Libav
rtsp.c
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1 /*
2  * RTSP/SDP client
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
31 #include "avformat.h"
32 #include "avio_internal.h"
33 
34 #if HAVE_POLL_H
35 #include <poll.h>
36 #endif
37 #include "internal.h"
38 #include "network.h"
39 #include "os_support.h"
40 #include "http.h"
41 #include "rtsp.h"
42 
43 #include "rtpdec.h"
44 #include "rtpproto.h"
45 #include "rdt.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
48 #include "url.h"
49 #include "rtpenc.h"
50 #include "mpegts.h"
51 
52 /* Timeout values for socket poll, in ms,
53  * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
60 
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 
65 #define RTSP_FLAG_OPTS(name, longname) \
66  { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67  { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
68 
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70  { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71  { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72  { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73  { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 
75 #define RTSP_REORDERING_OPTS() \
76  { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
77 
79  { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80  FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81  { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82  { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83  { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84  { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85  { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86  RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87  { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
88  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89  { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90  { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91  { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
93  { NULL },
94 };
95 
96 static const AVOption sdp_options[] = {
97  RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
98  { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
99  { "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
100  RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102  { NULL },
103 };
104 
105 static const AVOption rtp_options[] = {
106  RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108  { NULL },
109 };
110 
111 static void get_word_until_chars(char *buf, int buf_size,
112  const char *sep, const char **pp)
113 {
114  const char *p;
115  char *q;
116 
117  p = *pp;
118  p += strspn(p, SPACE_CHARS);
119  q = buf;
120  while (!strchr(sep, *p) && *p != '\0') {
121  if ((q - buf) < buf_size - 1)
122  *q++ = *p;
123  p++;
124  }
125  if (buf_size > 0)
126  *q = '\0';
127  *pp = p;
128 }
129 
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
131  const char **pp)
132 {
133  if (**pp == '/') (*pp)++;
134  get_word_until_chars(buf, buf_size, sep, pp);
135 }
136 
137 static void get_word(char *buf, int buf_size, const char **pp)
138 {
139  get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
140 }
141 
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
147 {
148  char buf[256];
149 
150  p += strspn(p, SPACE_CHARS);
151  if (!av_stristart(p, "npt=", &p))
152  return;
153 
154  *start = AV_NOPTS_VALUE;
155  *end = AV_NOPTS_VALUE;
156 
157  get_word_sep(buf, sizeof(buf), "-", &p);
158  av_parse_time(start, buf, 1);
159  if (*p == '-') {
160  p++;
161  get_word_sep(buf, sizeof(buf), "-", &p);
162  av_parse_time(end, buf, 1);
163  }
164 }
165 
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
167 {
168  struct addrinfo hints = { 0 }, *ai = NULL;
169  hints.ai_flags = AI_NUMERICHOST;
170  if (getaddrinfo(buf, NULL, &hints, &ai))
171  return -1;
172  memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
173  freeaddrinfo(ai);
174  return 0;
175 }
176 
177 #if CONFIG_RTPDEC
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179  RTSPStream *rtsp_st, AVCodecContext *codec)
180 {
181  if (!handler)
182  return;
183  if (codec)
184  codec->codec_id = handler->codec_id;
185  rtsp_st->dynamic_handler = handler;
186  if (handler->alloc) {
187  rtsp_st->dynamic_protocol_context = handler->alloc();
188  if (!rtsp_st->dynamic_protocol_context)
189  rtsp_st->dynamic_handler = NULL;
190  }
191 }
192 
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195  AVStream *st, RTSPStream *rtsp_st,
196  int payload_type, const char *p)
197 {
198  AVCodecContext *codec = st->codec;
199  char buf[256];
200  int i;
201  AVCodec *c;
202  const char *c_name;
203 
204  /* See if we can handle this kind of payload.
205  * The space should normally not be there but some Real streams or
206  * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207  * have a trailing space. */
208  get_word_sep(buf, sizeof(buf), "/ ", &p);
209  if (payload_type < RTP_PT_PRIVATE) {
210  /* We are in a standard case
211  * (from http://www.iana.org/assignments/rtp-parameters). */
212  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
213  }
214 
215  if (codec->codec_id == AV_CODEC_ID_NONE) {
216  RTPDynamicProtocolHandler *handler =
218  init_rtp_handler(handler, rtsp_st, codec);
219  /* If no dynamic handler was found, check with the list of standard
220  * allocated types, if such a stream for some reason happens to
221  * use a private payload type. This isn't handled in rtpdec.c, since
222  * the format name from the rtpmap line never is passed into rtpdec. */
223  if (!rtsp_st->dynamic_handler)
224  codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
225  }
226 
227  c = avcodec_find_decoder(codec->codec_id);
228  if (c && c->name)
229  c_name = c->name;
230  else
231  c_name = "(null)";
232 
233  get_word_sep(buf, sizeof(buf), "/", &p);
234  i = atoi(buf);
235  switch (codec->codec_type) {
236  case AVMEDIA_TYPE_AUDIO:
237  av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240  if (i > 0) {
241  codec->sample_rate = i;
242  avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243  get_word_sep(buf, sizeof(buf), "/", &p);
244  i = atoi(buf);
245  if (i > 0)
246  codec->channels = i;
247  }
248  av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
249  codec->sample_rate);
250  av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251  codec->channels);
252  break;
253  case AVMEDIA_TYPE_VIDEO:
254  av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
255  if (i > 0)
256  avpriv_set_pts_info(st, 32, 1, i);
257  break;
258  default:
259  break;
260  }
261  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262  rtsp_st->dynamic_handler->init(s, st->index,
263  rtsp_st->dynamic_protocol_context);
264  return 0;
265 }
266 
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268  * is broken out as a function because it is used in rtp_h264.c, which is
269  * forthcoming. */
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271  char *value, int value_size)
272 {
273  *p += strspn(*p, SPACE_CHARS);
274  if (**p) {
275  get_word_sep(attr, attr_size, "=", p);
276  if (**p == '=')
277  (*p)++;
278  get_word_sep(value, value_size, ";", p);
279  if (**p == ';')
280  (*p)++;
281  return 1;
282  }
283  return 0;
284 }
285 
286 typedef struct SDPParseState {
287  /* SDP only */
288  struct sockaddr_storage default_ip;
289  int default_ttl;
290  int skip_media;
291  int nb_default_include_source_addrs;
292  struct RTSPSource **default_include_source_addrs;
293  int nb_default_exclude_source_addrs;
294  struct RTSPSource **default_exclude_source_addrs;
295  int seen_rtpmap;
296  int seen_fmtp;
297  char delayed_fmtp[2048];
298 } SDPParseState;
299 
300 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
301  struct RTSPSource ***dest, int *dest_count)
302 {
303  RTSPSource *rtsp_src, *rtsp_src2;
304  int i;
305  for (i = 0; i < count; i++) {
306  rtsp_src = addrs[i];
307  rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
308  if (!rtsp_src2)
309  continue;
310  memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
311  dynarray_add(dest, dest_count, rtsp_src2);
312  }
313 }
314 
315 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
316  int payload_type, const char *line)
317 {
318  int i;
319 
320  for (i = 0; i < rt->nb_rtsp_streams; i++) {
321  RTSPStream *rtsp_st = rt->rtsp_streams[i];
322  if (rtsp_st->sdp_payload_type == payload_type &&
323  rtsp_st->dynamic_handler &&
324  rtsp_st->dynamic_handler->parse_sdp_a_line) {
325  rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
326  rtsp_st->dynamic_protocol_context, line);
327  }
328  }
329 }
330 
331 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
332  int letter, const char *buf)
333 {
334  RTSPState *rt = s->priv_data;
335  char buf1[64], st_type[64];
336  const char *p;
337  enum AVMediaType codec_type;
338  int payload_type;
339  AVStream *st;
340  RTSPStream *rtsp_st;
341  RTSPSource *rtsp_src;
342  struct sockaddr_storage sdp_ip;
343  int ttl;
344 
345  av_dlog(s, "sdp: %c='%s'\n", letter, buf);
346 
347  p = buf;
348  if (s1->skip_media && letter != 'm')
349  return;
350  switch (letter) {
351  case 'c':
352  get_word(buf1, sizeof(buf1), &p);
353  if (strcmp(buf1, "IN") != 0)
354  return;
355  get_word(buf1, sizeof(buf1), &p);
356  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
357  return;
358  get_word_sep(buf1, sizeof(buf1), "/", &p);
359  if (get_sockaddr(buf1, &sdp_ip))
360  return;
361  ttl = 16;
362  if (*p == '/') {
363  p++;
364  get_word_sep(buf1, sizeof(buf1), "/", &p);
365  ttl = atoi(buf1);
366  }
367  if (s->nb_streams == 0) {
368  s1->default_ip = sdp_ip;
369  s1->default_ttl = ttl;
370  } else {
371  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
372  rtsp_st->sdp_ip = sdp_ip;
373  rtsp_st->sdp_ttl = ttl;
374  }
375  break;
376  case 's':
377  av_dict_set(&s->metadata, "title", p, 0);
378  break;
379  case 'i':
380  if (s->nb_streams == 0) {
381  av_dict_set(&s->metadata, "comment", p, 0);
382  break;
383  }
384  break;
385  case 'm':
386  /* new stream */
387  s1->skip_media = 0;
388  s1->seen_fmtp = 0;
389  s1->seen_rtpmap = 0;
390  codec_type = AVMEDIA_TYPE_UNKNOWN;
391  get_word(st_type, sizeof(st_type), &p);
392  if (!strcmp(st_type, "audio")) {
393  codec_type = AVMEDIA_TYPE_AUDIO;
394  } else if (!strcmp(st_type, "video")) {
395  codec_type = AVMEDIA_TYPE_VIDEO;
396  } else if (!strcmp(st_type, "application")) {
397  codec_type = AVMEDIA_TYPE_DATA;
398  }
399  if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
400  s1->skip_media = 1;
401  return;
402  }
403  rtsp_st = av_mallocz(sizeof(RTSPStream));
404  if (!rtsp_st)
405  return;
406  rtsp_st->stream_index = -1;
407  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
408 
409  rtsp_st->sdp_ip = s1->default_ip;
410  rtsp_st->sdp_ttl = s1->default_ttl;
411 
412  copy_default_source_addrs(s1->default_include_source_addrs,
413  s1->nb_default_include_source_addrs,
414  &rtsp_st->include_source_addrs,
415  &rtsp_st->nb_include_source_addrs);
416  copy_default_source_addrs(s1->default_exclude_source_addrs,
417  s1->nb_default_exclude_source_addrs,
418  &rtsp_st->exclude_source_addrs,
419  &rtsp_st->nb_exclude_source_addrs);
420 
421  get_word(buf1, sizeof(buf1), &p); /* port */
422  rtsp_st->sdp_port = atoi(buf1);
423 
424  get_word(buf1, sizeof(buf1), &p); /* protocol */
425  if (!strcmp(buf1, "udp"))
427  else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
