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  if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
931  break;
932  }
933 }
934 
935 static void handle_rtp_info(RTSPState *rt, const char *url,
936  uint32_t seq, uint32_t rtptime)
937 {
938  int i;
939  if (!rtptime || !url[0])
940  return;
941  if (rt->transport != RTSP_TRANSPORT_RTP)
942  return;
943  for (i = 0; i < rt->nb_rtsp_streams; i++) {
944  RTSPStream *rtsp_st = rt->rtsp_streams[i];
945  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
946  if (!rtpctx)
947  continue;
948  if (!strcmp(rtsp_st->control_url, url)) {
949  rtpctx->base_timestamp = rtptime;
950  break;
951  }
952  }
953 }
954 
955 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
956 {
957  int read = 0;
958  char key[20], value[1024], url[1024] = "";
959  uint32_t seq = 0, rtptime = 0;
960 
961  for (;;) {
962  p += strspn(p, SPACE_CHARS);
963  if (!*p)
964  break;
965  get_word_sep(key, sizeof(key), "=", &p);
966  if (*p != '=')
967  break;
968  p++;
969  get_word_sep(value, sizeof(value), ";, ", &p);
970  read++;
971  if (!strcmp(key, "url"))
972  av_strlcpy(url, value, sizeof(url));
973  else if (!strcmp(key, "seq"))
974  seq = strtoul(value, NULL, 10);
975  else if (!strcmp(key, "rtptime"))
976  rtptime = strtoul(value, NULL, 10);
977  if (*p == ',') {
978  handle_rtp_info(rt, url, seq, rtptime);
979  url[0] = '\0';
980  seq = rtptime = 0;
981  read = 0;
982  }
983  if (*p)
984  p++;
985  }
986  if (read > 0)
987  handle_rtp_info(rt, url, seq, rtptime);
988 }
989 
990 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
991  RTSPState *rt, const char *method)
992 {
993  const char *p;
994 
995  /* NOTE: we do case independent match for broken servers */
996  p = buf;
997  if (av_stristart(p, "Session:", &p)) {
998  int t;
999  get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1000  if (av_stristart(p, ";timeout=", &p) &&
1001  (t = strtol(p, NULL, 10)) > 0) {
1002  reply->timeout = t;
1003  }
1004  } else if (av_stristart(p, "Content-Length:", &p)) {
1005  reply->content_length = strtol(p, NULL, 10);
1006  } else if (av_stristart(p, "Transport:", &p)) {
1007  rtsp_parse_transport(reply, p);
1008  } else if (av_stristart(p, "CSeq:", &p)) {
1009  reply->seq = strtol(p, NULL, 10);
1010  } else if (av_stristart(p, "Range:", &p)) {
1011  rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1012  } else if (av_stristart(p, "RealChallenge1:", &p)) {
1013  p += strspn(p, SPACE_CHARS);
1014  av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1015  } else if (av_stristart(p, "Server:", &p)) {
1016  p += strspn(p, SPACE_CHARS);
1017  av_strlcpy(reply->server, p, sizeof(reply->server));
1018  } else if (av_stristart(p, "Notice:", &p) ||
1019  av_stristart(p, "X-Notice:", &p)) {
1020  reply->notice = strtol(p, NULL, 10);
1021  } else if (av_stristart(p, "Location:", &p)) {
1022  p += strspn(p, SPACE_CHARS);
1023  av_strlcpy(reply->location, p , sizeof(reply->location));
1024  } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1025  p += strspn(p, SPACE_CHARS);
1026  ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1027  } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1028  p += strspn(p, SPACE_CHARS);
1029  ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1030  } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1031  p += strspn(p, SPACE_CHARS);
1032  if (method && !strcmp(method, "DESCRIBE"))
1033  av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1034  } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1035  p += strspn(p, SPACE_CHARS);
1036  if (method && !strcmp(method, "PLAY"))
1037  rtsp_parse_rtp_info(rt, p);
1038  } else if (av_stristart(p, "Public:", &p) && rt) {
1039  if (strstr(p, "GET_PARAMETER") &&
1040  method && !strcmp(method, "OPTIONS"))
1041  rt->get_parameter_supported = 1;
1042  } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1043  p += strspn(p, SPACE_CHARS);
1044  rt->accept_dynamic_rate = atoi(p);
1045  } else if (av_stristart(p, "Content-Type:", &p)) {
1046  p += strspn(p, SPACE_CHARS);
1047  av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1048  }
1049 }
1050 
1051 /* skip a RTP/TCP interleaved packet */
1053 {
1054  RTSPState *rt = s->priv_data;
1055  int ret, len, len1;
1056  uint8_t buf[1024];
1057 
1058  ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1059  if (ret != 3)
1060  return;
1061  len = AV_RB16(buf + 1);
1062 
1063  av_dlog(s, "skipping RTP packet len=%d\n", len);
1064 
1065  /* skip payload */
1066  while (len > 0) {
1067  len1 = len;
1068  if (len1 > sizeof(buf))
1069  len1 = sizeof(buf);
1070  ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1071  if (ret != len1)
1072  return;
1073  len -= len1;
1074  }
1075 }
1076 
1078  unsigned char **content_ptr,
1079  int return_on_interleaved_data, const char *method)
1080 {
1081  RTSPState *rt = s->priv_data;
1082  char buf[4096], buf1[1024], *q;
1083  unsigned char ch;
1084  const char *p;
1085  int ret, content_length, line_count = 0, request = 0;
1086  unsigned char *content = NULL;
1087 
1088 start:
1089  line_count = 0;
1090  request = 0;
1091  content = NULL;
1092  memset(reply, 0, sizeof(*reply));
1093 
1094  /* parse reply (XXX: use buffers) */
1095  rt->last_reply[0] = '\0';
1096  for (;;) {
1097  q = buf;
1098  for (;;) {
1099  ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1100  av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1101  if (ret != 1)
1102  return AVERROR_EOF;
1103  if (ch == '\n')
1104  break;
1105  if (ch == '$' && q == buf) {
1106  if (return_on_interleaved_data) {
1107  return 1;
1108  } else
1110  } else if (ch != '\r') {
