans_pref = "x-pn-tng";
1434  else if (rt->transport == RTSP_TRANSPORT_RAW)
1435  trans_pref = "RAW/RAW";
1436  else
1437  trans_pref = "RTP/AVP";
1438 
1439  /* default timeout: 1 minute */
1440  rt->timeout = 60;
1441 
1442  /* Choose a random starting offset within the first half of the
1443  * port range, to allow for a number of ports to try even if the offset
1444  * happens to be at the end of the random range. */
1445  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1446  /* even random offset */
1447  port_off -= port_off & 0x01;
1448 
1449  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1450  char transport[2048];
1451 
1452  /*
1453  * WMS serves all UDP data over a single connection, the RTX, which
1454  * isn't necessarily the first in the SDP but has to be the first
1455  * to be set up, else the second/third SETUP will fail with a 461.
1456  */
1457  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1458  rt->server_type == RTSP_SERVER_WMS) {
1459  if (i == 0) {
1460  /* rtx first */
1461  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1462  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1463  if (len >= 4 &&
1464  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1465  "/rtx"))
1466  break;
1467  }
1468  if (rtx == rt->nb_rtsp_streams)
1469  return -1; /* no RTX found */
1470  rtsp_st = rt->rtsp_streams[rtx];
1471  } else
1472  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1473  } else
1474  rtsp_st = rt->rtsp_streams[i];
1475 
1476  /* RTP/UDP */
1477  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1478  char buf[256];
1479 
1480  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1481  port = reply->transports[0].client_port_min;
1482  goto have_port;
1483  }
1484 
1485  /* first try in specified port range */
1486  while (j <= rt->rtp_port_max) {
1487  AVDictionary *opts = map_to_opts(rt);
1488 
1489  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1490  "?localport=%d", j);
1491  /* we will use two ports per rtp stream (rtp and rtcp) */
1492  j += 2;
1495 
1496  av_dict_free(&opts);
1497 
1498  if (!err)
1499  goto rtp_opened;
1500  }
1501  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1502  err = AVERROR(EIO);
1503  goto fail;
1504 
1505  rtp_opened:
1506  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1507  have_port:
1508  snprintf(transport, sizeof(transport) - 1,
1509  "%s/UDP;", trans_pref);
1510  if (rt->server_type != RTSP_SERVER_REAL)
1511  av_strlcat(transport, "unicast;", sizeof(transport));
1512  av_strlcatf(transport, sizeof(transport),
1513  "client_port=%d", port);
1514  if (rt->transport == RTSP_TRANSPORT_RTP &&
1515  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1516  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1517  }
1518 
1519  /* RTP/TCP */
1520  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1521  /* For WMS streams, the application streams are only used for
1522  * UDP. When trying to set it up for TCP streams, the server
1523  * will return an error. Therefore, we skip those streams. */
1524  if (rt->server_type == RTSP_SERVER_WMS &&
1525  (rtsp_st->stream_index < 0 ||
1526  s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1528  continue;
1529  snprintf(transport, sizeof(transport) - 1,
1530  "%s/TCP;", trans_pref);
1531  if (rt->transport != RTSP_TRANSPORT_RDT)
1532  av_strlcat(transport, "unicast;", sizeof(transport));
1533  av_strlcatf(transport, sizeof(transport),
1534  "interleaved=%d-%d",
1535  interleave, interleave + 1);
1536  interleave += 2;
1537  }
1538 
1539  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1540  snprintf(transport, sizeof(transport) - 1,
1541  "%s/UDP;multicast", trans_pref);
1542  }
1543  if (s->oformat) {
1544  av_strlcat(transport, ";mode=record", sizeof(transport));
1545  } else if (rt->server_type == RTSP_SERVER_REAL ||
1547  av_strlcat(transport, ";mode=play", sizeof(transport));
1548  snprintf(cmd, sizeof(cmd),
1549  "Transport: %s\r\n",
1550  transport);
1551  if (rt->accept_dynamic_rate)
1552  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1553  if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1554  char real_res[41], real_csum[9];
1555  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1556  real_challenge);
1557  av_strlcatf(cmd, sizeof(cmd),
1558  "If-Match: %s\r\n"
1559  "RealChallenge2: %s, sd=%s\r\n",
1560  rt->session_id, real_res, real_csum);
1561  }
1562  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1563  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1564  err = 1;
1565  goto fail;
1566  } else if (reply->status_code != RTSP_STATUS_OK ||
1567  reply->nb_transports != 1) {
1569  goto fail;
1570  }
1571 
1572  /* XXX: same protocol for all streams is required */
1573  if (i > 0) {
1574  if (reply->transports[0].lower_transport != rt->lower_transport ||
1575  reply->transports[0].transport != rt->transport) {
1576  err = AVERROR_INVALIDDATA;
1577  goto fail;
1578  }
1579  } else {
1580  rt->lower_transport = reply->transports[0].lower_transport;
1581  rt->transport = reply->transports[0].transport;
1582  }
1583 
1584  /* Fail if the server responded with another lower transport mode
1585  * than what we requested. */
1586  if (reply->transports[0].lower_transport != lower_transport) {
1587  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1588  err = AVERROR_INVALIDDATA;
1589  goto fail;
1590  }
1591 
1592  switch(reply->transports[0].lower_transport) {
1594  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1595  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1596  break;
1597 
1598  case RTSP_LOWER_TRANSPORT_UDP: {
1599  char url[1024], options[30] = "";
1600  const char *peer = host;
1601 
1602  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1603  av_strlcpy(options, "?connect=1", sizeof(options));
1604  /* Use source address if specified */
1605  if (reply->transports[0].source[0])
1606  peer = reply->transports[0].source;
1607  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1608  reply->transports[0].server_port_min, "%s", options);
1609  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1610  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1611  err = AVERROR_INVALIDDATA;
1612  goto fail;
1613  }
1614  break;
1615  }
1617  char url[1024], namebuf[50], optbuf[20] = "";
1618  struct sockaddr_storage addr;
1619  int port, ttl;
1620  AVDictionary *opts = map_to_opts(rt);
1621 
1622  if (reply->transports[0].destination.ss_family) {