428  rtsp_st->feedback = 1;
429 
430  /* XXX: handle list of formats */
431  get_word(buf1, sizeof(buf1), &p); /* format list */
432  rtsp_st->sdp_payload_type = atoi(buf1);
433 
434  if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
435  /* no corresponding stream */
436  if (rt->transport == RTSP_TRANSPORT_RAW) {
437  if (!rt->ts && CONFIG_RTPDEC)
438  rt->ts = ff_mpegts_parse_open(s);
439  } else {
440  RTPDynamicProtocolHandler *handler;
441  handler = ff_rtp_handler_find_by_id(
443  init_rtp_handler(handler, rtsp_st, NULL);
444  if (handler && handler->init)
445  handler->init(s, -1, rtsp_st->dynamic_protocol_context);
446  }
447  } else if (rt->server_type == RTSP_SERVER_WMS &&
448  codec_type == AVMEDIA_TYPE_DATA) {
449  /* RTX stream, a stream that carries all the other actual
450  * audio/video streams. Don't expose this to the callers. */
451  } else {
452  st = avformat_new_stream(s, NULL);
453  if (!st)
454  return;
455  st->id = rt->nb_rtsp_streams - 1;
456  rtsp_st->stream_index = st->index;
457  st->codec->codec_type = codec_type;
458  if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
459  RTPDynamicProtocolHandler *handler;
460  /* if standard payload type, we can find the codec right now */
462  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
463  st->codec->sample_rate > 0)
464  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
465  /* Even static payload types may need a custom depacketizer */
466  handler = ff_rtp_handler_find_by_id(
467  rtsp_st->sdp_payload_type, st->codec->codec_type);
468  init_rtp_handler(handler, rtsp_st, st->codec);
469  if (handler && handler->init)
470  handler->init(s, st->index,
471  rtsp_st->dynamic_protocol_context);
472  }
473  }
474  /* put a default control url */
475  av_strlcpy(rtsp_st->control_url, rt->control_uri,
476  sizeof(rtsp_st->control_url));
477  break;
478  case 'a':
479  if (av_strstart(p, "control:", &p)) {
480  if (s->nb_streams == 0) {
481  if (!strncmp(p, "rtsp://", 7))
482  av_strlcpy(rt->control_uri, p,
483  sizeof(rt->control_uri));
484  } else {
485  char proto[32];
486  /* get the control url */
487  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
488 
489  /* XXX: may need to add full url resolution */
490  av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
491  NULL, NULL, 0, p);
492  if (proto[0] == '\0') {
493  /* relative control URL */
494  if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
495  av_strlcat(rtsp_st->control_url, "/",
496  sizeof(rtsp_st->control_url));
497  av_strlcat(rtsp_st->control_url, p,
498  sizeof(rtsp_st->control_url));
499  } else
500  av_strlcpy(rtsp_st->control_url, p,
501  sizeof(rtsp_st->control_url));
502  }
503  } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
504  /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
505  get_word(buf1, sizeof(buf1), &p);
506  payload_type = atoi(buf1);
507  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
508  if (rtsp_st->stream_index >= 0) {
509  st = s->streams[rtsp_st->stream_index];
510  sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
511  }
512  s1->seen_rtpmap = 1;
513  if (s1->seen_fmtp) {
514  parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
515  }
516  } else if (av_strstart(p, "fmtp:", &p) ||
517  av_strstart(p, "framesize:", &p)) {
518  // let dynamic protocol handlers have a stab at the line.
519  get_word(buf1, sizeof(buf1), &p);
520  payload_type = atoi(buf1);
521  if (s1->seen_rtpmap) {
522  parse_fmtp(s, rt, payload_type, buf);
523  } else {
524  s1->seen_fmtp = 1;
525  av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
526  }
527  } else if (av_strstart(p, "range:", &p)) {
528  int64_t start, end;
529 
530  // this is so that seeking on a streamed file can work.
531  rtsp_parse_range_npt(p, &start, &end);
532  s->start_time = start;
533  /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
534  s->duration = (end == AV_NOPTS_VALUE) ?
535  AV_NOPTS_VALUE : end - start;
536  } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
537  if (atoi(p) == 1)
539  } else if (av_strstart(p, "SampleRate:integer;", &p) &&
540  s->nb_streams > 0) {
541  st = s->streams[s->nb_streams - 1];
542  st->codec->sample_rate = atoi(p);
543  } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
544  // RFC 4568
545  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
546  get_word(buf1, sizeof(buf1), &p); // ignore tag
547  get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
548  p += strspn(p, SPACE_CHARS);
549  if (av_strstart(p, "inline:", &p))
550  get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
551  } else if (av_strstart(p, "source-filter:", &p)) {
552  int exclude = 0;
553  get_word(buf1, sizeof(buf1), &p);
554  if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
555  return;
556  exclude = !strcmp(buf1, "excl");
557 
558  get_word(buf1, sizeof(buf1), &p);
559  if (strcmp(buf1, "IN") != 0)
560  return;
561  get_word(buf1, sizeof(buf1), &p);
562  if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
563  return;
564  // not checking that the destination address actually matches or is wildcard
565  get_word(buf1, sizeof(buf1), &p);
566 
567  while (*p != '\0') {
568  rtsp_src = av_mallocz(sizeof(*rtsp_src));
569  if (!rtsp_src)
570  return;
571  get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
572  if (exclude) {
573  if (s->nb_streams == 0) {
574  dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
575  } else {
576  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
577  dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
578  }
579  } else {
580  if (s->nb_streams == 0) {
581  dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
582  } else {
583  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
584  dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
585  }
586  }
587  }
588  } else {
589  if (rt->server_type == RTSP_SERVER_WMS)
591  if (s->nb_streams > 0) {
592  rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593 
594  if (rt->server_type == RTSP_SERVER_REAL)
595  ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
596 
597  if (rtsp_st->dynamic_handler &&
599  rtsp_st->dynamic_handler->parse_sdp_a_line(s,
600  rtsp_st->stream_index,
601  rtsp_st->dynamic_protocol_context, buf);
602  }
603  }
604  break;
605  }
606 }
607 
608 int ff_sdp_parse(AVFormatContext *s, const char *content)
609 {
610  RTSPState *rt = s->priv_data;
611  const char *p;
612  int letter, i;
613  /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
614  * contain long SDP lines containing complete ASF Headers (several
615  * kB) or arrays of MDPR (RM stream descriptor) headers plus
616  * "rulebooks" describing their properties. Therefore, the SDP line
617  * buffer is large.
618  *
619  * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
620  * in rtpdec_xiph.c. */
621  char buf[16384], *q;
622  SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
623 
624  p = content;
625  for (;;) {
626  p += strspn(p, SPACE_CHARS);
627  letter = *p;
628  if (letter == '\0')
629  break;
630  p++;
631  if (*p != '=')
632  goto next_line;
633  p++;
634  /* get the content */
635  q = buf;
636  while (*p != '\n' && *p != '\r' && *p != '\0') {
637  if ((q - buf) < sizeof(buf) - 1)
638  *q++ = *p;
639  p++;
640  }
641  *q = '\0';
642  sdp_parse_line(s, s1, letter, buf);
643  next_line:
644  while (*p != '\n' && *p != '\0')
645  p++;
646  if (*p == '\n')
647  p++;
648  }
649 
650  for (i = 0; i < s1->nb_default_include_source_addrs; i++)
651  av_free(s1->default_include_source_addrs[i]);
652  av_freep(&s1->default_include_source_addrs);
653  for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
654  av_free(s1->default_exclude_source_addrs[i]);
655  av_freep(&s1->default_exclude_source_addrs);
656 
657  rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
658  if (!rt->p) return AVERROR(ENOMEM);
659  return 0;
660 }
661 #endif /* CONFIG_RTPDEC */
662 
663 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
664 {
665  RTSPState *rt = s->priv_data;
666  int i;
667 
668  for (i = 0; i < rt->nb_rtsp_streams; i++) {
669  RTSPStream *rtsp_st = rt->rtsp_streams[i];
670  if (!rtsp_st)
671  continue;
672  if (rtsp_st->transport_priv) {
673  if (s->oformat) {
674  AVFormatContext *rtpctx = rtsp_st->transport_priv;
675  av_write_trailer(rtpctx);
677  uint8_t *ptr;
678  if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
679  ff_rtsp_tcp_write_packet(s, rtsp_st);
680  avio_close_dyn_buf(rtpctx->pb, &ptr);
681  av_free(ptr);
682  } else {
683  avio_close(rtpctx->pb);
684  }
685  avformat_free_context(rtpctx);
686  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
688  else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
690  }
691  rtsp_st->transport_priv = NULL;
692  if (rtsp_st->rtp_handle)
693  ffurl_close(rtsp_st->rtp_handle);
694  rtsp_st->rtp_handle = NULL;
695  }
696 }
697 
698 /* close and free RTSP streams */
700 {
701  RTSPState *rt = s->priv_data;
702  int i, j;
703  RTSPStream *rtsp_st;
704 
705  ff_rtsp_undo_setup(s, 0);
706  for (i = 0; i < rt->nb_rtsp_streams; i++) {
707  rtsp_st = rt->rtsp_streams[i];
708  if (rtsp_st) {
709  if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
710  rtsp_st->dynamic_handler->free(
711  rtsp_st->dynamic_protocol_context);
712  for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
713  av_free(rtsp_st->include_source_addrs[j]);
714  av_freep(&rtsp_st->include_source_addrs);
715  for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
716  av_free(rtsp_st->exclude_source_addrs[j]);
717  av_freep(&rtsp_st->exclude_source_addrs);
718 
719  av_free(rtsp_st);
720  }
721  }
722  av_free(rt->rtsp_streams);
723  if (rt->asf_ctx) {
725  }
726  if (rt->ts && CONFIG_RTPDEC)
728  av_free(rt->p);
729  av_free(rt->recvbuf);
730 }
731 
733 {
734  RTSPState *rt = s->priv_data;
735  AVStream *st = NULL;
736  int reordering_queue_size = rt->reordering_queue_size;
737  if (reordering_queue_size < 0) {
739  reordering_queue_size = 0;
740  else
741  reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
742  }
743 
744  /* open the RTP context */
745  if (rtsp_st->stream_index >= 0)
746  st = s->streams[rtsp_st->stream_index];
747  if (!st)
749 
750  if (s->oformat && CONFIG_RTSP_MUXER) {
751  int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
752  s, st, rtsp_st->rtp_handle,
754  rtsp_st->stream_index);
755  /* Ownership of rtp_handle is passed to the rtp mux context */
756  rtsp_st->rtp_handle = NULL;
757  if (ret < 0)
758  return ret;
759  st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
760  } else if (rt->transport == RTSP_TRANSPORT_RAW) {
761  return 0; // Don't need to open any parser here
762  } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
763  rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
764  rtsp_st->dynamic_protocol_context,
765  rtsp_st->dynamic_handler);
766  else if (CONFIG_RTPDEC)
767  rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
768  rtsp_st->sdp_payload_type,
769  reordering_queue_size);
770 
771  if (!rtsp_st->transport_priv) {
772  return AVERROR(ENOMEM);
773  } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
774  if (rtsp_st->dynamic_handler) {
776  rtsp_st->dynamic_protocol_context,
777  rtsp_st->dynamic_handler);
778  }
779  if (rtsp_st->crypto_suite[0])
781  rtsp_st->crypto_suite,
782  rtsp_st->crypto_params);
783  }
784 
785  return 0;
786 }
787 
788 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