1111  if ((q - buf) < sizeof(buf) - 1)
1112  *q++ = ch;
1113  }
1114  }
1115  *q = '\0';
1116 
1117  av_dlog(s, "line='%s'\n", buf);
1118 
1119  /* test if last line */
1120  if (buf[0] == '\0')
1121  break;
1122  p = buf;
1123  if (line_count == 0) {
1124  /* get reply code */
1125  get_word(buf1, sizeof(buf1), &p);
1126  if (!strncmp(buf1, "RTSP/", 5)) {
1127  get_word(buf1, sizeof(buf1), &p);
1128  reply->status_code = atoi(buf1);
1129  av_strlcpy(reply->reason, p, sizeof(reply->reason));
1130  } else {
1131  av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1132  get_word(buf1, sizeof(buf1), &p); // object
1133  request = 1;
1134  }
1135  } else {
1136  ff_rtsp_parse_line(reply, p, rt, method);
1137  av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1138  av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1139  }
1140  line_count++;
1141  }
1142 
1143  if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1144  av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1145 
1146  content_length = reply->content_length;
1147  if (content_length > 0) {
1148  /* leave some room for a trailing '\0' (useful for simple parsing) */
1149  content = av_malloc(content_length + 1);
1150  if (!content)
1151  return AVERROR(ENOMEM);
1152  ffurl_read_complete(rt->rtsp_hd, content, content_length);
1153  content[content_length] = '\0';
1154  }
1155  if (content_ptr)
1156  *content_ptr = content;
1157  else
1158  av_free(content);
1159 
1160  if (request) {
1161  char buf[1024];
1162  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1163  const char* ptr = buf;
1164 
1165  if (!strcmp(reply->reason, "OPTIONS")) {
1166  snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1167  if (reply->seq)
1168  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1169  if (reply->session_id[0])
1170  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1171  reply->session_id);
1172  } else {
1173  snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1174  }
1175  av_strlcat(buf, "\r\n", sizeof(buf));
1176 
1177  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1178  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1179  ptr = base64buf;
1180  }
1181  ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1182 
1183  rt->last_cmd_time = av_gettime();
1184  /* Even if the request from the server had data, it is not the data
1185  * that the caller wants or expects. The memory could also be leaked
1186  * if the actual following reply has content data. */
1187  if (content_ptr)
1188  av_freep(content_ptr);
1189  /* If method is set, this is called from ff_rtsp_send_cmd,
1190  * where a reply to exactly this request is awaited. For
1191  * callers from within packet receiving, we just want to
1192  * return to the caller and go back to receiving packets. */
1193  if (method)
1194  goto start;
1195  return 0;
1196  }
1197 
1198  if (rt->seq != reply->seq) {
1199  av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1200  rt->seq, reply->seq);
1201  }
1202 
1203  /* EOS */
1204  if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1205  reply->notice == 2104 /* Start-of-Stream Reached */ ||
1206  reply->notice == 2306 /* Continuous Feed Terminated */) {
1207  rt->state = RTSP_STATE_IDLE;
1208  } else if (reply->notice >= 4400 && reply->notice < 5500) {
1209  return AVERROR(EIO); /* data or server error */
1210  } else if (reply->notice == 2401 /* Ticket Expired */ ||
1211  (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1212  return AVERROR(EPERM);
1213 
1214  return 0;
1215 }
1216 
1230 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1231  const char *method, const char *url,
1232  const char *headers,
1233  const unsigned char *send_content,
1234  int send_content_length)
1235 {
1236  RTSPState *rt = s->priv_data;
1237  char buf[4096], *out_buf;
1238  char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1239 
1240  /* Add in RTSP headers */
1241  out_buf = buf;
1242  rt->seq++;
1243  snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1244  if (headers)
1245  av_strlcat(buf, headers, sizeof(buf));
1246  av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1247  av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1248  if (rt->session_id[0] != '\0' && (!headers ||
1249  !strstr(headers, "\nIf-Match:"))) {
1250  av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1251  }
1252  if (rt->auth[0]) {
1253  char *str = ff_http_auth_create_response(&rt->auth_state,
1254  rt->auth, url, method);
1255  if (str)
1256  av_strlcat(buf, str, sizeof(buf));
1257  av_free(str);
1258  }
1259  if (send_content_length > 0 && send_content)
1260  av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1261  av_strlcat(buf, "\r\n", sizeof(buf));
1262 
1263  /* base64 encode rtsp if tunneling */
1264  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1265  av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1266  out_buf = base64buf;
1267  }
1268 
1269  av_dlog(s, "Sending:\n%s--\n", buf);
1270 
1271  ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1272  if (send_content_length > 0 && send_content) {
1273  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1274  av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1275  "with content data not supported\n");
1276  return AVERROR_PATCHWELCOME;
1277  }
1278  ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1279  }
1280  rt->last_cmd_time = av_gettime();
1281 
1282  return 0;
1283 }
1284 
1285 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1286  const char *url, const char *headers)
1287 {
1288  return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1289 }
1290 