1623  addr = reply->transports[0].destination;
1624  port = reply->transports[0].port_min;
1625  ttl = reply->transports[0].ttl;
1626  } else {
1627  addr = rtsp_st->sdp_ip;
1628  port = rtsp_st->sdp_port;
1629  ttl = rtsp_st->sdp_ttl;
1630  }
1631  if (ttl > 0)
1632  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1633  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1634  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1635  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1636  port, "%s", optbuf);
1639  av_dict_free(&opts);
1640 
1641  if (err < 0) {
1642  err = AVERROR_INVALIDDATA;
1643  goto fail;
1644  }
1645  break;
1646  }
1647  }
1648 
1649  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1650  goto fail;
1651  }
1652 
1653  if (rt->nb_rtsp_streams && reply->timeout > 0)
1654  rt->timeout = reply->timeout;
1655 
1656  if (rt->server_type == RTSP_SERVER_REAL)
1657  rt->need_subscription = 1;
1658 
1659  return 0;
1660 
1661 fail:
1662  ff_rtsp_undo_setup(s, 0);
1663  return err;
1664 }
1665 
1667 {
1668  RTSPState *rt = s->priv_data;
1669  if (rt->rtsp_hd_out != rt->rtsp_hd)
1670  ffurl_closep(&rt->rtsp_hd_out);
1671  rt->rtsp_hd_out = NULL;
1672  ffurl_closep(&rt->rtsp_hd);
1673 }
1674 
1676 {
1677  RTSPState *rt = s->priv_data;
1678  char proto[128], host[1024], path[1024];
1679  char tcpname[1024], cmd[2048], auth[128];
1680  const char *lower_rtsp_proto = "tcp";
1681  int port, err, tcp_fd;
1682  RTSPMessageHeader reply1, *reply = &reply1;
1683  int lower_transport_mask = 0;
1684  int default_port = RTSP_DEFAULT_PORT;
1685  int https_tunnel = 0;
1686  char real_challenge[64] = "";
1687  struct sockaddr_storage peer;
1688  socklen_t peer_len = sizeof(peer);
1689 
1690  if (rt->rtp_port_max < rt->rtp_port_min) {
1691  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1692  "than min port %d\n", rt->rtp_port_max,
1693  rt->rtp_port_min);
1694  return AVERROR(EINVAL);
1695  }
1696 
1697  if (!ff_network_init())
1698  return AVERROR(EIO);
1699 
1700  if (s->max_delay < 0) /* Not set by the caller */
1702 
1705  (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1706  https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1709  }
1710  /* Only pass through valid flags from here */
1712 
1713 redirect:
1714  memset(&reply1, 0, sizeof(reply1));
1715  /* extract hostname and port */
1716  av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1717  host, sizeof(host), &port, path, sizeof(path), s->url);
1718 
1719  if (!strcmp(proto, "rtsps")) {
1720  lower_rtsp_proto = "tls";
1721  default_port = RTSPS_DEFAULT_PORT;
1723  }
1724 
1725  if (*auth) {
1726  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1727  }
1728  if (port < 0)
1729  port = default_port;
1730 
1731  lower_transport_mask = rt->lower_transport_mask;
1732 
1733  if (!lower_transport_mask)
1734  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1735 
1736  if (s->oformat) {
1737  /* Only UDP or TCP - UDP multicast isn't supported. */
1738  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1739  (1 << RTSP_LOWER_TRANSPORT_TCP);
1740  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1741  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1742  "only UDP and TCP are supported for output.\n");
1743  err = AVERROR(EINVAL);
1744  goto fail;
1745  }
1746  }
1747 
1748  /* Construct the URI used in request; this is similar to s->url,
1749  * but with authentication credentials removed and RTSP specific options
1750  * stripped out. */
1751  ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1752  host, port, "%s", path);
1753 
1754  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1755  /* set up initial handshake for tunneling */
1756  char httpname[1024];
1757  char sessioncookie[17];
1758  char headers[1024];
1760 
1761  av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1762 
1763  ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1764  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1766 
1767  /* GET requests */
1768  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1769  &s->interrupt_callback) < 0) {
1770  err = AVERROR(EIO);
1771  goto fail;
1772  }
1773 
1774  /* generate GET headers */
1775  snprintf(headers, sizeof(headers),
1776  "x-sessioncookie: %s\r\n"
1777  "Accept: application/x-rtsp-tunnelled\r\n"
1778  "Pragma: no-cache\r\n"
1779  "Cache-Control: no-cache\r\n",
1780  sessioncookie);
1781  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1782 
1783  if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1785  if (!rt->rtsp_hd->protocol_whitelist) {
1786  err = AVERROR(ENOMEM);
1787  goto fail;
1788  }
1789  }
1790 
1791  /* complete the connection */
1792  if (ffurl_connect(rt->rtsp_hd, &options)) {
1793  av_dict_free(&options);
1794  err = AVERROR(EIO);
1795  goto fail;
1796  }
1797 
1798  /* POST requests */
1799  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1800  &s->interrupt_callback) < 0 ) {
1801  err = AVERROR(EIO);
1802  goto fail;
1803  }
1804 
1805  /* generate POST headers */
1806  snprintf(headers, sizeof(headers),
1807  "x-sessioncookie: %s\r\n"
1808  "Content-Type: application/x-rtsp-tunnelled\r\n"
1809  "Pragma: no-cache\r\n"
1810  "Cache-Control: no-cache\r\n"
1811  "Content-Length: 32767\r\n"
1812  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1813  sessioncookie);
1814  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1815  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1816  av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1817 
1818  /* Initialize the authentication state for the POST session. The HTTP
1819  * protocol implementation doesn't properly handle multi-pass
1820  * authentication for POST requests, since it would require one of
1821  * the following:
1822  * - implementing Expect: 100-continue, which many HTTP servers
1823  * don't support anyway, even less the RTSP servers that do HTTP
1824  * tunneling
1825  * - sending the whole POST data until getting a 401 reply specifying
1826  * what authentication method to use, then resending all that data
1827  * - waiting for potential 401 replies directly after sending the
1828  * POST header (waiting for some unspecified time)
1829  * Therefore, we copy the full auth state, which works for both basic
1830  * and digest. (For digest, we would have to synchronize the nonce
1831  * count variable between the two sessions, if we'd do more requests
1832  * with the original session, though.)