789 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
790 {
791  const char *q;
792  char *p;
793  int v;
794 
795  q = *pp;
796  q += strspn(q, SPACE_CHARS);
797  v = strtol(q, &p, 10);
798  if (*p == '-') {
799  p++;
800  *min_ptr = v;
801  v = strtol(p, &p, 10);
802  *max_ptr = v;
803  } else {
804  *min_ptr = v;
805  *max_ptr = v;
806  }
807  *pp = p;
808 }
809 
810 /* XXX: only one transport specification is parsed */
811 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
812 {
813  char transport_protocol[16];
814  char profile[16];
815  char lower_transport[16];
816  char parameter[16];
817  RTSPTransportField *th;
818  char buf[256];
819 
820  reply->nb_transports = 0;
821 
822  for (;;) {
823  p += strspn(p, SPACE_CHARS);
824  if (*p == '\0')
825  break;
826 
827  th = &reply->transports[reply->nb_transports];
828 
829  get_word_sep(transport_protocol, sizeof(transport_protocol),
830  "/", &p);
831  if (!av_strcasecmp (transport_protocol, "rtp")) {
832  get_word_sep(profile, sizeof(profile), "/;,", &p);
833  lower_transport[0] = '\0';
834  /* rtp/avp/<protocol> */
835  if (*p == '/') {
836  get_word_sep(lower_transport, sizeof(lower_transport),
837  ";,", &p);
838  }
840  } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
841  !av_strcasecmp (transport_protocol, "x-real-rdt")) {
842  /* x-pn-tng/<protocol> */
843  get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
844  profile[0] = '\0';
846  } else if (!av_strcasecmp(transport_protocol, "raw")) {
847  get_word_sep(profile, sizeof(profile), "/;,", &p);
848  lower_transport[0] = '\0';
849  /* raw/raw/<protocol> */
850  if (*p == '/') {
851  get_word_sep(lower_transport, sizeof(lower_transport),
852  ";,", &p);
853  }
855  }
856  if (!av_strcasecmp(lower_transport, "TCP"))
858  else
860 
861  if (*p == ';')
862  p++;
863  /* get each parameter */
864  while (*p != '\0' && *p != ',') {
865  get_word_sep(parameter, sizeof(parameter), "=;,", &p);
866  if (!strcmp(parameter, "port")) {
867  if (*p == '=') {
868  p++;
869  rtsp_parse_range(&th->port_min, &th->port_max, &p);
870  }
871  } else if (!strcmp(parameter, "client_port")) {
872  if (*p == '=') {
873  p++;
874  rtsp_parse_range(&th->client_port_min,
875  &th->client_port_max, &p);
876  }
877  } else if (!strcmp(parameter, "server_port")) {
878  if (*p == '=') {
879  p++;
880  rtsp_parse_range(&th->server_port_min,
881  &th->server_port_max, &p);
882  }
883  } else if (!strcmp(parameter, "interleaved")) {
884  if (*p == '=') {
885  p++;
886  rtsp_parse_range(&th->interleaved_min,
887  &th->interleaved_max, &p);
888  }
889  } else if (!strcmp(parameter, "multicast")) {
892  } else if (!strcmp(parameter, "ttl")) {
893  if (*p == '=') {
894  char *end;
895  p++;
896  th->ttl = strtol(p, &end, 10);
897  p = end;
898  }
899  } else if (!strcmp(parameter, "destination")) {
900  if (*p == '=') {
901  p++;
902  get_word_sep(buf, sizeof(buf), ";,", &p);
903  get_sockaddr(buf, &th->destination);
904  }
905  } else if (!strcmp(parameter, "source")) {
906  if (*p == '=') {
907  p++;
908  get_word_sep(buf, sizeof(buf), ";,", &p);
909  av_strlcpy(th->source, buf, sizeof(th->source));
910  }
911  } else if (!strcmp(parameter, "mode")) {
912  if (*p == '=') {
913  p++;
914  get_word_sep(buf, sizeof(buf), ";, ", &p);
915  if (!strcmp(buf, "record") ||
916  !strcmp(buf, "receive"))
917  th->mode_record = 1;
918  }
919  }
920 
921  while (*p != ';' && *p != '\0' && *p != ',')
922  p++;
923  if (*p == ';')
924  p++;
925  }
926  if (*p == ',')
927  p++;
928 
929  reply->nb_transports++;
930  }
931 }
932 
933 static void handle_rtp_info(RTSPState *rt, const char *url,
934  uint32_t seq, uint32_t rtptime)
935 {
936  int i;
937  if (!rtptime || !url[0])
938  return;
939  if (rt->transport != RTSP_TRANSPORT_RTP)
940  return;
941  for (i = 0; i < rt->nb_rtsp_streams; i++) {
942  RTSPStream *rtsp_st = rt->rtsp_streams[i];
943  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
944  if (!rtpctx)
945  continue;
946  if (!strcmp(rtsp_st->control_url, url)) {
947  rtpctx->base_timestamp = rtptime;
948  break;
949  }
950  }
951 }
952 
953 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
954 {
955  int read = 0;
956  char key[20], value[1024], url[1024] = "";
957  uint32_t seq = 0, rtptime = 0;
958 
959  for (;;) {
960  p += strspn(p, SPACE_CHARS);
961  if (!*p)
962  break;
963  get_word_sep(key, sizeof(key), "=", &p);
964  if (*p != '=')
965  break;
966  p++;
967  get_word_sep(value, sizeof(value), ";, ", &p);
968  read++;
969  if (!strcmp(key, "url"))
970  av_strlcpy(url, value, sizeof(url));
971  else if (!strcmp(key, "seq"))
972  seq = strtoul(value, NULL, 10);
973  else if (!strcmp(key, "rtptime"))
974  rtptime = strtoul(value, NULL, 10);
975  if (*p == ',') {
976  handle_rtp_info(rt, url, seq, rtptime);
977  url[0] = '\0';
978  seq = rtptime = 0;
979  read = 0;
980  }
981  if (*p)
982  p++;
983  }
984  if (read > 0)
985  handle_rtp_info(rt, url, seq, rtptime);
986 }
987 
988 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
989  RTSPState *rt, const char *method)
990 {
991  const char *p;
992 
993  /* NOTE: we do case independent match for broken servers */
994  p = buf;
995  if (av_stristart(p, "Session:", &p)) {
996  int t;
997  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
998  if (av_stristart(p, ";timeout=", &p) &&
999  (t = strtol(p, NULL, 10)) > 0) {
1000  reply->timeout = t;
1001  }
1002  } else if (av_stristart(p, "Content-Length:", &p)) {
1003  reply->content_length = strtol(p, NULL, 10);
1004  } else if (av_stristart(p, "Transport:", &p)) {
1005  rtsp_parse_transport(reply, p);
1006  } else if (av_stristart(p, "CSeq:", &p)) {
1007  reply->seq = strtol(p, NULL, 10);
1008  } else if (av_stristart(p, "Range:", &p)) {
1009  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1010  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1011  p += strspn(p, SPACE_CHARS);
1012  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1013  } else if (av_stristart(p, "Server:", &p)) {
1014  p += strspn(p, SPACE_CHARS);
1015  av_strlcpy(reply->server, p, sizeof(reply->server));
1016  } else if (av_stristart(p, "Notice:", &p) ||
1017  av_stristart(p, "X-Notice:", &p)) {
1018  reply->notice = strtol(p, NULL, 10);
1019  } else if (av_stristart(p, "Location:", &p)) {
1020  p += strspn(p, SPACE_CHARS);
1021  av_strlcpy(reply->location, p , sizeof(reply->location));
1022  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1023  p += strspn(p, SPACE_CHARS);
1024  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1025  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1026  p += strspn(p, SPACE_CHARS);
1027  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1028  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1029  p += strspn(p, SPACE_CHARS);
1030  if (method && !strcmp(method, "DESCRIBE"))
1031  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1032  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1033  p += strspn(p, SPACE_CHARS);
1034  if (method && !strcmp(method, "PLAY"))
1035  rtsp_parse_rtp_info(rt, p);
1036  } else if (av_stristart(p, "Public:", &p) && rt) {
1037  if (strstr(p, "GET_PARAMETER") &&
1038  method && !strcmp(method, "OPTIONS"))
1039  rt->get_parameter_supported = 1;
1040  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1041  p += strspn(p, SPACE_CHARS);
1042  rt->accept_dynamic_rate = atoi(p);
1043  } else if (av_stristart(p, "Content-Type:", &p)) {
1044  p += strspn(p, SPACE_CHARS);
1045  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1046  }
1047 }
1048 
1049 /* skip a RTP/TCP interleaved packet */
1051 {
1052  RTSPState *rt = s->priv_data;
1053  int ret, len, len1;
1054  uint8_t buf[1024];
1055 
1056  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1057  if (ret != 3)
1058  return;
1059  len = AV_RB16(buf + 1);
1060 
1061  av_dlog(s, "skipping RTP packet len=%d\n", len);
1062 
1063  /* skip payload */
1064  while (len > 0) {
1065  len1 = len;
1066  if (len1 > sizeof(buf))
1067  len1 = sizeof(buf);
1068  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1069  if (ret != len1)
1070  return;
1071  len -= len1;
1072  }
1073 }
1074 
1076  unsigned char **content_ptr,
1077  int return_on_interleaved_data, const char *method)
1078 {
1079  RTSPState *rt = s->priv_data;
1080  char buf[4096], buf1[1024], *q;
1081  unsigned char ch;
1082  const char *p;
1083  int ret, content_length, line_count = 0, request = 0;
1084  unsigned char *content = NULL;
1085 
1086 start:
1087  line_count = 0;
1088  request = 0;
1089  content = NULL;
1090  memset(reply, 0, sizeof(*reply));
1091 
1092  /* parse reply (XXX: use buffers) */
1093  rt->last_reply[0] = '\0';
1094  for (;;) {
1095  q = buf;
1096  for (;;) {
1097  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1098  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1099  if (ret != 1)
1100  return AVERROR_EOF;
1101  if (ch == '\n')
1102  break;
1103  if (ch == '$') {
1104  /* XXX: only parse it if first char on line ? */
1105  if (return_on_interleaved_data) {
1106  return 1;
1107  } else
1109  } else if (ch != '\r') {
1110  if ((q - buf) < sizeof(buf) - 1)
1111  *q++ = ch;
1112  }
1113  }
1114  *q = '\0';
1115 
1116  av_dlog(s, "line='%s'\n", buf);
1117 
1118  /* test if last line */
1119  if (buf[0] == '\0')
1120  break;
1121  p = buf;
1122  if (line_count == 0) {
1123  /* get reply code */
1124  get_word(buf1, sizeof(buf1), &p);
1125  if (!strncmp(buf1, "RTSP/", 5)) {
1126  get_word(buf1, sizeof(buf1), &p);
1127  reply->status_code = atoi(buf1);
1128  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1129  } else {
1130  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1131  get_word(buf1, sizeof(buf1), &p); // object
1132  request = 1;
1133  }
1134  } else {
1135  ff_rtsp_parse_line(reply, p, rt, method);
1136  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1137  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1138  }
1139  line_count++;
1140  }
1141 
1142  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1143  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1144 
1145  content_length = reply->content_length;
1146  if (content_length > 0) {
1147  /* leave some room for a trailing '\0' (useful for simple parsing) */
1148  content = av_malloc(content_length + 1);
1149  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1150  content[content_length] = '\0';
1151  }
1152  if (content_ptr)
1153  *content_ptr = content;
1154  else
1155  av_free(content);
1156 
1157  if (request) {
1158  char buf[1024];
1159  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1160  const char* ptr = buf;
1161 
1162  if (!strcmp(reply->reason, "OPTIONS")) {
1163  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1164  if (reply->seq)
1165  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1166  if (reply->session_id[0])
1167  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1168  reply->session_id);
1169  } else {
1170  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1171  }
1172  av_strlcat(buf, "\r\n", sizeof(buf));
1173 
1174  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1175  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1176  ptr = base64buf;
1177  }
1178  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1179 
1180  rt->last_cmd_time = av_gettime();
1181  /* Even if the request from the server had data, it is not the data