1291 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1292  const char *headers, RTSPMessageHeader *reply,
1293  unsigned char **content_ptr)
1294 {
1295  return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1296  content_ptr, NULL, 0);
1297 }
1298 
1300  const char *method, const char *url,
1301  const char *header,
1302  RTSPMessageHeader *reply,
1303  unsigned char **content_ptr,
1304  const unsigned char *send_content,
1305  int send_content_length)
1306 {
1307  RTSPState *rt = s->priv_data;
1308  HTTPAuthType cur_auth_type;
1309  int ret, attempts = 0;
1310 
1311 retry:
1312  cur_auth_type = rt->auth_state.auth_type;
1313  if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1314  send_content,
1315  send_content_length)))
1316  return ret;
1317 
1318  if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1319  return ret;
1320  attempts++;
1321 
1322  if (reply->status_code == 401 &&
1323  (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1324  rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1325  goto retry;
1326 
1327  if (reply->status_code > 400){
1328  av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1329  method,
1330  reply->status_code,
1331  reply->reason);
1332  av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1333  }
1334 
1335  return 0;
1336 }
1337 
1338 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1339  int lower_transport, const char *real_challenge)
1340 {
1341  RTSPState *rt = s->priv_data;
1342  int rtx = 0, j, i, err, interleave = 0, port_off;
1343  RTSPStream *rtsp_st;
1344  RTSPMessageHeader reply1, *reply = &reply1;
1345  char cmd[2048];
1346  const char *trans_pref;
1347 
1348  if (rt->transport == RTSP_TRANSPORT_RDT)
1349  trans_pref = "x-pn-tng";
1350  else if (rt->transport == RTSP_TRANSPORT_RAW)
1351  trans_pref = "RAW/RAW";
1352  else
1353  trans_pref = "RTP/AVP";
1354 
1355  /* default timeout: 1 minute */
1356  rt->timeout = 60;
1357 
1358  /* for each stream, make the setup request */
1359  /* XXX: we assume the same server is used for the control of each
1360  * RTSP stream */
1361 
1362  /* Choose a random starting offset within the first half of the
1363  * port range, to allow for a number of ports to try even if the offset
1364  * happens to be at the end of the random range. */
1365  port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1366  /* even random offset */
1367  port_off -= port_off & 0x01;
1368 
1369  for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1370  char transport[2048];
1371 
1372  /*
1373  * WMS serves all UDP data over a single connection, the RTX, which
1374  * isn't necessarily the first in the SDP but has to be the first
1375  * to be set up, else the second/third SETUP will fail with a 461.
1376  */
1377  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1378  rt->server_type == RTSP_SERVER_WMS) {
1379  if (i == 0) {
1380  /* rtx first */
1381  for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1382  int len = strlen(rt->rtsp_streams[rtx]->control_url);
1383  if (len >= 4 &&
1384  !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1385  "/rtx"))
1386  break;
1387  }
1388  if (rtx == rt->nb_rtsp_streams)
1389  return -1; /* no RTX found */
1390  rtsp_st = rt->rtsp_streams[rtx];
1391  } else
1392  rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1393  } else
1394  rtsp_st = rt->rtsp_streams[i];
1395 
1396  /* RTP/UDP */
1397  if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1398  char buf[256];
1399 
1400  if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1401  port = reply->transports[0].client_port_min;
1402  goto have_port;
1403  }
1404 
1405  /* first try in specified port range */
1406  while (j <= rt->rtp_port_max) {
1407  ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1408  "?localport=%d", j);
1409  /* we will use two ports per rtp stream (rtp and rtcp) */
1410  j += 2;
1411  if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1412  &s->interrupt_callback, NULL))
1413  goto rtp_opened;
1414  }
1415 
1416  av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1417  err = AVERROR(EIO);
1418  goto fail;
1419 
1420  rtp_opened:
1421  port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1422  have_port:
1423  snprintf(transport, sizeof(transport) - 1,
1424  "%s/UDP;", trans_pref);
1425  if (rt->server_type != RTSP_SERVER_REAL)
1426  av_strlcat(transport, "unicast;", sizeof(transport));
1427  av_strlcatf(transport, sizeof(transport),
1428  "client_port=%d", port);
1429  if (rt->transport == RTSP_TRANSPORT_RTP &&
1430  !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1431  av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1432  }
1433 
1434  /* RTP/TCP */
1435  else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1436  /* For WMS streams, the application streams are only used for
1437  * UDP. When trying to set it up for TCP streams, the server
1438  * will return an error. Therefore, we skip those streams. */
1439  if (rt->server_type == RTSP_SERVER_WMS &&
1440  (rtsp_st->stream_index < 0 ||
1441  s->streams[rtsp_st->stream_index]->codec->codec_type ==
1443  continue;
1444  snprintf(transport, sizeof(transport) - 1,
1445  "%s/TCP;", trans_pref);
1446  if (rt->transport != RTSP_TRANSPORT_RDT)
1447  av_strlcat(transport, "unicast;", sizeof(transport));
1448  av_strlcatf(transport, sizeof(transport),
1449  "interleaved=%d-%d",
1450  interleave, interleave + 1);
1451  interleave += 2;
1452  }
1453 
1454  else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1455  snprintf(transport, sizeof(transport) - 1,
1456  "%s/UDP;multicast", trans_pref);
1457  }
1458  if (s->oformat) {