1833  */
1835 
1836  /* complete the connection */
1837  if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1838  av_dict_free(&options);
1839  err = AVERROR(EIO);
1840  goto fail;
1841  }
1842  av_dict_free(&options);
1843  } else {
1844  int ret;
1845  /* open the tcp connection */
1846  ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1847  host, port,
1848  "?timeout=%d", rt->stimeout);
1849  if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1851  err = ret;
1852  goto fail;
1853  }
1854  rt->rtsp_hd_out = rt->rtsp_hd;
1855  }
1856  rt->seq = 0;
1857 
1858  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1859  if (tcp_fd < 0) {
1860  err = tcp_fd;
1861  goto fail;
1862  }
1863  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1864  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1865  NULL, 0, NI_NUMERICHOST);
1866  }
1867 
1868  /* request options supported by the server; this also detects server
1869  * type */
1870  for (rt->server_type = RTSP_SERVER_RTP;;) {
1871  cmd[0] = 0;
1872  if (rt->server_type == RTSP_SERVER_REAL)
1873  av_strlcat(cmd,
1874  /*
1875  * The following entries are required for proper
1876  * streaming from a Realmedia server. They are
1877  * interdependent in some way although we currently
1878  * don't quite understand how. Values were copied
1879  * from mplayer SVN r23589.
1880  * ClientChallenge is a 16-byte ID in hex
1881  * CompanyID is a 16-byte ID in base64
1882  */
1883  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1884  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1885  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1886  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1887  sizeof(cmd));
1888  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1889  if (reply->status_code != RTSP_STATUS_OK) {
1891  goto fail;
1892  }
1893 
1894  /* detect server type if not standard-compliant RTP */
1895  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1897  continue;
1898  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1900  } else if (rt->server_type == RTSP_SERVER_REAL)
1901  strcpy(real_challenge, reply->real_challenge);
1902  break;
1903  }
1904 
1905  if (CONFIG_RTSP_DEMUXER && s->iformat)
1906  err = ff_rtsp_setup_input_streams(s, reply);
1907  else if (CONFIG_RTSP_MUXER)
1908  err = ff_rtsp_setup_output_streams(s, host);
1909  else
1910  av_assert0(0);
1911  if (err)
1912  goto fail;
1913 
1914  do {
1915  int lower_transport = ff_log2_tab[lower_transport_mask &
1916  ~(lower_transport_mask - 1)];
1917 
1918  if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1919  && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1920  lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1921 
1922  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1923  rt->server_type == RTSP_SERVER_REAL ?
1924  real_challenge : NULL);
1925  if (err < 0)
1926  goto fail;
1927  lower_transport_mask &= ~(1 << lower_transport);
1928  if (lower_transport_mask == 0 && err == 1) {
1929  err = AVERROR(EPROTONOSUPPORT);
1930  goto fail;
1931  }
1932  } while (err);
1933 
1934  rt->lower_transport_mask = lower_transport_mask;
1935  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1936  rt->state = RTSP_STATE_IDLE;
1937  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1938  return 0;
1939  fail:
1942  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1943  char *new_url = av_strdup(reply->location);
1944  if (!new_url) {
1945  err = AVERROR(ENOMEM);
1946  goto fail2;
1947  }
1948  ff_format_set_url(s, new_url);
1949  rt->session_id[0] = '\0';
1950  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1951  reply->status_code,
1952  s->url);
1953  goto redirect;
1954  }
1955  fail2:
1956  ff_network_close();
1957  return err;
1958 }
1959 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1960 
1961 #if CONFIG_RTPDEC
1962 static int parse_rtsp_message(AVFormatContext *s)
1963 {
1964  RTSPState *rt = s->priv_data;
1965  int ret;
1966 
1967  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1968  if (rt->state == RTSP_STATE_STREAMING) {
1970  return AVERROR_EOF;
1971  else
1973  "Unable to answer to TEARDOWN\n");
1974  } else
1975  return 0;
1976  } else {
1977  RTSPMessageHeader reply;
1978  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1979  if (ret < 0)
1980  return ret;
1981  /* XXX: parse message */
1982  if (rt->state != RTSP_STATE_STREAMING)
1983  return 0;
1984  }
1985 
1986  return 0;
1987 }
1988 
1989 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1990  uint8_t *buf, int buf_size, int64_t wait_end)
1991 {
1992  RTSPState *rt = s->priv_data;
1993  RTSPStream *rtsp_st;
1994  int n, i, ret, timeout_cnt = 0;
1995  struct pollfd *p = rt->p;
1996  int *fds = NULL, fdsnum, fdsidx;
1997 
1998  if (!p) {
1999  p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
2000  if (!p)
2001  return AVERROR(ENOMEM);
2002 
2003  if (rt->rtsp_hd) {
2004  p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2005  p[rt->max_p++].events = POLLIN;
2006  }
2007  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2008  rtsp_st = rt->rtsp_streams[i];
2009  if (rtsp_st->rtp_handle) {
2010  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2011  &fds, &fdsnum)) {
2012  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2013  return ret;
2014  }
2015  if (fdsnum != 2) {
2016  av_log(s, AV_LOG_ERROR,
2017  "Number of fds %d not supported\n", fdsnum);
2018  return AVERROR_INVALIDDATA;
2019  }