1182  * that the caller wants or expects. The memory could also be leaked
1183  * if the actual following reply has content data. */
1184  if (content_ptr)
1185  av_freep(content_ptr);
1186  /* If method is set, this is called from ff_rtsp_send_cmd,
1187  * where a reply to exactly this request is awaited. For
1188  * callers from within packet receiving, we just want to
1189  * return to the caller and go back to receiving packets. */
1190  if (method)
1191  goto start;
1192  return 0;
1193  }
1194 
1195  if (rt->seq != reply->seq) {
1196  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1197  rt->seq, reply->seq);
1198  }
1199 
1200  /* EOS */
1201  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1202  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1203  reply->notice == 2306 /* Continuous Feed Terminated */) {
1204  rt->state = RTSP_STATE_IDLE;
1205  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1206  return AVERROR(EIO); /* data or server error */
1207  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1208  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1209  return AVERROR(EPERM);
1210 
1211  return 0;
1212 }
1213 
1227 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1228  const char *method, const char *url,
1229  const char *headers,
1230  const unsigned char *send_content,
1231  int send_content_length)
1232 {
1233  RTSPState *rt = s->priv_data;
1234  char buf[4096], *out_buf;
1235  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1236 
1237  /* Add in RTSP headers */
1238  out_buf = buf;
1239  rt->seq++;
1240  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1241  if (headers)
1242  av_strlcat(buf, headers, sizeof(buf));
1243  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1244  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1245  if (rt->session_id[0] != '\0' && (!headers ||
1246  !strstr(headers, "\nIf-Match:"))) {
1247  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1248  }
1249  if (rt->auth[0]) {
1250  char *str = ff_http_auth_create_response(&rt->auth_state,
1251  rt->auth, url, method);
1252  if (str)
1253  av_strlcat(buf, str, sizeof(buf));
1254  av_free(str);
1255  }
1256  if (send_content_length > 0 && send_content)
1257  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1258  av_strlcat(buf, "\r\n", sizeof(buf));
1259 
1260  /* base64 encode rtsp if tunneling */
1261  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1262  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1263  out_buf = base64buf;
1264  }
1265 
1266  av_dlog(s, "Sending:\n%s--\n", buf);
1267 
1268  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1269  if (send_content_length > 0 && send_content) {
1270  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1271  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1272  "with content data not supported\n");
1273  return AVERROR_PATCHWELCOME;
1274  }
1275  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1276  }
1277  rt->last_cmd_time = av_gettime();
1278 
1279  return 0;
1280 }
1281 
1282 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1283  const char *url, const char *headers)
1284 {
1285  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1286 }
1287 
1288 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1289  const char *headers, RTSPMessageHeader *reply,
1290  unsigned char **content_ptr)
1291 {
1292  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1293  content_ptr, NULL, 0);
1294 }
1295 
1297  const char *method, const char *url,
1298  const char *header,
1299  RTSPMessageHeader *reply,
1300  unsigned char **content_ptr,
1301  const unsigned char *send_content,
1302  int send_content_length)
1303 {
1304  RTSPState *rt = s->priv_data;
1305  HTTPAuthType cur_auth_type;
1306  int ret, attempts = 0;
1307 
1308 retry:
1309  cur_auth_type = rt->auth_state.auth_type;
1310  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1311  send_content,
1312  send_content_length)))
1313  return ret;
1314 
1315  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1316  return ret;
1317  attempts++;
1318 
1319  if (reply->status_code == 401 &&
1320  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1321  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1322  goto retry;
1323 
1324  if (reply->status_code > 400){
1325  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1326  method,
1327  reply->status_code,
1328  reply->reason);
1329  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1330  }
1331 
1332  return 0;
1333 }
1334 
1335 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1336  int lower_transport, const char *real_challenge)
1337 {
1338  RTSPState *rt = s->priv_data;
1339  int rtx = 0, j, i, err, interleave = 0, port_off;
1340  RTSPStream *rtsp_st;
1341  RTSPMessageHeader reply1, *reply = &reply1;
1342  char cmd[2048];
1343  const char *trans_pref;
1344 
1345  if (rt->transport == RTSP_TRANSPORT_RDT)
1346  trans_pref = "x-pn-tng";
1347  else if (rt->transport == RTSP_TRANSPORT_RAW)
1348  trans_pref = "RAW/RAW";
1349  else
1350  trans_pref = "RTP/AVP";
1351 
1352  /* default timeout: 1 minute */
1353  rt->timeout = 60;
1354 
1355  /* for each stream, make the setup request */
1356  /* XXX: we assume the same server is used for the control of each
1357  * RTSP stream */
1358 
1359  /* Choose a random starting offset within the first half of the
1360  * port range, to allow for a number of ports to try even if the offset
1361  * happens to be at the end of the random range. */
1362  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1363  /* even random offset */
1364  port_off -= port_off & 0x01;
1365 
1366  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1367  char transport[2048];
1368 
1369  /*
1370  * WMS serves all UDP data over a single connection, the RTX, which
1371  * isn't necessarily the first in the SDP but has to be the first
1372  * to be set up, else the second/third SETUP will fail with a 461.
1373  */
1374  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1375  rt->server_type == RTSP_SERVER_WMS) {
1376  if (i == 0) {
1377  /* rtx first */
1378  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1379  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1380  if (len >= 4 &&
1381  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1382  "/rtx"))
1383  break;
1384  }
1385  if (rtx == rt->nb_rtsp_streams)
1386  return -1; /* no RTX found */
1387  rtsp_st = rt->rtsp_streams[rtx];
1388  } else
1389  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1390  } else
1391  rtsp_st = rt->rtsp_streams[i];
1392 
1393  /* RTP/UDP */
1394  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1395  char buf[256];
1396 
1397  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1398  port = reply->transports[0].client_port_min;
1399  goto have_port;
1400  }
1401 
1402  /* first try in specified port range */
1403  while (j <= rt->rtp_port_max) {
1404  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1405  "?localport=%d", j);
1406  /* we will use two ports per rtp stream (rtp and rtcp) */
1407  j += 2;
1408  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1409  &s->interrupt_callback, NULL))
1410  goto rtp_opened;
1411  }
1412 
1413  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1414  err = AVERROR(EIO);
1415  goto fail;
1416 
1417  rtp_opened:
1418  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1419  have_port:
1420  snprintf(transport, sizeof(transport) - 1,
1421  "%s/UDP;", trans_pref);
1422  if (rt->server_type != RTSP_SERVER_REAL)
1423  av_strlcat(transport, "unicast;", sizeof(transport));
1424  av_strlcatf(transport, sizeof(transport),
1425  "client_port=%d", port);
1426  if (rt->transport == RTSP_TRANSPORT_RTP &&
1427  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1428  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1429  }
1430 
1431  /* RTP/TCP */
1432  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1433  /* For WMS streams, the application streams are only used for
1434  * UDP. When trying to set it up for TCP streams, the server
1435  * will return an error. Therefore, we skip those streams. */
1436  if (rt->server_type == RTSP_SERVER_WMS &&
1437  (rtsp_st->stream_index < 0 ||
1438  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1440  continue;
1441  snprintf(transport, sizeof(transport) - 1,
1442  "%s/TCP;", trans_pref);
1443  if (rt->transport != RTSP_TRANSPORT_RDT)
1444  av_strlcat(transport, "unicast;", sizeof(transport));
1445  av_strlcatf(transport, sizeof(transport),
1446  "interleaved=%d-%d",
1447  interleave, interleave + 1);
1448  interleave += 2;
1449  }
1450 
1451  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1452  snprintf(transport, sizeof(transport) - 1,
1453  "%s/UDP;multicast", trans_pref);
1454  }
1455  if (s->oformat) {
1456  av_strlcat(transport, ";mode=record", sizeof(transport));
1457  } else if (rt->server_type == RTSP_SERVER_REAL ||
1459  av_strlcat(transport, ";mode=play", sizeof(transport));
1460  snprintf(cmd, sizeof(cmd),
1461  "Transport: %s\r\n",
1462  transport);
1463  if (rt->accept_dynamic_rate)
1464  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1465  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1466  char real_res[41], real_csum[9];
1467  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1468  real_challenge);
1469  av_strlcatf(cmd, sizeof(cmd),
1470  "If-Match: %s\r\n"
1471  "RealChallenge2: %s, sd=%s\r\n",
1472  rt->session_id, real_res, real_csum);
1473  }
1474  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1475  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1476  err = 1;
1477  goto fail;
1478  } else if (reply->status_code != RTSP_STATUS_OK ||
1479  reply->nb_transports != 1) {
1480  err = AVERROR_INVALIDDATA;
1481  goto fail;
1482  }
1483 
1484  /* XXX: same protocol for all streams is required */
1485  if (i > 0) {
1486  if (reply->transports[0].lower_transport != rt->lower_transport ||
1487  reply->transports[0].transport != rt->transport) {
1488  err = AVERROR_INVALIDDATA;
1489  goto fail;
1490  }
1491  } else {
1492  rt->lower_transport = reply->transports[0].lower_transport;
1493  rt->transport = reply->transports[0].transport;
1494  }
1495 
1496  /* Fail if the server responded with another lower transport mode
1497  * than what we requested. */
1498  if (reply->transports[0].lower_transport != lower_transport) {
1499  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1500  err = AVERROR_INVALIDDATA;
1501  goto fail;
1502  }
1503 
1504  switch(reply->transports[0].lower_transport) {
1506  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1507  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1508  break;
1509 
1510  case RTSP_LOWER_TRANSPORT_UDP: {
1511  char url[1024], options[30] = "";
1512  const char *peer = host;
1513 
1514  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1515  av_strlcpy(options, "?connect=1", sizeof(options));
1516  /* Use source address if specified */
1517  if (reply->transports[0].source[0])
1518  peer = reply->transports[0].source;
1519  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1520  reply->transports[0].server_port_min, "%s", options);
1521  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1522  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1523  err = AVERROR_INVALIDDATA;
1524  goto fail;
1525  }
1526  /* Try to initialize the connection state in a
1527  * potential NAT router by sending dummy packets.