1459  av_strlcat(transport, ";mode=record", sizeof(transport));
1460  } else if (rt->server_type == RTSP_SERVER_REAL ||
1462  av_strlcat(transport, ";mode=play", sizeof(transport));
1463  snprintf(cmd, sizeof(cmd),
1464  "Transport: %s\r\n",
1465  transport);
1466  if (rt->accept_dynamic_rate)
1467  av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1468  if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1469  char real_res[41], real_csum[9];
1470  ff_rdt_calc_response_and_checksum(real_res, real_csum,
1471  real_challenge);
1472  av_strlcatf(cmd, sizeof(cmd),
1473  "If-Match: %s\r\n"
1474  "RealChallenge2: %s, sd=%s\r\n",
1475  rt->session_id, real_res, real_csum);
1476  }
1477  ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1478  if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1479  err = 1;
1480  goto fail;
1481  } else if (reply->status_code != RTSP_STATUS_OK ||
1482  reply->nb_transports != 1) {
1483  err = AVERROR_INVALIDDATA;
1484  goto fail;
1485  }
1486 
1487  /* XXX: same protocol for all streams is required */
1488  if (i > 0) {
1489  if (reply->transports[0].lower_transport != rt->lower_transport ||
1490  reply->transports[0].transport != rt->transport) {
1491  err = AVERROR_INVALIDDATA;
1492  goto fail;
1493  }
1494  } else {
1495  rt->lower_transport = reply->transports[0].lower_transport;
1496  rt->transport = reply->transports[0].transport;
1497  }
1498 
1499  /* Fail if the server responded with another lower transport mode
1500  * than what we requested. */
1501  if (reply->transports[0].lower_transport != lower_transport) {
1502  av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1503  err = AVERROR_INVALIDDATA;
1504  goto fail;
1505  }
1506 
1507  switch(reply->transports[0].lower_transport) {
1509  rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1510  rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1511  break;
1512 
1513  case RTSP_LOWER_TRANSPORT_UDP: {
1514  char url[1024], options[30] = "";
1515  const char *peer = host;
1516 
1517  if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1518  av_strlcpy(options, "?connect=1", sizeof(options));
1519  /* Use source address if specified */
1520  if (reply->transports[0].source[0])
1521  peer = reply->transports[0].source;
1522  ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1523  reply->transports[0].server_port_min, "%s", options);
1524  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1525  ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1526  err = AVERROR_INVALIDDATA;
1527  goto fail;
1528  }
1529  /* Try to initialize the connection state in a
1530  * potential NAT router by sending dummy packets.
1531  * RTP/RTCP dummy packets are used for RDT, too.
1532  */
1533  if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1534  CONFIG_RTPDEC)
1536  break;
1537  }
1539  char url[1024], namebuf[50], optbuf[20] = "";
1540  struct sockaddr_storage addr;
1541  int port, ttl;
1542 
1543  if (reply->transports[0].destination.ss_family) {
1544  addr = reply->transports[0].destination;
1545  port = reply->transports[0].port_min;
1546  ttl = reply->transports[0].ttl;
1547  } else {
1548  addr = rtsp_st->sdp_ip;
1549  port = rtsp_st->sdp_port;
1550  ttl = rtsp_st->sdp_ttl;
1551  }
1552  if (ttl > 0)
1553  snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1554  getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1555  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1556  ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1557  port, "%s", optbuf);
1558  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1559  &s->interrupt_callback, NULL) < 0) {
1560  err = AVERROR_INVALIDDATA;
1561  goto fail;
1562  }
1563  break;
1564  }
1565  }
1566 
1567  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1568  goto fail;
1569  }
1570 
1571  if (rt->nb_rtsp_streams && reply->timeout > 0)
1572  rt->timeout = reply->timeout;
1573 
1574  if (rt->server_type == RTSP_SERVER_REAL)
1575  rt->need_subscription = 1;
1576 
1577  return 0;
1578 
1579 fail:
1580  ff_rtsp_undo_setup(s, 0);
1581  return err;
1582 }
1583 
1585 {
1586  RTSPState *rt = s->priv_data;
1587  if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1588  ffurl_close(rt->rtsp_hd);
1589  rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1590 }
1591 
1593 {
1594  RTSPState *rt = s->priv_data;
1595  char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1596  int port, err, tcp_fd;
1597  RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1598  int lower_transport_mask = 0;
1599  char real_challenge[64] = "";
1600  struct sockaddr_storage peer;
1601  socklen_t peer_len = sizeof(peer);
1602 
1603  if (rt->rtp_port_max < rt->rtp_port_min) {
1604  av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1605  "than min port %d\n", rt->rtp_port_max,
1606  rt->rtp_port_min);
1607  return AVERROR(EINVAL);
1608  }
1609 
1610  if (!ff_network_init())
1611  return AVERROR(EIO);
1612 
1613  if (s->max_delay < 0) /* Not set by the caller */
1615 
1620  }
1621  /* Only pass through valid flags from here */
1623 
1624 redirect:
1625  lower_transport_mask = rt->lower_transport_mask;
1626  /* extract hostname and port */
1627  av_url_split(NULL, 0, auth, sizeof(auth),
1628  host, sizeof(host), &port, path, sizeof(path), s->filename);
1629  if (*auth) {
1630  av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1631  }
1632  if (port < 0)
1633  port = RTSP_DEFAULT_PORT;
1634 
1635  if (!lower_transport_mask)
1636  lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1637 
1638  if (s->oformat) {
1639  /* Only UDP or TCP - UDP multicast isn't supported. */