2020  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2021  p[rt->max_p].fd = fds[fdsidx];
2022  p[rt->max_p++].events = POLLIN;
2023  }
2024  av_freep(&fds);
2025  }
2026  }
2027  }
2028 
2029  for (;;) {
2031  return AVERROR_EXIT;
2032  if (wait_end && wait_end - av_gettime_relative() < 0)
2033  return AVERROR(EAGAIN);
2034  n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2035  if (n > 0) {
2036  int j = rt->rtsp_hd ? 1 : 0;
2037  timeout_cnt = 0;
2038  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2039  rtsp_st = rt->rtsp_streams[i];
2040  if (rtsp_st->rtp_handle) {
2041  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2042  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2043  if (ret > 0) {
2044  *prtsp_st = rtsp_st;
2045  return ret;
2046  }
2047  }
2048  j+=2;
2049  }
2050  }
2051 #if CONFIG_RTSP_DEMUXER
2052  if (rt->rtsp_hd && p[0].revents & POLLIN) {
2053  if ((ret = parse_rtsp_message(s)) < 0) {
2054  return ret;
2055  }
2056  }
2057 #endif
2058  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2059  return AVERROR(ETIMEDOUT);
2060  } else if (n < 0 && errno != EINTR)
2061  return AVERROR(errno);
2062  }
2063 }
2064 
2065 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2066  const uint8_t *buf, int len)
2067 {
2068  RTSPState *rt = s->priv_data;
2069  int i;
2070  if (len < 0)
2071  return len;
2072  if (rt->nb_rtsp_streams == 1) {
2073  *rtsp_st = rt->rtsp_streams[0];
2074  return len;
2075  }
2076  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2077  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2078  int no_ssrc = 0;
2079  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2080  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2081  if (!rtpctx)
2082  continue;
2083  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2084  *rtsp_st = rt->rtsp_streams[i];
2085  return len;
2086  }
2087  if (!rtpctx->ssrc)
2088  no_ssrc = 1;
2089  }
2090  if (no_ssrc) {
2092  "Unable to pick stream for packet - SSRC not known for "
2093  "all streams\n");
2094  return AVERROR(EAGAIN);
2095  }
2096  } else {
2097  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2098  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2099  *rtsp_st = rt->rtsp_streams[i];
2100  return len;
2101  }
2102  }
2103  }
2104  }
2105  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2106  return AVERROR(EAGAIN);
2107 }
2108 
2109 static int read_packet(AVFormatContext *s,
2110  RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2111  int64_t wait_end)
2112 {
2113  RTSPState *rt = s->priv_data;
2114  int len;
2115 
2116  switch(rt->lower_transport) {
2117  default:
2118 #if CONFIG_RTSP_DEMUXER
2120  len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2121  break;
2122 #endif
2125  len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2126  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2127  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2128  break;
2130  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2131  wait_end && wait_end < av_gettime_relative())
2132  len = AVERROR(EAGAIN);
2133  else
2134  len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2135  len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2136  if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2137  ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2138  break;
2139  }
2140 
2141  if (len == 0)
2142  return AVERROR_EOF;
2143 
2144  return len;
2145 }
2146 
2148 {
2149  RTSPState *rt = s->priv_data;
2150  int ret, len;
2151  RTSPStream *rtsp_st, *first_queue_st = NULL;
2152  int64_t wait_end = 0;
2153 
2154  if (rt->nb_byes == rt->nb_rtsp_streams)
2155  return AVERROR_EOF;
2156 
2157  /* get next frames from the same RTP packet */
2158  if (rt->cur_transport_priv) {
2159  if (rt->transport == RTSP_TRANSPORT_RDT) {
2160  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2161  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2162  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2163  } else if (CONFIG_RTPDEC && rt->ts) {
2164  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2165  if (ret >= 0) {
2166  rt->recvbuf_pos += ret;
2167  ret = rt->recvbuf_pos < rt->recvbuf_len;
2168  }
2169  } else
2170  ret = -1;
2171  if (ret == 0) {
2172  rt->cur_transport_priv = NULL;
2173  return 0;
2174  } else if (ret == 1) {
2175  return 0;
2176  } else
2177  rt->cur_transport_priv = NULL;
2178  }
2179 
2180 redo:
2181  if (rt->transport == RTSP_TRANSPORT_RTP) {
2182  int i;
2183  int64_t first_queue_time = 0;
2184  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2185  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2186  int64_t queue_time;
2187  if (!rtpctx)
2188  continue;
2189  queue_time = ff_rtp_queued_packet_time(rtpctx);
2190  if (queue_time && (queue_time - first_queue_time < 0 ||
2191  !first_queue_time)) {
2192  first_queue_time = queue_time;
2193  first_queue_st = rt->rtsp_streams[i];
2194  }
2195  }
2196  if (first_queue_time) {
2197  wait_end = first_queue_time + s->max_delay;
2198  } else {
2199  wait_end = 0;
2200  first_queue_st = NULL;
2201  }
2202  }
2203 
2204  /* read next RTP packet */
2205  if (!rt->recvbuf) {
2207  if (!rt->recvbuf)
2208  return AVERROR(ENOMEM);
2209  }
2210 
2211  len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2212  if (len == AVERROR(EAGAIN) && first_queue_st &&
2213  rt->transport == RTSP_TRANSPORT_RTP) {
2215  "max delay reached. need to consume packet\n");
2216  rtsp_st = first_queue_st;