1528  * RTP/RTCP dummy packets are used for RDT, too.
1529  */
1530  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1531  CONFIG_RTPDEC)
1533  break;
1534  }
1536  char url[1024], namebuf[50], optbuf[20] = "";
1537  struct sockaddr_storage addr;
1538  int port, ttl;
1539 
1540  if (reply->transports[0].destination.ss_family) {
1541  addr = reply->transports[0].destination;
1542  port = reply->transports[0].port_min;
1543  ttl = reply->transports[0].ttl;
1544  } else {
1545  addr = rtsp_st->sdp_ip;
1546  port = rtsp_st->sdp_port;
1547  ttl = rtsp_st->sdp_ttl;
1548  }
1549  if (ttl > 0)
1550  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1551  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1552  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1553  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1554  port, "%s", optbuf);
1555  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1556  &s->interrupt_callback, NULL) < 0) {
1557  err = AVERROR_INVALIDDATA;
1558  goto fail;
1559  }
1560  break;
1561  }
1562  }
1563 
1564  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1565  goto fail;
1566  }
1567 
1568  if (rt->nb_rtsp_streams && reply->timeout > 0)
1569  rt->timeout = reply->timeout;
1570 
1571  if (rt->server_type == RTSP_SERVER_REAL)
1572  rt->need_subscription = 1;
1573 
1574  return 0;
1575 
1576 fail:
1577  ff_rtsp_undo_setup(s, 0);
1578  return err;
1579 }
1580 
1582 {
1583  RTSPState *rt = s->priv_data;
1584  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1585  ffurl_close(rt->rtsp_hd);
1586  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1587 }
1588 
1590 {
1591  RTSPState *rt = s->priv_data;
1592  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1593  int port, err, tcp_fd;
1594  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1595  int lower_transport_mask = 0;
1596  char real_challenge[64] = "";
1597  struct sockaddr_storage peer;
1598  socklen_t peer_len = sizeof(peer);
1599 
1600  if (rt->rtp_port_max < rt->rtp_port_min) {
1601  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1602  "than min port %d\n", rt->rtp_port_max,
1603  rt->rtp_port_min);
1604  return AVERROR(EINVAL);
1605  }
1606 
1607  if (!ff_network_init())
1608  return AVERROR(EIO);
1609 
1610  if (s->max_delay < 0) /* Not set by the caller */
1612 
1617  }
1618  /* Only pass through valid flags from here */
1620 
1621 redirect:
1622  lower_transport_mask = rt->lower_transport_mask;
1623  /* extract hostname and port */
1624  av_url_split(NULL, 0, auth, sizeof(auth),
1625  host, sizeof(host), &port, path, sizeof(path), s->filename);
1626  if (*auth) {
1627  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1628  }
1629  if (port < 0)
1630  port = RTSP_DEFAULT_PORT;
1631 
1632  if (!lower_transport_mask)
1633  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1634 
1635  if (s->oformat) {
1636  /* Only UDP or TCP - UDP multicast isn't supported. */
1637  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1638  (1 << RTSP_LOWER_TRANSPORT_TCP);
1639  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1640  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1641  "only UDP and TCP are supported for output.\n");
1642  err = AVERROR(EINVAL);
1643  goto fail;
1644  }
1645  }
1646 
1647  /* Construct the URI used in request; this is similar to s->filename,
1648  * but with authentication credentials removed and RTSP specific options
1649  * stripped out. */
1650  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1651  host, port, "%s", path);
1652 
1653  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1654  /* set up initial handshake for tunneling */
1655  char httpname[1024];
1656  char sessioncookie[17];
1657  char headers[1024];
1658 
1659  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1660  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1662 
1663  /* GET requests */
1664  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1665  &s->interrupt_callback) < 0) {
1666  err = AVERROR(EIO);
1667  goto fail;
1668  }
1669 
1670  /* generate GET headers */
1671  snprintf(headers, sizeof(headers),
1672  "x-sessioncookie: %s\r\n"
1673  "Accept: application/x-rtsp-tunnelled\r\n"
1674  "Pragma: no-cache\r\n"
1675  "Cache-Control: no-cache\r\n",
1676  sessioncookie);
1677  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1678 
1679  /* complete the connection */
1680  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1681  err = AVERROR(EIO);
1682  goto fail;
1683  }
1684 
1685  /* POST requests */
1686  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1687  &s->interrupt_callback) < 0 ) {
1688  err = AVERROR(EIO);
1689  goto fail;
1690  }
1691 
1692  /* generate POST headers */
1693  snprintf(headers, sizeof(headers),
1694  "x-sessioncookie: %s\r\n"
1695  "Content-Type: application/x-rtsp-tunnelled\r\n"
1696  "Pragma: no-cache\r\n"
1697  "Cache-Control: no-cache\r\n"
1698  "Content-Length: 32767\r\n"
1699  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1700  sessioncookie);
1701  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1702  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1703 
1704  /* Initialize the authentication state for the POST session. The HTTP
1705  * protocol implementation doesn't properly handle multi-pass
1706  * authentication for POST requests, since it would require one of
1707  * the following:
1708  * - implementing Expect: 100-continue, which many HTTP servers
1709  * don't support anyway, even less the RTSP servers that do HTTP
1710  * tunneling
1711  * - sending the whole POST data until getting a 401 reply specifying
1712  * what authentication method to use, then resending all that data
1713  * - waiting for potential 401 replies directly after sending the
1714  * POST header (waiting for some unspecified time)
1715  * Therefore, we copy the full auth state, which works for both basic
1716  * and digest. (For digest, we would have to synchronize the nonce
1717  * count variable between the two sessions, if we'd do more requests
1718  * with the original session, though.)
1719  */
1721 
1722  /* complete the connection */
1723  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1724  err = AVERROR(EIO);
1725  goto fail;
1726  }
1727  } else {
1728  /* open the tcp connection */
1729  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1730  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1731  &s->interrupt_callback, NULL) < 0) {
1732  err = AVERROR(EIO);
1733  goto fail;
1734  }
1735  rt->rtsp_hd_out = rt->rtsp_hd;
1736  }
1737  rt->seq = 0;
1738 
1739  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1740  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1741  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1742  NULL, 0, NI_NUMERICHOST);
1743  }
1744 
1745  /* request options supported by the server; this also detects server
1746  * type */
1747  for (rt->server_type = RTSP_SERVER_RTP;;) {
1748  cmd[0] = 0;
1749  if (rt->server_type == RTSP_SERVER_REAL)
1750  av_strlcat(cmd,
1751  /*
1752  * The following entries are required for proper
1753  * streaming from a Realmedia server. They are
1754  * interdependent in some way although we currently
1755  * don't quite understand how. Values were copied
1756  * from mplayer SVN r23589.
1757  * ClientChallenge is a 16-byte ID in hex
1758  * CompanyID is a 16-byte ID in base64
1759  */
1760  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1761  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1762  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1763  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1764  sizeof(cmd));
1765  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1766  if (reply->status_code != RTSP_STATUS_OK) {
1767  err = AVERROR_INVALIDDATA;
1768  goto fail;
1769  }
1770 
1771  /* detect server type if not standard-compliant RTP */
1772  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1774  continue;
1775  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1777  } else if (rt->server_type == RTSP_SERVER_REAL)
1778  strcpy(real_challenge, reply->real_challenge);
1779  break;
1780  }
1781 
1782  if (s->iformat && CONFIG_RTSP_DEMUXER)
1783  err = ff_rtsp_setup_input_streams(s, reply);
1784  else if (CONFIG_RTSP_MUXER)
1785  err = ff_rtsp_setup_output_streams(s, host);
1786  if (err)
1787  goto fail;
1788 
1789  do {
1790  int lower_transport = ff_log2_tab[lower_transport_mask &
1791  ~(lower_transport_mask - 1)];
1792 
1793  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1794  rt->server_type == RTSP_SERVER_REAL ?