1640  lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1641  (1 << RTSP_LOWER_TRANSPORT_TCP);
1642  if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1643  av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1644  "only UDP and TCP are supported for output.\n");
1645  err = AVERROR(EINVAL);
1646  goto fail;
1647  }
1648  }
1649 
1650  /* Construct the URI used in request; this is similar to s->filename,
1651  * but with authentication credentials removed and RTSP specific options
1652  * stripped out. */
1653  ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1654  host, port, "%s", path);
1655 
1656  if (rt->control_transport == RTSP_MODE_TUNNEL) {
1657  /* set up initial handshake for tunneling */
1658  char httpname[1024];
1659  char sessioncookie[17];
1660  char headers[1024];
1661 
1662  ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1663  snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1665 
1666  /* GET requests */
1667  if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1668  &s->interrupt_callback) < 0) {
1669  err = AVERROR(EIO);
1670  goto fail;
1671  }
1672 
1673  /* generate GET headers */
1674  snprintf(headers, sizeof(headers),
1675  "x-sessioncookie: %s\r\n"
1676  "Accept: application/x-rtsp-tunnelled\r\n"
1677  "Pragma: no-cache\r\n"
1678  "Cache-Control: no-cache\r\n",
1679  sessioncookie);
1680  av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1681 
1682  /* complete the connection */
1683  if (ffurl_connect(rt->rtsp_hd, NULL)) {
1684  err = AVERROR(EIO);
1685  goto fail;
1686  }
1687 
1688  /* POST requests */
1689  if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1690  &s->interrupt_callback) < 0 ) {
1691  err = AVERROR(EIO);
1692  goto fail;
1693  }
1694 
1695  /* generate POST headers */
1696  snprintf(headers, sizeof(headers),
1697  "x-sessioncookie: %s\r\n"
1698  "Content-Type: application/x-rtsp-tunnelled\r\n"
1699  "Pragma: no-cache\r\n"
1700  "Cache-Control: no-cache\r\n"
1701  "Content-Length: 32767\r\n"
1702  "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1703  sessioncookie);
1704  av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1705  av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1706 
1707  /* Initialize the authentication state for the POST session. The HTTP
1708  * protocol implementation doesn't properly handle multi-pass
1709  * authentication for POST requests, since it would require one of
1710  * the following:
1711  * - implementing Expect: 100-continue, which many HTTP servers
1712  * don't support anyway, even less the RTSP servers that do HTTP
1713  * tunneling
1714  * - sending the whole POST data until getting a 401 reply specifying
1715  * what authentication method to use, then resending all that data
1716  * - waiting for potential 401 replies directly after sending the
1717  * POST header (waiting for some unspecified time)
1718  * Therefore, we copy the full auth state, which works for both basic
1719  * and digest. (For digest, we would have to synchronize the nonce
1720  * count variable between the two sessions, if we'd do more requests
1721  * with the original session, though.)
1722  */
1724 
1725  /* complete the connection */
1726  if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1727  err = AVERROR(EIO);
1728  goto fail;
1729  }
1730  } else {
1731  /* open the tcp connection */
1732  ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1733  if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1734  &s->interrupt_callback, NULL) < 0) {
1735  err = AVERROR(EIO);
1736  goto fail;
1737  }
1738  rt->rtsp_hd_out = rt->rtsp_hd;
1739  }
1740  rt->seq = 0;
1741 
1742  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1743  if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1744  getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1745  NULL, 0, NI_NUMERICHOST);
1746  }
1747 
1748  /* request options supported by the server; this also detects server
1749  * type */
1750  for (rt->server_type = RTSP_SERVER_RTP;;) {
1751  cmd[0] = 0;
1752  if (rt->server_type == RTSP_SERVER_REAL)
1753  av_strlcat(cmd,
1754  /*
1755  * The following entries are required for proper
1756  * streaming from a Realmedia server. They are
1757  * interdependent in some way although we currently
1758  * don't quite understand how. Values were copied
1759  * from mplayer SVN r23589.
1760  * ClientChallenge is a 16-byte ID in hex
1761  * CompanyID is a 16-byte ID in base64
1762  */
1763  "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1764  "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1765  "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1766  "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1767  sizeof(cmd));
1768  ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1769  if (reply->status_code != RTSP_STATUS_OK) {
1770  err = AVERROR_INVALIDDATA;
1771  goto fail;
1772  }
1773 
1774  /* detect server type if not standard-compliant RTP */
1775  if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1777  continue;
1778  } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1780  } else if (rt->server_type == RTSP_SERVER_REAL)
1781  strcpy(real_challenge, reply->real_challenge);
1782  break;
1783  }
1784 
1785  if (s->iformat && CONFIG_RTSP_DEMUXER)
1786  err = ff_rtsp_setup_input_streams(s, reply);
1787  else if (CONFIG_RTSP_MUXER)
1788  err = ff_rtsp_setup_output_streams(s, host);
1789  if (err)
1790  goto fail;
1791 
1792  do {
1793  int lower_transport = ff_log2_tab[lower_transport_mask &
1794  ~(lower_transport_mask - 1)];
1795 
1796  err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1797  rt->server_type == RTSP_SERVER_REAL ?