2217  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2218  goto end;
2219  }
2220  if (len < 0)
2221  return len;
2222 
2223  if (rt->transport == RTSP_TRANSPORT_RDT) {
2224  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2225  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2226  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2227  if (rtsp_st->feedback) {
2228  AVIOContext *pb = NULL;
2230  pb = s->pb;
2231  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2232  }
2233  if (ret < 0) {
2234  /* Either bad packet, or a RTCP packet. Check if the
2235  * first_rtcp_ntp_time field was initialized. */
2236  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2237  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2238  /* first_rtcp_ntp_time has been initialized for this stream,
2239  * copy the same value to all other uninitialized streams,
2240  * in order to map their timestamp origin to the same ntp time
2241  * as this one. */
2242  int i;
2243  AVStream *st = NULL;
2244  if (rtsp_st->stream_index >= 0)
2245  st = s->streams[rtsp_st->stream_index];
2246  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2247  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2248  AVStream *st2 = NULL;
2249  if (rt->rtsp_streams[i]->stream_index >= 0)
2250  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2251  if (rtpctx2 && st && st2 &&
2252  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2253  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2254  rtpctx2->rtcp_ts_offset = av_rescale_q(
2255  rtpctx->rtcp_ts_offset, st->time_base,
2256  st2->time_base);
2257  }
2258  }
2259  // Make real NTP start time available in AVFormatContext
2260  if (s->start_time_realtime == AV_NOPTS_VALUE) {
2261  s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2262  if (rtpctx->st) {
2263  s->start_time_realtime -=
2264  av_rescale (rtpctx->rtcp_ts_offset,
2265  (uint64_t) rtpctx->st->time_base.num * 1000000,
2266  rtpctx->st->time_base.den);
2267  }
2268  }
2269  }
2270  if (ret == -RTCP_BYE) {
2271  rt->nb_byes++;
2272 
2273  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2274  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2275 
2276  if (rt->nb_byes == rt->nb_rtsp_streams)
2277  return AVERROR_EOF;
2278  }
2279  }
2280  } else if (CONFIG_RTPDEC && rt->ts) {
2281  ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2282  if (ret >= 0) {
2283  if (ret < len) {
2284  rt->recvbuf_len = len;
2285  rt->recvbuf_pos = ret;
2286  rt->cur_transport_priv = rt->ts;
2287  return 1;
2288  } else {
2289  ret = 0;
2290  }
2291  }
2292  } else {
2293  return AVERROR_INVALIDDATA;
2294  }
2295 end:
2296  if (ret < 0)
2297  goto redo;
2298  if (ret == 1)
2299  /* more packets may follow, so we save the RTP context */
2300  rt->cur_transport_priv = rtsp_st->transport_priv;
2301 
2302  return ret;
2303 }
2304 #endif /* CONFIG_RTPDEC */
2305 
2306 #if CONFIG_SDP_DEMUXER
2307 static int sdp_probe(const AVProbeData *p1)
2308 {
2309  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2310 
2311  /* we look for a line beginning "c=IN IP" */
2312  while (p < p_end && *p != '\0') {
2313  if (sizeof("c=IN IP") - 1 < p_end - p &&
2314  av_strstart(p, "c=IN IP", NULL))
2315  return AVPROBE_SCORE_EXTENSION;
2316 
2317  while (p < p_end - 1 && *p != '\n') p++;
2318  if (++p >= p_end)
2319  break;
2320  if (*p == '\r')
2321  p++;
2322  }
2323  return 0;
2324 }
2325 
2326 static void append_source_addrs(char *buf, int size, const char *name,
2327  int count, struct RTSPSource **addrs)
2328 {
2329  int i;
2330  if (!count)
2331  return;
2332  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2333  for (i = 1; i < count; i++)
2334  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2335 }
2336 
2337 static int sdp_read_header(AVFormatContext *s)
2338 {
2339  RTSPState *rt = s->priv_data;
2340  RTSPStream *rtsp_st;
2341  int size, i, err;
2342  char *content;
2343  char url[1024];
2344 
2345  if (!ff_network_init())
2346  return AVERROR(EIO);
2347 
2348  if (s->max_delay < 0) /* Not set by the caller */
2350  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2352 
2353  /* read the whole sdp file */
2354  /* XXX: better loading */
2355  content = av_malloc(SDP_MAX_SIZE);
2356  if (!content)
2357  return AVERROR(ENOMEM);
2358  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2359  if (size <= 0) {
2360  av_free(content);
2361  return AVERROR_INVALIDDATA;
2362  }
2363  content[size] ='\0';
2364 
2365  err = ff_sdp_parse(s, content);
2366  av_freep(&content);
2367  if (err) goto fail;
2368 
2369  /* open each RTP stream */
2370  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2371  char namebuf[50];
2372  rtsp_st = rt->rtsp_streams[i];
2373 
2374  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2375  AVDictionary *opts = map_to_opts(rt);
2376 
2377  err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2378  sizeof(rtsp_st->sdp_ip),
2379  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2380  if (err) {
2381  av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2382  err = AVERROR(EIO);
2383  av_dict_free(&opts);
2384  goto fail;
2385  }
2386  ff_url_join(url, sizeof(url), "rtp", NULL,
2387  namebuf, rtsp_st->sdp_port,
2388  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2389  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2390  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2391  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2392 
2393  append_source_addrs(url, sizeof(url), "sources",