1795  real_challenge : NULL);
1796  if (err < 0)
1797  goto fail;
1798  lower_transport_mask &= ~(1 << lower_transport);
1799  if (lower_transport_mask == 0 && err == 1) {
1800  err = AVERROR(EPROTONOSUPPORT);
1801  goto fail;
1802  }
1803  } while (err);
1804 
1805  rt->lower_transport_mask = lower_transport_mask;
1806  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1807  rt->state = RTSP_STATE_IDLE;
1808  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1809  return 0;
1810  fail:
1813  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1814  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1815  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1816  reply->status_code,
1817  s->filename);
1818  goto redirect;
1819  }
1820  ff_network_close();
1821  return err;
1822 }
1823 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1824 
1825 #if CONFIG_RTPDEC
1826 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1827  uint8_t *buf, int buf_size, int64_t wait_end)
1828 {
1829  RTSPState *rt = s->priv_data;
1830  RTSPStream *rtsp_st;
1831  int n, i, ret, tcp_fd, timeout_cnt = 0;
1832  int max_p = 0;
1833  struct pollfd *p = rt->p;
1834  int *fds = NULL, fdsnum, fdsidx;
1835 
1836  for (;;) {
1838  return AVERROR_EXIT;
1839  if (wait_end && wait_end - av_gettime() < 0)
1840  return AVERROR(EAGAIN);
1841  max_p = 0;
1842  if (rt->rtsp_hd) {
1843  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1844  p[max_p].fd = tcp_fd;
1845  p[max_p++].events = POLLIN;
1846  } else {
1847  tcp_fd = -1;
1848  }
1849  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1850  rtsp_st = rt->rtsp_streams[i];
1851  if (rtsp_st->rtp_handle) {
1852  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1853  &fds, &fdsnum)) {
1854  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1855  return ret;
1856  }
1857  if (fdsnum != 2) {
1858  av_log(s, AV_LOG_ERROR,
1859  "Number of fds %d not supported\n", fdsnum);
1860  return AVERROR_INVALIDDATA;
1861  }
1862  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1863  p[max_p].fd = fds[fdsidx];
1864  p[max_p++].events = POLLIN;
1865  }
1866  av_free(fds);
1867  }
1868  }
1869  n = poll(p, max_p, POLL_TIMEOUT_MS);
1870  if (n > 0) {
1871  int j = 1 - (tcp_fd == -1);
1872  timeout_cnt = 0;
1873  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1874  rtsp_st = rt->rtsp_streams[i];
1875  if (rtsp_st->rtp_handle) {
1876  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1877  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1878  if (ret > 0) {
1879  *prtsp_st = rtsp_st;
1880  return ret;
1881  }
1882  }
1883  j+=2;
1884  }
1885  }
1886 #if CONFIG_RTSP_DEMUXER
1887  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1888  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1889  if (rt->state == RTSP_STATE_STREAMING) {
1891  return AVERROR_EOF;
1892  else
1894  "Unable to answer to TEARDOWN\n");
1895  } else
1896  return 0;
1897  } else {
1898  RTSPMessageHeader reply;
1899  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1900  if (ret < 0)
1901  return ret;
1902  /* XXX: parse message */
1903  if (rt->state != RTSP_STATE_STREAMING)
1904  return 0;
1905  }
1906  }
1907 #endif
1908  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1909  return AVERROR(ETIMEDOUT);
1910  } else if (n < 0 && errno != EINTR)
1911  return AVERROR(errno);
1912  }
1913 }
1914 
1915 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1916  const uint8_t *buf, int len)
1917 {
1918  RTSPState *rt = s->priv_data;
1919  int i;
1920  if (len < 0)
1921  return len;
1922  if (rt->nb_rtsp_streams == 1) {
1923  *rtsp_st = rt->rtsp_streams[0];
1924  return len;
1925  }
1926  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1927  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1928  int no_ssrc = 0;
1929  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1930  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1931  if (!rtpctx)
1932  continue;
1933  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1934  *rtsp_st = rt->rtsp_streams[i];
1935  return len;
1936  }
1937  if (!rtpctx->ssrc)
1938  no_ssrc = 1;
1939  }
1940  if (no_ssrc) {
1942  "Unable to pick stream for packet - SSRC not known for "
1943  "all streams\n");
1944  return AVERROR(EAGAIN);
1945  }
1946  } else {
1947  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1948  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1949  *rtsp_st = rt->rtsp_streams[i];
1950  return len;
1951  }
1952  }
1953  }
1954  }
1955  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1956  return AVERROR(EAGAIN);
1957 }
1958 
1960 {
1961  RTSPState *rt = s->priv_data;
1962  int ret, len;
1963  RTSPStream *rtsp_st, *first_queue_st = NULL;
1964  int64_t wait_end = 0;
1965 
1966  if (rt->nb_byes == rt->nb_rtsp_streams)
1967  return AVERROR_EOF;
1968 
1969  /* get next frames from the same RTP packet */
1970  if (rt->cur_transport_priv) {
1971  if (rt->transport == RTSP_TRANSPORT_RDT) {
1972  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1973  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1974  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1975  } else if (rt->ts && CONFIG_RTPDEC) {
1976  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1977  if (ret >= 0) {
1978  rt->recvbuf_pos += ret;
1979  ret = rt->recvbuf_pos < rt->recvbuf_len;
1980  }
1981  } else
1982  ret = -1;
1983  if (ret == 0) {
1984  rt->cur_transport_priv = NULL;
1985  return 0;
1986  } else if (ret == 1) {
1987  return 0;
1988  } else
1989  rt->cur_transport_priv = NULL;
1990  }
1991 
1992 redo:
1993  if (rt->transport == RTSP_TRANSPORT_RTP) {
1994  int i;
1995  int64_t first_queue_time = 0;
1996  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1997  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1998  int64_t queue_time;
1999  if (!rtpctx)
2000  continue;
2001  queue_time = ff_rtp_queued_packet_time(rtpctx);
2002  if (queue_time && (queue_time - first_queue_time < 0 ||
2003  !first_queue_time)) {
2004  first_queue_time = queue_time;
2005  first_queue_st = rt->rtsp_streams[i];
2006  }
2007  }
2008  if (first_queue_time) {
2009  wait_end = first_queue_time + s->max_delay;
2010  } else {
2011  wait_end = 0;
2012  first_queue_st = NULL;
2013  }
2014  }
2015 
2016  /* read next RTP packet */
2017  if (!rt->recvbuf) {
2019  if (!rt->recvbuf)
2020  return AVERROR(ENOMEM);
2021  }
2022 
2023  switch(rt->lower_transport) {
2024  default:
2025 #if CONFIG_RTSP_DEMUXER
2027  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2028  break;
2029 #endif
2032  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2033  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2034  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2035  break;
2037  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2038  wait_end && wait_end < av_gettime())
2039  len = AVERROR(EAGAIN);
2040  else
2041  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2042  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2043  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2045  break;
2046  }
2047  if (len == AVERROR(EAGAIN) && first_queue_st &&
2048  rt->transport == RTSP_TRANSPORT_RTP) {
2049  rtsp_st = first_queue_st;
2050  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2051  goto end;
2052  }
2053  if (len < 0)
2054  return len;
2055  if (len == 0)
2056  return AVERROR_EOF;
2057  if (rt->transport == RTSP_TRANSPORT_RDT) {
2058  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2059  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2060  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2061  if (rtsp_st->feedback) {
2062  AVIOContext *pb = NULL;
2064  pb = s->pb;
2065  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2066  }
2067  if (ret < 0) {
2068  /* Either bad packet, or a RTCP packet. Check if the
2069  * first_rtcp_ntp_time field was initialized. */
2070  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2071  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2072  /* first_rtcp_ntp_time has been initialized for this stream,
2073  * copy the same value to all other uninitialized streams,
2074  * in order to map their timestamp origin to the same ntp time
2075  * as this one. */
2076  int i;
2077  AVStream *st = NULL;
2078  if (rtsp_st->stream_index >= 0)
2079  st = s->streams[rtsp_st->stream_index];
2080  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2081  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2082  AVStream *st2 = NULL;
2083  if (rt->rtsp_streams[i]->stream_index >= 0)
2084  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2085  if (rtpctx2 && st && st2 &&
2086  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2087  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2088  rtpctx2->rtcp_ts_offset = av_rescale_q(
2089  rtpctx->rtcp_ts_offset, st->time_base,
2090  st2->time_base);
2091  }
2092  }
2093  }
2094  if (ret == -RTCP_BYE) {
2095  rt->nb_byes++;
2096 
2097  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2098  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2099 
2100  if (rt->nb_byes == rt->nb_rtsp_streams)
2101  return AVERROR_EOF;
2102  }
2103  }
2104  } else if (rt->ts && CONFIG_RTPDEC) {
2105  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2106  if (ret >= 0) {
2107  if (ret < len) {
2108  rt->recvbuf_len = len;
2109  rt->recvbuf_pos = ret;
2110  rt->cur_transport_priv = rt->ts;
2111  return 1;
2112  } else {
2113  ret = 0;
2114  }
2115  }
2116  } else {
2117  return AVERROR_INVALIDDATA;
2118  }
2119 end:
2120  if (ret < 0)
2121  goto redo;
2122  if (ret == 1)
2123  /* more packets may follow, so we save the RTP context */
2124  rt->cur_transport_priv = rtsp_st->transport_priv;
2125 
2126  return ret;
2127 }
2128 #endif /* CONFIG_RTPDEC */
2129 
2130 #if CONFIG_SDP_DEMUXER
2131 static int sdp_probe(AVProbeData *p1)
2132 {
2133  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2134 
2135  /* we look for a line beginning "c=IN IP" */
2136  while (p < p_end && *p != '\0') {
2137  if (p + sizeof("c=IN IP") - 1 < p_end &&
2138  av_strstart(p, "c=IN IP", NULL))
2139  return AVPROBE_SCORE_EXTENSION;
2140 
2141  while (p < p_end - 1 && *p != '\n') p++;
2142  if (++p >= p_end)
2143  break;
2144  if (*p == '\r')
2145  p++;
2146  }
2147  return 0;
2148 }
2149 
2150 static void append_source_addrs(char *buf, int size, const char *name,
2151  int count, struct RTSPSource **addrs)
2152 {
2153  int i;
2154  if (!count)
2155  return;
2156  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2157  for (i = 1; i < count; i++)
2158  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2159 }
2160 
2161 static int sdp_read_header(AVFormatContext *s)
2162 {
2163  RTSPState *rt = s->priv_data;
2164  RTSPStream *rtsp_st;
2165  int size, i, err;
2166  char *content;
2167  char url[1024];
2168 
2169  if (!ff_network_init())
2170  return AVERROR(EIO);
2171 
2172  if (s->max_delay < 0) /* Not set by the caller */
2174  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2176 
2177  /* read the whole sdp file */
2178  /* XXX: better loading */
2179  content = av_malloc(SDP_MAX_SIZE);
2180  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2181  if (size <= 0) {
2182  av_free(content);
2183  return AVERROR_INVALIDDATA;
2184  }
2185  content[size] ='\0';
2186 
2187  err = ff_sdp_parse(s, content);
2188  av_free(content);
2189  if (err) goto fail;
2190 
2191  /* open each RTP stream */
2192  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2193  char namebuf[50];
2194  rtsp_st = rt->rtsp_streams[i];
2195 
2196  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2197  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2198  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2199  ff_url_join(url, sizeof(url), "rtp", NULL,
2200  namebuf, rtsp_st->sdp_port,
2201  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2202  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2203  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2204  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2205 
2206  append_source_addrs(url, sizeof(url), "sources",
2207  rtsp_st->nb_include_source_addrs,
2208  rtsp_st->include_source_addrs);