1798  real_challenge : NULL);
1799  if (err < 0)
1800  goto fail;
1801  lower_transport_mask &= ~(1 << lower_transport);
1802  if (lower_transport_mask == 0 && err == 1) {
1803  err = AVERROR(EPROTONOSUPPORT);
1804  goto fail;
1805  }
1806  } while (err);
1807 
1808  rt->lower_transport_mask = lower_transport_mask;
1809  av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1810  rt->state = RTSP_STATE_IDLE;
1811  rt->seek_timestamp = 0; /* default is to start stream at position zero */
1812  return 0;
1813  fail:
1816  if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1817  av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1818  av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1819  reply->status_code,
1820  s->filename);
1821  goto redirect;
1822  }
1823  ff_network_close();
1824  return err;
1825 }
1826 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1827 
1828 #if CONFIG_RTPDEC
1829 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1830  uint8_t *buf, int buf_size, int64_t wait_end)
1831 {
1832  RTSPState *rt = s->priv_data;
1833  RTSPStream *rtsp_st;
1834  int n, i, ret, tcp_fd, timeout_cnt = 0;
1835  int max_p = 0;
1836  struct pollfd *p = rt->p;
1837  int *fds = NULL, fdsnum, fdsidx;
1838 
1839  for (;;) {
1841  return AVERROR_EXIT;
1842  if (wait_end && wait_end - av_gettime() < 0)
1843  return AVERROR(EAGAIN);
1844  max_p = 0;
1845  if (rt->rtsp_hd) {
1846  tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1847  p[max_p].fd = tcp_fd;
1848  p[max_p++].events = POLLIN;
1849  } else {
1850  tcp_fd = -1;
1851  }
1852  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1853  rtsp_st = rt->rtsp_streams[i];
1854  if (rtsp_st->rtp_handle) {
1855  if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1856  &fds, &fdsnum)) {
1857  av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1858  return ret;
1859  }
1860  if (fdsnum != 2) {
1861  av_log(s, AV_LOG_ERROR,
1862  "Number of fds %d not supported\n", fdsnum);
1863  return AVERROR_INVALIDDATA;
1864  }
1865  for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1866  p[max_p].fd = fds[fdsidx];
1867  p[max_p++].events = POLLIN;
1868  }
1869  av_free(fds);
1870  }
1871  }
1872  n = poll(p, max_p, POLL_TIMEOUT_MS);
1873  if (n > 0) {
1874  int j = 1 - (tcp_fd == -1);
1875  timeout_cnt = 0;
1876  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1877  rtsp_st = rt->rtsp_streams[i];
1878  if (rtsp_st->rtp_handle) {
1879  if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1880  ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1881  if (ret > 0) {
1882  *prtsp_st = rtsp_st;
1883  return ret;
1884  }
1885  }
1886  j+=2;
1887  }
1888  }
1889 #if CONFIG_RTSP_DEMUXER
1890  if (tcp_fd != -1 && p[0].revents & POLLIN) {
1891  if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1892  if (rt->state == RTSP_STATE_STREAMING) {
1894  return AVERROR_EOF;
1895  else
1897  "Unable to answer to TEARDOWN\n");
1898  } else
1899  return 0;
1900  } else {
1901  RTSPMessageHeader reply;
1902  ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1903  if (ret < 0)
1904  return ret;
1905  /* XXX: parse message */
1906  if (rt->state != RTSP_STATE_STREAMING)
1907  return 0;
1908  }
1909  }
1910 #endif
1911  } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1912  return AVERROR(ETIMEDOUT);
1913  } else if (n < 0 && errno != EINTR)
1914  return AVERROR(errno);
1915  }
1916 }
1917 
1918 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1919  const uint8_t *buf, int len)
1920 {
1921  RTSPState *rt = s->priv_data;
1922  int i;
1923  if (len < 0)
1924  return len;
1925  if (rt->nb_rtsp_streams == 1) {
1926  *rtsp_st = rt->rtsp_streams[0];
1927  return len;
1928  }
1929  if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1930  if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1931  int no_ssrc = 0;
1932  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1933  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1934  if (!rtpctx)
1935  continue;
1936  if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1937  *rtsp_st = rt->rtsp_streams[i];
1938  return len;
1939  }
1940  if (!rtpctx->ssrc)
1941  no_ssrc = 1;
1942  }
1943  if (no_ssrc) {
1945  "Unable to pick stream for packet - SSRC not known for "
1946  "all streams\n");
1947  return AVERROR(EAGAIN);
1948  }
1949  } else {
1950  for (i = 0; i < rt->nb_rtsp_streams; i++) {
1951  if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1952  *rtsp_st = rt->rtsp_streams[i];
1953  return len;
1954  }
1955  }
1956  }
1957  }
1958  av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1959  return AVERROR(EAGAIN);
1960 }
1961 
1963 {
1964  RTSPState *rt = s->priv_data;
1965  int ret, len;
1966  RTSPStream *rtsp_st, *first_queue_st = NULL;
1967  int64_t wait_end = 0;
1968 
1969  if (rt->nb_byes == rt->nb_rtsp_streams)
1970  return AVERROR_EOF;
1971 
1972  /* get next frames from the same RTP packet */
1973  if (rt->cur_transport_priv) {
1974  if (rt->transport == RTSP_TRANSPORT_RDT) {
1975  ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1976  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1977  ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1978  } else if (rt->ts && CONFIG_RTPDEC) {
1979  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1980  if (ret >= 0) {
1981  rt->recvbuf_pos += ret;
1982  ret = rt->recvbuf_pos < rt->recvbuf_len;
1983  }
1984  } else
1985  ret = -1;
1986  if (ret == 0) {
1987  rt->cur_transport_priv = NULL;
1988  return 0;
1989  } else if (ret == 1) {
1990  return 0;
1991  } else
1992  rt->cur_transport_priv = NULL;
1993  }
1994 
1995 redo:
1996  if (rt->transport == RTSP_TRANSPORT_RTP) {
1997  int i;
1998  int64_t first_queue_time = 0;
1999  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2000  RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2001  int64_t queue_time;
2002  if (!rtpctx)
2003  continue;
2004  queue_time = ff_rtp_queued_packet_time(rtpctx);
2005  if (queue_time && (queue_time - first_queue_time < 0 ||
2006  !first_queue_time)) {
2007  first_queue_time = queue_time;