2394  rtsp_st->nb_include_source_addrs,
2395  rtsp_st->include_source_addrs);
2396  append_source_addrs(url, sizeof(url), "block",
2397  rtsp_st->nb_exclude_source_addrs,
2398  rtsp_st->exclude_source_addrs);
2399  err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2401 
2402  av_dict_free(&opts);
2403 
2404  if (err < 0) {
2405  err = AVERROR_INVALIDDATA;
2406  goto fail;
2407  }
2408  }
2409  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2410  goto fail;
2411  }
2412  return 0;
2413 fail:
2415  ff_network_close();
2416  return err;
2417 }
2418 
2419 static int sdp_read_close(AVFormatContext *s)
2420 {
2422  ff_network_close();
2423  return 0;
2424 }
2425 
2426 static const AVClass sdp_demuxer_class = {
2427  .class_name = "SDP demuxer",
2428  .item_name = av_default_item_name,
2429  .option = sdp_options,
2430  .version = LIBAVUTIL_VERSION_INT,
2431 };
2432 
2434  .name = "sdp",
2435  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2436  .priv_data_size = sizeof(RTSPState),
2437  .read_probe = sdp_probe,
2438  .read_header = sdp_read_header,
2440  .read_close = sdp_read_close,
2441  .priv_class = &sdp_demuxer_class,
2442 };
2443 #endif /* CONFIG_SDP_DEMUXER */
2444 
2445 #if CONFIG_RTP_DEMUXER
2446 static int rtp_probe(const AVProbeData *p)
2447 {
2448  if (av_strstart(p->filename, "rtp:", NULL))
2449  return AVPROBE_SCORE_MAX;
2450  return 0;
2451 }
2452 
2453 static int rtp_read_header(AVFormatContext *s)
2454 {
2455  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2456  char host[500], filters_buf[1000];
2457  int ret, port;
2458  URLContext* in = NULL;
2459  int payload_type;
2460  AVCodecParameters *par = NULL;
2461  struct sockaddr_storage addr;
2462  AVIOContext pb;
2463  socklen_t addrlen = sizeof(addr);
2464  RTSPState *rt = s->priv_data;
2465  const char *p;
2466  AVBPrint sdp;
2467 
2468  if (!ff_network_init())
2469  return AVERROR(EIO);
2470 
2471  ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2473  if (ret)
2474  goto fail;
2475 
2476  while (1) {
2477  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2478  if (ret == AVERROR(EAGAIN))
2479  continue;
2480  if (ret < 0)
2481  goto fail;
2482  if (ret < 12) {
2483  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2484  continue;
2485  }
2486 
2487  if ((recvbuf[0] & 0xc0) != 0x80) {
2488  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2489  "received\n");
2490  continue;
2491  }
2492 
2493  if (RTP_PT_IS_RTCP(recvbuf[1]))
2494  continue;
2495 
2496  payload_type = recvbuf[1] & 0x7f;
2497  break;
2498  }
2499  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2500  ffurl_closep(&in);
2501 
2502  par = avcodec_parameters_alloc();
2503  if (!par) {
2504  ret = AVERROR(ENOMEM);
2505  goto fail;
2506  }
2507 
2508  if (ff_rtp_get_codec_info(par, payload_type)) {
2509  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2510  "without an SDP file describing it\n",
2511  payload_type);
2512  goto fail;
2513  }
2514  if (par->codec_type != AVMEDIA_TYPE_DATA) {
2515  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2516  "properly you need an SDP file "
2517  "describing it\n");
2518  }
2519 
2520  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2521  NULL, 0, s->url);
2522 
2524  av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2525  addr.ss_family == AF_INET ? 4 : 6, host);
2526 
2527  p = strchr(s->url, '?');
2528  if (p) {
2529  static const char filters[][2][8] = { { "sources", "incl" },
2530  { "block", "excl" } };
2531  int i;
2532  char *q;
2533  for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2534  if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2535  q = filters_buf;
2536  while ((q = strchr(q, ',')) != NULL)
2537  *q = ' ';
2538  av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2539  filters[i][1],
2540  addr.ss_family == AF_INET ? 4 : 6, host,
2541  filters_buf);
2542  }
2543  }
2544  }
2545 
2546  av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2547  par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2548  par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2549  port, payload_type);
2550  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2551  if (!av_bprint_is_complete(&sdp))
2552  goto fail_nobuf;
2554 
2555  ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2556  s->pb = &pb;
2557 
2558  /* sdp_read_header initializes this again */
2559  ff_network_close();
2560 
2561  rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2562 
2563  ret = sdp_read_header(s);
2564  s->pb = NULL;
2565  av_bprint_finalize(&sdp, NULL);
2566  return ret;
2567 
2568 fail_nobuf:
2569  ret = AVERROR(ENOMEM);
2570  av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2571  av_bprint_finalize(&sdp, NULL);
2572 fail:
2574  ffurl_closep(&in);
2575  ff_network_close();
2576  return ret;
2577 }
2578 
2579 static const AVClass rtp_demuxer_class = {
2580  .class_name = "RTP demuxer",
2581  .item_name = av_default_item_name,
2582  .option = rtp_options,
2583  .version = LIBAVUTIL_VERSION_INT,
2584 };
2585 
2587  .name = "rtp",
2588  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2589  .priv_data_size = sizeof(RTSPState),
2590  .read_probe = rtp_probe,
2591  .read_header = rtp_read_header,
2593  .read_close = sdp_read_close,
2594  .flags = AVFMT_NOFILE,
2595  .priv_class = &rtp_demuxer_class,
2596 };
2597 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:274
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a &#39;$&#39;, stream length and stre...