2209  append_source_addrs(url, sizeof(url), "block",
2210  rtsp_st->nb_exclude_source_addrs,
2211  rtsp_st->exclude_source_addrs);
2212  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2213  &s->interrupt_callback, NULL) < 0) {
2214  err = AVERROR_INVALIDDATA;
2215  goto fail;
2216  }
2217  }
2218  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2219  goto fail;
2220  }
2221  return 0;
2222 fail:
2224  ff_network_close();
2225  return err;
2226 }
2227 
2228 static int sdp_read_close(AVFormatContext *s)
2229 {
2231  ff_network_close();
2232  return 0;
2233 }
2234 
2235 static const AVClass sdp_demuxer_class = {
2236  .class_name = "SDP demuxer",
2237  .item_name = av_default_item_name,
2238  .option = sdp_options,
2239  .version = LIBAVUTIL_VERSION_INT,
2240 };
2241 
2242 AVInputFormat ff_sdp_demuxer = {
2243  .name = "sdp",
2244  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2245  .priv_data_size = sizeof(RTSPState),
2246  .read_probe = sdp_probe,
2247  .read_header = sdp_read_header,
2249  .read_close = sdp_read_close,
2250  .priv_class = &sdp_demuxer_class,
2251 };
2252 #endif /* CONFIG_SDP_DEMUXER */
2253 
2254 #if CONFIG_RTP_DEMUXER
2255 static int rtp_probe(AVProbeData *p)
2256 {
2257  if (av_strstart(p->filename, "rtp:", NULL))
2258  return AVPROBE_SCORE_MAX;
2259  return 0;
2260 }
2261 
2262 static int rtp_read_header(AVFormatContext *s)
2263 {
2264  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2265  char host[500], sdp[500];
2266  int ret, port;
2267  URLContext* in = NULL;
2268  int payload_type;
2269  AVCodecContext codec = { 0 };
2270  struct sockaddr_storage addr;
2271  AVIOContext pb;
2272  socklen_t addrlen = sizeof(addr);
2273  RTSPState *rt = s->priv_data;
2274 
2275  if (!ff_network_init())
2276  return AVERROR(EIO);
2277 
2278  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2279  &s->interrupt_callback, NULL);
2280  if (ret)
2281  goto fail;
2282 
2283  while (1) {
2284  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2285  if (ret == AVERROR(EAGAIN))
2286  continue;
2287  if (ret < 0)
2288  goto fail;
2289  if (ret < 12) {
2290  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2291  continue;
2292  }
2293 
2294  if ((recvbuf[0] & 0xc0) != 0x80) {
2295  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2296  "received\n");
2297  continue;
2298  }
2299 
2300  if (RTP_PT_IS_RTCP(recvbuf[1]))
2301  continue;
2302 
2303  payload_type = recvbuf[1] & 0x7f;
2304  break;
2305  }
2306  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2307  ffurl_close(in);
2308  in = NULL;
2309 
2310  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2311  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2312  "without an SDP file describing it\n",
2313  payload_type);
2314  goto fail;
2315  }
2316  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2317  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2318  "properly you need an SDP file "
2319  "describing it\n");
2320  }
2321 
2322  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2323  NULL, 0, s->filename);
2324 
2325  snprintf(sdp, sizeof(sdp),
2326  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2327  addr.ss_family == AF_INET ? 4 : 6, host,
2328  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2329  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2330  port, payload_type);
2331  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2332 
2333  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2334  s->pb = &pb;
2335 
2336  /* sdp_read_header initializes this again */
2337  ff_network_close();
2338 
2339  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2340 
2341  ret = sdp_read_header(s);
2342  s->pb = NULL;
2343  return ret;
2344 
2345 fail:
2346  if (in)
2347  ffurl_close(in);
2348  ff_network_close();
2349  return ret;
2350 }
2351 
2352 static const AVClass rtp_demuxer_class = {
2353  .class_name = "RTP demuxer",
2354  .item_name = av_default_item_name,
2355  .option = rtp_options,
2356  .version = LIBAVUTIL_VERSION_INT,
2357 };
2358 
2359 AVInputFormat ff_rtp_demuxer = {
2360  .name = "rtp",
2361  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2362  .priv_data_size = sizeof(RTSPState),
2363  .read_probe = rtp_probe,
2364  .read_header = rtp_read_header,
2366  .read_close = sdp_read_close,
2367  .flags = AVFMT_NOFILE,
2368  .priv_class = &rtp_demuxer_class,
2369 };
2370 #endif /* CONFIG_RTP_DEMUXER */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:272
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:92
void av_url_split(char *proto, int proto_size, char *authorization, int authorization_size, char *hostname, int hostname_size, int *port_ptr, char *path, int path_size, const char *url)
Split a URL string into components.
Definition: utils.c:2705
char crypto_suite[40]
Definition: rtsp.h:457
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:386
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
Parse a Windows Media Server-specific SDP line.
Definition: rtpdec_asf.c:96
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:527
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
Bytestream IO Context.
Definition: avio.h:68
Realmedia Data Transport.
Definition: rtsp.h:58
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:54
int ff_rtp_get_local_rtp_port(URLContext *h)
Return the local rtp port used by the RTP connection.
Definition: rtpproto.c:498
int size
#define RTP_MAX_PACKET_LENGTH
Definition: rtpdec.h:36
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
Send a dummy packet on both port pairs to set up the connection state in potential NAT routers...
Definition: rtpdec.c:352
AVIOInterruptCB interrupt_callback
Custom interrupt callbacks for the I/O layer.
Definition: avformat.h:1163
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
Definition: aviobuf.c:966
AVOption.
Definition: opt.h:234
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:114
HTTPAuthType
Authentication types, ordered from weakest to strongest.
Definition: httpauth.h:28
char content_type[64]
Content type header.
Definition: rtsp.h:186
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:129
const char * filename
Definition: avformat.h:396
static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
Parse a string p in the form of Range:npt=xx-xx, and determine the start and end time.
Definition: rtsp.c:146
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:316
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:2821
int av_parse_time(int64_t *timeval, const char *timestr, int duration)
Parse timestr and return in *time a corresponding number of microseconds.
Definition: parseutils.c:482
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:276
#define RTSP_DEFAULT_PORT
Definition: rtsp.h:72
Windows Media server.
Definition: rtsp.h:208
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:353
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:732
int ffurl_connect(URLContext *uc, AVDictionary **options)
Connect an URLContext that has been allocated by ffurl_alloc.
Definition: avio.c:159
int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse RDT-style packet data (header + media data).
Definition: rdt.c:335
int index
stream index in AVFormatContext
Definition: avformat.h:700
#define AVIO_FLAG_READ
read-only
Definition: avio.h:292
char location[4096]
the "Location:" field.
Definition: rtsp.h:151
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:293
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int mode_record
transport set to record data
Definition: rtsp.h:111
enum AVMediaType codec_type
Definition: rtp.c:36
int av_strncasecmp(const char *a, const char *b, size_t n)
Locale-independent case-insensitive compare.
Definition: avstring.c:166
void ff_network_close(void)
Definition: network.c:150
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:143
initialized and sending/receiving data
Definition: rtsp.h:196
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:269
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define RTSP_FLAG_RTCP_TO_SOURCE
Send RTCP packets to the source address of received packets.
Definition: rtsp.h:406
#define RTSP_RTP_PORT_MAX
Definition: rtsp.h:78
#define freeaddrinfo
Definition: network.h:183
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:437
int ctx_flags
Flags signalling stream properties.
Definition: avformat.h:971
#define RTSP_FLAG_LISTEN
Wait for incoming connections.
Definition: rtsp.h:404
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:244
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:238
const char * ff_rtp_enc_name(int payload_type)
Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given p...
Definition: rtp.c:131
AVCodec.
Definition: avcodec.h:2796
#define AI_NUMERICHOST
Definition: network.h:152
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:120
This describes the server response to each RTSP command.
Definition: rtsp.h:126
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:488
#define RECVBUF_SIZE
Definition: rtsp.c:58
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:141
Format I/O context.
Definition: avformat.h:922
#define RTP_PT_PRIVATE
Definition: rtp.h:77
enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
Return the codec id for the given encoding name and codec type.
Definition: rtp.c:142
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:206
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:396
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:38
int recvbuf_len
Definition: rtsp.h:322
Public dictionary API.
int av_stristart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str independent of case.
Definition: avstring.c:43
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:358
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:147
Opaque data information usually continuous.
Definition: avutil.h:189
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:108
int(* init)(AVFormatContext *s, int st_index, PayloadContext *priv_data)
Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null...
Definition: rtpdec.h:124
int ff_network_init(void)
Definition: network.c:123
#define AVFMTCTX_NOHEADER
signal that no header is present (streams are added dynamically)
Definition: avformat.h:901
AVOptions.
miscellaneous OS support macros and functions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:455
#define AV_RB32
Definition: intreadwrite.h:130
uint16_t ss_family
Definition: network.h:93
PayloadContext *(* alloc)(void)
Allocate any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:129
int id
Format-specific stream ID.
Definition: avformat.h:706
#define POLL_TIMEOUT_MS
Definition: rtsp.c:54
#define DEFAULT_REORDERING_DELAY
Definition: rtsp.c:59
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
void(* free)(PayloadContext *protocol_data)
Free any data needed by the rtp parsing for this dynamic data.
Definition: rtpdec.h:131
const char * name
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:2518
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:990
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:371
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:327
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:420
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
static int flags
Definition: log.c:44
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:130
#define AVERROR_EOF
End of file.
Definition: error.h:51
void ff_http_init_auth_state(URLContext *dest, const URLContext *src)
Initialize the authentication state based on another HTTP URLContext.
Definition: http.c:127
#define CONFIG_RTPDEC
Definition: config.h:420
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:139
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
const uint8_t ff_log2_tab[256]
Definition: log2_tab.c:21
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:452
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
Normal RTSP.
Definition: rtsp.h:68
#define CONFIG_RTSP_DEMUXER
Definition: config.h:893
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:133
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:452
struct AVOutputFormat * oformat
The output container format.