2008  first_queue_st = rt->rtsp_streams[i];
2009  }
2010  }
2011  if (first_queue_time) {
2012  wait_end = first_queue_time + s->max_delay;
2013  } else {
2014  wait_end = 0;
2015  first_queue_st = NULL;
2016  }
2017  }
2018 
2019  /* read next RTP packet */
2020  if (!rt->recvbuf) {
2022  if (!rt->recvbuf)
2023  return AVERROR(ENOMEM);
2024  }
2025 
2026  switch(rt->lower_transport) {
2027  default:
2028 #if CONFIG_RTSP_DEMUXER
2030  len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2031  break;
2032 #endif
2035  len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2036  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2037  ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2038  break;
2040  if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2041  wait_end && wait_end < av_gettime())
2042  len = AVERROR(EAGAIN);
2043  else
2044  len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2045  len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2046  if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2048  break;
2049  }
2050  if (len == AVERROR(EAGAIN) && first_queue_st &&
2051  rt->transport == RTSP_TRANSPORT_RTP) {
2052  rtsp_st = first_queue_st;
2053  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2054  goto end;
2055  }
2056  if (len < 0)
2057  return len;
2058  if (len == 0)
2059  return AVERROR_EOF;
2060  if (rt->transport == RTSP_TRANSPORT_RDT) {
2061  ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2062  } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2063  ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2064  if (rtsp_st->feedback) {
2065  AVIOContext *pb = NULL;
2067  pb = s->pb;
2068  ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2069  }
2070  if (ret < 0) {
2071  /* Either bad packet, or a RTCP packet. Check if the
2072  * first_rtcp_ntp_time field was initialized. */
2073  RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2074  if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2075  /* first_rtcp_ntp_time has been initialized for this stream,
2076  * copy the same value to all other uninitialized streams,
2077  * in order to map their timestamp origin to the same ntp time
2078  * as this one. */
2079  int i;
2080  AVStream *st = NULL;
2081  if (rtsp_st->stream_index >= 0)
2082  st = s->streams[rtsp_st->stream_index];
2083  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2084  RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2085  AVStream *st2 = NULL;
2086  if (rt->rtsp_streams[i]->stream_index >= 0)
2087  st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2088  if (rtpctx2 && st && st2 &&
2089  rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2090  rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2091  rtpctx2->rtcp_ts_offset = av_rescale_q(
2092  rtpctx->rtcp_ts_offset, st->time_base,
2093  st2->time_base);
2094  }
2095  }
2096  }
2097  if (ret == -RTCP_BYE) {
2098  rt->nb_byes++;
2099 
2100  av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2101  rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2102 
2103  if (rt->nb_byes == rt->nb_rtsp_streams)
2104  return AVERROR_EOF;
2105  }
2106  }
2107  } else if (rt->ts && CONFIG_RTPDEC) {
2108  ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2109  if (ret >= 0) {
2110  if (ret < len) {
2111  rt->recvbuf_len = len;
2112  rt->recvbuf_pos = ret;
2113  rt->cur_transport_priv = rt->ts;
2114  return 1;
2115  } else {
2116  ret = 0;
2117  }
2118  }
2119  } else {
2120  return AVERROR_INVALIDDATA;
2121  }
2122 end:
2123  if (ret < 0)
2124  goto redo;
2125  if (ret == 1)
2126  /* more packets may follow, so we save the RTP context */
2127  rt->cur_transport_priv = rtsp_st->transport_priv;
2128 
2129  return ret;
2130 }
2131 #endif /* CONFIG_RTPDEC */
2132 
2133 #if CONFIG_SDP_DEMUXER
2134 static int sdp_probe(AVProbeData *p1)
2135 {
2136  const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2137 
2138  /* we look for a line beginning "c=IN IP" */
2139  while (p < p_end && *p != '\0') {
2140  if (p + sizeof("c=IN IP") - 1 < p_end &&
2141  av_strstart(p, "c=IN IP", NULL))
2142  return AVPROBE_SCORE_EXTENSION;
2143 
2144  while (p < p_end - 1 && *p != '\n') p++;
2145  if (++p >= p_end)
2146  break;
2147  if (*p == '\r')
2148  p++;
2149  }
2150  return 0;
2151 }
2152 
2153 static void append_source_addrs(char *buf, int size, const char *name,
2154  int count, struct RTSPSource **addrs)
2155 {
2156  int i;
2157  if (!count)
2158  return;
2159  av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2160  for (i = 1; i < count; i++)
2161  av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2162 }
2163 
2164 static int sdp_read_header(AVFormatContext *s)
2165 {
2166  RTSPState *rt = s->priv_data;
2167  RTSPStream *rtsp_st;
2168  int size, i, err;
2169  char *content;
2170  char url[1024];
2171 
2172  if (!ff_network_init())
2173  return AVERROR(EIO);
2174 
2175  if (s->max_delay < 0) /* Not set by the caller */
2177  if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2179 
2180  /* read the whole sdp file */
2181  /* XXX: better loading */
2182  content = av_malloc(SDP_MAX_SIZE);
2183  size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2184  if (size <= 0) {
2185  av_free(content);
2186  return AVERROR_INVALIDDATA;
2187  }
2188  content[size] ='\0';
2189 
2190  err = ff_sdp_parse(s, content);
2191  av_free(content);
2192  if (err) goto fail;
2193 
2194  /* open each RTP stream */
2195  for (i = 0; i < rt->nb_rtsp_streams; i++) {
2196  char namebuf[50];
2197  rtsp_st = rt->rtsp_streams[i];
2198 
2199  if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2200  getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2201  namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2202  ff_url_join(url, sizeof(url), "rtp", NULL,
2203  namebuf, rtsp_st->sdp_port,
2204  "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2205  rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2206  rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2207  rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2208 