Definition: rtsp.h:94
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:4802
char crypto_suite[40]
Definition: rtsp.h:478
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:389
#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:579
Bytestream IO Context.
Definition: avio.h:161
Realmedia Data Transport.
Definition: rtsp.h:59
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
void av_bprintf(AVBPrint *buf, const char *fmt,...)
Definition: bprint.c:94
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:518
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:1604
int size
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:307
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1629
AVOption.
Definition: opt.h:246
HTTPS tunneled.
Definition: rtsp.h:45
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:116
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:188
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
const char * filename
Definition: avformat.h:442
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:178
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:318
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:4948
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:587
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:423
const char * desc
Definition: nvenc.c:79
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:73
#define CONFIG_RTPDEC
Definition: config.h:677
Windows Media server.
Definition: rtsp.h:210
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:355
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:807
MpegTSContext * avpriv_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:3322
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:167
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:130
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
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:59
int index
stream index in AVFormatContext
Definition: avformat.h:877
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
#define AVIO_FLAG_READ
read-only
Definition: avio.h:674
char * user_agent
User-Agent string.
Definition: rtsp.h:409
char location[4096]
the "Location:" field.
Definition: rtsp.h:153
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:675
int mode_record
transport set to record data
Definition: rtsp.h:113
enum AVMediaType codec_type
Definition: rtp.c:37
int avio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:694
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:116
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:145
initialized and sending/receiving data
Definition: rtsp.h:198
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:271
const char * key
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:423
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:80
#define freeaddrinfo
Definition: network.h:218
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:455
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:1400
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:421
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:246
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:240
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:187
This struct describes the properties of an encoded stream.
Definition: codec_par.h:52
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.
int av_bprint_finalize(AVBPrint *buf, char **ret_str)
Finalize a print buffer.
Definition: bprint.c:235
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:122
This describes the server response to each RTSP command.
Definition: rtsp.h:128
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream &#39;st&#39;.
Definition: rtpdec.c:536
#define RECVBUF_SIZE
Definition: rtsp.c:60
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:143
Format I/O context.
Definition: avformat.h:1351
#define RTP_PT_PRIVATE
Definition: rtp.h:77
#define COMMON_OPTS()
Definition: rtsp.c:78
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:208
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:404
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:426
int recvbuf_len
Definition: rtsp.h:324
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:179
#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:361
#define CONFIG_RTSP_DEMUXER
Definition: config.h:2295
Standards-compliant RTP.
Definition: rtsp.h:58
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:149
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:75
#define av_malloc(s)
Opaque data information usually continuous.
Definition: avutil.h:203
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:110
static int get_sockaddr(AVFormatContext *s, const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:200
int ff_network_init(void)
Definition: network.c:58
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:1295
AVOptions.
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:473
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:116
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:92
int id
Format-specific stream ID.
Definition: avformat.h:883
enum AVStreamParseType need_parsing
Definition: avformat.h:1094
#define POLL_TIMEOUT_MS
Definition: rtsp.c:56
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:61
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:4526
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1419
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:374
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:329
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:438
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:46
char * protocol_whitelist
&#39;,&#39; separated list of allowed protocols.
Definition: avformat.h:1911
#define CONFIG_RTSP_MUXER
Definition: config.h:2523
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:132
#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:181
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
int av_find_info_tag(char *arg, int arg_size, const char *tag1, const char *info)
Attempt to find a specific tag in a URL.
Definition: parseutils.c:751
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:463
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:69
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 &#39;transports&#39; variable below
Definition: rtsp.h:135
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:625
AVInputFormat ff_rtp_demuxer
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:178
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:2104
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:78
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:458
Private data for the RTSP demuxer.
Definition: rtsp.h:219
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:256
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:294
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
#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:1591
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:635
#define AV_BPRINT_SIZE_UNLIMITED
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:251
const char * protocol_whitelist
Definition: url.h:49
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
ff_const59 struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:1363
void av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max)
Definition: bprint.c:69
char * url
input or output URL.
Definition: avformat.h:1447
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const AVOption ff_rtsp_options[]
Definition: rtsp.c:84
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:203
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:183
Definition: graph2dot.c:48
URLContext * rtsp_hd
Definition: rtsp.h:221
simple assert() macros that are a bit more flexible than ISO C assert().
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:332
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:456
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:572
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
int64_t rtcp_ts_offset
Definition: rtpdec.h:181
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:123
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:226
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:165
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:443
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:444
int seq
RTSP command sequence number.
Definition: rtsp.h:242
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:443
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:340
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1407
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:422
AVDictionary * opts
Definition: movenc.c:50
#define NI_NUMERICHOST
Definition: network.h:195
#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:308
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
const char * name
Definition: qsvenc.c:46
int recvbuf_pos
Definition: rtsp.h:323
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:197
int nb_rtsp_streams
number of items in the &#39;rtsp_streams&#39; variable
Definition: rtsp.h:224
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:284
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
int content_length
length of the data following this header
Definition: rtsp.h:130
int max_streams
The maximum number of streams.