Definition: avformat.h:941
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:176
const OptionDef options[]
Definition: avconv_opt.c:2187
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, char *attr, char *value)
Definition: rtpdec_latm.c:148
#define RTSP_DEFAULT_AUDIO_SAMPLERATE
Definition: rtsp.h:76
void ff_rdt_parse_close(RDTDemuxContext *s)
Definition: rdt.c:78
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:129
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:440
Private data for the RTSP demuxer.
Definition: rtsp.h:217
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:254
int ffurl_alloc(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb)
Create a URLContext for accessing to the resource indicated by url, but do not initiate the connectio...
Definition: avio.c:183
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
AVDictionary * metadata
Metadata that applies to the whole file.
Definition: avformat.h:1130
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:186
int ffurl_get_multi_file_handle(URLContext *h, int **handles, int *numhandles)
Return the file descriptors associated with this URL.
Definition: avio.c:359
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:249
#define AV_RB16
Definition: intreadwrite.h:53
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:150
int avio_close(AVIOContext *s)
Close the resource accessed by the AVIOContext s and free it.
Definition: aviobuf.c:800
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:144
const AVOption ff_rtsp_options[]
Definition: rtsp.c:78
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:181
Definition: graph2dot.c:49
URLContext * rtsp_hd
Definition: rtsp.h:219
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:168
const char * name
Name of the codec implementation.
Definition: avcodec.h:2803
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:330
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:438
int ffio_read_partial(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:511
char * av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size)
Encode data to base64 and null-terminate.
Definition: base64.c:72
int64_t rtcp_ts_offset
Definition: rtpdec.h:180
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:81
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:110
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:224
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:163
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
Initialize a codec context based on the payload type.
Definition: rtp.c:70
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:425
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:718
MpegTSContext * ff_mpegts_parse_open(AVFormatContext *s)
Definition: mpegts.c:2230
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:398
int seq
RTSP command sequence number.
Definition: rtsp.h:240
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:397
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:338
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:978
#define RTSP_FLAG_CUSTOM_IO
Do all IO via the AVIOContext.
Definition: rtsp.h:405
#define NI_NUMERICHOST
Definition: network.h:160
#define LIBAVFORMAT_IDENT
Definition: version.h:44
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:306
int recvbuf_pos
Definition: rtsp.h:321
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:64
char filename[1024]
input or output filename
Definition: avformat.h:998
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:222
int64_t first_rtcp_ntp_time
Definition: rtpdec.h:178
#define AV_BASE64_SIZE(x)
Calculate the output size needed to base64-encode x bytes.
Definition: base64.h:59
#define FFMIN(a, b)
Definition: common.h:57
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:282
int av_strcasecmp(const char *a, const char *b)
Definition: avstring.c:156
static int read_probe(AVProbeData *pd)
Definition: jvdec.c:55
int content_length
length of the data following this header
Definition: rtsp.h:128
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: [;ti...
Definition: rtsp.h:171
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:74
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:87
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:111
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:139
static void get_word(char *buf, int buf_size, const char **pp)
Definition: rtsp.c:137
char crypto_params[100]
Definition: rtsp.h:458
Usually treated as AVMEDIA_TYPE_DATA.
Definition: avutil.h:186
int(* parse_sdp_a_line)(AVFormatContext *s, int st_index, PayloadContext *priv_data, const char *line)
Parse the a= line from the sdp field.
Definition: rtpdec.h:126
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:352
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:52
#define ENC
Definition: rtsp.c:63
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:435
int ff_mpegts_parse_packet(MpegTSContext *ts, AVPacket *pkt, const uint8_t *buf, int len)
Definition: mpegts.c:2246
Raw data (over UDP)
Definition: rtsp.h:59
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:320
int stale
Auth ok, but needs to be resent with a new nonce.
Definition: httpauth.h:71
int sdp_payload_type
payload type
Definition: rtsp.h:442
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:520
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:439
static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
Definition: rtsp.c:166
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:544
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:37
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:420
Stream structure.
Definition: avformat.h:699
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
Definition: error.h:57
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
int nb_byes
Definition: rtsp.h:335
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:261
NULL
Definition: eval.c:55
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:411
#define AV_LOG_INFO
Standard information.
Definition: log.h:134
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:436
enum AVMediaType codec_type
Definition: avcodec.h:1058
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:663
int ff_check_interrupt(AVIOInterruptCB *cb)
Check if the user has requested to interrup a blocking function associated with cb.
Definition: avio.c:381
enum AVCodecID codec_id
Definition: avcodec.h:1067
int rtp_port_max
Definition: rtsp.h:386
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:1791
AVIOContext * pb
I/O context.
Definition: avformat.h:964
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:381
av_default_item_name
Definition: dnxhdenc.c:52
int server_port_max
Definition: rtsp.h:104
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:73
main external API structure.
Definition: avcodec.h:1050
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:1780
#define RTSP_FLAG_OPTS(name, longname)
Definition: rtsp.c:65
RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler)
Allocate and init the RDT parsing context.
Definition: rdt.c:55
#define RTSP_FLAG_FILTER_SRC
Filter incoming UDP packets - receive packets only from the right source address and port...
Definition: rtsp.h:399
enum AVCodecID codec_id
Definition: rtpdec.h:118
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:257
Definition: url.h:41
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
static int read_packet(AVFormatContext *ctx, AVPacket *pkt)
Definition: libcdio.c:114
#define AVIO_FLAG_READ_WRITE
read-write pseudo flag
Definition: avio.h:294
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:68
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:376
int client_port_max
Definition: rtsp.h:100
Describe the class of an AVClass context structure.
Definition: log.h:33
#define SDP_MAX_SIZE
Definition: rtsp.c:57
void ff_real_parse_sdp_a_line(AVFormatContext *s, int stream_index, const char *line)
Parse a server-related SDP line.
Definition: rdt.c:513
#define SPACE_CHARS
Definition: internal.h:160
void * priv_data
Definition: url.h:44
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:451
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:278
not initialized
Definition: rtsp.h:195
int64_t range_end
Definition: rtsp.h:137
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:117
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:154
AVMediaType
Definition: avutil.h:185
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:99
#define RTSP_MEDIATYPE_OPTS(name, longname)
Definition: rtsp.c:69
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:699
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:714
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:699
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
Definition: rtpdec.c:249
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:402
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:2445
This structure contains the data a format has to probe a file.
Definition: avformat.h:395
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS
Definition: rtsp.h:75
misc parsing utilities
char * ff_http_auth_create_response(HTTPAuthState *state, const char *auth, const char *path, const char *method)
Definition: httpauth.c:242
size_t av_strlcat(char *dst, const char *src, size_t size)
Append the string src to the string dst, but to a total length of no more than size - 1 bytes...
Definition: avstring.c:91
int interleaved_max
Definition: rtsp.h:92
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:266
int ffurl_close(URLContext *h)
Close the resource accessed by the URLContext h, and free the memory used by it.
Definition: avio.c:297
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:137
int64_t start_time
Position of the first frame of the component, in AV_TIME_BASE fractional seconds. ...
Definition: avformat.h:1007
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:230
#define DEC
Definition: rtsp.c:62
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:404
int av_strstart(const char *str, const char *pfx, const char **ptr)
Return non-zero if pfx is a prefix of str.
Definition: avstring.c:32
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
#define getaddrinfo
Definition: network.h:182
Main libavformat public API header.
static const AVOption sdp_options[]
Definition: rtsp.c:96
void ff_mpegts_parse_close(MpegTSContext *ts)
Definition: mpegts.c:2271
int ff_rtp_chain_mux_open(AVFormatContext **out, AVFormatContext *s, AVStream *st, URLContext *handle, int packet_size, int idx)
Definition: rtpenc_chain.c:29
uint32_t ssrc
Definition: rtpdec.h:151
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:409
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:98
int ffio_init_context(AVIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
Definition: aviobuf.c:70
int ffurl_open(URLContext **puc, const char *filename, int flags, const AVIOInterruptCB *int_cb, AVDictionary **options)
Create an URLContext for accessing to the resource indicated by url, and open it. ...
Definition: avio.c:211
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:287
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:448
int ffurl_read_complete(URLContext *h, unsigned char *buf, int size)
Read as many bytes as possible (up to size), calling the read function multiple times if necessary...
Definition: avio.c:269
void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge)
Calculate the response (RealChallenge2 in the RTSP header) to the challenge (RealChallenge1 in the RT...
Definition: rdt.c:94
#define RTSP_REORDERING_OPTS()
Definition: rtsp.c:75
struct AVInputFormat * iformat
The input container format.
Definition: avformat.h:934
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:2496
void ff_http_auth_handle_header(HTTPAuthState *state, const char *key, const char *value)
Definition: httpauth.c:90
uint32_t base_timestamp
Definition: rtpdec.h:154
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
#define getnameinfo
Definition: network.h:184
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp)
Definition: rtsp.c:130
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:275
int len
#define RTSP_RTP_PORT_MIN
Definition: rtsp.h:77
int channels
number of audio channels
Definition: avcodec.h:1792
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:431
void * priv_data
Format private data.
Definition: avformat.h:950
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:568
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:822
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:441
#define MAX_TIMEOUTS
Definition: rtsp.c:56
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:583
int ai_flags
Definition: network.h:103
int64_t duration
Duration of the stream, in AV_TIME_BASE fractional seconds.
Definition: avformat.h:1017
HTTPAuthType auth_type
The currently chosen auth type.
Definition: httpauth.h:59
Realmedia-style server.
Definition: rtsp.h:207
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:343
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:525
unbuffered private I/O API
uint32_t av_get_random_seed(void)
Get random data.
Definition: random_seed.c:95
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:741
int interleaved_max
Definition: rtsp.h:429
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:809
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:113
int ff_rtp_set_remote_url(URLContext *h, const char *uri)
If no filename is given to av_open_input_file because you want to get the local port first...
Definition: rtpproto.c:63
#define RTP_REORDER_QUEUE_DEFAULT_SIZE
Definition: rtpdec.h:38
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:429
This structure stores compressed data.
Definition: avcodec.h:950
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:104
#define CONFIG_RTSP_MUXER
Definition: config.h:1252
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
int av_opt_set(void *obj, const char *name, const char *val, int search_flags)
Definition: opt.c:210
static const AVOption rtp_options[]
Definition: rtsp.c:105
int ffurl_read(URLContext *h, unsigned char *buf, int size)
Read up to size bytes from the resource accessed by h, and store the read bytes in buf...
Definition: avio.c:262
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:205
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:421
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:228
#define OFFSET(x)
Definition: rtsp.c:61
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:96
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:422
No authentication specified.
Definition: httpauth.h:29
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:100