2209  append_source_addrs(url, sizeof(url), "sources",
2210  rtsp_st->nb_include_source_addrs,
2211  rtsp_st->include_source_addrs);
2212  append_source_addrs(url, sizeof(url), "block",
2213  rtsp_st->nb_exclude_source_addrs,
2214  rtsp_st->exclude_source_addrs);
2215  if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2216  &s->interrupt_callback, NULL) < 0) {
2217  err = AVERROR_INVALIDDATA;
2218  goto fail;
2219  }
2220  }
2221  if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2222  goto fail;
2223  }
2224  return 0;
2225 fail:
2227  ff_network_close();
2228  return err;
2229 }
2230 
2231 static int sdp_read_close(AVFormatContext *s)
2232 {
2234  ff_network_close();
2235  return 0;
2236 }
2237 
2238 static const AVClass sdp_demuxer_class = {
2239  .class_name = "SDP demuxer",
2240  .item_name = av_default_item_name,
2241  .option = sdp_options,
2242  .version = LIBAVUTIL_VERSION_INT,
2243 };
2244 
2245 AVInputFormat ff_sdp_demuxer = {
2246  .name = "sdp",
2247  .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2248  .priv_data_size = sizeof(RTSPState),
2249  .read_probe = sdp_probe,
2250  .read_header = sdp_read_header,
2252  .read_close = sdp_read_close,
2253  .priv_class = &sdp_demuxer_class,
2254 };
2255 #endif /* CONFIG_SDP_DEMUXER */
2256 
2257 #if CONFIG_RTP_DEMUXER
2258 static int rtp_probe(AVProbeData *p)
2259 {
2260  if (av_strstart(p->filename, "rtp:", NULL))
2261  return AVPROBE_SCORE_MAX;
2262  return 0;
2263 }
2264 
2265 static int rtp_read_header(AVFormatContext *s)
2266 {
2267  uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2268  char host[500], sdp[500];
2269  int ret, port;
2270  URLContext* in = NULL;
2271  int payload_type;
2272  AVCodecContext codec = { 0 };
2273  struct sockaddr_storage addr;
2274  AVIOContext pb;
2275  socklen_t addrlen = sizeof(addr);
2276  RTSPState *rt = s->priv_data;
2277 
2278  if (!ff_network_init())
2279  return AVERROR(EIO);
2280 
2281  ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2282  &s->interrupt_callback, NULL);
2283  if (ret)
2284  goto fail;
2285 
2286  while (1) {
2287  ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2288  if (ret == AVERROR(EAGAIN))
2289  continue;
2290  if (ret < 0)
2291  goto fail;
2292  if (ret < 12) {
2293  av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2294  continue;
2295  }
2296 
2297  if ((recvbuf[0] & 0xc0) != 0x80) {
2298  av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2299  "received\n");
2300  continue;
2301  }
2302 
2303  if (RTP_PT_IS_RTCP(recvbuf[1]))
2304  continue;
2305 
2306  payload_type = recvbuf[1] & 0x7f;
2307  break;
2308  }
2309  getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2310  ffurl_close(in);
2311  in = NULL;
2312 
2313  if (ff_rtp_get_codec_info(&codec, payload_type)) {
2314  av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2315  "without an SDP file describing it\n",
2316  payload_type);
2317  goto fail;
2318  }
2319  if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2320  av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2321  "properly you need an SDP file "
2322  "describing it\n");
2323  }
2324 
2325  av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2326  NULL, 0, s->filename);
2327 
2328  snprintf(sdp, sizeof(sdp),
2329  "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2330  addr.ss_family == AF_INET ? 4 : 6, host,
2331  codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2332  codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2333  port, payload_type);
2334  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2335 
2336  ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2337  s->pb = &pb;
2338 
2339  /* sdp_read_header initializes this again */
2340  ff_network_close();
2341 
2342  rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2343 
2344  ret = sdp_read_header(s);
2345  s->pb = NULL;
2346  return ret;
2347 
2348 fail:
2349  if (in)
2350  ffurl_close(in);
2351  ff_network_close();
2352  return ret;
2353 }
2354 
2355 static const AVClass rtp_demuxer_class = {
2356  .class_name = "RTP demuxer",
2357  .item_name = av_default_item_name,
2358  .option = rtp_options,
2359  .version = LIBAVUTIL_VERSION_INT,
2360 };
2361 
2362 AVInputFormat ff_rtp_demuxer = {
2363  .name = "rtp",
2364  .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2365  .priv_data_size = sizeof(RTSPState),
2366  .read_probe = rtp_probe,
2367  .read_header = rtp_read_header,
2369  .read_close = sdp_read_close,
2370  .flags = AVFMT_NOFILE,
2371  .priv_class = &rtp_demuxer_class,
2372 };
2373 #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:2713
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:2829
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:2812
#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
#define CONFIG_RTPDEC
Definition: config.h:439
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
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:73
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:2521
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 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
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:145
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:169
const char * name
Name of the codec implementation.
Definition: avcodec.h:2819
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:2250
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:2266
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
#define CONFIG_RTSP_MUXER
Definition: config.h:1271
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:1807
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
#define CONFIG_RTSP_DEMUXER
Definition: config.h:912
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:2291
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:2499
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:1808
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:589
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
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:212
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