Definition: avformat.h:1953
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:173
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:76
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:89
RTSP over HTTP (tunneling)
Definition: rtsp.h:70
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:143
#define s(width, name)
Definition: cbs_vp9.c:257
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
void ff_format_set_url(AVFormatContext *s, char *url)
Set AVFormatContext url field to the provided pointer.
Definition: utils.c:5855
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:169
static int av_bprint_is_complete(const AVBPrint *buf)
Test if the print buffer is complete (not truncated).
Definition: bprint.h:185
AVDictionary * metadata
Definition: avformat.h:940
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:466
char crypto_params[100]
Definition: rtsp.h:479
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:200
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, const RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:628
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
#define ENC
Definition: rtsp.c:65
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:453
Raw data (over UDP)
Definition: rtsp.h:60
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:322
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
void(* close)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:133
#define FF_ARRAY_ELEMS(a)
int sdp_payload_type
payload type
Definition: rtsp.h:460
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:457
ff_const59 struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:1370
void ffio_free_dyn_buf(AVIOContext **s)
Free a dynamic buffer.
Definition: aviobuf.c:1431
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:545
int ffurl_closep(URLContext **hh)
Close the resource accessed by the URLContext h, and free the memory used by it.
Definition: avio.c:446
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:468
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:38
int nb_byes
Definition: rtsp.h:337
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:263
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:429
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:454
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:253
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:739
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrupt a blocking function associated with cb.
Definition: avio.c:666
int rtp_port_max
Definition: rtsp.h:389
#define NTP_OFFSET
Definition: internal.h:243
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1393
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:384
AVInputFormat ff_sdp_demuxer
int server_port_max
Definition: rtsp.h:106
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:67
static av_always_inline void RENAME() interleave(TYPE *dst, TYPE *src0, TYPE *src1, int w2, int add, int shift)
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:476
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:416
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:259
Definition: url.h:38
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
#define RTSPS_DEFAULT_PORT
Definition: rtsp.h:74
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:676
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:70
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:379
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
double value
Definition: eval.c:98
int client_port_max
Definition: rtsp.h:102
Describe the class of an AVClass context structure.
Definition: log.h:67
#define SDP_MAX_SIZE
Definition: rtsp.c:59
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:343
void * priv_data
Definition: url.h:41
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:469
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:280
#define gai_strerror
Definition: network.h:225
not initialized
Definition: rtsp.h:197
int64_t range_end
Definition: rtsp.h:139
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:119
int avpriv_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:3343
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:156
AVMediaType
Definition: avutil.h:199
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:71
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:753
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:748
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:771
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&#39;t hear from them...
Definition: rtpdec.c:299
#define s1
Definition: regdef.h:38
const char * name
Name of the codec described by this descriptor.
Definition: codec_desc.h:46
#define snprintf
Definition: snprintf.h:34
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:451
int max_p
Definition: rtsp.h:356
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4455
int buffer_size
Definition: rtsp.h:412
This structure contains the data a format has to probe a file.
Definition: avformat.h:441
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:77
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:245
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:94
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
mfxU16 profile
Definition: qsvenc.c:45
This struct describes the properties of a single codec described by an AVCodecID. ...
Definition: codec_desc.h:38
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
#define flags(name, subs,...)
Definition: cbs_av1.c:565
enum RTSPServerType server_type
brand of server that we&#39;re talking to; e.g.
Definition: rtsp.h:268
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:139
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1456
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:232
static int read_probe(const AVProbeData *pd)
Definition: jvdec.c:55
int sample_rate
Audio only.
Definition: codec_par.h:170
#define DEC
Definition: rtsp.c:64
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:453
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: utils.c:2094
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:217
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:114
const OptionDef options[]
Definition: ffmpeg_opt.c:3388
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:130
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:458
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:76
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:289
int av_dict_set_int(AVDictionary **pm, const char *key, int64_t value, int flags)
Convenience wrapper for av_dict_set that converts the value to a string and stores it...
Definition: dict.c:147
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:416
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:60
char default_lang[4]
Definition: rtsp.h:411
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4498
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&#39;re reading data interleave...
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:399
#define getnameinfo
Definition: network.h:219
#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:162
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:277
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:79
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:449
void * priv_data
Format private data.
Definition: avformat.h:1379
static const struct PPFilter filters[]
Definition: postprocess.c:134
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:591
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:880
const AVCodecDescriptor * avcodec_descriptor_get(enum AVCodecID id)
Definition: codec_desc.c:3394
int channels
Audio only.
Definition: codec_par.h:166
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:459
#define MAX_TIMEOUTS
Definition: rtsp.c:58
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1251
char * protocol_blacklist
&#39;,&#39; separated list of disallowed protocols.
Definition: avformat.h:1946
int ai_flags
Definition: network.h:138
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1466
Realmedia-style server.
Definition: rtsp.h:209
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:345
#define av_freep(p)
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:650
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:120
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1023
#define av_malloc_array(a, b)
int pkt_size
Definition: rtsp.h:413
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:905
int interleaved_max
Definition: rtsp.h:447
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:867
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:115
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:101
void avpriv_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:3368
AVStream * st
Definition: rtpdec.h:151
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with a matching codec ID.
Definition: rtpdec.c:160
#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:447
This structure stores compressed data.
Definition: packet.h:332
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:1170
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:106
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:465
static const AVOption rtp_options[]
Definition: rtsp.c:123
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:409
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:439
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define OFFSET(x)
Definition: rtsp.c:63
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with the specified name.
Definition: rtpdec.c:146
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:98
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:440
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:102